Merge "Change docs to include aaudio/Aaudio.h"
diff --git a/.clang-format b/.clang-format
new file mode 100644
index 0000000..a7614d2
--- /dev/null
+++ b/.clang-format
@@ -0,0 +1,13 @@
+BasedOnStyle: Google
+Standard: Cpp11
+AccessModifierOffset: -2
+AllowShortFunctionsOnASingleLine: Inline
+ColumnLimit: 100
+CommentPragmas: NOLINT:.*
+DerivePointerAlignment: false
+IncludeBlocks: Preserve
+IndentWidth: 4
+ContinuationIndentWidth: 8
+PointerAlignment: Left
+TabWidth: 4
+UseTab: Never
diff --git a/camera/CameraBase.cpp b/camera/CameraBase.cpp
index 03439fd..24c9108 100644
--- a/camera/CameraBase.cpp
+++ b/camera/CameraBase.cpp
@@ -68,6 +68,9 @@
unavailablePhysicalIds16.push_back(String16(id8));
}
res = parcel->writeString16Vector(unavailablePhysicalIds16);
+ if (res != OK) return res;
+
+ res = parcel->writeString16(String16(clientPackage));
return res;
}
@@ -86,6 +89,12 @@
for (auto& id16 : unavailablePhysicalIds16) {
unavailablePhysicalIds.push_back(String8(id16));
}
+
+ String16 tempClientPackage;
+ res = parcel->readString16(&tempClientPackage);
+ if (res != OK) return res;
+ clientPackage = String8(tempClientPackage);
+
return res;
}
diff --git a/camera/OWNERS b/camera/OWNERS
index d6b95da..a4ab616 100644
--- a/camera/OWNERS
+++ b/camera/OWNERS
@@ -1,8 +1,8 @@
+
+# Bug component: 41727
+arakesh@google.com
epeev@google.com
etalvala@google.com
jchowdhary@google.com
shuzhenwang@google.com
-yinchiayeh@google.com
-# backup owner
-cychen@google.com
-zhijunhe@google.com
+ruchamk@google.com
diff --git a/camera/include/camera/CameraBase.h b/camera/include/camera/CameraBase.h
index e156994..8e53968 100644
--- a/camera/include/camera/CameraBase.h
+++ b/camera/include/camera/CameraBase.h
@@ -85,11 +85,17 @@
*/
std::vector<String8> unavailablePhysicalIds;
+ /**
+ * Client package name if camera is open, otherwise not applicable
+ */
+ String8 clientPackage;
+
virtual status_t writeToParcel(android::Parcel* parcel) const;
virtual status_t readFromParcel(const android::Parcel* parcel);
- CameraStatus(String8 id, int32_t s, const std::vector<String8>& unavailSubIds) :
- cameraId(id), status(s), unavailablePhysicalIds(unavailSubIds) {}
+ CameraStatus(String8 id, int32_t s, const std::vector<String8>& unavailSubIds,
+ const String8& clientPkg) : cameraId(id), status(s),
+ unavailablePhysicalIds(unavailSubIds), clientPackage(clientPkg) {}
CameraStatus() : status(ICameraServiceListener::STATUS_PRESENT) {}
};
diff --git a/camera/ndk/impl/ACameraManager.h b/camera/ndk/impl/ACameraManager.h
index ccbfaa9..da887a2 100644
--- a/camera/ndk/impl/ACameraManager.h
+++ b/camera/ndk/impl/ACameraManager.h
@@ -139,6 +139,8 @@
return !(*this == other);
}
bool operator < (const Callback& other) const {
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wordered-compare-function-pointers"
if (*this == other) return false;
if (mContext != other.mContext) return mContext < other.mContext;
if (mPhysicalCamAvailable != other.mPhysicalCamAvailable) {
@@ -152,6 +154,7 @@
}
if (mAvailable != other.mAvailable) return mAvailable < other.mAvailable;
return mUnavailable < other.mUnavailable;
+#pragma GCC diagnostic pop
}
bool operator > (const Callback& other) const {
return (*this != other && !(*this < other));
diff --git a/camera/ndk/include/camera/NdkCameraMetadataTags.h b/camera/ndk/include/camera/NdkCameraMetadataTags.h
index 86781e5..c353b2d 100644
--- a/camera/ndk/include/camera/NdkCameraMetadataTags.h
+++ b/camera/ndk/include/camera/NdkCameraMetadataTags.h
@@ -4578,6 +4578,25 @@
*
* <p>Also defines the direction of rolling shutter readout, which is from top to bottom in
* the sensor's coordinate system.</p>
+ * <p>Starting with Android API level 32, camera clients that query the orientation via
+ * <a href="https://developer.android.com/reference/android/hardware/camera2/CameraCharacteristics.html#get">CameraCharacteristics#get</a> on foldable devices which
+ * include logical cameras can receive a value that can dynamically change depending on the
+ * device/fold state.
+ * Clients are advised to not cache or store the orientation value of such logical sensors.
+ * In case repeated queries to CameraCharacteristics are not preferred, then clients can
+ * also access the entire mapping from device state to sensor orientation in
+ * <a href="https://developer.android.com/reference/android/hardware/camera2/params/DeviceStateOrientationMap.html">DeviceStateOrientationMap</a>.
+ * Do note that a dynamically changing sensor orientation value in camera characteristics
+ * will not be the best way to establish the orientation per frame. Clients that want to
+ * know the sensor orientation of a particular captured frame should query the
+ * ACAMERA_LOGICAL_MULTI_CAMERA_ACTIVE_PHYSICAL_ID from the corresponding capture result and
+ * check the respective physical camera orientation.</p>
+ * <p>Native camera clients must query ACAMERA_INFO_DEVICE_STATE_ORIENTATIONS for the mapping
+ * between device state and camera sensor orientation. Dynamic updates to the sensor
+ * orientation are not supported in this code path.</p>
+ *
+ * @see ACAMERA_INFO_DEVICE_STATE_ORIENTATIONS
+ * @see ACAMERA_LOGICAL_MULTI_CAMERA_ACTIVE_PHYSICAL_ID
*/
ACAMERA_SENSOR_ORIENTATION = // int32
ACAMERA_SENSOR_START + 14,
@@ -6284,6 +6303,21 @@
*/
ACAMERA_INFO_VERSION = // byte
ACAMERA_INFO_START + 1,
+ /**
+ *
+ * <p>Type: int64[2*n]</p>
+ *
+ * <p>This tag may appear in:
+ * <ul>
+ * <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
+ * </ul></p>
+ *
+ * <p>HAL must populate the array with
+ * (hardware::camera::provider::V2_5::DeviceState, sensorOrientation) pairs for each
+ * supported device state bitwise combination.</p>
+ */
+ ACAMERA_INFO_DEVICE_STATE_ORIENTATIONS = // int64[2*n]
+ ACAMERA_INFO_START + 3,
ACAMERA_INFO_END,
/**
diff --git a/camera/ndk/ndk_vendor/impl/ACameraManager.h b/camera/ndk/ndk_vendor/impl/ACameraManager.h
index 8359bb1..4663529 100644
--- a/camera/ndk/ndk_vendor/impl/ACameraManager.h
+++ b/camera/ndk/ndk_vendor/impl/ACameraManager.h
@@ -136,6 +136,8 @@
return !(*this == other);
}
bool operator < (const Callback& other) const {
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wordered-compare-function-pointers"
if (*this == other) return false;
if (mContext != other.mContext) return mContext < other.mContext;
if (mAvailable != other.mAvailable) return mAvailable < other.mAvailable;
@@ -146,6 +148,7 @@
if (mPhysicalCamUnavailable != other.mPhysicalCamUnavailable)
return mPhysicalCamUnavailable < other.mPhysicalCamUnavailable;
return mUnavailable < other.mUnavailable;
+#pragma GCC diagnostic pop
}
bool operator > (const Callback& other) const {
return (*this != other && !(*this < other));
diff --git a/cmds/screenrecord/Android.bp b/cmds/screenrecord/Android.bp
index 359a835..d0b3ce0 100644
--- a/cmds/screenrecord/Android.bp
+++ b/cmds/screenrecord/Android.bp
@@ -55,12 +55,6 @@
"libGLESv2",
],
- include_dirs: [
- "frameworks/av/media/libstagefright",
- "frameworks/av/media/libstagefright/include",
- "frameworks/native/include/media/openmax",
- ],
-
cflags: [
"-Werror",
"-Wall",
diff --git a/drm/mediadrm/plugins/TEST_MAPPING b/drm/mediadrm/plugins/TEST_MAPPING
index 87becb6..fd4ef95 100644
--- a/drm/mediadrm/plugins/TEST_MAPPING
+++ b/drm/mediadrm/plugins/TEST_MAPPING
@@ -1,19 +1,19 @@
{
"presubmit": [
{
- "name": "CtsMediaTestCases",
+ "name": "CtsMediaDrmTestCases",
"options" : [
{
"include-annotation": "android.platform.test.annotations.Presubmit"
},
{
- "include-filter": "android.media.cts.MediaDrmClearkeyTest"
+ "include-filter": "android.mediadrm.cts.MediaDrmClearkeyTest"
},
{
- "include-filter": "android.media.cts.MediaDrmMetricsTest"
+ "include-filter": "android.mediadrm.cts.MediaDrmMetricsTest"
},
{
- "include-filter": "android.media.cts.NativeMediaDrmClearkeyTest"
+ "include-filter": "android.mediadrm.cts.NativeMediaDrmClearkeyTest"
}
]
}
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 200e92d..bd6db55 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -53,6 +53,83 @@
//EL_FIXME 20 seconds may not be enough and must be reconciled with new obtainBuffer implementation
#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 // assuming up to a maximum of 20 seconds of offloaded
+// for audio_track_cblk_t::mState, to match TrackBase.h
+static inline constexpr int CBLK_STATE_IDLE = 0;
+static inline constexpr int CBLK_STATE_PAUSING = 7;
+
+/**
+ * MirroredVariable is a local variable which simultaneously updates
+ * a mirrored storage location. This is useful for server side variables
+ * where a local copy is kept, but a client visible copy is offered through shared memory.
+ *
+ * We use std::atomic as the default container class to access this memory.
+ */
+template <typename T, template <typename> class Container = std::atomic>
+class MirroredVariable {
+ template <typename C>
+ struct Constraints {
+ // If setMirror is used with a different type U != T passed in,
+ // as a general rule, the Container must issue a memcpy to read or write
+ // (or its equivalent) to avoid possible strict aliasing issues.
+ // The memcpy also avoids gaps in structs and alignment issues with different types.
+ static constexpr bool ok_ = false; // Containers must specify constraints.
+ };
+ template <typename X>
+ struct Constraints<std::atomic<X>> {
+ // Atomics force read and write to memory.
+ static constexpr bool ok = std::is_same_v<X, T> ||
+ (std::atomic<X>::is_always_lock_free // no additional locking
+ && sizeof(std::atomic<X>) == sizeof(X) // layout identical to X.
+ && (std::is_arithmetic_v<X> || std::is_enum_v<X>)); // No gaps in the layout.
+ };
+
+static_assert(Constraints<Container<T>>::ok);
+public:
+ explicit MirroredVariable(const T& t) : t_{t} {}
+
+ // implicit conversion operator
+ operator T() const {
+ return t_;
+ }
+
+ MirroredVariable& operator=(const T& t) {
+ t_ = t;
+ if (mirror_ != nullptr) {
+ *mirror_ = t;
+ }
+ return *this;
+ }
+
+ template <typename U>
+ void setMirror(Container<U> *other_mirror) {
+ // Much of the concern is with T != U, however there are additional concerns
+ // when storage uses shared memory between processes. For atomics, it must be
+ // lock free.
+ static_assert(sizeof(U) == sizeof(T));
+ static_assert(alignof(U) == alignof(T));
+ static_assert(Constraints<Container<U>>::ok);
+ static_assert(sizeof(Container<U>) == sizeof(Container<T>));
+ static_assert(alignof(Container<U>) == alignof(Container<T>));
+ auto mirror = reinterpret_cast<Container<T>*>(other_mirror);
+ if (mirror_ != mirror) {
+ mirror_ = mirror;
+ if (mirror != nullptr) {
+ *mirror = t_;
+ }
+ }
+ }
+
+ void clear() {
+ mirror_ = nullptr;
+ }
+
+ MirroredVariable& operator&() const = delete;
+
+protected:
+ T t_{};
+ Container<T>* mirror_ = nullptr;
+};
+
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
// in continuously incrementing frame units, take modulo buffer size, which must be a power of 2
@@ -188,6 +265,8 @@
volatile int32_t mFlags; // combinations of CBLK_*
+ std::atomic<int32_t> mState; // current TrackBase state.
+
public:
union {
AudioTrackSharedStreaming mStreaming;
@@ -198,6 +277,9 @@
// Cache line boundary (32 bytes)
};
+// TODO: ensure standard layout.
+// static_assert(std::is_standard_layout_v<audio_track_cblk_t>);
+
// ----------------------------------------------------------------------------
// Proxy for shared memory control block, to isolate callers from needing to know the details.
@@ -323,6 +405,7 @@
return mEpoch;
}
+ int32_t getState() const { return mCblk->mState; }
uint32_t getBufferSizeInFrames() const { return mBufferSizeInFrames; }
// See documentation for AudioTrack::setBufferSizeInFrames()
uint32_t setBufferSizeInFrames(uint32_t requestedSize);
diff --git a/media/audioserver/Android.bp b/media/audioserver/Android.bp
index be25ffb..e4fc4de 100644
--- a/media/audioserver/Android.bp
+++ b/media/audioserver/Android.bp
@@ -26,6 +26,7 @@
shared_libs: [
"libaaudioservice",
+ "libaudioclient",
"libaudioflinger",
"libaudiopolicyservice",
"libaudioprocessing",
@@ -41,7 +42,6 @@
"libpowermanager",
"libutils",
"libvibrator",
-
],
// TODO check if we still need all of these include directories
diff --git a/media/audioserver/main_audioserver.cpp b/media/audioserver/main_audioserver.cpp
index 8ee1efb..726a6be 100644
--- a/media/audioserver/main_audioserver.cpp
+++ b/media/audioserver/main_audioserver.cpp
@@ -17,11 +17,17 @@
#define LOG_TAG "audioserver"
//#define LOG_NDEBUG 0
+#include <algorithm>
+
#include <fcntl.h>
#include <sys/prctl.h>
#include <sys/wait.h>
#include <cutils/properties.h>
+#include <android/media/audio/common/AudioMMapPolicy.h>
+#include <android/media/audio/common/AudioMMapPolicyInfo.h>
+#include <android/media/audio/common/AudioMMapPolicyType.h>
+#include <android/media/IAudioFlingerService.h>
#include <binder/IPCThreadState.h>
#include <binder/ProcessState.h>
#include <binder/IServiceManager.h>
@@ -30,7 +36,6 @@
#include <utils/Log.h>
// from include_dirs
-#include "aaudio/AAudioTesting.h" // aaudio_policy_t, AAUDIO_PROP_MMAP_POLICY, AAUDIO_POLICY_*
#include "AudioFlinger.h"
#include "AudioPolicyService.h"
#include "AAudioService.h"
@@ -39,6 +44,10 @@
using namespace android;
+using android::media::audio::common::AudioMMapPolicy;
+using android::media::audio::common::AudioMMapPolicyInfo;
+using android::media::audio::common::AudioMMapPolicyType;
+
int main(int argc __unused, char **argv)
{
// TODO: update with refined parameters
@@ -146,10 +155,24 @@
// AAudioService should only be used in OC-MR1 and later.
// And only enable the AAudioService if the system MMAP policy explicitly allows it.
// This prevents a client from misusing AAudioService when it is not supported.
- aaudio_policy_t mmapPolicy = property_get_int32(AAUDIO_PROP_MMAP_POLICY,
- AAUDIO_POLICY_NEVER);
- if (mmapPolicy == AAUDIO_POLICY_AUTO || mmapPolicy == AAUDIO_POLICY_ALWAYS) {
+ // If we cannot get audio flinger here, there must be some serious problems. In that case,
+ // attempting to call audio flinger on a null pointer could make the process crash
+ // and attract attentions.
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ std::vector<AudioMMapPolicyInfo> policyInfos;
+ status_t status = af->getMmapPolicyInfos(
+ AudioMMapPolicyType::DEFAULT, &policyInfos);
+ // Initialize aaudio service when querying mmap policy succeeds and
+ // any of the policy supports MMAP.
+ if (status == NO_ERROR &&
+ std::any_of(policyInfos.begin(), policyInfos.end(), [](const auto& info) {
+ return info.mmapPolicy == AudioMMapPolicy::AUTO ||
+ info.mmapPolicy == AudioMMapPolicy::ALWAYS;
+ })) {
AAudioService::instantiate();
+ } else {
+ ALOGD("Do not init aaudio service, status %d, policy info size %zu",
+ status, policyInfos.size());
}
ProcessState::self()->startThreadPool();
diff --git a/media/codec2/components/aac/C2SoftAacDec.cpp b/media/codec2/components/aac/C2SoftAacDec.cpp
index 342d771..57cdcd0 100644
--- a/media/codec2/components/aac/C2SoftAacDec.cpp
+++ b/media/codec2/components/aac/C2SoftAacDec.cpp
@@ -287,6 +287,7 @@
mOutputDelayRingBufferWritePos = 0;
mOutputDelayRingBufferReadPos = 0;
mOutputDelayRingBufferFilled = 0;
+ mOutputDelayRingBuffer.reset();
mBuffersInfo.clear();
status_t status = UNKNOWN_ERROR;
@@ -308,10 +309,7 @@
aacDecoder_Close(mAACDecoder);
mAACDecoder = nullptr;
}
- if (mOutputDelayRingBuffer) {
- delete[] mOutputDelayRingBuffer;
- mOutputDelayRingBuffer = nullptr;
- }
+ mOutputDelayRingBuffer.reset();
}
status_t C2SoftAacDec::initDecoder() {
@@ -327,7 +325,7 @@
mOutputDelayCompensated = 0;
mOutputDelayRingBufferSize = 2048 * MAX_CHANNEL_COUNT * kNumDelayBlocksMax;
- mOutputDelayRingBuffer = new short[mOutputDelayRingBufferSize];
+ mOutputDelayRingBuffer.reset(new short[mOutputDelayRingBufferSize]);
mOutputDelayRingBufferWritePos = 0;
mOutputDelayRingBufferReadPos = 0;
mOutputDelayRingBufferFilled = 0;
diff --git a/media/codec2/components/aac/C2SoftAacDec.h b/media/codec2/components/aac/C2SoftAacDec.h
index 986187c..a03fc70 100644
--- a/media/codec2/components/aac/C2SoftAacDec.h
+++ b/media/codec2/components/aac/C2SoftAacDec.h
@@ -93,7 +93,7 @@
bool mEndOfOutput;
int32_t mOutputDelayCompensated;
int32_t mOutputDelayRingBufferSize;
- short *mOutputDelayRingBuffer;
+ std::unique_ptr<short[]> mOutputDelayRingBuffer;
int32_t mOutputDelayRingBufferWritePos;
int32_t mOutputDelayRingBufferReadPos;
int32_t mOutputDelayRingBufferFilled;
diff --git a/media/codec2/components/mp3/C2SoftMp3Dec.cpp b/media/codec2/components/mp3/C2SoftMp3Dec.cpp
index 50690af..149c6ee 100644
--- a/media/codec2/components/mp3/C2SoftMp3Dec.cpp
+++ b/media/codec2/components/mp3/C2SoftMp3Dec.cpp
@@ -413,7 +413,7 @@
mConfig->inputBufferCurrentLength = (inSize - inPos);
mConfig->inputBufferMaxLength = 0;
mConfig->inputBufferUsedLength = 0;
- mConfig->outputFrameSize = (calOutSize - outSize);
+ mConfig->outputFrameSize = (calOutSize - outSize) / sizeof(int16_t);
mConfig->pOutputBuffer = reinterpret_cast<int16_t *> (wView.data() + outSize);
ERROR_CODE decoderErr;
diff --git a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
index 60d6b44..81f4679 100644
--- a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
+++ b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
@@ -228,7 +228,6 @@
const std::shared_ptr<IntfImpl> &intfImpl)
: SimpleC2Component(std::make_shared<SimpleInterface<IntfImpl>>(name, id, intfImpl)),
mIntf(intfImpl),
- mDecHandle(nullptr),
mOutputBuffer{},
mInitialized(false) {
}
@@ -243,13 +242,9 @@
}
c2_status_t C2SoftMpeg4Dec::onStop() {
- if (mDecHandle) {
- if (mInitialized) {
- PVCleanUpVideoDecoder(mDecHandle);
- mInitialized = false;
- }
- delete mDecHandle;
- mDecHandle = nullptr;
+ if (mInitialized) {
+ PVCleanUpVideoDecoder(&mVideoDecControls);
+ mInitialized = false;
}
for (int32_t i = 0; i < kNumOutputBuffers; ++i) {
if (mOutputBuffer[i]) {
@@ -279,7 +274,7 @@
c2_status_t C2SoftMpeg4Dec::onFlush_sm() {
if (mInitialized) {
- if (PV_TRUE != PVResetVideoDecoder(mDecHandle)) {
+ if (PV_TRUE != PVResetVideoDecoder(&mVideoDecControls)) {
return C2_CORRUPTED;
}
}
@@ -294,14 +289,8 @@
#else
mIsMpeg4 = false;
#endif
- if (!mDecHandle) {
- mDecHandle = new tagvideoDecControls;
- }
- if (!mDecHandle) {
- ALOGE("mDecHandle is null");
- return NO_MEMORY;
- }
- memset(mDecHandle, 0, sizeof(tagvideoDecControls));
+
+ memset(&mVideoDecControls, 0, sizeof(tagvideoDecControls));
/* TODO: bring these values to 352 and 288. It cannot be done as of now
* because, h263 doesn't seem to allow port reconfiguration. In OMX, the
@@ -357,10 +346,6 @@
}
c2_status_t C2SoftMpeg4Dec::ensureDecoderState(const std::shared_ptr<C2BlockPool> &pool) {
- if (!mDecHandle) {
- ALOGE("not supposed to be here, invalid decoder context");
- return C2_CORRUPTED;
- }
mOutputBufferSize = align(mIntf->getMaxWidth(), 16) * align(mIntf->getMaxHeight(), 16) * 3 / 2;
for (int32_t i = 0; i < kNumOutputBuffers; ++i) {
@@ -391,10 +376,10 @@
bool C2SoftMpeg4Dec::handleResChange(const std::unique_ptr<C2Work> &work) {
uint32_t disp_width, disp_height;
- PVGetVideoDimensions(mDecHandle, (int32 *)&disp_width, (int32 *)&disp_height);
+ PVGetVideoDimensions(&mVideoDecControls, (int32 *)&disp_width, (int32 *)&disp_height);
uint32_t buf_width, buf_height;
- PVGetBufferDimensions(mDecHandle, (int32 *)&buf_width, (int32 *)&buf_height);
+ PVGetBufferDimensions(&mVideoDecControls, (int32 *)&buf_width, (int32 *)&buf_height);
CHECK_LE(disp_width, buf_width);
CHECK_LE(disp_height, buf_height);
@@ -415,13 +400,14 @@
}
if (!mIsMpeg4) {
- PVCleanUpVideoDecoder(mDecHandle);
+ PVCleanUpVideoDecoder(&mVideoDecControls);
uint8_t *vol_data[1]{};
int32_t vol_size = 0;
if (!PVInitVideoDecoder(
- mDecHandle, vol_data, &vol_size, 1, mIntf->getMaxWidth(), mIntf->getMaxHeight(), H263_MODE)) {
+ &mVideoDecControls, vol_data, &vol_size, 1, mIntf->getMaxWidth(),
+ mIntf->getMaxHeight(), H263_MODE)) {
ALOGE("Error in PVInitVideoDecoder H263_MODE while resChanged was set to true");
mSignalledError = true;
work->result = C2_CORRUPTED;
@@ -511,7 +497,7 @@
uint32_t *start_code = (uint32_t *)bitstream;
bool volHeader = *start_code == 0xB0010000;
if (volHeader) {
- PVCleanUpVideoDecoder(mDecHandle);
+ PVCleanUpVideoDecoder(&mVideoDecControls);
mInitialized = false;
}
@@ -526,7 +512,7 @@
}
MP4DecodingMode mode = (mIsMpeg4) ? MPEG4_MODE : H263_MODE;
if (!PVInitVideoDecoder(
- mDecHandle, vol_data, &vol_size, 1,
+ &mVideoDecControls, vol_data, &vol_size, 1,
mIntf->getMaxWidth(), mIntf->getMaxHeight(), mode)) {
ALOGE("PVInitVideoDecoder failed. Unsupported content?");
mSignalledError = true;
@@ -534,7 +520,7 @@
return;
}
mInitialized = true;
- MP4DecodingMode actualMode = PVGetDecBitstreamMode(mDecHandle);
+ MP4DecodingMode actualMode = PVGetDecBitstreamMode(&mVideoDecControls);
if (mode != actualMode) {
ALOGE("Decoded mode not same as actual mode of the decoder");
mSignalledError = true;
@@ -542,7 +528,7 @@
return;
}
- PVSetPostProcType(mDecHandle, 0);
+ PVSetPostProcType(&mVideoDecControls, 0);
if (handleResChange(work)) {
ALOGI("Setting width and height");
C2StreamPictureSizeInfo::output size(0u, mWidth, mHeight);
@@ -579,7 +565,7 @@
return;
}
- uint32_t yFrameSize = sizeof(uint8) * mDecHandle->size;
+ uint32_t yFrameSize = sizeof(uint8) * mVideoDecControls.size;
if (mOutputBufferSize < yFrameSize * 3 / 2){
ALOGE("Too small output buffer: %zu bytes", mOutputBufferSize);
mSignalledError = true;
@@ -588,7 +574,7 @@
}
if (!mFramesConfigured) {
- PVSetReferenceYUV(mDecHandle,mOutputBuffer[1]);
+ PVSetReferenceYUV(&mVideoDecControls,mOutputBuffer[1]);
mFramesConfigured = true;
}
@@ -599,7 +585,7 @@
uint8_t *bitstreamTmp = bitstream;
uint32_t timestamp = workIndex;
if (PVDecodeVopHeader(
- mDecHandle, &bitstreamTmp, ×tamp, &tmpInSize,
+ &mVideoDecControls, &bitstreamTmp, ×tamp, &tmpInSize,
&header_info, &useExtTimestamp,
mOutputBuffer[mNumSamplesOutput & 1]) != PV_TRUE) {
ALOGE("failed to decode vop header.");
@@ -631,7 +617,7 @@
continue;
}
- if (PVDecodeVopBody(mDecHandle, &tmpInSize) != PV_TRUE) {
+ if (PVDecodeVopBody(&mVideoDecControls, &tmpInSize) != PV_TRUE) {
ALOGE("failed to decode video frame.");
mSignalledError = true;
work->result = C2_CORRUPTED;
diff --git a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.h b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.h
index 716a095..fed04c9 100644
--- a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.h
+++ b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.h
@@ -19,8 +19,8 @@
#include <SimpleC2Component.h>
+#include <mp4dec_api.h>
-struct tagvideoDecControls;
namespace android {
@@ -54,7 +54,7 @@
bool handleResChange(const std::unique_ptr<C2Work> &work);
std::shared_ptr<IntfImpl> mIntf;
- tagvideoDecControls *mDecHandle;
+ tagvideoDecControls mVideoDecControls;
std::shared_ptr<C2GraphicBlock> mOutBlock;
uint8_t *mOutputBuffer[kNumOutputBuffers];
size_t mOutputBufferSize;
diff --git a/media/codec2/components/vpx/C2SoftVpxDec.cpp b/media/codec2/components/vpx/C2SoftVpxDec.cpp
index 2953d90..45e2ca8 100644
--- a/media/codec2/components/vpx/C2SoftVpxDec.cpp
+++ b/media/codec2/components/vpx/C2SoftVpxDec.cpp
@@ -149,8 +149,16 @@
#else
addParameter(
DefineParam(mProfileLevel, C2_PARAMKEY_PROFILE_LEVEL)
- .withConstValue(new C2StreamProfileLevelInfo::input(0u,
- C2Config::PROFILE_UNUSED, C2Config::LEVEL_UNUSED))
+ .withDefault(new C2StreamProfileLevelInfo::input(0u,
+ C2Config::PROFILE_VP8_0, C2Config::LEVEL_UNUSED))
+ .withFields({
+ C2F(mProfileLevel, profile).equalTo(
+ PROFILE_VP8_0
+ ),
+ C2F(mProfileLevel, level).equalTo(
+ LEVEL_UNUSED),
+ })
+ .withSetter(ProfileLevelSetter, mSize)
.build());
#endif
diff --git a/media/codec2/components/vpx/C2SoftVpxEnc.h b/media/codec2/components/vpx/C2SoftVpxEnc.h
index c98b802..926b2e9 100644
--- a/media/codec2/components/vpx/C2SoftVpxEnc.h
+++ b/media/codec2/components/vpx/C2SoftVpxEnc.h
@@ -324,21 +324,35 @@
.withConstValue(new C2StreamIntraRefreshTuning::output(
0u, C2Config::INTRA_REFRESH_DISABLED, 0.))
.build());
-
+#ifdef VP9
addParameter(
- DefineParam(mProfileLevel, C2_PARAMKEY_PROFILE_LEVEL)
- .withDefault(new C2StreamProfileLevelInfo::output(
- 0u, PROFILE_VP9_0, LEVEL_VP9_4_1))
- .withFields({
- C2F(mProfileLevel, profile).equalTo(
- PROFILE_VP9_0
- ),
- C2F(mProfileLevel, level).equalTo(
- LEVEL_VP9_4_1),
- })
- .withSetter(ProfileLevelSetter)
- .build());
-
+ DefineParam(mProfileLevel, C2_PARAMKEY_PROFILE_LEVEL)
+ .withDefault(new C2StreamProfileLevelInfo::output(
+ 0u, PROFILE_VP9_0, LEVEL_VP9_4_1))
+ .withFields({
+ C2F(mProfileLevel, profile).equalTo(
+ PROFILE_VP9_0
+ ),
+ C2F(mProfileLevel, level).equalTo(
+ LEVEL_VP9_4_1),
+ })
+ .withSetter(ProfileLevelSetter)
+ .build());
+#else
+ addParameter(
+ DefineParam(mProfileLevel, C2_PARAMKEY_PROFILE_LEVEL)
+ .withDefault(new C2StreamProfileLevelInfo::output(
+ 0u, PROFILE_VP8_0, LEVEL_UNUSED))
+ .withFields({
+ C2F(mProfileLevel, profile).equalTo(
+ PROFILE_VP8_0
+ ),
+ C2F(mProfileLevel, level).equalTo(
+ LEVEL_UNUSED),
+ })
+ .withSetter(ProfileLevelSetter)
+ .build());
+#endif
addParameter(
DefineParam(mRequestSync, C2_PARAMKEY_REQUEST_SYNC_FRAME)
.withDefault(new C2StreamRequestSyncFrameTuning::output(0u, C2_FALSE))
diff --git a/media/codec2/fuzzer/C2Fuzzer.cpp b/media/codec2/fuzzer/C2Fuzzer.cpp
index e35ee48..e469d8b 100644
--- a/media/codec2/fuzzer/C2Fuzzer.cpp
+++ b/media/codec2/fuzzer/C2Fuzzer.cpp
@@ -239,17 +239,17 @@
}
void Codec2Fuzzer::decodeFrames(const uint8_t* data, size_t size) {
- mBufferSource = new BufferSource(data, size);
- if (!mBufferSource) {
+ std::unique_ptr<BufferSource> bufferSource = std::make_unique<BufferSource>(data, size);
+ if (!bufferSource) {
return;
}
- mBufferSource->parse();
+ bufferSource->parse();
c2_status_t status = C2_OK;
size_t numFrames = 0;
- while (!mBufferSource->isEos()) {
+ while (!bufferSource->isEos()) {
uint8_t* frame = nullptr;
size_t frameSize = 0;
- FrameData frameData = mBufferSource->getFrame();
+ FrameData frameData = bufferSource->getFrame();
frame = std::get<0>(frameData);
frameSize = std::get<1>(frameData);
@@ -298,7 +298,6 @@
mConditionalVariable.wait_for(waitForDecodeComplete, kC2FuzzerTimeOut, [this] { return mEos; });
std::list<std::unique_ptr<C2Work>> c2flushedWorks;
mComponent->flush_sm(C2Component::FLUSH_COMPONENT, &c2flushedWorks);
- delete mBufferSource;
}
extern "C" int LLVMFuzzerTestOneInput(const uint8_t* data, size_t size) {
diff --git a/media/codec2/fuzzer/C2Fuzzer.h b/media/codec2/fuzzer/C2Fuzzer.h
index d5ac81a..da76885 100644
--- a/media/codec2/fuzzer/C2Fuzzer.h
+++ b/media/codec2/fuzzer/C2Fuzzer.h
@@ -104,7 +104,6 @@
static constexpr size_t kMarkerSuffixSize = 3;
};
- BufferSource* mBufferSource;
bool mEos = false;
C2BlockPool::local_id_t mBlockPoolId;
diff --git a/media/codec2/hidl/1.0/vts/.clang-format b/media/codec2/hidl/1.0/vts/.clang-format
deleted file mode 120000
index 136279c..0000000
--- a/media/codec2/hidl/1.0/vts/.clang-format
+++ /dev/null
@@ -1 +0,0 @@
-../../../../../../../build/soong/scripts/system-clang-format
\ No newline at end of file
diff --git a/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioDecTest.cpp b/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioDecTest.cpp
index 74b099c..d47ef67 100644
--- a/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioDecTest.cpp
+++ b/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioDecTest.cpp
@@ -100,8 +100,10 @@
ASSERT_NE(mLinearPool, nullptr);
std::vector<std::unique_ptr<C2Param>> queried;
- mComponent->query({}, {C2PortMediaTypeSetting::input::PARAM_TYPE}, C2_DONT_BLOCK, &queried);
- ASSERT_GT(queried.size(), 0);
+ c2_status_t c2err = mComponent->query({}, {C2PortMediaTypeSetting::input::PARAM_TYPE},
+ C2_DONT_BLOCK, &queried);
+ ASSERT_EQ(c2err, C2_OK) << "Query media type failed";
+ ASSERT_EQ(queried.size(), 1) << "Size of the vector returned is invalid";
mMime = ((C2PortMediaTypeSetting::input*)queried[0].get())->m.value;
@@ -277,24 +279,20 @@
};
std::vector<std::unique_ptr<C2Param>> inParams;
c2_status_t status = component->query({}, indices, C2_DONT_BLOCK, &inParams);
- if (status != C2_OK && inParams.size() == 0) {
- ALOGE("Query media type failed => %d", status);
- ASSERT_TRUE(false);
- } else {
- size_t offset = sizeof(C2Param);
- for (size_t i = 0; i < inParams.size(); ++i) {
- C2Param* param = inParams[i].get();
- bitStreamInfo[i] = *(int32_t*)((uint8_t*)param + offset);
- }
- if (mime.find("3gpp") != std::string::npos) {
- ASSERT_EQ(bitStreamInfo[0], 8000);
- ASSERT_EQ(bitStreamInfo[1], 1);
- } else if (mime.find("amr-wb") != std::string::npos) {
- ASSERT_EQ(bitStreamInfo[0], 16000);
- ASSERT_EQ(bitStreamInfo[1], 1);
- } else if (mime.find("gsm") != std::string::npos) {
- ASSERT_EQ(bitStreamInfo[0], 8000);
- }
+ ASSERT_EQ(status, C2_OK) << "Query sample rate and channel count info failed";
+ ASSERT_EQ(inParams.size(), indices.size()) << "Size of the vector returned is invalid";
+
+ bitStreamInfo[0] = C2StreamSampleRateInfo::output::From(inParams[0].get())->value;
+ bitStreamInfo[1] = C2StreamChannelCountInfo::output::From(inParams[1].get())->value;
+ if (mime.find("3gpp") != std::string::npos) {
+ ASSERT_EQ(bitStreamInfo[0], 8000);
+ ASSERT_EQ(bitStreamInfo[1], 1);
+ } else if (mime.find("amr-wb") != std::string::npos) {
+ ASSERT_EQ(bitStreamInfo[0], 16000);
+ ASSERT_EQ(bitStreamInfo[1], 1);
+ } else if (mime.find("gsm") != std::string::npos) {
+ ASSERT_EQ(bitStreamInfo[0], 8000);
+ ASSERT_EQ(bitStreamInfo[1], 1);
}
}
@@ -467,6 +465,11 @@
if (mMime.find("raw") != std::string::npos) {
bitStreamInfo[0] = 8000;
bitStreamInfo[1] = 1;
+ } else if ((mMime.find("g711-alaw") != std::string::npos) ||
+ (mMime.find("g711-mlaw") != std::string::npos)) {
+ // g711 test data is all 1-channel and has no embedded config info.
+ bitStreamInfo[0] = 8000;
+ bitStreamInfo[1] = 1;
} else {
ASSERT_NO_FATAL_FAILURE(getInputChannelInfo(mComponent, mMime, bitStreamInfo));
}
diff --git a/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioEncTest.cpp b/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioEncTest.cpp
index c487fa3..bd7ec0d 100644
--- a/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioEncTest.cpp
+++ b/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioEncTest.cpp
@@ -73,26 +73,22 @@
ASSERT_NE(mLinearPool, nullptr);
std::vector<std::unique_ptr<C2Param>> queried;
- mComponent->query({}, {C2PortMediaTypeSetting::output::PARAM_TYPE}, C2_DONT_BLOCK,
- &queried);
- ASSERT_GT(queried.size(), 0);
+ c2_status_t c2err = mComponent->query({}, {C2PortMediaTypeSetting::output::PARAM_TYPE},
+ C2_DONT_BLOCK, &queried);
+ ASSERT_EQ(c2err, C2_OK) << "Query media type failed";
+ ASSERT_EQ(queried.size(), 1) << "Size of the vector returned is invalid";
mMime = ((C2PortMediaTypeSetting::output*)queried[0].get())->m.value;
mEos = false;
mCsd = false;
mFramesReceived = 0;
+ mEncoderFrameSize = 0;
mWorkResult = C2_OK;
mOutputSize = 0u;
- getInputMaxBufSize();
-
- c2_status_t status = getChannelCount(&mNumChannels);
- ASSERT_EQ(status, C2_OK) << "Unable to get supported channel count";
-
- status = getSampleRate(&mSampleRate);
- ASSERT_EQ(status, C2_OK) << "Unable to get supported sample rate";
-
- status = getSamplesPerFrame(mNumChannels, &mSamplesPerFrame);
- ASSERT_EQ(status, C2_OK) << "Unable to get supported number of samples per frame";
+ ASSERT_NO_FATAL_FAILURE(getInputMaxBufSize());
+ ASSERT_NO_FATAL_FAILURE(getChannelCount(&mNumChannels));
+ ASSERT_NO_FATAL_FAILURE(getSampleRate(&mSampleRate));
+ ASSERT_NO_FATAL_FAILURE(getSamplesPerFrame(mNumChannels, &mSamplesPerFrame));
getFile(mNumChannels, mSampleRate);
}
@@ -108,9 +104,9 @@
// Get the test parameters from GetParam call.
virtual void getParams() {}
- c2_status_t getChannelCount(int32_t* nChannels);
- c2_status_t getSampleRate(int32_t* nSampleRate);
- c2_status_t getSamplesPerFrame(int32_t nChannels, int32_t* samplesPerFrame);
+ void getChannelCount(int32_t* nChannels);
+ void getSampleRate(int32_t* nSampleRate);
+ void getSamplesPerFrame(int32_t nChannels, int32_t* samplesPerFrame);
void getFile(int32_t channelCount, int32_t sampleRate);
@@ -146,6 +142,7 @@
uint32_t mFramesReceived;
int32_t mInputMaxBufSize;
uint64_t mOutputSize;
+ uint32_t mEncoderFrameSize;
std::list<uint64_t> mFlushedIndices;
C2BlockPool::local_id_t mBlockPoolId;
@@ -173,21 +170,13 @@
// In encoder components, fetch the size of input buffer allocated
void getInputMaxBufSize() {
- int32_t bitStreamInfo[1] = {0};
std::vector<std::unique_ptr<C2Param>> inParams;
c2_status_t status = mComponent->query({}, {C2StreamMaxBufferSizeInfo::input::PARAM_TYPE},
C2_DONT_BLOCK, &inParams);
- if (status != C2_OK && inParams.size() == 0) {
- ALOGE("Query MaxBufferSizeInfo failed => %d", status);
- ASSERT_TRUE(false);
- } else {
- size_t offset = sizeof(C2Param);
- for (size_t i = 0; i < inParams.size(); ++i) {
- C2Param* param = inParams[i].get();
- bitStreamInfo[i] = *(int32_t*)((uint8_t*)param + offset);
- }
- }
- mInputMaxBufSize = bitStreamInfo[0];
+ ASSERT_EQ(status, C2_OK) << "Query max buffer size info failed";
+ ASSERT_EQ(inParams.size(), 1) << "Size of the vector returned is invalid";
+
+ mInputMaxBufSize = C2StreamMaxBufferSizeInfo::input::From(inParams[0].get())->value;
}
};
@@ -243,17 +232,15 @@
return false;
}
-c2_status_t Codec2AudioEncHidlTestBase::getChannelCount(int32_t* nChannels) {
+void Codec2AudioEncHidlTestBase::getChannelCount(int32_t* nChannels) {
std::unique_ptr<C2StreamChannelCountInfo::input> channelCount =
std::make_unique<C2StreamChannelCountInfo::input>();
std::vector<C2FieldSupportedValuesQuery> validValueInfos = {
C2FieldSupportedValuesQuery::Current(
C2ParamField(channelCount.get(), &C2StreamChannelCountInfo::value))};
c2_status_t c2err = mComponent->querySupportedValues(validValueInfos, C2_DONT_BLOCK);
- if (c2err != C2_OK || validValueInfos.size() != 1u) {
- ALOGE("querySupportedValues_vb failed for channelCount");
- return c2err;
- }
+ ASSERT_EQ(c2err, C2_OK) << "Query channel count info failed";
+ ASSERT_EQ(validValueInfos.size(), 1) << "Size of the vector returned is invalid";
// setting default value of channelCount
*nChannels = 1;
@@ -280,47 +267,45 @@
break;
}
default:
+ ASSERT_TRUE(false) << "Unsupported type: " << c2FSV.type;
break;
}
- return C2_OK;
+ return;
}
-c2_status_t Codec2AudioEncHidlTestBase::getSampleRate(int32_t* nSampleRate) {
+void Codec2AudioEncHidlTestBase::getSampleRate(int32_t* nSampleRate) {
// Use the default sample rate for mComponents
std::vector<std::unique_ptr<C2Param>> queried;
c2_status_t c2err = mComponent->query({}, {C2StreamSampleRateInfo::input::PARAM_TYPE},
C2_DONT_BLOCK, &queried);
- if (c2err != C2_OK || queried.size() == 0) return c2err;
+ ASSERT_EQ(c2err, C2_OK) << "Query sample rate info failed";
+ ASSERT_EQ(queried.size(), 1) << "Size of the vector returned is invalid";
- size_t offset = sizeof(C2Param);
- C2Param* param = queried[0].get();
- *nSampleRate = *(int32_t*)((uint8_t*)param + offset);
-
- return C2_OK;
+ *nSampleRate = C2StreamSampleRateInfo::input::From(queried[0].get())->value;
+ return;
}
-c2_status_t Codec2AudioEncHidlTestBase::getSamplesPerFrame(int32_t nChannels,
- int32_t* samplesPerFrame) {
+void Codec2AudioEncHidlTestBase::getSamplesPerFrame(int32_t nChannels, int32_t* samplesPerFrame) {
std::vector<std::unique_ptr<C2Param>> queried;
c2_status_t c2err = mComponent->query({}, {C2StreamAudioFrameSizeInfo::input::PARAM_TYPE},
C2_DONT_BLOCK, &queried);
- size_t offset = sizeof(C2Param);
- uint32_t maxInputSize = 0;
- if (c2err == C2_OK && queried.size()) {
- C2Param* param = queried[0].get();
- maxInputSize = *(uint32_t*)((uint8_t*)param + offset);
+
+ if (c2err == C2_OK && queried.size() == 1) {
+ mEncoderFrameSize = C2StreamAudioFrameSizeInfo::input::From(queried[0].get())->value;
+ if (mEncoderFrameSize) {
+ *samplesPerFrame = mEncoderFrameSize;
+ return;
+ }
}
- if (0 == maxInputSize) {
- c2err = mComponent->query({}, {C2StreamMaxBufferSizeInfo::input::PARAM_TYPE}, C2_DONT_BLOCK,
- &queried);
- if (c2err != C2_OK || queried.size() == 0) return c2err;
+ c2err = mComponent->query({}, {C2StreamMaxBufferSizeInfo::input::PARAM_TYPE}, C2_DONT_BLOCK,
+ &queried);
+ ASSERT_EQ(c2err, C2_OK) << "Query max buffer size info failed";
+ ASSERT_EQ(queried.size(), 1) << "Size of the vector returned is invalid";
- C2Param* param = queried[0].get();
- maxInputSize = *(uint32_t*)((uint8_t*)param + offset);
- }
+ uint32_t maxInputSize = C2StreamMaxBufferSizeInfo::input::From(queried[0].get())->value;
*samplesPerFrame = std::min((maxInputSize / (nChannels * 2)), kMaxSamplesPerFrame);
- return C2_OK;
+ return;
}
// LookUpTable of clips and metadata for component testing
@@ -450,10 +435,13 @@
ALOGV("EncodeTest");
if (mDisableTest) GTEST_SKIP() << "Test is disabled";
bool signalEOS = std::get<2>(GetParam());
- // Ratio w.r.t to mInputMaxBufSize
- int32_t inputMaxBufRatio = std::get<3>(GetParam());
- mSamplesPerFrame = ((mInputMaxBufSize / inputMaxBufRatio) / (mNumChannels * 2));
-
+ // Set samples per frame based on inputMaxBufRatio if component does not
+ // advertise supported frame size
+ if (!mEncoderFrameSize) {
+ // Ratio w.r.t to mInputMaxBufSize
+ int32_t inputMaxBufRatio = std::get<3>(GetParam());
+ mSamplesPerFrame = ((mInputMaxBufSize / inputMaxBufRatio) / (mNumChannels * 2));
+ }
ALOGV("signalEOS %d mInputMaxBufSize %d mSamplesPerFrame %d", signalEOS, mInputMaxBufSize,
mSamplesPerFrame);
@@ -613,12 +601,11 @@
std::vector<std::unique_ptr<C2Param>> inParams;
c2_status_t c2_status = mComponent->query({}, {C2StreamChannelCountInfo::input::PARAM_TYPE},
C2_DONT_BLOCK, &inParams);
- ASSERT_TRUE(!c2_status && inParams.size())
- << "Query configured channelCount failed => %d" << c2_status;
+ ASSERT_EQ(c2_status, C2_OK) << "Query channel count info failed";
+ ASSERT_EQ(inParams.size(), 1) << "Size of the vector returned is invalid";
- size_t offset = sizeof(C2Param);
- C2Param* param = inParams[0].get();
- int32_t channelCount = *(int32_t*)((uint8_t*)param + offset);
+ int32_t channelCount = C2StreamChannelCountInfo::input::From(inParams[0].get())->value;
+
if (channelCount != nChannels) {
std::cout << "[ WARN ] Test Skipped for ChannelCount " << nChannels << "\n";
continue;
@@ -702,13 +689,11 @@
std::vector<std::unique_ptr<C2Param>> inParams;
c2_status_t c2_status = mComponent->query({}, {C2StreamSampleRateInfo::input::PARAM_TYPE},
C2_DONT_BLOCK, &inParams);
+ ASSERT_EQ(c2_status, C2_OK) << "Query sample rate info failed";
+ ASSERT_EQ(inParams.size(), 1) << "Size of the vector returned is invalid";
- ASSERT_TRUE(!c2_status && inParams.size())
- << "Query configured SampleRate failed => %d" << c2_status;
- size_t offset = sizeof(C2Param);
- C2Param* param = inParams[0].get();
- int32_t configuredSampleRate = *(int32_t*)((uint8_t*)param + offset);
-
+ int32_t configuredSampleRate =
+ C2StreamSampleRateInfo::input::From(inParams[0].get())->value;
if (configuredSampleRate != nSampleRate) {
std::cout << "[ WARN ] Test Skipped for SampleRate " << nSampleRate << "\n";
continue;
diff --git a/media/codec2/hidl/1.0/vts/functional/component/VtsHalMediaC2V1_0TargetComponentTest.cpp b/media/codec2/hidl/1.0/vts/functional/component/VtsHalMediaC2V1_0TargetComponentTest.cpp
index ffec897..275a721 100644
--- a/media/codec2/hidl/1.0/vts/functional/component/VtsHalMediaC2V1_0TargetComponentTest.cpp
+++ b/media/codec2/hidl/1.0/vts/functional/component/VtsHalMediaC2V1_0TargetComponentTest.cpp
@@ -61,6 +61,7 @@
public:
virtual void SetUp() override {
getParams();
+ mDisableTest = false;
mEos = false;
mClient = android::Codec2Client::CreateFromService(mInstanceName.c_str());
ASSERT_NE(mClient, nullptr);
@@ -73,6 +74,14 @@
for (int i = 0; i < MAX_INPUT_BUFFERS; ++i) {
mWorkQueue.emplace_back(new C2Work);
}
+
+ C2SecureModeTuning secureModeTuning{};
+ mComponent->query({&secureModeTuning}, {}, C2_MAY_BLOCK, nullptr);
+ if (secureModeTuning.value != C2Config::SM_UNPROTECTED) {
+ mDisableTest = true;
+ }
+
+ if (mDisableTest) std::cout << "[ WARN ] Test Disabled \n";
}
virtual void TearDown() override {
@@ -105,6 +114,7 @@
std::string mInstanceName;
std::string mComponentName;
bool mEos;
+ bool mDisableTest;
std::mutex mQueueLock;
std::condition_variable mQueueCondition;
std::list<std::unique_ptr<C2Work>> mWorkQueue;
@@ -324,6 +334,7 @@
};
TEST_P(Codec2ComponentInputTests, InputBufferTest) {
+ if (mDisableTest) GTEST_SKIP() << "Test is disabled";
description("Tests for different inputs");
uint32_t flags = std::get<2>(GetParam());
diff --git a/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoDecTest.cpp b/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoDecTest.cpp
index 95a4674..67873fa 100644
--- a/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoDecTest.cpp
+++ b/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoDecTest.cpp
@@ -123,8 +123,10 @@
ASSERT_NE(mLinearPool, nullptr);
std::vector<std::unique_ptr<C2Param>> queried;
- mComponent->query({}, {C2PortMediaTypeSetting::input::PARAM_TYPE}, C2_DONT_BLOCK, &queried);
- ASSERT_GT(queried.size(), 0);
+ c2_status_t c2err = mComponent->query({}, {C2PortMediaTypeSetting::input::PARAM_TYPE},
+ C2_DONT_BLOCK, &queried);
+ ASSERT_EQ(c2err, C2_OK) << "Query media type failed";
+ ASSERT_EQ(queried.size(), 1) << "Size of the vector returned is invalid";
mMime = ((C2PortMediaTypeSetting::input*)queried[0].get())->m.value;
mEos = false;
diff --git a/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoEncTest.cpp b/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoEncTest.cpp
index a6507e7..8305feb 100644
--- a/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoEncTest.cpp
+++ b/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoEncTest.cpp
@@ -74,9 +74,10 @@
ASSERT_NE(mGraphicPool, nullptr);
std::vector<std::unique_ptr<C2Param>> queried;
- mComponent->query({}, {C2PortMediaTypeSetting::output::PARAM_TYPE}, C2_DONT_BLOCK,
- &queried);
- ASSERT_GT(queried.size(), 0);
+ c2_status_t c2err = mComponent->query({}, {C2PortMediaTypeSetting::output::PARAM_TYPE},
+ C2_DONT_BLOCK, &queried);
+ ASSERT_EQ(c2err, C2_OK) << "Query media type failed";
+ ASSERT_EQ(queried.size(), 1) << "Size of the vector returned is invalid";
mMime = ((C2PortMediaTypeSetting::output*)queried[0].get())->m.value;
std::cout << "mime : " << mMime << "\n";
@@ -531,9 +532,17 @@
<< " resetting num BFrames to 0\n";
mConfigBPictures = false;
} else {
- size_t offset = sizeof(C2Param);
- C2Param* param = inParams[0].get();
- int32_t numBFrames = *(int32_t*)((uint8_t*)param + offset);
+ int32_t numBFrames = 0;
+ C2StreamGopTuning::output* gop = C2StreamGopTuning::output::From(inParams[0].get());
+ if (gop && gop->flexCount() >= 1) {
+ for (size_t i = 0; i < gop->flexCount(); ++i) {
+ const C2GopLayerStruct& layer = gop->m.values[i];
+ if (layer.type_ == C2Config::picture_type_t(P_FRAME | B_FRAME)) {
+ numBFrames = layer.count;
+ break;
+ }
+ }
+ }
if (!numBFrames) {
std::cout << "[ WARN ] Bframe not supported for " << mComponentName
@@ -817,6 +826,10 @@
TEST_P(Codec2VideoEncHidlTest, AdaptiveBitrateTest) {
description("Encodes input file for different bitrates");
if (mDisableTest) GTEST_SKIP() << "Test is disabled";
+ if (mMime != "video/avc" && mMime != "video/hevc" && mMime != "video/x-vnd.on2.vp8" &&
+ mMime != "video/x-vnd.on2.vp9") {
+ GTEST_SKIP() << "AdaptiveBitrateTest is enabled only for avc, hevc, vp8 and vp9 encoders";
+ }
std::ifstream eleStream;
eleStream.open(mInputFile, std::ifstream::binary);
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index 82460c9..44a2c5b 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -1332,8 +1332,8 @@
}
}
- // set channel-mask
if (config->mDomain & Config::IS_AUDIO) {
+ // set channel-mask
int32_t mask;
if (msg->findInt32(KEY_CHANNEL_MASK, &mask)) {
if (config->mDomain & Config::IS_ENCODER) {
@@ -1342,6 +1342,15 @@
config->mOutputFormat->setInt32(KEY_CHANNEL_MASK, mask);
}
}
+
+ // set PCM encoding
+ int32_t pcmEncoding = kAudioEncodingPcm16bit;
+ msg->findInt32(KEY_PCM_ENCODING, &pcmEncoding);
+ if (encoder) {
+ config->mInputFormat->setInt32("android._config-pcm-encoding", pcmEncoding);
+ } else {
+ config->mOutputFormat->setInt32("android._config-pcm-encoding", pcmEncoding);
+ }
}
std::unique_ptr<C2Param> colorTransferRequestParam;
@@ -1421,6 +1430,10 @@
}
}
+ if (config->mTunneled) {
+ config->mOutputFormat->setInt32("android._tunneled", 1);
+ }
+
ALOGD("setup formats input: %s",
config->mInputFormat->debugString().c_str());
ALOGD("setup formats output: %s",
@@ -1896,9 +1909,11 @@
{
Mutexed<std::unique_ptr<Config>>::Locked configLocked(mConfig);
const std::unique_ptr<Config> &config = *configLocked;
+ sp<ANativeWindow> nativeWindow = static_cast<ANativeWindow *>(surface.get());
+ status_t err = OK;
+
if (config->mTunneled && config->mSidebandHandle != nullptr) {
- sp<ANativeWindow> nativeWindow = static_cast<ANativeWindow *>(surface.get());
- status_t err = native_window_set_sideband_stream(
+ err = native_window_set_sideband_stream(
nativeWindow.get(),
const_cast<native_handle_t *>(config->mSidebandHandle->handle()));
if (err != OK) {
@@ -1906,6 +1921,15 @@
nativeWindow.get(), config->mSidebandHandle->handle(), err);
return err;
}
+ } else {
+ // Explicitly reset the sideband handle of the window for
+ // non-tunneled video in case the window was previously used
+ // for a tunneled video playback.
+ err = native_window_set_sideband_stream(nativeWindow.get(), nullptr);
+ if (err != OK) {
+ ALOGE("native_window_set_sideband_stream(nullptr) failed! (err %d).", err);
+ return err;
+ }
}
}
return mChannel->setSurface(surface);
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 8194bb8..23a326f 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -19,6 +19,7 @@
#include <utils/Log.h>
#include <algorithm>
+#include <atomic>
#include <list>
#include <numeric>
@@ -155,6 +156,7 @@
input->pipelineDelay = 0u;
input->numSlots = kSmoothnessFactor;
input->numExtraSlots = 0u;
+ input->lastFlushIndex = 0u;
}
{
Mutexed<Output>::Locked output(mOutput);
@@ -1122,6 +1124,7 @@
input->numSlots = numInputSlots;
input->extraBuffers.flush();
input->numExtraSlots = 0u;
+ input->lastFlushIndex = mFrameIndex.load(std::memory_order_relaxed);
if (audioEncoder && encoderFrameSize && sampleRate && channelCount) {
input->frameReassembler.init(
pool,
@@ -1385,6 +1388,12 @@
}
}
}
+
+ int32_t tunneled = 0;
+ if (!outputFormat->findInt32("android._tunneled", &tunneled)) {
+ tunneled = 0;
+ }
+ mTunneled = (tunneled != 0);
}
// Set up pipeline control. This has to be done after mInputBuffers and
@@ -1529,6 +1538,7 @@
ALOGV("[%s] flush", mName);
std::vector<uint64_t> indices;
std::list<std::unique_ptr<C2Work>> configs;
+ mInput.lock()->lastFlushIndex = mFrameIndex.load(std::memory_order_relaxed);
for (const std::unique_ptr<C2Work> &work : flushedWork) {
indices.push_back(work->input.ordinal.frameIndex.peeku());
if (!(work->input.flags & C2FrameData::FLAG_CODEC_CONFIG)) {
@@ -1595,12 +1605,18 @@
}
std::shared_ptr<C2Buffer> buffer =
mPipelineWatcher.lock()->onInputBufferReleased(frameIndex, arrayIndex);
- bool newInputSlotAvailable;
+ bool newInputSlotAvailable = false;
{
Mutexed<Input>::Locked input(mInput);
- newInputSlotAvailable = input->buffers->expireComponentBuffer(buffer);
- if (!newInputSlotAvailable) {
- (void)input->extraBuffers.expireComponentBuffer(buffer);
+ if (input->lastFlushIndex >= frameIndex) {
+ ALOGD("[%s] Ignoring stale input buffer done callback: "
+ "last flush index = %lld, frameIndex = %lld",
+ mName, input->lastFlushIndex.peekll(), (long long)frameIndex);
+ } else {
+ newInputSlotAvailable = input->buffers->expireComponentBuffer(buffer);
+ if (!newInputSlotAvailable) {
+ (void)input->extraBuffers.expireComponentBuffer(buffer);
+ }
}
}
if (newInputSlotAvailable) {
@@ -1887,10 +1903,21 @@
}
}
+ bool drop = false;
+ if (worklet->output.flags & C2FrameData::FLAG_DROP_FRAME) {
+ ALOGV("[%s] onWorkDone: drop buffer but keep metadata", mName);
+ drop = true;
+ }
+
if (notifyClient && !buffer && !flags) {
- ALOGV("[%s] onWorkDone: Not reporting output buffer (%lld)",
- mName, work->input.ordinal.frameIndex.peekull());
- notifyClient = false;
+ if (mTunneled && drop && outputFormat) {
+ ALOGV("[%s] onWorkDone: Keep tunneled, drop frame with format change (%lld)",
+ mName, work->input.ordinal.frameIndex.peekull());
+ } else {
+ ALOGV("[%s] onWorkDone: Not reporting output buffer (%lld)",
+ mName, work->input.ordinal.frameIndex.peekull());
+ notifyClient = false;
+ }
}
if (buffer) {
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index 5a2aca2..26eef30 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -273,6 +273,7 @@
size_t numExtraSlots;
uint32_t inputDelay;
uint32_t pipelineDelay;
+ c2_cntr64_t lastFlushIndex;
FrameReassembler frameReassembler;
};
@@ -323,6 +324,8 @@
return mCrypto != nullptr || mDescrambler != nullptr;
}
std::atomic_bool mSendEncryptedInfoBuffer;
+
+ std::atomic_bool mTunneled;
};
// Conversion of a c2_status_t value to a status_t value may depend on the
diff --git a/media/codec2/sfplugin/CCodecBuffers.cpp b/media/codec2/sfplugin/CCodecBuffers.cpp
index d94d0a2..20f2ecf 100644
--- a/media/codec2/sfplugin/CCodecBuffers.cpp
+++ b/media/codec2/sfplugin/CCodecBuffers.cpp
@@ -21,6 +21,7 @@
#include <C2PlatformSupport.h>
#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/MediaDefs.h>
#include <media/stagefright/MediaCodec.h>
#include <media/stagefright/MediaCodecConstants.h>
#include <media/stagefright/SkipCutBuffer.h>
@@ -100,7 +101,10 @@
if (img->mNumPlanes > 0 && img->mType != img->MEDIA_IMAGE_TYPE_UNKNOWN) {
int32_t stride = img->mPlane[0].mRowInc;
mFormatWithImageData->setInt32(KEY_STRIDE, stride);
- ALOGD("[%s] updating stride = %d", mName, stride);
+ mFormatWithImageData->setInt32(KEY_WIDTH, img->mWidth);
+ mFormatWithImageData->setInt32(KEY_HEIGHT, img->mHeight);
+ ALOGD("[%s] updating stride = %d, width: %d, height: %d",
+ mName, stride, img->mWidth, img->mHeight);
if (img->mNumPlanes > 1 && stride > 0) {
int64_t offsetDelta =
(int64_t)img->mPlane[1].mOffset - (int64_t)img->mPlane[0].mOffset;
@@ -201,6 +205,56 @@
mSkipCutBuffer = new SkipCutBuffer(skip, cut, mChannelCount);
}
+bool OutputBuffers::convert(
+ const std::shared_ptr<C2Buffer> &src, sp<Codec2Buffer> *dst) {
+ if (!src || src->data().type() != C2BufferData::LINEAR) {
+ return false;
+ }
+ int32_t configEncoding = kAudioEncodingPcm16bit;
+ int32_t codecEncoding = kAudioEncodingPcm16bit;
+ if (mFormat->findInt32("android._codec-pcm-encoding", &codecEncoding)
+ && mFormat->findInt32("android._config-pcm-encoding", &configEncoding)) {
+ if (mSrcEncoding != codecEncoding || mDstEncoding != configEncoding) {
+ if (codecEncoding != configEncoding) {
+ mDataConverter = AudioConverter::Create(
+ (AudioEncoding)codecEncoding, (AudioEncoding)configEncoding);
+ ALOGD_IF(mDataConverter, "[%s] Converter created from %d to %d",
+ mName, codecEncoding, configEncoding);
+ mFormatWithConverter = mFormat->dup();
+ mFormatWithConverter->setInt32(KEY_PCM_ENCODING, configEncoding);
+ } else {
+ mDataConverter = nullptr;
+ mFormatWithConverter = nullptr;
+ }
+ mSrcEncoding = codecEncoding;
+ mDstEncoding = configEncoding;
+ }
+ if (int encoding; !mFormat->findInt32(KEY_PCM_ENCODING, &encoding)
+ || encoding != mDstEncoding) {
+ }
+ }
+ if (!mDataConverter) {
+ return false;
+ }
+ sp<MediaCodecBuffer> srcBuffer = ConstLinearBlockBuffer::Allocate(mFormat, src);
+ if (!srcBuffer) {
+ return false;
+ }
+ if (!*dst) {
+ *dst = new Codec2Buffer(
+ mFormat,
+ new ABuffer(mDataConverter->targetSize(srcBuffer->size())));
+ }
+ sp<MediaCodecBuffer> dstBuffer = *dst;
+ status_t err = mDataConverter->convert(srcBuffer, dstBuffer);
+ if (err != OK) {
+ ALOGD("[%s] buffer conversion failed: %d", mName, err);
+ return false;
+ }
+ dstBuffer->setFormat(mFormatWithConverter);
+ return true;
+}
+
void OutputBuffers::clearStash() {
mPending.clear();
mReorderStash.clear();
@@ -1082,7 +1136,7 @@
return err;
}
c2Buffer->setFormat(mFormat);
- if (!c2Buffer->copy(buffer)) {
+ if (!convert(buffer, &c2Buffer) && !c2Buffer->copy(buffer)) {
ALOGD("[%s] copy buffer failed", mName);
return WOULD_BLOCK;
}
@@ -1198,9 +1252,12 @@
const std::shared_ptr<C2Buffer> &buffer,
size_t *index,
sp<MediaCodecBuffer> *clientBuffer) {
- sp<Codec2Buffer> newBuffer = wrap(buffer);
- if (newBuffer == nullptr) {
- return NO_MEMORY;
+ sp<Codec2Buffer> newBuffer;
+ if (!convert(buffer, &newBuffer)) {
+ newBuffer = wrap(buffer);
+ if (newBuffer == nullptr) {
+ return NO_MEMORY;
+ }
}
newBuffer->setFormat(mFormat);
*index = mImpl.assignSlot(newBuffer);
diff --git a/media/codec2/sfplugin/CCodecBuffers.h b/media/codec2/sfplugin/CCodecBuffers.h
index 995d3a4..c8e9930 100644
--- a/media/codec2/sfplugin/CCodecBuffers.h
+++ b/media/codec2/sfplugin/CCodecBuffers.h
@@ -18,9 +18,11 @@
#define CCODEC_BUFFERS_H_
+#include <optional>
#include <string>
#include <C2Config.h>
+#include <DataConverter.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/MediaCodecBuffer.h>
@@ -382,6 +384,14 @@
*/
void submit(const sp<MediaCodecBuffer> &buffer);
+ /**
+ * Apply DataConverter from |src| to |*dst| if needed. If |*dst| is nullptr,
+ * a new buffer is allocated.
+ *
+ * Returns true if conversion was needed and executed; false otherwise.
+ */
+ bool convert(const std::shared_ptr<C2Buffer> &src, sp<Codec2Buffer> *dst);
+
private:
// SkipCutBuffer
int32_t mDelay;
@@ -391,6 +401,12 @@
void setSkipCutBuffer(int32_t skip, int32_t cut);
+ // DataConverter
+ sp<DataConverter> mDataConverter;
+ sp<AMessage> mFormatWithConverter;
+ std::optional<int32_t> mSrcEncoding;
+ std::optional<int32_t> mDstEncoding;
+
// Output stash
// Struct for an entry in the output stash (mPending and mReorderStash)
diff --git a/media/codec2/sfplugin/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index c275187..03418d9 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -657,24 +657,29 @@
add(ConfigMapper(KEY_SAMPLE_RATE, C2_PARAMKEY_CODED_SAMPLE_RATE, "value")
.limitTo(D::AUDIO & D::CODED));
- add(ConfigMapper(KEY_PCM_ENCODING, C2_PARAMKEY_PCM_ENCODING, "value")
+ auto pcmEncodingMapper = [](C2Value v) -> C2Value {
+ int32_t value;
+ C2Config::pcm_encoding_t to;
+ if (v.get(&value) && C2Mapper::map(value, &to)) {
+ return to;
+ }
+ return C2Value();
+ };
+ auto pcmEncodingReverse = [](C2Value v) -> C2Value {
+ C2Config::pcm_encoding_t value;
+ int32_t to;
+ using C2ValueType=typename _c2_reduce_enum_to_underlying_type<decltype(value)>::type;
+ if (v.get((C2ValueType*)&value) && C2Mapper::map(value, &to)) {
+ return to;
+ }
+ return C2Value();
+ };
+ add(ConfigMapper(KEY_PCM_ENCODING, C2_PARAMKEY_PCM_ENCODING, "value")
.limitTo(D::AUDIO)
- .withMappers([](C2Value v) -> C2Value {
- int32_t value;
- C2Config::pcm_encoding_t to;
- if (v.get(&value) && C2Mapper::map(value, &to)) {
- return to;
- }
- return C2Value();
- }, [](C2Value v) -> C2Value {
- C2Config::pcm_encoding_t value;
- int32_t to;
- using C2ValueType=typename _c2_reduce_enum_to_underlying_type<decltype(value)>::type;
- if (v.get((C2ValueType*)&value) && C2Mapper::map(value, &to)) {
- return to;
- }
- return C2Value();
- }));
+ .withMappers(pcmEncodingMapper, pcmEncodingReverse));
+ add(ConfigMapper("android._codec-pcm-encoding", C2_PARAMKEY_PCM_ENCODING, "value")
+ .limitTo(D::AUDIO & D::READ)
+ .withMappers(pcmEncodingMapper, pcmEncodingReverse));
add(ConfigMapper(KEY_IS_ADTS, C2_PARAMKEY_AAC_PACKAGING, "value")
.limitTo(D::AUDIO & D::CODED)
diff --git a/media/codec2/vndk/C2PlatformStorePluginLoader.cpp b/media/codec2/vndk/C2PlatformStorePluginLoader.cpp
index bee028a..2a888a8 100644
--- a/media/codec2/vndk/C2PlatformStorePluginLoader.cpp
+++ b/media/codec2/vndk/C2PlatformStorePluginLoader.cpp
@@ -59,13 +59,14 @@
c2_status_t C2PlatformStorePluginLoader::createBlockPool(
::C2Allocator::id_t allocatorId, ::C2BlockPool::local_id_t blockPoolId,
- std::shared_ptr<C2BlockPool>* pool) {
+ std::shared_ptr<C2BlockPool>* pool,
+ std::function<void(C2BlockPool *)> deleter) {
if (mCreateBlockPool == nullptr) {
ALOGD("Handle or CreateBlockPool symbol is null");
return C2_NOT_FOUND;
}
- std::shared_ptr<::C2BlockPool> ptr(mCreateBlockPool(allocatorId, blockPoolId));
+ std::shared_ptr<::C2BlockPool> ptr(mCreateBlockPool(allocatorId, blockPoolId), deleter);
if (ptr) {
*pool = ptr;
return C2_OK;
@@ -75,14 +76,16 @@
}
c2_status_t C2PlatformStorePluginLoader::createAllocator(
- ::C2Allocator::id_t allocatorId, std::shared_ptr<C2Allocator>* const allocator) {
+ ::C2Allocator::id_t allocatorId,
+ std::shared_ptr<C2Allocator>* const allocator,
+ std::function<void(C2Allocator *)> deleter) {
if (mCreateAllocator == nullptr) {
ALOGD("Handle or CreateAllocator symbol is null");
return C2_NOT_FOUND;
}
c2_status_t res = C2_CORRUPTED;
- allocator->reset(mCreateAllocator(allocatorId, &res));
+ allocator->reset(mCreateAllocator(allocatorId, &res), deleter);
if (res != C2_OK) {
ALOGD("Failed to CreateAllocator by id: %u, res: %d", allocatorId, res);
allocator->reset();
diff --git a/media/codec2/vndk/C2Store.cpp b/media/codec2/vndk/C2Store.cpp
index c07c09e..1660c38 100644
--- a/media/codec2/vndk/C2Store.cpp
+++ b/media/codec2/vndk/C2Store.cpp
@@ -443,6 +443,7 @@
public:
_C2BlockPoolCache() : mBlockPoolSeqId(C2BlockPool::PLATFORM_START + 1) {}
+private:
c2_status_t _createBlockPool(
C2PlatformAllocatorStore::id_t allocatorId,
std::vector<std::shared_ptr<const C2Component>> components,
@@ -456,14 +457,19 @@
if (allocatorId == C2AllocatorStore::DEFAULT_LINEAR) {
allocatorId = GetPreferredLinearAllocatorId(GetCodec2PoolMask());
}
+ auto deleter = [this, poolId](C2BlockPool *pool) {
+ std::unique_lock lock(mMutex);
+ mBlockPools.erase(poolId);
+ mComponents.erase(poolId);
+ delete pool;
+ };
switch(allocatorId) {
case C2PlatformAllocatorStore::ION: /* also ::DMABUFHEAP */
res = allocatorStore->fetchAllocator(
C2PlatformAllocatorStore::ION, &allocator);
if (res == C2_OK) {
- std::shared_ptr<C2BlockPool> ptr =
- std::make_shared<C2PooledBlockPool>(
- allocator, poolId);
+ std::shared_ptr<C2BlockPool> ptr(
+ new C2PooledBlockPool(allocator, poolId), deleter);
*pool = ptr;
mBlockPools[poolId] = ptr;
mComponents[poolId].insert(
@@ -475,9 +481,8 @@
res = allocatorStore->fetchAllocator(
C2PlatformAllocatorStore::BLOB, &allocator);
if (res == C2_OK) {
- std::shared_ptr<C2BlockPool> ptr =
- std::make_shared<C2PooledBlockPool>(
- allocator, poolId);
+ std::shared_ptr<C2BlockPool> ptr(
+ new C2PooledBlockPool(allocator, poolId), deleter);
*pool = ptr;
mBlockPools[poolId] = ptr;
mComponents[poolId].insert(
@@ -490,8 +495,8 @@
res = allocatorStore->fetchAllocator(
C2AllocatorStore::DEFAULT_GRAPHIC, &allocator);
if (res == C2_OK) {
- std::shared_ptr<C2BlockPool> ptr =
- std::make_shared<C2PooledBlockPool>(allocator, poolId);
+ std::shared_ptr<C2BlockPool> ptr(
+ new C2PooledBlockPool(allocator, poolId), deleter);
*pool = ptr;
mBlockPools[poolId] = ptr;
mComponents[poolId].insert(
@@ -503,9 +508,8 @@
res = allocatorStore->fetchAllocator(
C2PlatformAllocatorStore::BUFFERQUEUE, &allocator);
if (res == C2_OK) {
- std::shared_ptr<C2BlockPool> ptr =
- std::make_shared<C2BufferQueueBlockPool>(
- allocator, poolId);
+ std::shared_ptr<C2BlockPool> ptr(
+ new C2BufferQueueBlockPool(allocator, poolId), deleter);
*pool = ptr;
mBlockPools[poolId] = ptr;
mComponents[poolId].insert(
@@ -517,7 +521,7 @@
// Try to create block pool from platform store plugins.
std::shared_ptr<C2BlockPool> ptr;
res = C2PlatformStorePluginLoader::GetInstance()->createBlockPool(
- allocatorId, poolId, &ptr);
+ allocatorId, poolId, &ptr, deleter);
if (res == C2_OK) {
*pool = ptr;
mBlockPools[poolId] = ptr;
@@ -530,17 +534,20 @@
return res;
}
+public:
c2_status_t createBlockPool(
C2PlatformAllocatorStore::id_t allocatorId,
std::vector<std::shared_ptr<const C2Component>> components,
std::shared_ptr<C2BlockPool> *pool) {
+ std::unique_lock lock(mMutex);
return _createBlockPool(allocatorId, components, mBlockPoolSeqId++, pool);
}
- bool getBlockPool(
+ c2_status_t getBlockPool(
C2BlockPool::local_id_t blockPoolId,
std::shared_ptr<const C2Component> component,
std::shared_ptr<C2BlockPool> *pool) {
+ std::unique_lock lock(mMutex);
// TODO: use one iterator for multiple blockpool type scalability.
std::shared_ptr<C2BlockPool> ptr;
auto it = mBlockPools.find(blockPoolId);
@@ -558,14 +565,22 @@
});
if (found != mComponents[blockPoolId].end()) {
*pool = ptr;
- return true;
+ return C2_OK;
}
}
}
- return false;
+ // TODO: remove this. this is temporary
+ if (blockPoolId == C2BlockPool::PLATFORM_START) {
+ return _createBlockPool(
+ C2PlatformAllocatorStore::BUFFERQUEUE, {component}, blockPoolId, pool);
+ }
+ return C2_NOT_FOUND;
}
private:
+ // Deleter needs to hold this mutex, and there is a small chance that deleter
+ // is invoked while the mutex is held.
+ std::recursive_mutex mMutex;
C2BlockPool::local_id_t mBlockPoolSeqId;
std::map<C2BlockPool::local_id_t, std::weak_ptr<C2BlockPool>> mBlockPools;
@@ -574,7 +589,6 @@
static std::unique_ptr<_C2BlockPoolCache> sBlockPoolCache =
std::make_unique<_C2BlockPoolCache>();
-static std::mutex sBlockPoolCacheMutex;
} // anynymous namespace
@@ -582,15 +596,12 @@
C2BlockPool::local_id_t id, std::shared_ptr<const C2Component> component,
std::shared_ptr<C2BlockPool> *pool) {
pool->reset();
- std::lock_guard<std::mutex> lock(sBlockPoolCacheMutex);
std::shared_ptr<C2AllocatorStore> allocatorStore = GetCodec2PlatformAllocatorStore();
std::shared_ptr<C2Allocator> allocator;
c2_status_t res = C2_NOT_FOUND;
if (id >= C2BlockPool::PLATFORM_START) {
- if (sBlockPoolCache->getBlockPool(id, component, pool)) {
- return C2_OK;
- }
+ return sBlockPoolCache->getBlockPool(id, component, pool);
}
switch (id) {
@@ -606,11 +617,6 @@
*pool = std::make_shared<C2BasicGraphicBlockPool>(allocator);
}
break;
- // TODO: remove this. this is temporary
- case C2BlockPool::PLATFORM_START:
- res = sBlockPoolCache->_createBlockPool(
- C2PlatformAllocatorStore::BUFFERQUEUE, {component}, id, pool);
- break;
default:
break;
}
@@ -623,7 +629,6 @@
std::shared_ptr<C2BlockPool> *pool) {
pool->reset();
- std::lock_guard<std::mutex> lock(sBlockPoolCacheMutex);
return sBlockPoolCache->createBlockPool(allocatorId, components, pool);
}
@@ -633,7 +638,6 @@
std::shared_ptr<C2BlockPool> *pool) {
pool->reset();
- std::lock_guard<std::mutex> lock(sBlockPoolCacheMutex);
return sBlockPoolCache->createBlockPool(allocatorId, {component}, pool);
}
diff --git a/media/codec2/vndk/include/C2PlatformStorePluginLoader.h b/media/codec2/vndk/include/C2PlatformStorePluginLoader.h
index 4c10643..73d1b5e 100644
--- a/media/codec2/vndk/include/C2PlatformStorePluginLoader.h
+++ b/media/codec2/vndk/include/C2PlatformStorePluginLoader.h
@@ -61,9 +61,11 @@
* \retval C2_NOT_FOUND the extension symbol was not found.
* \retval C2_BAD_INDEX the input allocatorId is not defined in platform store extension.
*/
- c2_status_t createBlockPool(::C2Allocator::id_t allocatorId,
- ::C2BlockPool::local_id_t blockPoolId,
- std::shared_ptr<C2BlockPool>* pool);
+ c2_status_t createBlockPool(
+ ::C2Allocator::id_t allocatorId,
+ ::C2BlockPool::local_id_t blockPoolId,
+ std::shared_ptr<C2BlockPool>* pool,
+ std::function<void(C2BlockPool *)> deleter = std::default_delete<C2BlockPool>());
/**
* Creates allocator from platform store extension.
@@ -81,8 +83,10 @@
* \retval C2_BAD_INDEX the input allocatorId is not defined in platform store extension.
* \retval C2_NO_MEMORY not enough memory to create the allocator
*/
- c2_status_t createAllocator(::C2Allocator::id_t allocatorId,
- std::shared_ptr<C2Allocator>* const allocator);
+ c2_status_t createAllocator(
+ ::C2Allocator::id_t allocatorId,
+ std::shared_ptr<C2Allocator>* const allocator,
+ std::function<void(C2Allocator *)> deleter = std::default_delete<C2Allocator>());
private:
explicit C2PlatformStorePluginLoader(const char *libPath);
diff --git a/media/codecs/amrwb/enc/Android.bp b/media/codecs/amrwb/enc/Android.bp
index cc72eb7..d945531 100644
--- a/media/codecs/amrwb/enc/Android.bp
+++ b/media/codecs/amrwb/enc/Android.bp
@@ -139,11 +139,6 @@
},
},
- include_dirs: [
- "frameworks/av/include",
- "frameworks/av/media/libstagefright/include",
- ],
-
local_include_dirs: ["src"],
export_include_dirs: ["inc"],
diff --git a/media/codecs/m4v_h263/enc/src/fastidct.cpp b/media/codecs/m4v_h263/enc/src/fastidct.cpp
index 688effc..ec1b28f 100644
--- a/media/codecs/m4v_h263/enc/src/fastidct.cpp
+++ b/media/codecs/m4v_h263/enc/src/fastidct.cpp
@@ -76,6 +76,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_col2(Short *blk)
{
int32 x0, x1, x3, x5, x7;//, x8;
@@ -102,6 +104,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_col3(Short *blk)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -137,6 +141,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_col4(Short *blk)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -180,6 +186,8 @@
}
#ifndef SMALL_DCT
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_col0x40(Short *blk)
{
int32 x1, x3, x5, x7;//, x8;
@@ -230,6 +238,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_col0x10(Short *blk)
{
int32 x1, x3, x5, x7;
@@ -256,6 +266,8 @@
#endif /* SMALL_DCT */
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_col(Short *blk)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -368,6 +380,8 @@
return;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row2Inter(Short *blk, UChar *rec, Int lx)
{
int32 x0, x1, x2, x4, x5;
@@ -427,6 +441,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row3Inter(Short *blk, UChar *rec, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -497,6 +513,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row4Inter(Short *blk, UChar *rec, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -573,6 +591,8 @@
}
#ifndef SMALL_DCT
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row0x40Inter(Short *blk, UChar *rec, Int lx)
{
int32 x1, x2, x4, x5;
@@ -686,6 +706,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row0x10Inter(Short *blk, UChar *rec, Int lx)
{
int32 x1, x3, x5, x7;
@@ -741,6 +763,8 @@
#endif /* SMALL_DCT */
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_rowInter(Short *blk, UChar *rec, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -864,6 +888,8 @@
return;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row2Intra(Short *blk, UChar *rec, Int lx)
{
int32 x0, x1, x2, x4, x5;
@@ -919,6 +945,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row3Intra(Short *blk, UChar *rec, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -985,6 +1013,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row4Intra(Short *blk, UChar *rec, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -1058,6 +1088,8 @@
}
#ifndef SMALL_DCT
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row0x40Intra(Short *blk, UChar *rec, Int lx)
{
int32 x1, x2, x4, x5;
@@ -1166,6 +1198,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row0x10Intra(Short *blk, UChar *rec, Int lx)
{
int32 x1, x3, x5, x7;
@@ -1218,6 +1252,8 @@
}
#endif /* SMALL_DCT */
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_rowIntra(Short *blk, UChar *rec, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -1364,6 +1400,8 @@
return;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row2zmv(Short *blk, UChar *rec, UChar *pred, Int lx)
{
int32 x0, x1, x2, x4, x5;
@@ -1424,6 +1462,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row3zmv(Short *blk, UChar *rec, UChar *pred, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -1495,6 +1535,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row4zmv(Short *blk, UChar *rec, UChar *pred, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -1572,6 +1614,8 @@
}
#ifndef SMALL_DCT
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row0x40zmv(Short *blk, UChar *rec, UChar *pred, Int lx)
{
int32 x1, x2, x4, x5;
@@ -1687,6 +1731,8 @@
return ;
}
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_row0x10zmv(Short *blk, UChar *rec, UChar *pred, Int lx)
{
int32 x1, x3, x5, x7;
@@ -1743,6 +1789,8 @@
#endif /* SMALL_DCT */
+/* Ignoring overflows as idct function expects and uses overflows */
+__attribute__((no_sanitize("signed-integer-overflow")))
void idct_rowzmv(Short *blk, UChar *rec, UChar *pred, Int lx)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
diff --git a/media/codecs/m4v_h263/fuzzer/mpeg4_h263_dec_fuzzer.cpp b/media/codecs/m4v_h263/fuzzer/mpeg4_h263_dec_fuzzer.cpp
index 912c821..5e613d9 100644
--- a/media/codecs/m4v_h263/fuzzer/mpeg4_h263_dec_fuzzer.cpp
+++ b/media/codecs/m4v_h263/fuzzer/mpeg4_h263_dec_fuzzer.cpp
@@ -50,7 +50,7 @@
private:
tagvideoDecControls *mDecHandle = nullptr;
- uint8_t *mOutputBuffer[kNumOutputBuffers];
+ uint8_t *mOutputBuffer[kNumOutputBuffers] = {};
bool mInitialized = false;
bool mFramesConfigured = false;
#ifdef MPEG4
diff --git a/media/codecs/mp3dec/Android.bp b/media/codecs/mp3dec/Android.bp
index 015b8b6..1ab0511 100644
--- a/media/codecs/mp3dec/Android.bp
+++ b/media/codecs/mp3dec/Android.bp
@@ -108,8 +108,6 @@
cfi: true,
},
- include_dirs: ["frameworks/av/media/libstagefright/include"],
-
header_libs: ["libstagefright_mp3dec_headers"],
export_header_lib_headers: ["libstagefright_mp3dec_headers"],
diff --git a/media/codecs/mp3dec/src/pvmp3_stereo_proc.cpp b/media/codecs/mp3dec/src/pvmp3_stereo_proc.cpp
index c04f7f3..b3ecc77 100644
--- a/media/codecs/mp3dec/src/pvmp3_stereo_proc.cpp
+++ b/media/codecs/mp3dec/src/pvmp3_stereo_proc.cpp
@@ -219,6 +219,7 @@
; FUNCTION CODE
----------------------------------------------------------------------------*/
+// deliberately plays near overflow points of int32
#if __has_attribute(no_sanitize)
__attribute__((no_sanitize("integer")))
#endif
diff --git a/media/extractors/Android.bp b/media/extractors/Android.bp
index 7513cb1..66585da 100644
--- a/media/extractors/Android.bp
+++ b/media/extractors/Android.bp
@@ -24,10 +24,6 @@
cc_defaults {
name: "extractor-defaults",
- include_dirs: [
- "frameworks/av/media/libstagefright/include",
- ],
-
shared_libs: [
"liblog",
],
diff --git a/media/extractors/aac/Android.bp b/media/extractors/aac/Android.bp
index 7bf3a13..a926422 100644
--- a/media/extractors/aac/Android.bp
+++ b/media/extractors/aac/Android.bp
@@ -21,6 +21,10 @@
srcs: ["AACExtractor.cpp"],
+ export_include_dirs: [
+ "include",
+ ],
+
static_libs: [
"libstagefright_foundation",
"libstagefright_metadatautils",
diff --git a/media/extractors/aac/AACExtractor.h b/media/extractors/aac/include/AACExtractor.h
similarity index 100%
rename from media/extractors/aac/AACExtractor.h
rename to media/extractors/aac/include/AACExtractor.h
diff --git a/media/extractors/amr/Android.bp b/media/extractors/amr/Android.bp
index 712360d..121b7a3 100644
--- a/media/extractors/amr/Android.bp
+++ b/media/extractors/amr/Android.bp
@@ -21,6 +21,10 @@
srcs: ["AMRExtractor.cpp"],
+ export_include_dirs: [
+ "include",
+ ],
+
static_libs: [
"libstagefright_foundation",
],
diff --git a/media/extractors/amr/AMRExtractor.h b/media/extractors/amr/include/AMRExtractor.h
similarity index 100%
rename from media/extractors/amr/AMRExtractor.h
rename to media/extractors/amr/include/AMRExtractor.h
diff --git a/media/extractors/flac/Android.bp b/media/extractors/flac/Android.bp
index 9a2a76b..fd51622 100644
--- a/media/extractors/flac/Android.bp
+++ b/media/extractors/flac/Android.bp
@@ -23,8 +23,8 @@
srcs: ["FLACExtractor.cpp"],
- include_dirs: [
- "external/flac/include",
+ export_include_dirs: [
+ "include",
],
shared_libs: [
diff --git a/media/extractors/flac/FLACExtractor.h b/media/extractors/flac/include/FLACExtractor.h
similarity index 100%
rename from media/extractors/flac/FLACExtractor.h
rename to media/extractors/flac/include/FLACExtractor.h
diff --git a/media/extractors/fuzzers/Android.bp b/media/extractors/fuzzers/Android.bp
index 0e54b58..490e195 100644
--- a/media/extractors/fuzzers/Android.bp
+++ b/media/extractors/fuzzers/Android.bp
@@ -80,11 +80,6 @@
defaults: ["extractor-fuzzer-defaults"],
host_supported: true,
- include_dirs: [
- "frameworks/av/media/extractors/mpeg2",
- "frameworks/av/media/libstagefright",
- ],
-
static_libs: [
"libstagefright_foundation_without_imemory",
"libstagefright_mpeg2support",
@@ -124,14 +119,6 @@
"mp4_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/mp4",
- ],
-
- header_libs: [
- "libaudioclient_headers",
- ],
-
static_libs: [
"libstagefright_id3",
"libstagefright_esds",
@@ -150,10 +137,6 @@
"wav_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/wav",
- ],
-
static_libs: [
"libfifo",
"libwavextractor",
@@ -173,10 +156,6 @@
"amr_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/amr",
- ],
-
static_libs: [
"libamrextractor",
],
@@ -193,10 +172,6 @@
"mkv_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/mkv",
- ],
-
static_libs: [
"libwebm",
"libstagefright_flacdec",
@@ -217,9 +192,6 @@
"ogg_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/ogg",
- ],
static_libs: [
"libstagefright_metadatautils",
@@ -265,10 +237,6 @@
"mp3_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/mp3",
- ],
-
static_libs: [
"libfifo",
"libmp3extractor",
@@ -285,10 +253,6 @@
"aac_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/aac",
- ],
-
static_libs: [
"libaacextractor",
"libstagefright_metadatautils",
@@ -304,10 +268,6 @@
"flac_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/flac",
- ],
-
static_libs: [
"libstagefright_metadatautils",
"libFLAC",
@@ -329,10 +289,6 @@
"midi_extractor_fuzzer.cpp",
],
- include_dirs: [
- "frameworks/av/media/extractors/midi",
- ],
-
static_libs: [
"libsonivox",
"libmedia_midiiowrapper",
diff --git a/media/extractors/midi/Android.bp b/media/extractors/midi/Android.bp
index 08a6fa0..feabf9e 100644
--- a/media/extractors/midi/Android.bp
+++ b/media/extractors/midi/Android.bp
@@ -23,6 +23,10 @@
srcs: ["MidiExtractor.cpp"],
+ export_include_dirs: [
+ "include",
+ ],
+
header_libs: [
"libmedia_datasource_headers",
],
diff --git a/media/extractors/midi/MidiExtractor.h b/media/extractors/midi/include/MidiExtractor.h
similarity index 100%
rename from media/extractors/midi/MidiExtractor.h
rename to media/extractors/midi/include/MidiExtractor.h
diff --git a/media/extractors/mkv/Android.bp b/media/extractors/mkv/Android.bp
index 54c5b27..98ce305 100644
--- a/media/extractors/mkv/Android.bp
+++ b/media/extractors/mkv/Android.bp
@@ -21,10 +21,8 @@
srcs: ["MatroskaExtractor.cpp"],
- include_dirs: [
- "external/flac/include",
- "external/libvpx/libwebm",
- "frameworks/av/media/libstagefright/flac/dec",
+ export_include_dirs: [
+ "include",
],
shared_libs: [
diff --git a/media/extractors/mkv/MatroskaExtractor.h b/media/extractors/mkv/include/MatroskaExtractor.h
similarity index 100%
rename from media/extractors/mkv/MatroskaExtractor.h
rename to media/extractors/mkv/include/MatroskaExtractor.h
diff --git a/media/extractors/mp3/Android.bp b/media/extractors/mp3/Android.bp
index 75b9b7b..396a13a 100644
--- a/media/extractors/mp3/Android.bp
+++ b/media/extractors/mp3/Android.bp
@@ -16,6 +16,10 @@
"XINGSeeker.cpp",
],
+ export_include_dirs: [
+ "include",
+ ],
+
static_libs: [
"libutils",
"libstagefright_id3",
diff --git a/media/extractors/mp3/MP3Extractor.h b/media/extractors/mp3/include/MP3Extractor.h
similarity index 100%
rename from media/extractors/mp3/MP3Extractor.h
rename to media/extractors/mp3/include/MP3Extractor.h
diff --git a/media/extractors/mp3/MP3Seeker.h b/media/extractors/mp3/include/MP3Seeker.h
similarity index 100%
rename from media/extractors/mp3/MP3Seeker.h
rename to media/extractors/mp3/include/MP3Seeker.h
diff --git a/media/extractors/mp3/VBRISeeker.h b/media/extractors/mp3/include/VBRISeeker.h
similarity index 100%
rename from media/extractors/mp3/VBRISeeker.h
rename to media/extractors/mp3/include/VBRISeeker.h
diff --git a/media/extractors/mp3/XINGSeeker.h b/media/extractors/mp3/include/XINGSeeker.h
similarity index 100%
rename from media/extractors/mp3/XINGSeeker.h
rename to media/extractors/mp3/include/XINGSeeker.h
diff --git a/media/extractors/mp4/Android.bp b/media/extractors/mp4/Android.bp
index 7fa6bfd..540d75d 100644
--- a/media/extractors/mp4/Android.bp
+++ b/media/extractors/mp4/Android.bp
@@ -15,6 +15,15 @@
],
}
+cc_library_headers {
+ name: "libmp4extractor_headers",
+ host_supported: true,
+
+ export_include_dirs: [
+ "include",
+ ],
+}
+
cc_library {
name: "libmp4extractor",
defaults: ["extractor-defaults"],
@@ -27,6 +36,10 @@
"SampleTable.cpp",
],
+ export_include_dirs: [
+ "include",
+ ],
+
static_libs: [
"libstagefright_esds",
"libstagefright_foundation",
diff --git a/media/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
index fbcd554..8836c47 100644
--- a/media/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/extractors/mp4/MPEG4Extractor.cpp
@@ -157,6 +157,7 @@
MediaBufferHelper *mBuffer;
+ size_t mSrcBufferSize;
uint8_t *mSrcBuffer;
bool mIsHeif;
@@ -5083,6 +5084,7 @@
mNALLengthSize(0),
mStarted(false),
mBuffer(NULL),
+ mSrcBufferSize(0),
mSrcBuffer(NULL),
mItemTable(itemTable),
mElstShiftStartTicks(elstShiftStartTicks),
@@ -5264,6 +5266,7 @@
// file probably specified a bad max size
return AMEDIA_ERROR_MALFORMED;
}
+ mSrcBufferSize = max_size;
mStarted = true;
@@ -5280,6 +5283,7 @@
mBuffer = NULL;
}
+ mSrcBufferSize = 0;
delete[] mSrcBuffer;
mSrcBuffer = NULL;
@@ -6467,13 +6471,19 @@
// Whole NAL units are returned but each fragment is prefixed by
// the start code (0x00 00 00 01).
ssize_t num_bytes_read = 0;
- num_bytes_read = mDataSource->readAt(offset, mSrcBuffer, size);
+ bool mSrcBufferFitsDataToRead = size <= mSrcBufferSize;
+ if (mSrcBufferFitsDataToRead) {
+ num_bytes_read = mDataSource->readAt(offset, mSrcBuffer, size);
+ } else {
+ // We are trying to read a sample larger than the expected max sample size.
+ // Fall through and let the failure be handled by the following if.
+ android_errorWriteLog(0x534e4554, "188893559");
+ }
if (num_bytes_read < (ssize_t)size) {
mBuffer->release();
mBuffer = NULL;
-
- return AMEDIA_ERROR_IO;
+ return mSrcBufferFitsDataToRead ? AMEDIA_ERROR_IO : AMEDIA_ERROR_MALFORMED;
}
uint8_t *dstData = (uint8_t *)mBuffer->data();
diff --git a/media/extractors/mp4/AC4Parser.h b/media/extractors/mp4/include/AC4Parser.h
similarity index 100%
rename from media/extractors/mp4/AC4Parser.h
rename to media/extractors/mp4/include/AC4Parser.h
diff --git a/media/extractors/mp4/ItemTable.h b/media/extractors/mp4/include/ItemTable.h
similarity index 100%
rename from media/extractors/mp4/ItemTable.h
rename to media/extractors/mp4/include/ItemTable.h
diff --git a/media/extractors/mp4/MPEG4Extractor.h b/media/extractors/mp4/include/MPEG4Extractor.h
similarity index 100%
rename from media/extractors/mp4/MPEG4Extractor.h
rename to media/extractors/mp4/include/MPEG4Extractor.h
diff --git a/media/extractors/mp4/SampleIterator.h b/media/extractors/mp4/include/SampleIterator.h
similarity index 100%
rename from media/extractors/mp4/SampleIterator.h
rename to media/extractors/mp4/include/SampleIterator.h
diff --git a/media/extractors/mp4/SampleTable.h b/media/extractors/mp4/include/SampleTable.h
similarity index 100%
rename from media/extractors/mp4/SampleTable.h
rename to media/extractors/mp4/include/SampleTable.h
diff --git a/media/extractors/mpeg2/Android.bp b/media/extractors/mpeg2/Android.bp
index 7e6247b..8faecae 100644
--- a/media/extractors/mpeg2/Android.bp
+++ b/media/extractors/mpeg2/Android.bp
@@ -38,6 +38,10 @@
"MPEG2TSExtractor.cpp",
],
+ export_include_dirs: [
+ "include",
+ ],
+
shared_libs: [
"libbase",
"libcgrouprc#29",
diff --git a/media/extractors/mpeg2/MPEG2PSExtractor.h b/media/extractors/mpeg2/include/MPEG2PSExtractor.h
similarity index 100%
rename from media/extractors/mpeg2/MPEG2PSExtractor.h
rename to media/extractors/mpeg2/include/MPEG2PSExtractor.h
diff --git a/media/extractors/mpeg2/MPEG2TSExtractor.h b/media/extractors/mpeg2/include/MPEG2TSExtractor.h
similarity index 100%
rename from media/extractors/mpeg2/MPEG2TSExtractor.h
rename to media/extractors/mpeg2/include/MPEG2TSExtractor.h
diff --git a/media/extractors/ogg/Android.bp b/media/extractors/ogg/Android.bp
index d7540c4..dc3c25c 100644
--- a/media/extractors/ogg/Android.bp
+++ b/media/extractors/ogg/Android.bp
@@ -22,8 +22,8 @@
srcs: ["OggExtractor.cpp"],
- include_dirs: [
- "external/tremolo",
+ export_include_dirs: [
+ "include",
],
header_libs: [
diff --git a/media/extractors/ogg/OggExtractor.h b/media/extractors/ogg/include/OggExtractor.h
similarity index 100%
rename from media/extractors/ogg/OggExtractor.h
rename to media/extractors/ogg/include/OggExtractor.h
diff --git a/media/extractors/tests/Android.bp b/media/extractors/tests/Android.bp
index 5d97d9a..3c3bbdc 100644
--- a/media/extractors/tests/Android.bp
+++ b/media/extractors/tests/Android.bp
@@ -45,14 +45,11 @@
"libdatasource",
"libwatchdog",
- "libstagefright",
"libstagefright_id3",
"libstagefright_flacdec",
"libstagefright_esds",
"libstagefright_mpeg2support",
- "libstagefright_mpeg2extractor",
"libstagefright_foundation_colorutils_ndk",
- "libstagefright_foundation",
"libstagefright_metadatautils",
"libmedia_midiiowrapper",
@@ -74,17 +71,14 @@
"libcutils",
"libmediandk",
"libmedia",
+ "libstagefright",
+ "libstagefright_foundation",
"libcrypto",
"libhidlmemory",
"libhidlbase",
"libbase",
],
- include_dirs: [
- "frameworks/av/media/extractors/",
- "frameworks/av/media/libstagefright/",
- ],
-
compile_multilib: "first",
cflags: [
diff --git a/media/extractors/tests/ExtractorUnitTest.cpp b/media/extractors/tests/ExtractorUnitTest.cpp
index 84ec1f2..2bd9c6a 100644
--- a/media/extractors/tests/ExtractorUnitTest.cpp
+++ b/media/extractors/tests/ExtractorUnitTest.cpp
@@ -27,18 +27,18 @@
#include <media/stagefright/MetaDataUtils.h>
#include <media/stagefright/foundation/OpusHeader.h>
-#include "aac/AACExtractor.h"
-#include "amr/AMRExtractor.h"
-#include "flac/FLACExtractor.h"
-#include "midi/MidiExtractor.h"
-#include "mkv/MatroskaExtractor.h"
-#include "mp3/MP3Extractor.h"
-#include "mp4/MPEG4Extractor.h"
-#include "mp4/SampleTable.h"
-#include "mpeg2/MPEG2PSExtractor.h"
-#include "mpeg2/MPEG2TSExtractor.h"
-#include "ogg/OggExtractor.h"
-#include "wav/WAVExtractor.h"
+#include <AACExtractor.h>
+#include <AMRExtractor.h>
+#include <FLACExtractor.h>
+#include <MidiExtractor.h>
+#include <MatroskaExtractor.h>
+#include <MP3Extractor.h>
+#include <MPEG4Extractor.h>
+#include <SampleTable.h>
+#include <MPEG2PSExtractor.h>
+#include <MPEG2TSExtractor.h>
+#include <OggExtractor.h>
+#include <WAVExtractor.h>
#include "ExtractorUnitTestEnvironment.h"
diff --git a/media/extractors/wav/Android.bp b/media/extractors/wav/Android.bp
index cc5e1c7..b7e2af3 100644
--- a/media/extractors/wav/Android.bp
+++ b/media/extractors/wav/Android.bp
@@ -22,8 +22,8 @@
srcs: ["WAVExtractor.cpp"],
- include_dirs: [
- "frameworks/av/media/libstagefright/include",
+ export_include_dirs: [
+ "include",
],
shared_libs: [
diff --git a/media/extractors/wav/WAVExtractor.h b/media/extractors/wav/include/WAVExtractor.h
similarity index 100%
rename from media/extractors/wav/WAVExtractor.h
rename to media/extractors/wav/include/WAVExtractor.h
diff --git a/media/janitors/reliability_mainline_OWNERS b/media/janitors/reliability_mainline_OWNERS
new file mode 100644
index 0000000..e4c4fc2
--- /dev/null
+++ b/media/janitors/reliability_mainline_OWNERS
@@ -0,0 +1,5 @@
+# Bug component: 1051309
+# go/android-media-relaibility
+
+essick@google.com
+nchalko@google.com
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index ed3b986..efa9941 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -444,6 +444,22 @@
};
typedef int32_t aaudio_content_type_t;
+enum {
+
+ /**
+ * Constant indicating the audio content associated with these attributes will follow the
+ * default platform behavior with regards to which content will be spatialized or not.
+ */
+ AAUDIO_SPATIALIZATION_BEHAVIOR_AUTO = 1,
+
+ /**
+ * Constant indicating the audio content associated with these attributes should never
+ * be spatialized.
+ */
+ AAUDIO_SPATIALIZATION_BEHAVIOR_NEVER = 2,
+};
+typedef int32_t aaudio_spatialization_behavior_t;
+
/**
* Defines the audio source.
* An audio source defines both a default physical source of audio signal, and a recording
@@ -985,6 +1001,37 @@
aaudio_content_type_t contentType) __INTRODUCED_IN(28);
/**
+ * Sets the behavior affecting whether spatialization will be used.
+ *
+ * The AAudio system will use this information to select whether the stream will go
+ * through a spatializer effect or not when the effect is supported and enabled.
+ *
+ * Available since API level 32.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param spatializationBehavior the desired behavior with regards to spatialization, eg.
+ * {@link #AAUDIO_SPATIALIZATION_BEHAVIOR_AUTO}
+ */
+AAUDIO_API void AAudioStreamBuilder_setSpatializationBehavior(AAudioStreamBuilder* builder,
+ aaudio_spatialization_behavior_t spatializationBehavior) __INTRODUCED_IN(32);
+
+/**
+ * Specifies whether the audio data of this output stream has already been processed for
+ * spatialization.
+ *
+ * If the stream has been processed for spatialization, setting this to true will prevent
+ * issues such as double-processing on platforms that will spatialize audio data.
+ *
+ * Available since API level 32.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param isSpatialized true if the content is already processed for binaural or transaural spatial
+ * rendering, false otherwise.
+ */
+AAUDIO_API void AAudioStreamBuilder_setIsContentSpatialized(AAudioStreamBuilder* builder,
+ bool isSpatialized) __INTRODUCED_IN(32);
+
+/**
* Set the input (capture) preset for the stream.
*
* The AAudio system will use this information to optimize the
@@ -1791,6 +1838,31 @@
__INTRODUCED_IN(28);
/**
+ * Return the spatialization behavior for the stream.
+ *
+ * If none was explicitly set, it will return the default
+ * {@link #AAUDIO_SPATIALIZATION_BEHAVIOR_AUTO} behavior.
+ *
+ * Available since API level 32.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return spatialization behavior, for example {@link #AAUDIO_SPATIALIZATION_BEHAVIOR_AUTO}
+ */
+AAUDIO_API aaudio_spatialization_behavior_t AAudioStream_getSpatializationBehavior(
+ AAudioStream* stream) __INTRODUCED_IN(32);
+
+/**
+ * Return whether the content of the stream is spatialized.
+ *
+ * Available since API level 32.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return true if the content is spatialized
+ */
+AAUDIO_API bool AAudioStream_isContentSpatialized(AAudioStream* stream) __INTRODUCED_IN(32);
+
+
+/**
* Return the input preset for the stream.
*
* Available since API level 28.
diff --git a/media/libaaudio/src/Android.bp b/media/libaaudio/src/Android.bp
index da59da6..01aaa2a 100644
--- a/media/libaaudio/src/Android.bp
+++ b/media/libaaudio/src/Android.bp
@@ -168,6 +168,7 @@
"libbinder",
"framework-permission-aidl-cpp",
"aaudio-aidl-cpp",
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"libaudioclient_aidl_conversion",
],
@@ -246,6 +247,7 @@
"binding/aidl/aaudio/IAAudioService.aidl",
],
imports: [
+ "android.media.audio.common.types",
"audioclient-types-aidl",
"shared-file-region-aidl",
"framework-permission-aidl",
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
index cf76225..b60bac2 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
@@ -29,7 +29,7 @@
using namespace aaudio;
-using android::media::AudioFormatDescription;
+using android::media::audio::common::AudioFormatDescription;
AAudioStreamConfiguration::AAudioStreamConfiguration(const StreamParameters& parcelable) {
setChannelMask(parcelable.channelMask);
@@ -46,6 +46,13 @@
setUsage(parcelable.usage);
static_assert(sizeof(aaudio_content_type_t) == sizeof(parcelable.contentType));
setContentType(parcelable.contentType);
+
+ static_assert(sizeof(aaudio_spatialization_behavior_t) ==
+ sizeof(parcelable.spatializationBehavior));
+ setSpatializationBehavior(parcelable.spatializationBehavior);
+ setIsContentSpatialized(parcelable.isContentSpatialized);
+
+
static_assert(sizeof(aaudio_input_preset_t) == sizeof(parcelable.inputPreset));
setInputPreset(parcelable.inputPreset);
setBufferCapacity(parcelable.bufferCapacity);
@@ -76,7 +83,8 @@
result.audioFormat = convAudioFormat.value();
} else {
result.audioFormat = AudioFormatDescription{};
- result.audioFormat.type = android::media::AudioFormatType::SYS_RESERVED_INVALID;
+ result.audioFormat.type =
+ android::media::audio::common::AudioFormatType::SYS_RESERVED_INVALID;
}
static_assert(sizeof(aaudio_direction_t) == sizeof(result.direction));
result.direction = getDirection();
diff --git a/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl b/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
index 3b274f2..983e193 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
@@ -16,7 +16,7 @@
package aaudio;
-import android.media.AudioFormatDescription;
+import android.media.audio.common.AudioFormatDescription;
parcelable StreamParameters {
int channelMask; // = AAUDIO_UNSPECIFIED;
@@ -27,6 +27,8 @@
int /* aaudio_direction_t */ direction; // = AAUDIO_DIRECTION_OUTPUT;
int /* aaudio_usage_t */ usage; // = AAUDIO_UNSPECIFIED;
int /* aaudio_content_type_t */ contentType; // = AAUDIO_UNSPECIFIED;
+ int /* aaudio_spatialization_behavior_t */spatializationBehavior; //= AAUDIO_UNSPECIFIED;
+ boolean isContentSpatialized; // = false;
int /* aaudio_input_preset_t */ inputPreset; // = AAUDIO_UNSPECIFIED;
int bufferCapacity; // = AAUDIO_UNSPECIFIED;
int /* aaudio_allowed_capture_policy_t */ allowedCapturePolicy; // = AAUDIO_UNSPECIFIED;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 8284be5..89d42bf 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -27,6 +27,7 @@
#include <aaudio/AAudio.h>
#include <cutils/properties.h>
+#include <media/AudioSystem.h>
#include <media/MediaMetricsItem.h>
#include <utils/Trace.h>
@@ -95,7 +96,7 @@
return result;
}
- const int32_t burstMinMicros = AAudioProperty_getHardwareBurstMinMicros();
+ const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
int32_t burstMicros = 0;
const audio_format_t requestedFormat = getFormat();
@@ -127,6 +128,8 @@
request.getConfiguration().setUsage(getUsage());
request.getConfiguration().setContentType(getContentType());
+ request.getConfiguration().setSpatializationBehavior(getSpatializationBehavior());
+ request.getConfiguration().setIsContentSpatialized(isContentSpatialized());
request.getConfiguration().setInputPreset(getInputPreset());
request.getConfiguration().setPrivacySensitive(isPrivacySensitive());
@@ -186,6 +189,8 @@
setUsage(configurationOutput.getUsage());
setContentType(configurationOutput.getContentType());
+ setSpatializationBehavior(configurationOutput.getSpatializationBehavior());
+ setIsContentSpatialized(configurationOutput.isContentSpatialized());
setInputPreset(configurationOutput.getInputPreset());
// Save device format so we can do format conversion and volume scaling together.
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 02e7f5f..90ff4a5 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -167,6 +167,18 @@
streamBuilder->setContentType(contentType);
}
+AAUDIO_API void AAudioStreamBuilder_setSpatializationBehavior(AAudioStreamBuilder* builder,
+ aaudio_spatialization_behavior_t spatializationBehavior) {
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+ streamBuilder->setSpatializationBehavior(spatializationBehavior);
+}
+
+AAUDIO_API void AAudioStreamBuilder_setIsContentSpatialized(AAudioStreamBuilder* builder,
+ bool isSpatialized) {
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+ streamBuilder->setIsContentSpatialized(isSpatialized);
+}
+
AAUDIO_API void AAudioStreamBuilder_setInputPreset(AAudioStreamBuilder* builder,
aaudio_input_preset_t inputPreset) {
AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
@@ -340,7 +352,8 @@
{
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
- return audioStream->waitForStateChange(inputState, nextState, timeoutNanoseconds);
+ android::sp<AudioStream> spAudioStream(audioStream);
+ return spAudioStream->waitForStateChange(inputState, nextState, timeoutNanoseconds);
}
// ============================================================
@@ -503,6 +516,19 @@
return audioStream->getContentType();
}
+AAUDIO_API aaudio_spatialization_behavior_t AAudioStream_getSpatializationBehavior(
+ AAudioStream* stream)
+{
+ AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+ return audioStream->getSpatializationBehavior();
+}
+
+AAUDIO_API bool AAudioStream_isContentSpatialized(AAudioStream* stream)
+{
+ AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+ return audioStream->isContentSpatialized();
+}
+
AAUDIO_API aaudio_input_preset_t AAudioStream_getInputPreset(AAudioStream* stream)
{
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index 84133f4..8b7b75e 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -41,6 +41,8 @@
mBufferCapacity = other.mBufferCapacity;
mUsage = other.mUsage;
mContentType = other.mContentType;
+ mSpatializationBehavior = other.mSpatializationBehavior;
+ mIsContentSpatialized = other.mIsContentSpatialized;
mInputPreset = other.mInputPreset;
mAllowedCapturePolicy = other.mAllowedCapturePolicy;
mIsPrivacySensitive = other.mIsPrivacySensitive;
@@ -158,6 +160,19 @@
// break;
}
+ switch (mSpatializationBehavior) {
+ case AAUDIO_UNSPECIFIED:
+ case AAUDIO_SPATIALIZATION_BEHAVIOR_AUTO:
+ case AAUDIO_SPATIALIZATION_BEHAVIOR_NEVER:
+ break; // valid
+ default:
+ ALOGD("spatialization behavior not valid = %d", mSpatializationBehavior);
+ return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ // break;
+ }
+
+ // no validation required for mIsContentSpatialized
+
switch (mInputPreset) {
case AAUDIO_UNSPECIFIED:
case AAUDIO_INPUT_PRESET_GENERIC:
@@ -287,6 +302,8 @@
ALOGD("mBufferCapacity = %6d", mBufferCapacity);
ALOGD("mUsage = %6d", mUsage);
ALOGD("mContentType = %6d", mContentType);
+ ALOGD("mSpatializationBehavior = %6d", mSpatializationBehavior);
+ ALOGD("mIsContentSpatialized = %s", mIsContentSpatialized ? "true" : "false");
ALOGD("mInputPreset = %6d", mInputPreset);
ALOGD("mAllowedCapturePolicy = %6d", mAllowedCapturePolicy);
ALOGD("mIsPrivacySensitive = %s", mIsPrivacySensitive ? "true" : "false");
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.h b/media/libaaudio/src/core/AAudioStreamParameters.h
index 7a6ed3e..cb998bf 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.h
+++ b/media/libaaudio/src/core/AAudioStreamParameters.h
@@ -97,6 +97,22 @@
mContentType = contentType;
}
+ aaudio_spatialization_behavior_t getSpatializationBehavior() const {
+ return mSpatializationBehavior;
+ }
+
+ void setSpatializationBehavior(aaudio_spatialization_behavior_t spatializationBehavior) {
+ mSpatializationBehavior = spatializationBehavior;
+ }
+
+ bool isContentSpatialized() const {
+ return mIsContentSpatialized;
+ }
+
+ void setIsContentSpatialized(bool isSpatialized) {
+ mIsContentSpatialized = isSpatialized;
+ }
+
aaudio_input_preset_t getInputPreset() const {
return mInputPreset;
}
@@ -183,6 +199,9 @@
aaudio_direction_t mDirection = AAUDIO_DIRECTION_OUTPUT;
aaudio_usage_t mUsage = AAUDIO_UNSPECIFIED;
aaudio_content_type_t mContentType = AAUDIO_UNSPECIFIED;
+ aaudio_spatialization_behavior_t mSpatializationBehavior
+ = AAUDIO_UNSPECIFIED;
+ bool mIsContentSpatialized = false;
aaudio_input_preset_t mInputPreset = AAUDIO_UNSPECIFIED;
int32_t mBufferCapacity = AAUDIO_UNSPECIFIED;
aaudio_allowed_capture_policy_t mAllowedCapturePolicy = AAUDIO_UNSPECIFIED;
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index ffc3b9d..73432af 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -92,6 +92,12 @@
if (mContentType == AAUDIO_UNSPECIFIED) {
mContentType = AAUDIO_CONTENT_TYPE_MUSIC;
}
+ mSpatializationBehavior = builder.getSpatializationBehavior();
+ // for consistency with other properties, note UNSPECIFIED is the same as AUTO
+ if (mSpatializationBehavior == AAUDIO_UNSPECIFIED) {
+ mSpatializationBehavior = AAUDIO_SPATIALIZATION_BEHAVIOR_AUTO;
+ }
+ mIsContentSpatialized = builder.isContentSpatialized();
mInputPreset = builder.getInputPreset();
if (mInputPreset == AAUDIO_UNSPECIFIED) {
mInputPreset = AAUDIO_INPUT_PRESET_VOICE_RECOGNITION;
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 8bb9b19..afb8551 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -253,6 +253,14 @@
return mContentType;
}
+ aaudio_spatialization_behavior_t getSpatializationBehavior() const {
+ return mSpatializationBehavior;
+ }
+
+ bool isContentSpatialized() const {
+ return mIsContentSpatialized;
+ }
+
aaudio_input_preset_t getInputPreset() const {
return mInputPreset;
}
@@ -594,6 +602,14 @@
mContentType = contentType;
}
+ void setSpatializationBehavior(aaudio_spatialization_behavior_t spatializationBehavior) {
+ mSpatializationBehavior = spatializationBehavior;
+ }
+
+ void setIsContentSpatialized(bool isContentSpatialized) {
+ mIsContentSpatialized = isContentSpatialized;
+ }
+
/**
* This should not be called after the open() call.
*/
@@ -651,6 +667,8 @@
aaudio_usage_t mUsage = AAUDIO_UNSPECIFIED;
aaudio_content_type_t mContentType = AAUDIO_UNSPECIFIED;
+ aaudio_spatialization_behavior_t mSpatializationBehavior = AAUDIO_UNSPECIFIED;
+ bool mIsContentSpatialized = false;
aaudio_input_preset_t mInputPreset = AAUDIO_UNSPECIFIED;
aaudio_allowed_capture_policy_t mAllowedCapturePolicy = AAUDIO_ALLOW_CAPTURE_BY_ALL;
bool mIsPrivacySensitive = false;
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index a8fd0d9..2be3d65 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -20,9 +20,14 @@
#include <new>
#include <stdint.h>
+#include <vector>
#include <aaudio/AAudio.h>
#include <aaudio/AAudioTesting.h>
+#include <android/media/audio/common/AudioMMapPolicy.h>
+#include <android/media/audio/common/AudioMMapPolicyInfo.h>
+#include <android/media/audio/common/AudioMMapPolicyType.h>
+#include <media/AudioSystem.h>
#include "binding/AAudioBinderClient.h"
#include "client/AudioStreamInternalCapture.h"
@@ -35,6 +40,10 @@
using namespace aaudio;
+using android::media::audio::common::AudioMMapPolicy;
+using android::media::audio::common::AudioMMapPolicyInfo;
+using android::media::audio::common::AudioMMapPolicyType;
+
#define AAUDIO_MMAP_POLICY_DEFAULT AAUDIO_POLICY_NEVER
#define AAUDIO_MMAP_EXCLUSIVE_POLICY_DEFAULT AAUDIO_POLICY_NEVER
@@ -86,6 +95,37 @@
return result;
}
+namespace {
+
+aaudio_policy_t aidl2legacy_aaudio_policy(AudioMMapPolicy aidl) {
+ switch (aidl) {
+ case AudioMMapPolicy::NEVER:
+ return AAUDIO_POLICY_NEVER;
+ case AudioMMapPolicy::AUTO:
+ return AAUDIO_POLICY_AUTO;
+ case AudioMMapPolicy::ALWAYS:
+ return AAUDIO_POLICY_ALWAYS;
+ case AudioMMapPolicy::UNSPECIFIED:
+ default:
+ return AAUDIO_UNSPECIFIED;
+ }
+}
+
+// The aaudio policy will be ALWAYS, NEVER, UNSPECIFIED only when all policy info are
+// ALWAYS, NEVER or UNSPECIFIED. Otherwise, the aaudio policy will be AUTO.
+aaudio_policy_t getAAudioPolicy(
+ const std::vector<AudioMMapPolicyInfo>& policyInfos) {
+ if (policyInfos.empty()) return AAUDIO_POLICY_AUTO;
+ for (size_t i = 1; i < policyInfos.size(); ++i) {
+ if (policyInfos.at(i).mmapPolicy != policyInfos.at(0).mmapPolicy) {
+ return AAUDIO_POLICY_AUTO;
+ }
+ }
+ return aidl2legacy_aaudio_policy(policyInfos.at(0).mmapPolicy);
+}
+
+} // namespace
+
// Try to open using MMAP path if that is allowed.
// Fall back to Legacy path if MMAP not available.
// Exact behavior is controlled by MMapPolicy.
@@ -104,25 +144,32 @@
return result;
}
+ std::vector<AudioMMapPolicyInfo> policyInfos;
// The API setting is the highest priority.
aaudio_policy_t mmapPolicy = AudioGlobal_getMMapPolicy();
// If not specified then get from a system property.
- if (mmapPolicy == AAUDIO_UNSPECIFIED) {
- mmapPolicy = AAudioProperty_getMMapPolicy();
+ if (mmapPolicy == AAUDIO_UNSPECIFIED && android::AudioSystem::getMmapPolicyInfo(
+ AudioMMapPolicyType::DEFAULT, &policyInfos) == NO_ERROR) {
+ mmapPolicy = getAAudioPolicy(policyInfos);
}
// If still not specified then use the default.
if (mmapPolicy == AAUDIO_UNSPECIFIED) {
mmapPolicy = AAUDIO_MMAP_POLICY_DEFAULT;
}
- int32_t mapExclusivePolicy = AAudioProperty_getMMapExclusivePolicy();
- if (mapExclusivePolicy == AAUDIO_UNSPECIFIED) {
- mapExclusivePolicy = AAUDIO_MMAP_EXCLUSIVE_POLICY_DEFAULT;
+ policyInfos.clear();
+ aaudio_policy_t mmapExclusivePolicy = AAUDIO_UNSPECIFIED;
+ if (android::AudioSystem::getMmapPolicyInfo(
+ AudioMMapPolicyType::EXCLUSIVE, &policyInfos) == NO_ERROR) {
+ mmapExclusivePolicy = getAAudioPolicy(policyInfos);
+ }
+ if (mmapExclusivePolicy == AAUDIO_UNSPECIFIED) {
+ mmapExclusivePolicy = AAUDIO_MMAP_EXCLUSIVE_POLICY_DEFAULT;
}
aaudio_sharing_mode_t sharingMode = getSharingMode();
if ((sharingMode == AAUDIO_SHARING_MODE_EXCLUSIVE)
- && (mapExclusivePolicy == AAUDIO_POLICY_NEVER)) {
+ && (mmapExclusivePolicy == AAUDIO_POLICY_NEVER)) {
ALOGD("%s() EXCLUSIVE sharing mode not supported. Use SHARED.", __func__);
sharingMode = AAUDIO_SHARING_MODE_SHARED;
setSharingMode(sharingMode);
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 118c004..b512b48 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -135,7 +135,9 @@
const audio_usage_t usage =
AAudioConvert_usageToInternal(builder.getUsage());
const audio_flags_mask_t attributesFlags =
- AAudioConvert_allowCapturePolicyToAudioFlagsMask(builder.getAllowedCapturePolicy());
+ AAudioConvert_allowCapturePolicyToAudioFlagsMask(builder.getAllowedCapturePolicy(),
+ builder.getSpatializationBehavior(),
+ builder.isContentSpatialized());
const audio_attributes_t attributes = {
.content_type = contentType,
@@ -215,7 +217,6 @@
mBlockAdapter = nullptr;
}
- setState(AAUDIO_STREAM_STATE_OPEN);
setDeviceId(mAudioTrack->getRoutedDeviceId());
aaudio_session_id_t actualSessionId =
@@ -248,6 +249,19 @@
"open() perfMode changed from %d to %d",
perfMode, actualPerformanceMode);
+ if (getState() != AAUDIO_STREAM_STATE_UNINITIALIZED) {
+ ALOGE("%s - Open canceled since state = %d", __func__, getState());
+ if (getState() == AAUDIO_STREAM_STATE_DISCONNECTED)
+ {
+ ALOGE("%s - Opening while state is disconnected", __func__);
+ safeReleaseClose();
+ return AAUDIO_ERROR_DISCONNECTED;
+ }
+ safeReleaseClose();
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+
+ setState(AAUDIO_STREAM_STATE_OPEN);
return AAUDIO_OK;
}
diff --git a/media/libaaudio/src/libaaudio.map.txt b/media/libaaudio/src/libaaudio.map.txt
index 8fa9e38..f45b816 100644
--- a/media/libaaudio/src/libaaudio.map.txt
+++ b/media/libaaudio/src/libaaudio.map.txt
@@ -26,6 +26,8 @@
AAudioStreamBuilder_setPackageName; # introduced=31
AAudioStreamBuilder_setAttributionTag; # introduced=31
AAudioStreamBuilder_setChannelMask; # introduced=32
+ AAudioStreamBuilder_setSpatializationBehavior; # introduced=32
+ AAudioStreamBuilder_setIsContentSpatialized; # introduced=32
AAudioStreamBuilder_openStream;
AAudioStreamBuilder_delete;
AAudioStream_close;
@@ -63,6 +65,8 @@
AAudioStream_isPrivacySensitive; # introduced=30
AAudioStream_release; # introduced=30
AAudioStream_getChannelMask; # introduced=32
+ AAudioStream_getSpatializationBehavior; # introduced=32
+ AAudioStream_isContentSpatialized; # introduced=32
local:
*;
};
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index d829934..a0952fe 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -235,20 +235,46 @@
}
audio_flags_mask_t AAudioConvert_allowCapturePolicyToAudioFlagsMask(
- aaudio_allowed_capture_policy_t policy) {
+ aaudio_allowed_capture_policy_t policy,
+ aaudio_spatialization_behavior_t spatializationBehavior,
+ bool isContentSpatialized) {
+ audio_flags_mask_t flagsMask = AUDIO_FLAG_NONE;
switch (policy) {
case AAUDIO_UNSPECIFIED:
case AAUDIO_ALLOW_CAPTURE_BY_ALL:
- return AUDIO_FLAG_NONE;
+ // flagsMask is not modified
+ break;
case AAUDIO_ALLOW_CAPTURE_BY_SYSTEM:
- return AUDIO_FLAG_NO_MEDIA_PROJECTION;
+ flagsMask = static_cast<audio_flags_mask_t>(flagsMask | AUDIO_FLAG_NO_MEDIA_PROJECTION);
+ break;
case AAUDIO_ALLOW_CAPTURE_BY_NONE:
- return static_cast<audio_flags_mask_t>(
+ flagsMask = static_cast<audio_flags_mask_t>(flagsMask |
AUDIO_FLAG_NO_MEDIA_PROJECTION | AUDIO_FLAG_NO_SYSTEM_CAPTURE);
+ break;
default:
- ALOGE("%s() 0x%08X unrecognized", __func__, policy);
- return AUDIO_FLAG_NONE; //
+ ALOGE("%s() 0x%08X unrecognized capture policy", __func__, policy);
+ // flagsMask is not modified
}
+
+ switch (spatializationBehavior) {
+ case AAUDIO_UNSPECIFIED:
+ case AAUDIO_SPATIALIZATION_BEHAVIOR_AUTO:
+ // flagsMask is not modified
+ break;
+ case AAUDIO_SPATIALIZATION_BEHAVIOR_NEVER:
+ flagsMask = static_cast<audio_flags_mask_t>(flagsMask | AUDIO_FLAG_NEVER_SPATIALIZE);
+ break;
+ default:
+ ALOGE("%s() 0x%08X unrecognized spatialization behavior",
+ __func__, spatializationBehavior);
+ // flagsMask is not modified
+ }
+
+ if (isContentSpatialized) {
+ flagsMask = static_cast<audio_flags_mask_t>(flagsMask | AUDIO_FLAG_CONTENT_SPATIALIZED);
+ }
+
+ return flagsMask;
}
audio_flags_mask_t AAudioConvert_privacySensitiveToAudioFlagsMask(
@@ -518,45 +544,6 @@
return AAUDIO_OK;
}
-static int32_t AAudioProperty_getMMapProperty(const char *propName,
- int32_t defaultValue,
- const char * caller) {
- int32_t prop = property_get_int32(propName, defaultValue);
- switch (prop) {
- case AAUDIO_UNSPECIFIED:
- case AAUDIO_POLICY_NEVER:
- case AAUDIO_POLICY_ALWAYS:
- case AAUDIO_POLICY_AUTO:
- break;
- default:
- ALOGE("%s: invalid = %d", caller, prop);
- prop = defaultValue;
- break;
- }
- return prop;
-}
-
-int32_t AAudioProperty_getMMapPolicy() {
- return AAudioProperty_getMMapProperty(AAUDIO_PROP_MMAP_POLICY,
- AAUDIO_UNSPECIFIED, __func__);
-}
-
-int32_t AAudioProperty_getMMapExclusivePolicy() {
- return AAudioProperty_getMMapProperty(AAUDIO_PROP_MMAP_EXCLUSIVE_POLICY,
- AAUDIO_UNSPECIFIED, __func__);
-}
-
-int32_t AAudioProperty_getMixerBursts() {
- const int32_t defaultBursts = 2; // arbitrary, use 2 for double buffered
- const int32_t maxBursts = 1024; // arbitrary
- int32_t prop = property_get_int32(AAUDIO_PROP_MIXER_BURSTS, defaultBursts);
- if (prop < 1 || prop > maxBursts) {
- ALOGE("AAudioProperty_getMixerBursts: invalid = %d", prop);
- prop = defaultBursts;
- }
- return prop;
-}
-
int32_t AAudioProperty_getWakeupDelayMicros() {
const int32_t minMicros = 0; // arbitrary
const int32_t defaultMicros = 200; // arbitrary, based on some observed jitter
@@ -587,18 +574,6 @@
return prop;
}
-int32_t AAudioProperty_getHardwareBurstMinMicros() {
- const int32_t defaultMicros = 1000; // arbitrary
- const int32_t maxMicros = 1000 * 1000; // arbitrary
- int32_t prop = property_get_int32(AAUDIO_PROP_HW_BURST_MIN_USEC, defaultMicros);
- if (prop < 1 || prop > maxMicros) {
- ALOGE("AAudioProperty_getHardwareBurstMinMicros: invalid = %d, use %d",
- prop, defaultMicros);
- prop = defaultMicros;
- }
- return prop;
-}
-
static int32_t AAudioProperty_getMMapOffsetMicros(const char *functionName,
const char *propertyName) {
const int32_t minMicros = -20000; // arbitrary
diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index 52e57da..b59ce1c 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h
@@ -91,7 +91,9 @@
* @return internal audio flags mask
*/
audio_flags_mask_t AAudioConvert_allowCapturePolicyToAudioFlagsMask(
- aaudio_allowed_capture_policy_t policy);
+ aaudio_allowed_capture_policy_t policy,
+ aaudio_spatialization_behavior_t spatializationBehavior,
+ bool isContentSpatialized);
audio_flags_mask_t AAudioConvert_privacySensitiveToAudioFlagsMask(
bool privacySensitive);
@@ -126,27 +128,6 @@
// Note that this code may be replaced by Settings or by some other system configuration tool.
/**
- * Read system property.
- * @return AAUDIO_UNSPECIFIED, AAUDIO_POLICY_NEVER or AAUDIO_POLICY_AUTO or AAUDIO_POLICY_ALWAYS
- */
-int32_t AAudioProperty_getMMapPolicy();
-#define AAUDIO_PROP_MMAP_POLICY "aaudio.mmap_policy"
-
-/**
- * Read system property.
- * @return AAUDIO_UNSPECIFIED, AAUDIO_POLICY_NEVER or AAUDIO_POLICY_AUTO or AAUDIO_POLICY_ALWAYS
- */
-int32_t AAudioProperty_getMMapExclusivePolicy();
-#define AAUDIO_PROP_MMAP_EXCLUSIVE_POLICY "aaudio.mmap_exclusive_policy"
-
-/**
- * Read system property.
- * @return number of bursts per AAudio service mixer cycle
- */
-int32_t AAudioProperty_getMixerBursts();
-#define AAUDIO_PROP_MIXER_BURSTS "aaudio.mixer_bursts"
-
-/**
* Read a system property that specifies the number of extra microseconds that a thread
* should sleep when waiting for another thread to service a FIFO. This is used
* to avoid the waking thread from being overly optimistic about the other threads
@@ -167,19 +148,6 @@
#define AAUDIO_PROP_MINIMUM_SLEEP_USEC "aaudio.minimum_sleep_usec"
/**
- * Read system property.
- * This is handy in case the DMA is bursting too quickly for the CPU to keep up.
- * For example, there may be a DMA burst every 100 usec but you only
- * want to feed the MMAP buffer every 2000 usec.
- *
- * This will affect the framesPerBurst for an MMAP stream.
- *
- * @return minimum number of microseconds for a MMAP HW burst
- */
-int32_t AAudioProperty_getHardwareBurstMinMicros();
-#define AAUDIO_PROP_HW_BURST_MIN_USEC "aaudio.hw_burst_min_usec"
-
-/**
* Read a system property that specifies an offset that will be added to MMAP timestamps.
* This can be used to correct bias in the timestamp.
* It can also be used to analyze the time distribution of the timestamp
diff --git a/media/libaaudio/tests/Android.bp b/media/libaaudio/tests/Android.bp
index 98e9727..ea00a5a 100644
--- a/media/libaaudio/tests/Android.bp
+++ b/media/libaaudio/tests/Android.bp
@@ -48,7 +48,7 @@
shared_libs: ["libaaudio_internal"],
}
-cc_test {
+cc_binary {
name: "test_timestamps",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_timestamps.cpp"],
@@ -60,121 +60,71 @@
name: "test_open_params",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_open_params.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
-cc_test {
+cc_binary {
name: "test_no_close",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_no_close.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
-cc_test {
+cc_binary {
name: "test_aaudio_recovery",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_recovery.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
-cc_test {
+cc_binary {
name: "test_n_streams",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_n_streams.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
-cc_test {
+cc_binary {
name: "test_bad_disconnect",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_bad_disconnect.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
cc_test {
name: "test_various",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_various.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
cc_test {
name: "test_session_id",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_session_id.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
-cc_test {
+cc_binary {
name: "test_aaudio_monkey",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_aaudio_monkey.cpp"],
header_libs: ["libaaudio_example_utils"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
cc_test {
name: "test_attributes",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_attributes.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
cc_test {
name: "test_interference",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_interference.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
cc_test {
@@ -196,28 +146,18 @@
],
}
-cc_test {
+cc_binary {
name: "test_return_stop",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_return_stop.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
cc_test {
name: "test_callback_race",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_callback_race.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
cc_test {
@@ -238,7 +178,7 @@
],
}
-cc_test {
+cc_binary {
name: "test_steal_exclusive",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_steal_exclusive.cpp"],
@@ -251,15 +191,9 @@
],
}
-
-cc_test {
+cc_binary {
name: "test_disconnect_race",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_disconnect_race.cpp"],
- shared_libs: [
- "libaaudio",
- "libbinder",
- "libcutils",
- "libutils",
- ],
+ shared_libs: ["libaaudio"],
}
diff --git a/media/libaaudio/tests/test_attributes.cpp b/media/libaaudio/tests/test_attributes.cpp
index d540866..b88d562 100644
--- a/media/libaaudio/tests/test_attributes.cpp
+++ b/media/libaaudio/tests/test_attributes.cpp
@@ -16,6 +16,10 @@
// Test AAudio attributes such as Usage, ContentType and InputPreset.
+// TODO Many of these tests are duplicates of CTS tests in
+// "test_aaudio_attributes.cpp". That other file is more current.
+// So these tests could be deleted.
+
#include <stdio.h>
#include <unistd.h>
@@ -91,7 +95,7 @@
aaudio_allowed_capture_policy_t expectedCapturePolicy =
(capturePolicy == DONT_SET || capturePolicy == AAUDIO_UNSPECIFIED)
? AAUDIO_ALLOW_CAPTURE_BY_ALL // default
- : preset;
+ : capturePolicy;
EXPECT_EQ(expectedCapturePolicy, AAudioStream_getAllowedCapturePolicy(aaudioStream));
bool expectedPrivacyMode =
@@ -132,10 +136,7 @@
AAUDIO_USAGE_ASSISTANCE_SONIFICATION,
AAUDIO_USAGE_GAME,
AAUDIO_USAGE_ASSISTANT,
- AAUDIO_SYSTEM_USAGE_EMERGENCY,
- AAUDIO_SYSTEM_USAGE_SAFETY,
- AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS,
- AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT
+ // Note that the AAUDIO_SYSTEM_USAGE_* values requires special permission.
};
static const aaudio_content_type_t sContentypes[] = {
diff --git a/media/libaaudio/tests/test_steal_exclusive.cpp b/media/libaaudio/tests/test_steal_exclusive.cpp
index 05c560d..5cb005c 100644
--- a/media/libaaudio/tests/test_steal_exclusive.cpp
+++ b/media/libaaudio/tests/test_steal_exclusive.cpp
@@ -110,6 +110,10 @@
mOpenDelayMillis = openDelayMillis;
}
+ void setCloseEnabled(bool enabled) {
+ mCloseEnabled = enabled;
+ }
+
void restartStream() {
int retriesLeft = mMaxRetries;
aaudio_result_t result;
@@ -189,10 +193,12 @@
std::lock_guard<std::mutex> lock(mLock);
aaudio_result_t result = AAUDIO_OK;
if (mStream != nullptr) {
- result = AAudioStream_close(mStream);
- if (result != AAUDIO_OK) {
- printf("AAudioStream_close returned %s\n",
- AAudio_convertResultToText(result));
+ if (mCloseEnabled) {
+ result = AAudioStream_close(mStream);
+ printf("AAudioStream_close() returned %s\n",
+ AAudio_convertResultToText(result));
+ } else {
+ printf("AAudioStream_close() DISABLED!\n");
}
mStream = nullptr;
}
@@ -287,6 +293,7 @@
std::string mName;
int mMaxRetries = 1;
int mOpenDelayMillis = 0;
+ bool mCloseEnabled = true;
};
// Callback function that fills the audio output buffer.
@@ -319,11 +326,12 @@
}
static void s_usage() {
- printf("test_steal_exclusive [-i] [-r{maxRetries}] [-d{delay}] -s\n");
+ printf("test_steal_exclusive [-i] [-r{maxRetries}] [-d{delay}] -s -c{flag}\n");
printf(" -i direction INPUT, otherwise OUTPUT\n");
- printf(" -d delay open by milliseconds, default = 0\n");
- printf(" -r max retries in the error callback, default = 1\n");
+ printf(" -d Delay open by milliseconds, default = 0\n");
+ printf(" -r max Retries in the error callback, default = 1\n");
printf(" -s try to open in SHARED mode\n");
+ printf(" -c enable or disabling Closing of the stream with 0/1, default = 1\n");
}
int main(int argc, char ** argv) {
@@ -334,6 +342,7 @@
int errorCount = 0;
int maxRetries = 1;
int openDelayMillis = 0;
+ bool closeEnabled = true;
aaudio_sharing_mode_t requestedSharingMode = AAUDIO_SHARING_MODE_EXCLUSIVE;
// Make printf print immediately so that debug info is not stuck
@@ -348,6 +357,9 @@
if (arg[0] == '-') {
char option = arg[1];
switch (option) {
+ case 'c':
+ closeEnabled = atoi(&arg[2]) != 0;
+ break;
case 'd':
openDelayMillis = atoi(&arg[2]);
break;
@@ -376,6 +388,8 @@
thief.setOpenDelayMillis(openDelayMillis);
victim.setMaxRetries(maxRetries);
thief.setMaxRetries(maxRetries);
+ victim.setCloseEnabled(closeEnabled);
+ thief.setCloseEnabled(closeEnabled);
result = victim.openAudioStream(direction, requestedSharingMode);
if (result != AAUDIO_OK) {
@@ -442,7 +456,7 @@
}
LOGI("Both streams running. Ask user to plug in headset. ====");
- printf("\n====\nPlease PLUG IN A HEADSET now!\n====\n\n");
+ printf("\n====\nPlease PLUG IN A HEADSET now! - OPTIONAL\n====\n\n");
if (result == AAUDIO_OK) {
const int watchLoops = DUET_DURATION_MSEC / SLEEP_DURATION_MSEC;
diff --git a/media/libaaudio/tests/test_various.cpp b/media/libaaudio/tests/test_various.cpp
index cbf863f..b68fc7b 100644
--- a/media/libaaudio/tests/test_various.cpp
+++ b/media/libaaudio/tests/test_various.cpp
@@ -25,6 +25,7 @@
#include <gtest/gtest.h>
#include <unistd.h>
+#include <thread>
// Callback function that does nothing.
aaudio_data_callback_result_t NoopDataCallbackProc(
@@ -51,6 +52,7 @@
}
constexpr int64_t NANOS_PER_MILLISECOND = 1000 * 1000;
+constexpr int64_t MICROS_PER_MILLISECOND = 1000;
void checkReleaseThenClose(aaudio_performance_mode_t perfMode,
aaudio_sharing_mode_t sharingMode,
@@ -762,6 +764,58 @@
checkCallbackOnce(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
}
+void waitForStateChangeToClosingorClosed(AAudioStream **stream, std::atomic<bool>* isReady)
+{
+ *isReady = true;
+ aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(*stream,
+ AAUDIO_STREAM_STATE_OPEN, &state,
+ 10000 * NANOS_PER_MILLISECOND));
+ if ((state != AAUDIO_STREAM_STATE_CLOSING) && (state != AAUDIO_STREAM_STATE_CLOSED)){
+ FAIL() << "ERROR - State not closing or closed. Current state: " <<
+ AAudio_convertStreamStateToText(state);
+ }
+}
+
+void testWaitForStateChangeClose(aaudio_performance_mode_t perfMode) {
+ AAudioStreamBuilder *aaudioBuilder = nullptr;
+ AAudioStream *aaudioStream = nullptr;
+
+ ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+ AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+ ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
+
+ // Verify Open State
+ aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+ AAUDIO_STREAM_STATE_UNKNOWN, &state,
+ 1000 * NANOS_PER_MILLISECOND));
+ EXPECT_EQ(AAUDIO_STREAM_STATE_OPEN, state);
+
+ std::atomic<bool> isWaitThreadReady{false};
+
+ // Spawn a new thread to wait for the state change
+ std::thread waitThread (waitForStateChangeToClosingorClosed, &aaudioStream,
+ &isWaitThreadReady);
+
+ // Wait for worker thread to be ready
+ while (!isWaitThreadReady) {
+ usleep(MICROS_PER_MILLISECOND);
+ }
+ // Sleep an additional millisecond to make sure waitForAudioThread is called
+ usleep(MICROS_PER_MILLISECOND);
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
+ waitThread.join();
+}
+
+TEST(test_various, wait_for_state_change_close_none) {
+ testWaitForStateChangeClose(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_various, wait_for_state_change_close_lowlat) {
+ testWaitForStateChangeClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}
+
// ************************************************************
struct WakeUpCallbackData {
void wakeOther() {
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
index b0eb7bd..5f63a69 100644
--- a/media/libaudioclient/AidlConversion.cpp
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -34,6 +34,36 @@
namespace android {
using base::unexpected;
+using media::audio::common::AudioChannelLayout;
+using media::audio::common::AudioConfig;
+using media::audio::common::AudioConfigBase;
+using media::audio::common::AudioContentType;
+using media::audio::common::AudioDevice;
+using media::audio::common::AudioDeviceAddress;
+using media::audio::common::AudioDeviceDescription;
+using media::audio::common::AudioDeviceType;
+using media::audio::common::AudioEncapsulationMetadataType;
+using media::audio::common::AudioEncapsulationMode;
+using media::audio::common::AudioEncapsulationType;
+using media::audio::common::AudioFormatDescription;
+using media::audio::common::AudioFormatType;
+using media::audio::common::AudioGain;
+using media::audio::common::AudioGainConfig;
+using media::audio::common::AudioGainMode;
+using media::audio::common::AudioMode;
+using media::audio::common::AudioOffloadInfo;
+using media::audio::common::AudioPortExt;
+using media::audio::common::AudioPortMixExt;
+using media::audio::common::AudioPortMixExtUseCase;
+using media::audio::common::AudioProfile;
+using media::audio::common::AudioSource;
+using media::audio::common::AudioStandard;
+using media::audio::common::AudioStreamType;
+using media::audio::common::AudioUsage;
+using media::audio::common::AudioUuid;
+using media::audio::common::ExtraAudioDescriptor;
+using media::audio::common::Int;
+using media::audio::common::PcmType;
namespace {
@@ -225,60 +255,6 @@
return std::string(legacy.c_str());
}
-// The legacy enum is unnamed. Thus, we use int32_t.
-ConversionResult<int32_t> aidl2legacy_AudioPortConfigType_int32_t(
- media::AudioPortConfigType aidl) {
- switch (aidl) {
- case media::AudioPortConfigType::SAMPLE_RATE:
- return AUDIO_PORT_CONFIG_SAMPLE_RATE;
- case media::AudioPortConfigType::CHANNEL_MASK:
- return AUDIO_PORT_CONFIG_CHANNEL_MASK;
- case media::AudioPortConfigType::FORMAT:
- return AUDIO_PORT_CONFIG_FORMAT;
- case media::AudioPortConfigType::GAIN:
- return AUDIO_PORT_CONFIG_GAIN;
- case media::AudioPortConfigType::FLAGS:
- return AUDIO_PORT_CONFIG_FLAGS;
- }
- return unexpected(BAD_VALUE);
-}
-
-// The legacy enum is unnamed. Thus, we use int32_t.
-ConversionResult<media::AudioPortConfigType> legacy2aidl_int32_t_AudioPortConfigType(
- int32_t legacy) {
- switch (legacy) {
- case AUDIO_PORT_CONFIG_SAMPLE_RATE:
- return media::AudioPortConfigType::SAMPLE_RATE;
- case AUDIO_PORT_CONFIG_CHANNEL_MASK:
- return media::AudioPortConfigType::CHANNEL_MASK;
- case AUDIO_PORT_CONFIG_FORMAT:
- return media::AudioPortConfigType::FORMAT;
- case AUDIO_PORT_CONFIG_GAIN:
- return media::AudioPortConfigType::GAIN;
- case AUDIO_PORT_CONFIG_FLAGS:
- return media::AudioPortConfigType::FLAGS;
- }
- return unexpected(BAD_VALUE);
-}
-
-ConversionResult<unsigned int> aidl2legacy_int32_t_config_mask(int32_t aidl) {
- return convertBitmask<unsigned int, int32_t, int, media::AudioPortConfigType>(
- aidl, aidl2legacy_AudioPortConfigType_int32_t,
- // AudioPortConfigType enum is index-based.
- indexToEnum_index<media::AudioPortConfigType>,
- // AUDIO_PORT_CONFIG_* flags are mask-based.
- enumToMask_bitmask<unsigned int, int>);
-}
-
-ConversionResult<int32_t> legacy2aidl_config_mask_int32_t(unsigned int legacy) {
- return convertBitmask<int32_t, unsigned int, media::AudioPortConfigType, int>(
- legacy, legacy2aidl_int32_t_AudioPortConfigType,
- // AUDIO_PORT_CONFIG_* flags are mask-based.
- indexToEnum_bitmask<unsigned>,
- // AudioPortConfigType enum is index-based.
- enumToMask_index<int32_t, media::AudioPortConfigType>);
-}
-
ConversionResult<audio_io_config_event_t> aidl2legacy_AudioIoConfigEvent_audio_io_config_event_t(
media::AudioIoConfigEvent aidl) {
switch (aidl) {
@@ -388,21 +364,21 @@
namespace {
namespace detail {
-using AudioChannelPair = std::pair<audio_channel_mask_t, media::AudioChannelLayout>;
+using AudioChannelPair = std::pair<audio_channel_mask_t, AudioChannelLayout>;
using AudioChannelPairs = std::vector<AudioChannelPair>;
-using AudioDevicePair = std::pair<audio_devices_t, media::AudioDeviceDescription>;
+using AudioDevicePair = std::pair<audio_devices_t, AudioDeviceDescription>;
using AudioDevicePairs = std::vector<AudioDevicePair>;
-using AudioFormatPair = std::pair<audio_format_t, media::AudioFormatDescription>;
+using AudioFormatPair = std::pair<audio_format_t, AudioFormatDescription>;
using AudioFormatPairs = std::vector<AudioFormatPair>;
}
const detail::AudioChannelPairs& getInAudioChannelPairs() {
static const detail::AudioChannelPairs pairs = {
-#define DEFINE_INPUT_LAYOUT(n) \
- { \
- AUDIO_CHANNEL_IN_##n, \
- media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::layoutMask>( \
- media::AudioChannelLayout::LAYOUT_##n) \
+#define DEFINE_INPUT_LAYOUT(n) \
+ { \
+ AUDIO_CHANNEL_IN_##n, \
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>( \
+ AudioChannelLayout::LAYOUT_##n) \
}
DEFINE_INPUT_LAYOUT(MONO),
@@ -421,11 +397,11 @@
const detail::AudioChannelPairs& getOutAudioChannelPairs() {
static const detail::AudioChannelPairs pairs = {
-#define DEFINE_OUTPUT_LAYOUT(n) \
- { \
- AUDIO_CHANNEL_OUT_##n, \
- media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::layoutMask>( \
- media::AudioChannelLayout::LAYOUT_##n) \
+#define DEFINE_OUTPUT_LAYOUT(n) \
+ { \
+ AUDIO_CHANNEL_OUT_##n, \
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>( \
+ AudioChannelLayout::LAYOUT_##n) \
}
DEFINE_OUTPUT_LAYOUT(MONO),
@@ -464,11 +440,11 @@
const detail::AudioChannelPairs& getVoiceAudioChannelPairs() {
static const detail::AudioChannelPairs pairs = {
-#define DEFINE_VOICE_LAYOUT(n) \
- { \
- AUDIO_CHANNEL_IN_VOICE_##n, \
- media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::voiceMask>( \
- media::AudioChannelLayout::VOICE_##n) \
+#define DEFINE_VOICE_LAYOUT(n) \
+ { \
+ AUDIO_CHANNEL_IN_VOICE_##n, \
+ AudioChannelLayout::make<AudioChannelLayout::Tag::voiceMask>( \
+ AudioChannelLayout::VOICE_##n) \
}
DEFINE_VOICE_LAYOUT(UPLINK_MONO),
DEFINE_VOICE_LAYOUT(DNLINK_MONO),
@@ -478,9 +454,9 @@
return pairs;
}
-media::AudioDeviceDescription make_AudioDeviceDescription(media::AudioDeviceType type,
+AudioDeviceDescription make_AudioDeviceDescription(AudioDeviceType type,
const std::string& connection = "") {
- media::AudioDeviceDescription result;
+ AudioDeviceDescription result;
result.type = type;
result.connection = connection;
return result;
@@ -488,7 +464,7 @@
void append_AudioDeviceDescription(detail::AudioDevicePairs& pairs,
audio_devices_t inputType, audio_devices_t outputType,
- media::AudioDeviceType inType, media::AudioDeviceType outType,
+ AudioDeviceType inType, AudioDeviceType outType,
const std::string& connection = "") {
pairs.push_back(std::make_pair(inputType, make_AudioDeviceDescription(inType, connection)));
pairs.push_back(std::make_pair(outputType, make_AudioDeviceDescription(outType, connection)));
@@ -498,198 +474,198 @@
static const detail::AudioDevicePairs pairs = []() {
detail::AudioDevicePairs pairs = {{
{
- AUDIO_DEVICE_NONE, media::AudioDeviceDescription{}
+ AUDIO_DEVICE_NONE, AudioDeviceDescription{}
},
{
AUDIO_DEVICE_OUT_EARPIECE, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_SPEAKER_EARPIECE)
+ AudioDeviceType::OUT_SPEAKER_EARPIECE)
},
{
AUDIO_DEVICE_OUT_SPEAKER, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_SPEAKER)
+ AudioDeviceType::OUT_SPEAKER)
},
{
AUDIO_DEVICE_OUT_WIRED_HEADPHONE, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_HEADPHONE,
- media::AudioDeviceDescription::CONNECTION_ANALOG())
+ AudioDeviceType::OUT_HEADPHONE,
+ AudioDeviceDescription::CONNECTION_ANALOG())
},
{
AUDIO_DEVICE_OUT_BLUETOOTH_SCO, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_BT_SCO())
+ AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_BT_SCO())
},
{
AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_CARKIT,
- media::AudioDeviceDescription::CONNECTION_BT_SCO())
+ AudioDeviceType::OUT_CARKIT,
+ AudioDeviceDescription::CONNECTION_BT_SCO())
},
{
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_HEADPHONE,
- media::AudioDeviceDescription::CONNECTION_BT_A2DP())
+ AudioDeviceType::OUT_HEADPHONE,
+ AudioDeviceDescription::CONNECTION_BT_A2DP())
},
{
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_SPEAKER,
- media::AudioDeviceDescription::CONNECTION_BT_A2DP())
+ AudioDeviceType::OUT_SPEAKER,
+ AudioDeviceDescription::CONNECTION_BT_A2DP())
},
{
AUDIO_DEVICE_OUT_TELEPHONY_TX, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_TELEPHONY_TX)
+ AudioDeviceType::OUT_TELEPHONY_TX)
},
{
AUDIO_DEVICE_OUT_AUX_LINE, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_LINE_AUX)
+ AudioDeviceType::OUT_LINE_AUX)
},
{
AUDIO_DEVICE_OUT_SPEAKER_SAFE, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_SPEAKER_SAFE)
+ AudioDeviceType::OUT_SPEAKER_SAFE)
},
{
AUDIO_DEVICE_OUT_HEARING_AID, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_HEARING_AID,
- media::AudioDeviceDescription::CONNECTION_WIRELESS())
+ AudioDeviceType::OUT_HEARING_AID,
+ AudioDeviceDescription::CONNECTION_WIRELESS())
},
{
AUDIO_DEVICE_OUT_ECHO_CANCELLER, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_ECHO_CANCELLER)
+ AudioDeviceType::OUT_ECHO_CANCELLER)
},
{
AUDIO_DEVICE_OUT_BLE_SPEAKER, make_AudioDeviceDescription(
- media::AudioDeviceType::OUT_SPEAKER,
- media::AudioDeviceDescription::CONNECTION_BT_LE())
+ AudioDeviceType::OUT_SPEAKER,
+ AudioDeviceDescription::CONNECTION_BT_LE())
},
// AUDIO_DEVICE_IN_AMBIENT and IN_COMMUNICATION are removed since they were deprecated.
{
AUDIO_DEVICE_IN_BUILTIN_MIC, make_AudioDeviceDescription(
- media::AudioDeviceType::IN_MICROPHONE)
+ AudioDeviceType::IN_MICROPHONE)
},
{
AUDIO_DEVICE_IN_BACK_MIC, make_AudioDeviceDescription(
- media::AudioDeviceType::IN_MICROPHONE_BACK)
+ AudioDeviceType::IN_MICROPHONE_BACK)
},
{
AUDIO_DEVICE_IN_TELEPHONY_RX, make_AudioDeviceDescription(
- media::AudioDeviceType::IN_TELEPHONY_RX)
+ AudioDeviceType::IN_TELEPHONY_RX)
},
{
AUDIO_DEVICE_IN_TV_TUNER, make_AudioDeviceDescription(
- media::AudioDeviceType::IN_TV_TUNER)
+ AudioDeviceType::IN_TV_TUNER)
},
{
AUDIO_DEVICE_IN_LOOPBACK, make_AudioDeviceDescription(
- media::AudioDeviceType::IN_LOOPBACK)
+ AudioDeviceType::IN_LOOPBACK)
},
{
AUDIO_DEVICE_IN_BLUETOOTH_BLE, make_AudioDeviceDescription(
- media::AudioDeviceType::IN_DEVICE,
- media::AudioDeviceDescription::CONNECTION_BT_LE())
+ AudioDeviceType::IN_DEVICE,
+ AudioDeviceDescription::CONNECTION_BT_LE())
},
{
AUDIO_DEVICE_IN_ECHO_REFERENCE, make_AudioDeviceDescription(
- media::AudioDeviceType::IN_ECHO_REFERENCE)
+ AudioDeviceType::IN_ECHO_REFERENCE)
}
}};
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_DEFAULT, AUDIO_DEVICE_OUT_DEFAULT,
- media::AudioDeviceType::IN_DEFAULT, media::AudioDeviceType::OUT_DEFAULT);
+ AudioDeviceType::IN_DEFAULT, AudioDeviceType::OUT_DEFAULT);
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_OUT_WIRED_HEADSET,
- media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
- media::AudioDeviceDescription::CONNECTION_ANALOG());
+ AudioDeviceType::IN_HEADSET, AudioDeviceType::OUT_HEADSET,
+ AudioDeviceDescription::CONNECTION_ANALOG());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET,
- media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
- media::AudioDeviceDescription::CONNECTION_BT_SCO());
+ AudioDeviceType::IN_HEADSET, AudioDeviceType::OUT_HEADSET,
+ AudioDeviceDescription::CONNECTION_BT_SCO());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_HDMI, AUDIO_DEVICE_OUT_HDMI,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_HDMI());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_HDMI());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- media::AudioDeviceType::IN_SUBMIX, media::AudioDeviceType::OUT_SUBMIX);
+ AudioDeviceType::IN_SUBMIX, AudioDeviceType::OUT_SUBMIX);
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,
- media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
- media::AudioDeviceDescription::CONNECTION_ANALOG_DOCK());
+ AudioDeviceType::IN_DOCK, AudioDeviceType::OUT_DOCK,
+ AudioDeviceDescription::CONNECTION_ANALOG());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,
- media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
- media::AudioDeviceDescription::CONNECTION_DIGITAL_DOCK());
+ AudioDeviceType::IN_DOCK, AudioDeviceType::OUT_DOCK,
+ AudioDeviceDescription::CONNECTION_USB());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_USB_ACCESSORY, AUDIO_DEVICE_OUT_USB_ACCESSORY,
- media::AudioDeviceType::IN_ACCESSORY, media::AudioDeviceType::OUT_ACCESSORY,
- media::AudioDeviceDescription::CONNECTION_USB());
+ AudioDeviceType::IN_ACCESSORY, AudioDeviceType::OUT_ACCESSORY,
+ AudioDeviceDescription::CONNECTION_USB());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_OUT_USB_DEVICE,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_USB());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_USB());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_FM_TUNER, AUDIO_DEVICE_OUT_FM,
- media::AudioDeviceType::IN_FM_TUNER, media::AudioDeviceType::OUT_FM);
+ AudioDeviceType::IN_FM_TUNER, AudioDeviceType::OUT_FM);
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_LINE, AUDIO_DEVICE_OUT_LINE,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_ANALOG());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_ANALOG());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_SPDIF, AUDIO_DEVICE_OUT_SPDIF,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_SPDIF());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_SPDIF());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_BT_A2DP());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_BT_A2DP());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_IP, AUDIO_DEVICE_OUT_IP,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_IP_V4());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_IP_V4());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_BUS, AUDIO_DEVICE_OUT_BUS,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_BUS());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_BUS());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_PROXY, AUDIO_DEVICE_OUT_PROXY,
- media::AudioDeviceType::IN_AFE_PROXY, media::AudioDeviceType::OUT_AFE_PROXY);
+ AudioDeviceType::IN_AFE_PROXY, AudioDeviceType::OUT_AFE_PROXY);
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_USB_HEADSET, AUDIO_DEVICE_OUT_USB_HEADSET,
- media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
- media::AudioDeviceDescription::CONNECTION_USB());
+ AudioDeviceType::IN_HEADSET, AudioDeviceType::OUT_HEADSET,
+ AudioDeviceDescription::CONNECTION_USB());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_HDMI_ARC, AUDIO_DEVICE_OUT_HDMI_ARC,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_HDMI_ARC());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_HDMI_ARC());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_HDMI_EARC, AUDIO_DEVICE_OUT_HDMI_EARC,
- media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
- media::AudioDeviceDescription::CONNECTION_HDMI_EARC());
+ AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE,
+ AudioDeviceDescription::CONNECTION_HDMI_EARC());
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_BLE_HEADSET, AUDIO_DEVICE_OUT_BLE_HEADSET,
- media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
- media::AudioDeviceDescription::CONNECTION_BT_LE());
+ AudioDeviceType::IN_HEADSET, AudioDeviceType::OUT_HEADSET,
+ AudioDeviceDescription::CONNECTION_BT_LE());
return pairs;
}();
return pairs;
}
-media::AudioFormatDescription make_AudioFormatDescription(media::AudioFormatType type) {
- media::AudioFormatDescription result;
+AudioFormatDescription make_AudioFormatDescription(AudioFormatType type) {
+ AudioFormatDescription result;
result.type = type;
return result;
}
-media::AudioFormatDescription make_AudioFormatDescription(media::PcmType pcm) {
- auto result = make_AudioFormatDescription(media::AudioFormatType::PCM);
+AudioFormatDescription make_AudioFormatDescription(PcmType pcm) {
+ auto result = make_AudioFormatDescription(AudioFormatType::PCM);
result.pcm = pcm;
return result;
}
-media::AudioFormatDescription make_AudioFormatDescription(const std::string& encoding) {
- media::AudioFormatDescription result;
+AudioFormatDescription make_AudioFormatDescription(const std::string& encoding) {
+ AudioFormatDescription result;
result.encoding = encoding;
return result;
}
-media::AudioFormatDescription make_AudioFormatDescription(media::PcmType transport,
+AudioFormatDescription make_AudioFormatDescription(PcmType transport,
const std::string& encoding) {
auto result = make_AudioFormatDescription(encoding);
result.pcm = transport;
@@ -700,31 +676,30 @@
static const detail::AudioFormatPairs pairs = {{
{
AUDIO_FORMAT_INVALID,
- make_AudioFormatDescription(media::AudioFormatType::SYS_RESERVED_INVALID)
+ make_AudioFormatDescription(AudioFormatType::SYS_RESERVED_INVALID)
},
{
- AUDIO_FORMAT_DEFAULT, media::AudioFormatDescription{}
+ AUDIO_FORMAT_DEFAULT, AudioFormatDescription{}
},
{
- AUDIO_FORMAT_PCM_16_BIT, make_AudioFormatDescription(media::PcmType::INT_16_BIT)
+ AUDIO_FORMAT_PCM_16_BIT, make_AudioFormatDescription(PcmType::INT_16_BIT)
},
{
- AUDIO_FORMAT_PCM_8_BIT, make_AudioFormatDescription(media::PcmType::UINT_8_BIT)
+ AUDIO_FORMAT_PCM_8_BIT, make_AudioFormatDescription(PcmType::UINT_8_BIT)
},
{
- AUDIO_FORMAT_PCM_32_BIT, make_AudioFormatDescription(media::PcmType::INT_32_BIT)
+ AUDIO_FORMAT_PCM_32_BIT, make_AudioFormatDescription(PcmType::INT_32_BIT)
},
{
- AUDIO_FORMAT_PCM_8_24_BIT, make_AudioFormatDescription(media::PcmType::FIXED_Q_8_24)
+ AUDIO_FORMAT_PCM_8_24_BIT, make_AudioFormatDescription(PcmType::FIXED_Q_8_24)
},
{
- AUDIO_FORMAT_PCM_FLOAT, make_AudioFormatDescription(media::PcmType::FLOAT_32_BIT)
+ AUDIO_FORMAT_PCM_FLOAT, make_AudioFormatDescription(PcmType::FLOAT_32_BIT)
},
{
- AUDIO_FORMAT_PCM_24_BIT_PACKED, make_AudioFormatDescription(media::PcmType::INT_24_BIT)
+ AUDIO_FORMAT_PCM_24_BIT_PACKED, make_AudioFormatDescription(PcmType::INT_24_BIT)
},
{
- // See the comment in MediaDefs.h.
AUDIO_FORMAT_MP3, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEG)
},
{
@@ -734,52 +709,41 @@
AUDIO_FORMAT_AMR_WB, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AMR_WB)
},
{
- // Note: in MediaDefs.cpp MEDIA_MIMETYPE_AUDIO_AAC = "audio/mp4a-latm".
- AUDIO_FORMAT_AAC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_FORMAT)
+ AUDIO_FORMAT_AAC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_MP4)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_MAIN, make_AudioFormatDescription("audio/aac.main")
+ AUDIO_FORMAT_AAC_MAIN, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_MAIN)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_LC, make_AudioFormatDescription("audio/aac.lc")
+ AUDIO_FORMAT_AAC_LC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_LC)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_SSR, make_AudioFormatDescription("audio/aac.ssr")
+ AUDIO_FORMAT_AAC_SSR, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_SSR)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_LTP, make_AudioFormatDescription("audio/aac.ltp")
+ AUDIO_FORMAT_AAC_LTP, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_LTP)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_HE_V1, make_AudioFormatDescription("audio/aac.he.v1")
+ AUDIO_FORMAT_AAC_HE_V1, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_HE_V1)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_SCALABLE, make_AudioFormatDescription("audio/aac.scalable")
+ AUDIO_FORMAT_AAC_SCALABLE,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_SCALABLE)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ERLC, make_AudioFormatDescription("audio/aac.erlc")
+ AUDIO_FORMAT_AAC_ERLC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ERLC)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_LD, make_AudioFormatDescription("audio/aac.ld")
+ AUDIO_FORMAT_AAC_LD, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_LD)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_HE_V2, make_AudioFormatDescription("audio/aac.he.v2")
+ AUDIO_FORMAT_AAC_HE_V2, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_HE_V2)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ELD, make_AudioFormatDescription("audio/aac.eld")
+ AUDIO_FORMAT_AAC_ELD, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ELD)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_XHE, make_AudioFormatDescription("audio/aac.xhe")
+ AUDIO_FORMAT_AAC_XHE, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_XHE)
},
// AUDIO_FORMAT_HE_AAC_V1 and HE_AAC_V2 are removed since they were deprecated long time
// ago.
@@ -796,7 +760,6 @@
AUDIO_FORMAT_E_AC3, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EAC3)
},
{
- // Note: not in the IANA registry.
AUDIO_FORMAT_E_AC3_JOC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EAC3_JOC)
},
{
@@ -809,13 +772,12 @@
// nested AudioFormatDescriptions. The legacy 'AUDIO_FORMAT_IEC61937' type doesn't
// specify the format of the encapsulated bitstream.
{
- // Note: not in the IANA registry.
AUDIO_FORMAT_IEC61937,
- make_AudioFormatDescription(media::PcmType::INT_16_BIT, "audio/x-iec61937")
+ make_AudioFormatDescription(PcmType::INT_16_BIT, MEDIA_MIMETYPE_AUDIO_IEC61937)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_DOLBY_TRUEHD, make_AudioFormatDescription("audio/vnd.dolby.truehd")
+ AUDIO_FORMAT_DOLBY_TRUEHD,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DOLBY_TRUEHD)
},
{
AUDIO_FORMAT_EVRC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EVRC)
@@ -830,11 +792,9 @@
AUDIO_FORMAT_EVRCNW, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EVRCNW)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADIF, make_AudioFormatDescription("audio/aac.adif")
+ AUDIO_FORMAT_AAC_ADIF, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADIF)
},
{
- // Note: not in the IANA registry.
AUDIO_FORMAT_WMA, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_WMA)
},
{
@@ -845,7 +805,6 @@
AUDIO_FORMAT_AMR_WB_PLUS, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AMR_WB_PLUS)
},
{
- // Note: not in the IANA registry.
AUDIO_FORMAT_MP2, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEG_LAYER_II)
},
{
@@ -859,7 +818,6 @@
AUDIO_FORMAT_FLAC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_FLAC)
},
{
- // Note: not in the IANA registry.
AUDIO_FORMAT_ALAC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_ALAC)
},
{
@@ -867,52 +825,49 @@
AUDIO_FORMAT_APE, make_AudioFormatDescription("audio/x-ape")
},
{
- // Note: not in the IANA registry.
AUDIO_FORMAT_AAC_ADTS, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_MAIN, make_AudioFormatDescription("audio/aac-adts.main")
+ AUDIO_FORMAT_AAC_ADTS_MAIN,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_MAIN)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_LC, make_AudioFormatDescription("audio/aac-adts.lc")
+ AUDIO_FORMAT_AAC_ADTS_LC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LC)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_SSR, make_AudioFormatDescription("audio/aac-adts.ssr")
+ AUDIO_FORMAT_AAC_ADTS_SSR,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_SSR)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_LTP, make_AudioFormatDescription("audio/aac-adts.ltp")
+ AUDIO_FORMAT_AAC_ADTS_LTP,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LTP)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_HE_V1, make_AudioFormatDescription("audio/aac-adts.he.v1")
+ AUDIO_FORMAT_AAC_ADTS_HE_V1,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_HE_V1)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_SCALABLE, make_AudioFormatDescription("audio/aac-adts.scalable")
+ AUDIO_FORMAT_AAC_ADTS_SCALABLE,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_SCALABLE)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_ERLC, make_AudioFormatDescription("audio/aac-adts.erlc")
+ AUDIO_FORMAT_AAC_ADTS_ERLC,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_ERLC)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_LD, make_AudioFormatDescription("audio/aac-adts.ld")
+ AUDIO_FORMAT_AAC_ADTS_LD, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LD)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_HE_V2, make_AudioFormatDescription("audio/aac-adts.he.v2")
+ AUDIO_FORMAT_AAC_ADTS_HE_V2,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_HE_V2)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_ELD, make_AudioFormatDescription("audio/aac-adts.eld")
+ AUDIO_FORMAT_AAC_ADTS_ELD,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_ELD)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_ADTS_XHE, make_AudioFormatDescription("audio/aac-adts.xhe")
+ AUDIO_FORMAT_AAC_ADTS_XHE,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS_XHE)
},
{
// Note: not in the IANA registry. "vnd.octel.sbc" is not BT SBC.
@@ -926,7 +881,6 @@
AUDIO_FORMAT_APTX_HD, make_AudioFormatDescription("audio/vnd.qcom.aptx.hd")
},
{
- // Note: not in the IANA registry. Matches MediaDefs.cpp.
AUDIO_FORMAT_AC4, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AC4)
},
{
@@ -934,35 +888,36 @@
AUDIO_FORMAT_LDAC, make_AudioFormatDescription("audio/vnd.sony.ldac")
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_MAT, make_AudioFormatDescription("audio/vnd.dolby.mat")
+ AUDIO_FORMAT_MAT, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DOLBY_MAT)
},
{
// Note: not in the IANA registry.
- AUDIO_FORMAT_MAT_1_0, make_AudioFormatDescription("audio/vnd.dolby.mat.1.0")
+ AUDIO_FORMAT_MAT_1_0,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DOLBY_MAT + std::string(".1.0"))
},
{
// Note: not in the IANA registry.
- AUDIO_FORMAT_MAT_2_0, make_AudioFormatDescription("audio/vnd.dolby.mat.2.0")
+ AUDIO_FORMAT_MAT_2_0,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DOLBY_MAT + std::string(".2.0"))
},
{
// Note: not in the IANA registry.
- AUDIO_FORMAT_MAT_2_1, make_AudioFormatDescription("audio/vnd.dolby.mat.2.1")
+ AUDIO_FORMAT_MAT_2_1,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DOLBY_MAT + std::string(".2.1"))
},
{
AUDIO_FORMAT_AAC_LATM, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_LATM_LC, make_AudioFormatDescription("audio/mp4a-latm.lc")
+ AUDIO_FORMAT_AAC_LATM_LC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_LATM_LC)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_LATM_HE_V1, make_AudioFormatDescription("audio/mp4a-latm.he.v1")
+ AUDIO_FORMAT_AAC_LATM_HE_V1,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_LATM_HE_V1)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_AAC_LATM_HE_V2, make_AudioFormatDescription("audio/mp4a-latm.he.v2")
+ AUDIO_FORMAT_AAC_LATM_HE_V2,
+ make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_LATM_HE_V2)
},
{
// Note: not in the IANA registry.
@@ -989,29 +944,23 @@
AUDIO_FORMAT_LC3, make_AudioFormatDescription("audio/x-lc3")
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_MPEGH, make_AudioFormatDescription("audio/x-mpegh")
+ AUDIO_FORMAT_MPEGH, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEGH_MHM1)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_MPEGH_BL_L3, make_AudioFormatDescription("audio/x-mpegh.bl.l3")
+ AUDIO_FORMAT_MPEGH_BL_L3, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEGH_BL_L3)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_MPEGH_BL_L4, make_AudioFormatDescription("audio/x-mpegh.bl.l4")
+ AUDIO_FORMAT_MPEGH_BL_L4, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEGH_BL_L4)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_MPEGH_LC_L3, make_AudioFormatDescription("audio/x-mpegh.lc.l3")
+ AUDIO_FORMAT_MPEGH_LC_L3, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEGH_LC_L3)
},
{
- // Note: not in the IANA registry.
- AUDIO_FORMAT_MPEGH_LC_L4, make_AudioFormatDescription("audio/x-mpegh.lc.l4")
+ AUDIO_FORMAT_MPEGH_LC_L4, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEGH_LC_L4)
},
{
- // Note: not in the IANA registry.
AUDIO_FORMAT_IEC60958,
- make_AudioFormatDescription(media::PcmType::INT_24_BIT, "audio/x-iec60958")
+ make_AudioFormatDescription(PcmType::INT_24_BIT, MEDIA_MIMETYPE_AUDIO_IEC60958)
},
{
AUDIO_FORMAT_DTS_UHD, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DTS_UHD)
@@ -1055,14 +1004,14 @@
} // namespace
ConversionResult<audio_channel_mask_t> aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
- const media::AudioChannelLayout& aidl, bool isInput) {
- using ReverseMap = std::unordered_map<media::AudioChannelLayout, audio_channel_mask_t>;
- using Tag = media::AudioChannelLayout::Tag;
+ const AudioChannelLayout& aidl, bool isInput) {
+ using ReverseMap = std::unordered_map<AudioChannelLayout, audio_channel_mask_t>;
+ using Tag = AudioChannelLayout::Tag;
static const ReverseMap mIn = make_ReverseMap(getInAudioChannelPairs());
static const ReverseMap mOut = make_ReverseMap(getOutAudioChannelPairs());
static const ReverseMap mVoice = make_ReverseMap(getVoiceAudioChannelPairs());
- auto convert = [](const media::AudioChannelLayout& aidl, const ReverseMap& m,
+ auto convert = [](const AudioChannelLayout& aidl, const ReverseMap& m,
const char* func, const char* type) -> ConversionResult<audio_channel_mask_t> {
if (auto it = m.find(aidl); it != m.end()) {
return it->second;
@@ -1100,16 +1049,16 @@
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioChannelLayout> legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
+ConversionResult<AudioChannelLayout> legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
audio_channel_mask_t legacy, bool isInput) {
- using DirectMap = std::unordered_map<audio_channel_mask_t, media::AudioChannelLayout>;
- using Tag = media::AudioChannelLayout::Tag;
+ using DirectMap = std::unordered_map<audio_channel_mask_t, AudioChannelLayout>;
+ using Tag = AudioChannelLayout::Tag;
static const DirectMap mInAndVoice = make_DirectMap(
getInAudioChannelPairs(), getVoiceAudioChannelPairs());
static const DirectMap mOut = make_DirectMap(getOutAudioChannelPairs());
auto convert = [](const audio_channel_mask_t legacy, const DirectMap& m,
- const char* func, const char* type) -> ConversionResult<media::AudioChannelLayout> {
+ const char* func, const char* type) -> ConversionResult<AudioChannelLayout> {
if (auto it = m.find(legacy); it != m.end()) {
return it->second;
} else {
@@ -1120,9 +1069,9 @@
};
if (legacy == AUDIO_CHANNEL_NONE) {
- return media::AudioChannelLayout{};
+ return AudioChannelLayout{};
} else if (legacy == AUDIO_CHANNEL_INVALID) {
- return media::AudioChannelLayout::make<Tag::invalid>(0);
+ return AudioChannelLayout::make<Tag::invalid>(0);
}
const audio_channel_representation_t repr = audio_channel_mask_get_representation(legacy);
@@ -1130,7 +1079,7 @@
if (audio_channel_mask_is_valid(legacy)) {
const int indexMask = VALUE_OR_RETURN(
convertIntegral<int>(audio_channel_mask_get_bits(legacy)));
- return media::AudioChannelLayout::make<Tag::indexMask>(indexMask);
+ return AudioChannelLayout::make<Tag::indexMask>(indexMask);
} else {
ALOGE("%s: legacy audio_channel_mask_t value 0x%x is invalid", __func__, legacy);
return unexpected(BAD_VALUE);
@@ -1146,8 +1095,8 @@
}
ConversionResult<audio_devices_t> aidl2legacy_AudioDeviceDescription_audio_devices_t(
- const media::AudioDeviceDescription& aidl) {
- static const std::unordered_map<media::AudioDeviceDescription, audio_devices_t> m =
+ const AudioDeviceDescription& aidl) {
+ static const std::unordered_map<AudioDeviceDescription, audio_devices_t> m =
make_ReverseMap(getAudioDevicePairs());
if (auto it = m.find(aidl); it != m.end()) {
return it->second;
@@ -1157,9 +1106,9 @@
}
}
-ConversionResult<media::AudioDeviceDescription> legacy2aidl_audio_devices_t_AudioDeviceDescription(
+ConversionResult<AudioDeviceDescription> legacy2aidl_audio_devices_t_AudioDeviceDescription(
audio_devices_t legacy) {
- static const std::unordered_map<audio_devices_t, media::AudioDeviceDescription> m =
+ static const std::unordered_map<audio_devices_t, AudioDeviceDescription> m =
make_DirectMap(getAudioDevicePairs());
if (auto it = m.find(legacy); it != m.end()) {
return it->second;
@@ -1170,9 +1119,61 @@
}
}
+status_t aidl2legacy_AudioDevice_audio_device(
+ const AudioDevice& aidl,
+ audio_devices_t* legacyType, char* legacyAddress) {
+ *legacyType = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.type));
+ return aidl2legacy_string(
+ aidl.address.get<AudioDeviceAddress::id>(),
+ legacyAddress, AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+status_t aidl2legacy_AudioDevice_audio_device(
+ const AudioDevice& aidl,
+ audio_devices_t* legacyType, String8* legacyAddress) {
+ *legacyType = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.type));
+ *legacyAddress = VALUE_OR_RETURN_STATUS(aidl2legacy_string_view_String8(
+ aidl.address.get<AudioDeviceAddress::id>()));
+ return OK;
+}
+
+status_t aidl2legacy_AudioDevice_audio_device(
+ const AudioDevice& aidl,
+ audio_devices_t* legacyType, std::string* legacyAddress) {
+ *legacyType = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.type));
+ *legacyAddress = aidl.address.get<AudioDeviceAddress::id>();
+ return OK;
+}
+
+ConversionResult<AudioDevice> legacy2aidl_audio_device_AudioDevice(
+ audio_devices_t legacyType, const char* legacyAddress) {
+ AudioDevice aidl;
+ aidl.type = VALUE_OR_RETURN(
+ legacy2aidl_audio_devices_t_AudioDeviceDescription(legacyType));
+ const std::string aidl_id = VALUE_OR_RETURN(
+ legacy2aidl_string(legacyAddress, AUDIO_DEVICE_MAX_ADDRESS_LEN));
+ aidl.address = AudioDeviceAddress::make<AudioDeviceAddress::id>(aidl_id);
+ return aidl;
+}
+
+ConversionResult<AudioDevice>
+legacy2aidl_audio_device_AudioDevice(
+ audio_devices_t legacyType, const String8& legacyAddress) {
+ AudioDevice aidl;
+ aidl.type = VALUE_OR_RETURN(
+ legacy2aidl_audio_devices_t_AudioDeviceDescription(legacyType));
+ const std::string aidl_id = VALUE_OR_RETURN(
+ legacy2aidl_String8_string(legacyAddress));
+ aidl.address = AudioDeviceAddress::make<AudioDeviceAddress::id>(aidl_id);
+ return aidl;
+}
+
ConversionResult<audio_format_t> aidl2legacy_AudioFormatDescription_audio_format_t(
- const media::AudioFormatDescription& aidl) {
- static const std::unordered_map<media::AudioFormatDescription, audio_format_t> m =
+ const AudioFormatDescription& aidl) {
+ static const std::unordered_map<AudioFormatDescription, audio_format_t> m =
make_ReverseMap(getAudioFormatPairs());
if (auto it = m.find(aidl); it != m.end()) {
return it->second;
@@ -1182,9 +1183,9 @@
}
}
-ConversionResult<media::AudioFormatDescription> legacy2aidl_audio_format_t_AudioFormatDescription(
+ConversionResult<AudioFormatDescription> legacy2aidl_audio_format_t_AudioFormatDescription(
audio_format_t legacy) {
- static const std::unordered_map<audio_format_t, media::AudioFormatDescription> m =
+ static const std::unordered_map<audio_format_t, AudioFormatDescription> m =
make_DirectMap(getAudioFormatPairs());
if (auto it = m.find(legacy); it != m.end()) {
return it->second;
@@ -1195,57 +1196,58 @@
}
}
-ConversionResult<audio_gain_mode_t> aidl2legacy_AudioGainMode_audio_gain_mode_t(media::AudioGainMode aidl) {
+ConversionResult<audio_gain_mode_t> aidl2legacy_AudioGainMode_audio_gain_mode_t(
+ AudioGainMode aidl) {
switch (aidl) {
- case media::AudioGainMode::JOINT:
+ case AudioGainMode::JOINT:
return AUDIO_GAIN_MODE_JOINT;
- case media::AudioGainMode::CHANNELS:
+ case AudioGainMode::CHANNELS:
return AUDIO_GAIN_MODE_CHANNELS;
- case media::AudioGainMode::RAMP:
+ case AudioGainMode::RAMP:
return AUDIO_GAIN_MODE_RAMP;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioGainMode> legacy2aidl_audio_gain_mode_t_AudioGainMode(audio_gain_mode_t legacy) {
+ConversionResult<AudioGainMode> legacy2aidl_audio_gain_mode_t_AudioGainMode(
+ audio_gain_mode_t legacy) {
switch (legacy) {
case AUDIO_GAIN_MODE_JOINT:
- return media::AudioGainMode::JOINT;
+ return AudioGainMode::JOINT;
case AUDIO_GAIN_MODE_CHANNELS:
- return media::AudioGainMode::CHANNELS;
+ return AudioGainMode::CHANNELS;
case AUDIO_GAIN_MODE_RAMP:
- return media::AudioGainMode::RAMP;
+ return AudioGainMode::RAMP;
}
return unexpected(BAD_VALUE);
}
ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t_mask(int32_t aidl) {
- return convertBitmask<audio_gain_mode_t, int32_t, audio_gain_mode_t, media::AudioGainMode>(
+ return convertBitmask<audio_gain_mode_t, int32_t, audio_gain_mode_t, AudioGainMode>(
aidl, aidl2legacy_AudioGainMode_audio_gain_mode_t,
// AudioGainMode is index-based.
- indexToEnum_index<media::AudioGainMode>,
+ indexToEnum_index<AudioGainMode>,
// AUDIO_GAIN_MODE_* constants are mask-based.
enumToMask_bitmask<audio_gain_mode_t, audio_gain_mode_t>);
}
ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t_mask(audio_gain_mode_t legacy) {
- return convertBitmask<int32_t, audio_gain_mode_t, media::AudioGainMode, audio_gain_mode_t>(
+ return convertBitmask<int32_t, audio_gain_mode_t, AudioGainMode, audio_gain_mode_t>(
legacy, legacy2aidl_audio_gain_mode_t_AudioGainMode,
// AUDIO_GAIN_MODE_* constants are mask-based.
indexToEnum_bitmask<audio_gain_mode_t>,
// AudioGainMode is index-based.
- enumToMask_index<int32_t, media::AudioGainMode>);
+ enumToMask_index<int32_t, AudioGainMode>);
}
ConversionResult<audio_gain_config> aidl2legacy_AudioGainConfig_audio_gain_config(
- const media::AudioGainConfig& aidl, media::AudioPortRole role, media::AudioPortType type) {
+ const AudioGainConfig& aidl, bool isInput) {
audio_gain_config legacy;
legacy.index = VALUE_OR_RETURN(convertIntegral<int>(aidl.index));
legacy.mode = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_gain_mode_t_mask(aidl.mode));
- const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
legacy.channel_mask = VALUE_OR_RETURN(
aidl2legacy_AudioChannelLayout_audio_channel_mask_t(aidl.channelMask, isInput));
- const bool isJoint = bitmaskIsSet(aidl.mode, media::AudioGainMode::JOINT);
+ const bool isJoint = bitmaskIsSet(aidl.mode, AudioGainMode::JOINT);
size_t numValues = isJoint ? 1
: isInput ? audio_channel_count_from_in_mask(legacy.channel_mask)
: audio_channel_count_from_out_mask(legacy.channel_mask);
@@ -1259,12 +1261,11 @@
return legacy;
}
-ConversionResult<media::AudioGainConfig> legacy2aidl_audio_gain_config_AudioGainConfig(
- const audio_gain_config& legacy, audio_port_role_t role, audio_port_type_t type) {
- media::AudioGainConfig aidl;
+ConversionResult<AudioGainConfig> legacy2aidl_audio_gain_config_AudioGainConfig(
+ const audio_gain_config& legacy, bool isInput) {
+ AudioGainConfig aidl;
aidl.index = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.index));
aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_int32_t_mask(legacy.mode));
- const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
aidl.channelMask = VALUE_OR_RETURN(
legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
const bool isJoint = (legacy.mode & AUDIO_GAIN_MODE_JOINT) != 0;
@@ -1496,175 +1497,175 @@
}
ConversionResult<audio_port_config_device_ext>
-aidl2legacy_AudioPortConfigDeviceExt_audio_port_config_device_ext(
- const media::AudioPortConfigDeviceExt& aidl) {
+aidl2legacy_AudioDevice_audio_port_config_device_ext(
+ const AudioDevice& aidl, const media::AudioPortDeviceExtSys& aidlDeviceExt) {
audio_port_config_device_ext legacy;
- legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
- legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.type));
- RETURN_IF_ERROR(aidl2legacy_string(aidl.address, legacy.address, AUDIO_DEVICE_MAX_ADDRESS_LEN));
+ legacy.hw_module = VALUE_OR_RETURN(
+ aidl2legacy_int32_t_audio_module_handle_t(aidlDeviceExt.hwModule));
+ RETURN_IF_ERROR(aidl2legacy_AudioDevice_audio_device(aidl, &legacy.type, legacy.address));
return legacy;
}
-ConversionResult<media::AudioPortConfigDeviceExt>
-legacy2aidl_audio_port_config_device_ext_AudioPortConfigDeviceExt(
- const audio_port_config_device_ext& legacy) {
- media::AudioPortConfigDeviceExt aidl;
- aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
- aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_AudioDeviceDescription(legacy.type));
- aidl.address = VALUE_OR_RETURN(
- legacy2aidl_string(legacy.address, AUDIO_DEVICE_MAX_ADDRESS_LEN));
- return aidl;
+status_t legacy2aidl_audio_port_config_device_ext_AudioDevice(
+ const audio_port_config_device_ext& legacy,
+ AudioDevice* aidl, media::AudioPortDeviceExtSys* aidlDeviceExt) {
+ aidlDeviceExt->hwModule = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
+ *aidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_device_AudioDevice(legacy.type, legacy.address));
+ return OK;
}
ConversionResult<audio_stream_type_t> aidl2legacy_AudioStreamType_audio_stream_type_t(
- media::AudioStreamType aidl) {
+ AudioStreamType aidl) {
switch (aidl) {
- case media::AudioStreamType::DEFAULT:
+ case AudioStreamType::INVALID:
+ break; // return error
+ case AudioStreamType::SYS_RESERVED_DEFAULT:
return AUDIO_STREAM_DEFAULT;
- case media::AudioStreamType::VOICE_CALL:
+ case AudioStreamType::VOICE_CALL:
return AUDIO_STREAM_VOICE_CALL;
- case media::AudioStreamType::SYSTEM:
+ case AudioStreamType::SYSTEM:
return AUDIO_STREAM_SYSTEM;
- case media::AudioStreamType::RING:
+ case AudioStreamType::RING:
return AUDIO_STREAM_RING;
- case media::AudioStreamType::MUSIC:
+ case AudioStreamType::MUSIC:
return AUDIO_STREAM_MUSIC;
- case media::AudioStreamType::ALARM:
+ case AudioStreamType::ALARM:
return AUDIO_STREAM_ALARM;
- case media::AudioStreamType::NOTIFICATION:
+ case AudioStreamType::NOTIFICATION:
return AUDIO_STREAM_NOTIFICATION;
- case media::AudioStreamType::BLUETOOTH_SCO:
+ case AudioStreamType::BLUETOOTH_SCO:
return AUDIO_STREAM_BLUETOOTH_SCO;
- case media::AudioStreamType::ENFORCED_AUDIBLE:
+ case AudioStreamType::ENFORCED_AUDIBLE:
return AUDIO_STREAM_ENFORCED_AUDIBLE;
- case media::AudioStreamType::DTMF:
+ case AudioStreamType::DTMF:
return AUDIO_STREAM_DTMF;
- case media::AudioStreamType::TTS:
+ case AudioStreamType::TTS:
return AUDIO_STREAM_TTS;
- case media::AudioStreamType::ACCESSIBILITY:
+ case AudioStreamType::ACCESSIBILITY:
return AUDIO_STREAM_ACCESSIBILITY;
- case media::AudioStreamType::ASSISTANT:
+ case AudioStreamType::ASSISTANT:
return AUDIO_STREAM_ASSISTANT;
- case media::AudioStreamType::REROUTING:
+ case AudioStreamType::SYS_RESERVED_REROUTING:
return AUDIO_STREAM_REROUTING;
- case media::AudioStreamType::PATCH:
+ case AudioStreamType::SYS_RESERVED_PATCH:
return AUDIO_STREAM_PATCH;
- case media::AudioStreamType::CALL_ASSISTANT:
+ case AudioStreamType::CALL_ASSISTANT:
return AUDIO_STREAM_CALL_ASSISTANT;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioStreamType> legacy2aidl_audio_stream_type_t_AudioStreamType(
+ConversionResult<AudioStreamType> legacy2aidl_audio_stream_type_t_AudioStreamType(
audio_stream_type_t legacy) {
switch (legacy) {
case AUDIO_STREAM_DEFAULT:
- return media::AudioStreamType::DEFAULT;
+ return AudioStreamType::SYS_RESERVED_DEFAULT;
case AUDIO_STREAM_VOICE_CALL:
- return media::AudioStreamType::VOICE_CALL;
+ return AudioStreamType::VOICE_CALL;
case AUDIO_STREAM_SYSTEM:
- return media::AudioStreamType::SYSTEM;
+ return AudioStreamType::SYSTEM;
case AUDIO_STREAM_RING:
- return media::AudioStreamType::RING;
+ return AudioStreamType::RING;
case AUDIO_STREAM_MUSIC:
- return media::AudioStreamType::MUSIC;
+ return AudioStreamType::MUSIC;
case AUDIO_STREAM_ALARM:
- return media::AudioStreamType::ALARM;
+ return AudioStreamType::ALARM;
case AUDIO_STREAM_NOTIFICATION:
- return media::AudioStreamType::NOTIFICATION;
+ return AudioStreamType::NOTIFICATION;
case AUDIO_STREAM_BLUETOOTH_SCO:
- return media::AudioStreamType::BLUETOOTH_SCO;
+ return AudioStreamType::BLUETOOTH_SCO;
case AUDIO_STREAM_ENFORCED_AUDIBLE:
- return media::AudioStreamType::ENFORCED_AUDIBLE;
+ return AudioStreamType::ENFORCED_AUDIBLE;
case AUDIO_STREAM_DTMF:
- return media::AudioStreamType::DTMF;
+ return AudioStreamType::DTMF;
case AUDIO_STREAM_TTS:
- return media::AudioStreamType::TTS;
+ return AudioStreamType::TTS;
case AUDIO_STREAM_ACCESSIBILITY:
- return media::AudioStreamType::ACCESSIBILITY;
+ return AudioStreamType::ACCESSIBILITY;
case AUDIO_STREAM_ASSISTANT:
- return media::AudioStreamType::ASSISTANT;
+ return AudioStreamType::ASSISTANT;
case AUDIO_STREAM_REROUTING:
- return media::AudioStreamType::REROUTING;
+ return AudioStreamType::SYS_RESERVED_REROUTING;
case AUDIO_STREAM_PATCH:
- return media::AudioStreamType::PATCH;
+ return AudioStreamType::SYS_RESERVED_PATCH;
case AUDIO_STREAM_CALL_ASSISTANT:
- return media::AudioStreamType::CALL_ASSISTANT;
+ return AudioStreamType::CALL_ASSISTANT;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<audio_source_t> aidl2legacy_AudioSourceType_audio_source_t(
- media::AudioSourceType aidl) {
+ConversionResult<audio_source_t> aidl2legacy_AudioSource_audio_source_t(
+ AudioSource aidl) {
switch (aidl) {
- case media::AudioSourceType::INVALID:
- // This value does not have an enum
+ case AudioSource::SYS_RESERVED_INVALID:
return AUDIO_SOURCE_INVALID;
- case media::AudioSourceType::DEFAULT:
+ case AudioSource::DEFAULT:
return AUDIO_SOURCE_DEFAULT;
- case media::AudioSourceType::MIC:
+ case AudioSource::MIC:
return AUDIO_SOURCE_MIC;
- case media::AudioSourceType::VOICE_UPLINK:
+ case AudioSource::VOICE_UPLINK:
return AUDIO_SOURCE_VOICE_UPLINK;
- case media::AudioSourceType::VOICE_DOWNLINK:
+ case AudioSource::VOICE_DOWNLINK:
return AUDIO_SOURCE_VOICE_DOWNLINK;
- case media::AudioSourceType::VOICE_CALL:
+ case AudioSource::VOICE_CALL:
return AUDIO_SOURCE_VOICE_CALL;
- case media::AudioSourceType::CAMCORDER:
+ case AudioSource::CAMCORDER:
return AUDIO_SOURCE_CAMCORDER;
- case media::AudioSourceType::VOICE_RECOGNITION:
+ case AudioSource::VOICE_RECOGNITION:
return AUDIO_SOURCE_VOICE_RECOGNITION;
- case media::AudioSourceType::VOICE_COMMUNICATION:
+ case AudioSource::VOICE_COMMUNICATION:
return AUDIO_SOURCE_VOICE_COMMUNICATION;
- case media::AudioSourceType::REMOTE_SUBMIX:
+ case AudioSource::REMOTE_SUBMIX:
return AUDIO_SOURCE_REMOTE_SUBMIX;
- case media::AudioSourceType::UNPROCESSED:
+ case AudioSource::UNPROCESSED:
return AUDIO_SOURCE_UNPROCESSED;
- case media::AudioSourceType::VOICE_PERFORMANCE:
+ case AudioSource::VOICE_PERFORMANCE:
return AUDIO_SOURCE_VOICE_PERFORMANCE;
- case media::AudioSourceType::ECHO_REFERENCE:
+ case AudioSource::ECHO_REFERENCE:
return AUDIO_SOURCE_ECHO_REFERENCE;
- case media::AudioSourceType::FM_TUNER:
+ case AudioSource::FM_TUNER:
return AUDIO_SOURCE_FM_TUNER;
- case media::AudioSourceType::HOTWORD:
+ case AudioSource::HOTWORD:
return AUDIO_SOURCE_HOTWORD;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioSourceType> legacy2aidl_audio_source_t_AudioSourceType(
+ConversionResult<AudioSource> legacy2aidl_audio_source_t_AudioSource(
audio_source_t legacy) {
switch (legacy) {
case AUDIO_SOURCE_INVALID:
- return media::AudioSourceType::INVALID;
+ return AudioSource::SYS_RESERVED_INVALID;
case AUDIO_SOURCE_DEFAULT:
- return media::AudioSourceType::DEFAULT;
+ return AudioSource::DEFAULT;
case AUDIO_SOURCE_MIC:
- return media::AudioSourceType::MIC;
+ return AudioSource::MIC;
case AUDIO_SOURCE_VOICE_UPLINK:
- return media::AudioSourceType::VOICE_UPLINK;
+ return AudioSource::VOICE_UPLINK;
case AUDIO_SOURCE_VOICE_DOWNLINK:
- return media::AudioSourceType::VOICE_DOWNLINK;
+ return AudioSource::VOICE_DOWNLINK;
case AUDIO_SOURCE_VOICE_CALL:
- return media::AudioSourceType::VOICE_CALL;
+ return AudioSource::VOICE_CALL;
case AUDIO_SOURCE_CAMCORDER:
- return media::AudioSourceType::CAMCORDER;
+ return AudioSource::CAMCORDER;
case AUDIO_SOURCE_VOICE_RECOGNITION:
- return media::AudioSourceType::VOICE_RECOGNITION;
+ return AudioSource::VOICE_RECOGNITION;
case AUDIO_SOURCE_VOICE_COMMUNICATION:
- return media::AudioSourceType::VOICE_COMMUNICATION;
+ return AudioSource::VOICE_COMMUNICATION;
case AUDIO_SOURCE_REMOTE_SUBMIX:
- return media::AudioSourceType::REMOTE_SUBMIX;
+ return AudioSource::REMOTE_SUBMIX;
case AUDIO_SOURCE_UNPROCESSED:
- return media::AudioSourceType::UNPROCESSED;
+ return AudioSource::UNPROCESSED;
case AUDIO_SOURCE_VOICE_PERFORMANCE:
- return media::AudioSourceType::VOICE_PERFORMANCE;
+ return AudioSource::VOICE_PERFORMANCE;
case AUDIO_SOURCE_ECHO_REFERENCE:
- return media::AudioSourceType::ECHO_REFERENCE;
+ return AudioSource::ECHO_REFERENCE;
case AUDIO_SOURCE_FM_TUNER:
- return media::AudioSourceType::FM_TUNER;
+ return AudioSource::FM_TUNER;
case AUDIO_SOURCE_HOTWORD:
- return media::AudioSourceType::HOTWORD;
+ return AudioSource::HOTWORD;
}
return unexpected(BAD_VALUE);
}
@@ -1680,8 +1681,8 @@
// This type is unnamed in the original definition, thus we name it here.
using audio_port_config_mix_ext_usecase = decltype(audio_port_config_mix_ext::usecase);
-ConversionResult<audio_port_config_mix_ext_usecase> aidl2legacy_AudioPortConfigMixExtUseCase(
- const media::AudioPortConfigMixExtUseCase& aidl, media::AudioPortRole role) {
+ConversionResult<audio_port_config_mix_ext_usecase> aidl2legacy_AudioPortMixExtUseCase(
+ const AudioPortMixExtUseCase& aidl, media::AudioPortRole role) {
audio_port_config_mix_ext_usecase legacy;
switch (role) {
@@ -1698,16 +1699,16 @@
case media::AudioPortRole::SINK:
// This is not a bug. A SINK role corresponds to the source field.
- legacy.source = VALUE_OR_RETURN(aidl2legacy_AudioSourceType_audio_source_t(
+ legacy.source = VALUE_OR_RETURN(aidl2legacy_AudioSource_audio_source_t(
VALUE_OR_RETURN(UNION_GET(aidl, source))));
return legacy;
}
LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
-ConversionResult<media::AudioPortConfigMixExtUseCase> legacy2aidl_AudioPortConfigMixExtUseCase(
+ConversionResult<AudioPortMixExtUseCase> legacy2aidl_AudioPortMixExtUseCase(
const audio_port_config_mix_ext_usecase& legacy, audio_port_role_t role) {
- media::AudioPortConfigMixExtUseCase aidl;
+ AudioPortMixExtUseCase aidl;
switch (role) {
case AUDIO_PORT_ROLE_NONE:
@@ -1721,52 +1722,53 @@
case AUDIO_PORT_ROLE_SINK:
// This is not a bug. A SINK role corresponds to the source field.
UNION_SET(aidl, source,
- VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSourceType(legacy.source)));
+ VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSource(legacy.source)));
return aidl;
}
LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
-ConversionResult<audio_port_config_mix_ext> aidl2legacy_AudioPortConfigMixExt(
- const media::AudioPortConfigMixExt& aidl, media::AudioPortRole role) {
+ConversionResult<audio_port_config_mix_ext> aidl2legacy_AudioPortMixExt(
+ const AudioPortMixExt& aidl, media::AudioPortRole role,
+ const media::AudioPortMixExtSys& aidlMixExt) {
audio_port_config_mix_ext legacy;
- legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
+ legacy.hw_module = VALUE_OR_RETURN(
+ aidl2legacy_int32_t_audio_module_handle_t(aidlMixExt.hwModule));
legacy.handle = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_io_handle_t(aidl.handle));
- legacy.usecase = VALUE_OR_RETURN(aidl2legacy_AudioPortConfigMixExtUseCase(aidl.usecase, role));
+ legacy.usecase = VALUE_OR_RETURN(aidl2legacy_AudioPortMixExtUseCase(aidl.usecase, role));
return legacy;
}
-ConversionResult<media::AudioPortConfigMixExt> legacy2aidl_AudioPortConfigMixExt(
- const audio_port_config_mix_ext& legacy, audio_port_role_t role) {
- media::AudioPortConfigMixExt aidl;
- aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
- aidl.handle = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(legacy.handle));
- aidl.usecase = VALUE_OR_RETURN(legacy2aidl_AudioPortConfigMixExtUseCase(legacy.usecase, role));
- return aidl;
+status_t legacy2aidl_AudioPortMixExt(
+ const audio_port_config_mix_ext& legacy, audio_port_role_t role,
+ AudioPortMixExt* aidl, media::AudioPortMixExtSys* aidlMixExt) {
+ aidlMixExt->hwModule = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
+ aidl->handle = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(legacy.handle));
+ aidl->usecase = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_AudioPortMixExtUseCase(legacy.usecase, role));
+ return OK;
}
ConversionResult<audio_port_config_session_ext>
-aidl2legacy_AudioPortConfigSessionExt_audio_port_config_session_ext(
- const media::AudioPortConfigSessionExt& aidl) {
+aidl2legacy_int32_t_audio_port_config_session_ext(int32_t aidl) {
audio_port_config_session_ext legacy;
- legacy.session = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_session_t(aidl.session));
+ legacy.session = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_session_t(aidl));
return legacy;
}
-ConversionResult<media::AudioPortConfigSessionExt>
-legacy2aidl_audio_port_config_session_ext_AudioPortConfigSessionExt(
+ConversionResult<int32_t>
+legacy2aidl_audio_port_config_session_ext_int32_t(
const audio_port_config_session_ext& legacy) {
- media::AudioPortConfigSessionExt aidl;
- aidl.session = VALUE_OR_RETURN(legacy2aidl_audio_session_t_int32_t(legacy.session));
- return aidl;
+ return legacy2aidl_audio_session_t_int32_t(legacy.session);
}
// This type is unnamed in the original definition, thus we name it here.
using audio_port_config_ext = decltype(audio_port_config::ext);
-ConversionResult<audio_port_config_ext> aidl2legacy_AudioPortConfigExt(
- const media::AudioPortConfigExt& aidl, media::AudioPortType type,
- media::AudioPortRole role) {
+ConversionResult<audio_port_config_ext> aidl2legacy_AudioPortExt_audio_port_config_ext(
+ const AudioPortExt& aidl, media::AudioPortType type,
+ media::AudioPortRole role, const media::AudioPortExtSys& aidlSys) {
audio_port_config_ext legacy;
switch (type) {
case media::AudioPortType::NONE:
@@ -1775,16 +1777,19 @@
return legacy;
case media::AudioPortType::DEVICE:
legacy.device = VALUE_OR_RETURN(
- aidl2legacy_AudioPortConfigDeviceExt_audio_port_config_device_ext(
- VALUE_OR_RETURN(UNION_GET(aidl, device))));
+ aidl2legacy_AudioDevice_audio_port_config_device_ext(
+ VALUE_OR_RETURN(UNION_GET(aidl, device)),
+ VALUE_OR_RETURN(UNION_GET(aidlSys, device))));
return legacy;
case media::AudioPortType::MIX:
legacy.mix = VALUE_OR_RETURN(
- aidl2legacy_AudioPortConfigMixExt(VALUE_OR_RETURN(UNION_GET(aidl, mix)), role));
+ aidl2legacy_AudioPortMixExt(
+ VALUE_OR_RETURN(UNION_GET(aidl, mix)), role,
+ VALUE_OR_RETURN(UNION_GET(aidlSys, mix))));
return legacy;
case media::AudioPortType::SESSION:
legacy.session = VALUE_OR_RETURN(
- aidl2legacy_AudioPortConfigSessionExt_audio_port_config_session_ext(
+ aidl2legacy_int32_t_audio_port_config_session_ext(
VALUE_OR_RETURN(UNION_GET(aidl, session))));
return legacy;
@@ -1792,97 +1797,114 @@
LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
-ConversionResult<media::AudioPortConfigExt> legacy2aidl_AudioPortConfigExt(
- const audio_port_config_ext& legacy, audio_port_type_t type, audio_port_role_t role) {
- media::AudioPortConfigExt aidl;
-
+status_t legacy2aidl_AudioPortExt(
+ const audio_port_config_ext& legacy, audio_port_type_t type, audio_port_role_t role,
+ AudioPortExt* aidl, media::AudioPortExtSys* aidlSys) {
switch (type) {
case AUDIO_PORT_TYPE_NONE:
- UNION_SET(aidl, unspecified, false);
- return aidl;
- case AUDIO_PORT_TYPE_DEVICE:
- UNION_SET(aidl, device,
- VALUE_OR_RETURN(
- legacy2aidl_audio_port_config_device_ext_AudioPortConfigDeviceExt(
- legacy.device)));
- return aidl;
- case AUDIO_PORT_TYPE_MIX:
- UNION_SET(aidl, mix,
- VALUE_OR_RETURN(legacy2aidl_AudioPortConfigMixExt(legacy.mix, role)));
- return aidl;
+ UNION_SET(*aidl, unspecified, false);
+ UNION_SET(*aidlSys, unspecified, false);
+ return OK;
+ case AUDIO_PORT_TYPE_DEVICE: {
+ AudioDevice device;
+ media::AudioPortDeviceExtSys deviceSys;
+ RETURN_STATUS_IF_ERROR(
+ legacy2aidl_audio_port_config_device_ext_AudioDevice(
+ legacy.device, &device, &deviceSys));
+ UNION_SET(*aidl, device, device);
+ UNION_SET(*aidlSys, device, deviceSys);
+ return OK;
+ }
+ case AUDIO_PORT_TYPE_MIX: {
+ AudioPortMixExt mix;
+ media::AudioPortMixExtSys mixSys;
+ RETURN_STATUS_IF_ERROR(legacy2aidl_AudioPortMixExt(legacy.mix, role, &mix, &mixSys));
+ UNION_SET(*aidl, mix, mix);
+ UNION_SET(*aidlSys, mix, mixSys);
+ return OK;
+ }
case AUDIO_PORT_TYPE_SESSION:
- UNION_SET(aidl, session,
- VALUE_OR_RETURN(
- legacy2aidl_audio_port_config_session_ext_AudioPortConfigSessionExt(
- legacy.session)));
- return aidl;
+ UNION_SET(*aidl, session, VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_port_config_session_ext_int32_t(legacy.session)));
+ UNION_SET(*aidlSys, unspecified, false);
+ return OK;
}
LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
ConversionResult<audio_port_config> aidl2legacy_AudioPortConfig_audio_port_config(
const media::AudioPortConfig& aidl) {
- audio_port_config legacy;
- legacy.id = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.id));
- legacy.role = VALUE_OR_RETURN(aidl2legacy_AudioPortRole_audio_port_role_t(aidl.role));
- legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioPortType_audio_port_type_t(aidl.type));
- legacy.config_mask = VALUE_OR_RETURN(aidl2legacy_int32_t_config_mask(aidl.configMask));
- if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::SAMPLE_RATE)) {
- legacy.sample_rate = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.sampleRate));
+ audio_port_config legacy{};
+ legacy.id = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.hal.id));
+ legacy.role = VALUE_OR_RETURN(aidl2legacy_AudioPortRole_audio_port_role_t(aidl.sys.role));
+ legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioPortType_audio_port_type_t(aidl.sys.type));
+ const bool isInput =
+ VALUE_OR_RETURN(direction(aidl.sys.role, aidl.sys.type)) == Direction::INPUT;
+ if (aidl.hal.sampleRate.has_value()) {
+ legacy.sample_rate = VALUE_OR_RETURN(
+ convertIntegral<unsigned int>(aidl.hal.sampleRate.value().value));
+ legacy.config_mask |= AUDIO_PORT_CONFIG_SAMPLE_RATE;
}
- if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::CHANNEL_MASK)) {
- const bool isInput = VALUE_OR_RETURN(direction(aidl.role, aidl.type)) == Direction::INPUT;
+ if (aidl.hal.channelMask.has_value()) {
legacy.channel_mask =
VALUE_OR_RETURN(
aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
- aidl.channelMask, isInput));
+ aidl.hal.channelMask.value(), isInput));
+ legacy.config_mask |= AUDIO_PORT_CONFIG_CHANNEL_MASK;
}
- if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::FORMAT)) {
+ if (aidl.hal.format.has_value()) {
legacy.format = VALUE_OR_RETURN(
- aidl2legacy_AudioFormatDescription_audio_format_t(aidl.format));
+ aidl2legacy_AudioFormatDescription_audio_format_t(aidl.hal.format.value()));
+ legacy.config_mask |= AUDIO_PORT_CONFIG_FORMAT;
}
- if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::GAIN)) {
- legacy.gain = VALUE_OR_RETURN(
- aidl2legacy_AudioGainConfig_audio_gain_config(aidl.gain, aidl.role, aidl.type));
+ if (aidl.hal.gain.has_value()) {
+ legacy.gain = VALUE_OR_RETURN(aidl2legacy_AudioGainConfig_audio_gain_config(
+ aidl.hal.gain.value(), isInput));
+ legacy.config_mask |= AUDIO_PORT_CONFIG_GAIN;
}
- if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::FLAGS)) {
+ if (aidl.sys.flags.has_value()) {
legacy.flags = VALUE_OR_RETURN(
- aidl2legacy_AudioIoFlags_audio_io_flags(aidl.flags, aidl.role, aidl.type));
+ aidl2legacy_AudioIoFlags_audio_io_flags(
+ aidl.sys.flags.value(), aidl.sys.role, aidl.sys.type));
+ legacy.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
}
- legacy.ext = VALUE_OR_RETURN(aidl2legacy_AudioPortConfigExt(aidl.ext, aidl.type, aidl.role));
+ legacy.ext = VALUE_OR_RETURN(
+ aidl2legacy_AudioPortExt_audio_port_config_ext(
+ aidl.hal.ext, aidl.sys.type, aidl.sys.role, aidl.sys.ext));
return legacy;
}
ConversionResult<media::AudioPortConfig> legacy2aidl_audio_port_config_AudioPortConfig(
const audio_port_config& legacy) {
media::AudioPortConfig aidl;
- aidl.id = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy.id));
- aidl.role = VALUE_OR_RETURN(legacy2aidl_audio_port_role_t_AudioPortRole(legacy.role));
- aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_port_type_t_AudioPortType(legacy.type));
- aidl.configMask = VALUE_OR_RETURN(legacy2aidl_config_mask_int32_t(legacy.config_mask));
+ aidl.hal.id = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy.id));
+ aidl.sys.role = VALUE_OR_RETURN(legacy2aidl_audio_port_role_t_AudioPortRole(legacy.role));
+ aidl.sys.type = VALUE_OR_RETURN(legacy2aidl_audio_port_type_t_AudioPortType(legacy.type));
+ const bool isInput = VALUE_OR_RETURN(
+ direction(legacy.role, legacy.type)) == Direction::INPUT;
if (legacy.config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- aidl.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
+ Int aidl_sampleRate;
+ aidl_sampleRate.value = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
+ aidl.hal.sampleRate = aidl_sampleRate;
}
if (legacy.config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- const bool isInput = VALUE_OR_RETURN(
- direction(legacy.role, legacy.type)) == Direction::INPUT;
- aidl.channelMask = VALUE_OR_RETURN(
+ aidl.hal.channelMask = VALUE_OR_RETURN(
legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
}
if (legacy.config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- aidl.format = VALUE_OR_RETURN(
+ aidl.hal.format = VALUE_OR_RETURN(
legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
}
if (legacy.config_mask & AUDIO_PORT_CONFIG_GAIN) {
- aidl.gain = VALUE_OR_RETURN(legacy2aidl_audio_gain_config_AudioGainConfig(
- legacy.gain, legacy.role, legacy.type));
+ aidl.hal.gain = VALUE_OR_RETURN(
+ legacy2aidl_audio_gain_config_AudioGainConfig(legacy.gain, isInput));
}
if (legacy.config_mask & AUDIO_PORT_CONFIG_FLAGS) {
- aidl.flags = VALUE_OR_RETURN(
+ aidl.sys.flags = VALUE_OR_RETURN(
legacy2aidl_audio_io_flags_AudioIoFlags(legacy.flags, legacy.role, legacy.type));
}
- aidl.ext =
- VALUE_OR_RETURN(legacy2aidl_AudioPortConfigExt(legacy.ext, legacy.type, legacy.role));
+ RETURN_IF_ERROR(legacy2aidl_AudioPortExt(legacy.ext, legacy.type, legacy.role,
+ &aidl.hal.ext, &aidl.sys.ext));
return aidl;
}
@@ -1987,137 +2009,139 @@
}
ConversionResult<audio_content_type_t>
-aidl2legacy_AudioContentType_audio_content_type_t(media::AudioContentType aidl) {
+aidl2legacy_AudioContentType_audio_content_type_t(AudioContentType aidl) {
switch (aidl) {
- case media::AudioContentType::UNKNOWN:
+ case AudioContentType::UNKNOWN:
return AUDIO_CONTENT_TYPE_UNKNOWN;
- case media::AudioContentType::SPEECH:
+ case AudioContentType::SPEECH:
return AUDIO_CONTENT_TYPE_SPEECH;
- case media::AudioContentType::MUSIC:
+ case AudioContentType::MUSIC:
return AUDIO_CONTENT_TYPE_MUSIC;
- case media::AudioContentType::MOVIE:
+ case AudioContentType::MOVIE:
return AUDIO_CONTENT_TYPE_MOVIE;
- case media::AudioContentType::SONIFICATION:
+ case AudioContentType::SONIFICATION:
return AUDIO_CONTENT_TYPE_SONIFICATION;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioContentType>
+ConversionResult<AudioContentType>
legacy2aidl_audio_content_type_t_AudioContentType(audio_content_type_t legacy) {
switch (legacy) {
case AUDIO_CONTENT_TYPE_UNKNOWN:
- return media::AudioContentType::UNKNOWN;
+ return AudioContentType::UNKNOWN;
case AUDIO_CONTENT_TYPE_SPEECH:
- return media::AudioContentType::SPEECH;
+ return AudioContentType::SPEECH;
case AUDIO_CONTENT_TYPE_MUSIC:
- return media::AudioContentType::MUSIC;
+ return AudioContentType::MUSIC;
case AUDIO_CONTENT_TYPE_MOVIE:
- return media::AudioContentType::MOVIE;
+ return AudioContentType::MOVIE;
case AUDIO_CONTENT_TYPE_SONIFICATION:
- return media::AudioContentType::SONIFICATION;
+ return AudioContentType::SONIFICATION;
}
return unexpected(BAD_VALUE);
}
ConversionResult<audio_usage_t>
-aidl2legacy_AudioUsage_audio_usage_t(media::AudioUsage aidl) {
+aidl2legacy_AudioUsage_audio_usage_t(AudioUsage aidl) {
switch (aidl) {
- case media::AudioUsage::UNKNOWN:
+ case AudioUsage::INVALID:
+ break; // return error
+ case AudioUsage::UNKNOWN:
return AUDIO_USAGE_UNKNOWN;
- case media::AudioUsage::MEDIA:
+ case AudioUsage::MEDIA:
return AUDIO_USAGE_MEDIA;
- case media::AudioUsage::VOICE_COMMUNICATION:
+ case AudioUsage::VOICE_COMMUNICATION:
return AUDIO_USAGE_VOICE_COMMUNICATION;
- case media::AudioUsage::VOICE_COMMUNICATION_SIGNALLING:
+ case AudioUsage::VOICE_COMMUNICATION_SIGNALLING:
return AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
- case media::AudioUsage::ALARM:
+ case AudioUsage::ALARM:
return AUDIO_USAGE_ALARM;
- case media::AudioUsage::NOTIFICATION:
+ case AudioUsage::NOTIFICATION:
return AUDIO_USAGE_NOTIFICATION;
- case media::AudioUsage::NOTIFICATION_TELEPHONY_RINGTONE:
+ case AudioUsage::NOTIFICATION_TELEPHONY_RINGTONE:
return AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
- case media::AudioUsage::NOTIFICATION_COMMUNICATION_REQUEST:
+ case AudioUsage::SYS_RESERVED_NOTIFICATION_COMMUNICATION_REQUEST:
return AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST;
- case media::AudioUsage::NOTIFICATION_COMMUNICATION_INSTANT:
+ case AudioUsage::SYS_RESERVED_NOTIFICATION_COMMUNICATION_INSTANT:
return AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT;
- case media::AudioUsage::NOTIFICATION_COMMUNICATION_DELAYED:
+ case AudioUsage::SYS_RESERVED_NOTIFICATION_COMMUNICATION_DELAYED:
return AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED;
- case media::AudioUsage::NOTIFICATION_EVENT:
+ case AudioUsage::NOTIFICATION_EVENT:
return AUDIO_USAGE_NOTIFICATION_EVENT;
- case media::AudioUsage::ASSISTANCE_ACCESSIBILITY:
+ case AudioUsage::ASSISTANCE_ACCESSIBILITY:
return AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
- case media::AudioUsage::ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AudioUsage::ASSISTANCE_NAVIGATION_GUIDANCE:
return AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE;
- case media::AudioUsage::ASSISTANCE_SONIFICATION:
+ case AudioUsage::ASSISTANCE_SONIFICATION:
return AUDIO_USAGE_ASSISTANCE_SONIFICATION;
- case media::AudioUsage::GAME:
+ case AudioUsage::GAME:
return AUDIO_USAGE_GAME;
- case media::AudioUsage::VIRTUAL_SOURCE:
+ case AudioUsage::VIRTUAL_SOURCE:
return AUDIO_USAGE_VIRTUAL_SOURCE;
- case media::AudioUsage::ASSISTANT:
+ case AudioUsage::ASSISTANT:
return AUDIO_USAGE_ASSISTANT;
- case media::AudioUsage::CALL_ASSISTANT:
+ case AudioUsage::CALL_ASSISTANT:
return AUDIO_USAGE_CALL_ASSISTANT;
- case media::AudioUsage::EMERGENCY:
+ case AudioUsage::EMERGENCY:
return AUDIO_USAGE_EMERGENCY;
- case media::AudioUsage::SAFETY:
+ case AudioUsage::SAFETY:
return AUDIO_USAGE_SAFETY;
- case media::AudioUsage::VEHICLE_STATUS:
+ case AudioUsage::VEHICLE_STATUS:
return AUDIO_USAGE_VEHICLE_STATUS;
- case media::AudioUsage::ANNOUNCEMENT:
+ case AudioUsage::ANNOUNCEMENT:
return AUDIO_USAGE_ANNOUNCEMENT;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioUsage>
+ConversionResult<AudioUsage>
legacy2aidl_audio_usage_t_AudioUsage(audio_usage_t legacy) {
switch (legacy) {
case AUDIO_USAGE_UNKNOWN:
- return media::AudioUsage::UNKNOWN;
+ return AudioUsage::UNKNOWN;
case AUDIO_USAGE_MEDIA:
- return media::AudioUsage::MEDIA;
+ return AudioUsage::MEDIA;
case AUDIO_USAGE_VOICE_COMMUNICATION:
- return media::AudioUsage::VOICE_COMMUNICATION;
+ return AudioUsage::VOICE_COMMUNICATION;
case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
- return media::AudioUsage::VOICE_COMMUNICATION_SIGNALLING;
+ return AudioUsage::VOICE_COMMUNICATION_SIGNALLING;
case AUDIO_USAGE_ALARM:
- return media::AudioUsage::ALARM;
+ return AudioUsage::ALARM;
case AUDIO_USAGE_NOTIFICATION:
- return media::AudioUsage::NOTIFICATION;
+ return AudioUsage::NOTIFICATION;
case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
- return media::AudioUsage::NOTIFICATION_TELEPHONY_RINGTONE;
+ return AudioUsage::NOTIFICATION_TELEPHONY_RINGTONE;
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
- return media::AudioUsage::NOTIFICATION_COMMUNICATION_REQUEST;
+ return AudioUsage::SYS_RESERVED_NOTIFICATION_COMMUNICATION_REQUEST;
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
- return media::AudioUsage::NOTIFICATION_COMMUNICATION_INSTANT;
+ return AudioUsage::SYS_RESERVED_NOTIFICATION_COMMUNICATION_INSTANT;
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
- return media::AudioUsage::NOTIFICATION_COMMUNICATION_DELAYED;
+ return AudioUsage::SYS_RESERVED_NOTIFICATION_COMMUNICATION_DELAYED;
case AUDIO_USAGE_NOTIFICATION_EVENT:
- return media::AudioUsage::NOTIFICATION_EVENT;
+ return AudioUsage::NOTIFICATION_EVENT;
case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
- return media::AudioUsage::ASSISTANCE_ACCESSIBILITY;
+ return AudioUsage::ASSISTANCE_ACCESSIBILITY;
case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
- return media::AudioUsage::ASSISTANCE_NAVIGATION_GUIDANCE;
+ return AudioUsage::ASSISTANCE_NAVIGATION_GUIDANCE;
case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
- return media::AudioUsage::ASSISTANCE_SONIFICATION;
+ return AudioUsage::ASSISTANCE_SONIFICATION;
case AUDIO_USAGE_GAME:
- return media::AudioUsage::GAME;
+ return AudioUsage::GAME;
case AUDIO_USAGE_VIRTUAL_SOURCE:
- return media::AudioUsage::VIRTUAL_SOURCE;
+ return AudioUsage::VIRTUAL_SOURCE;
case AUDIO_USAGE_ASSISTANT:
- return media::AudioUsage::ASSISTANT;
+ return AudioUsage::ASSISTANT;
case AUDIO_USAGE_CALL_ASSISTANT:
- return media::AudioUsage::CALL_ASSISTANT;
+ return AudioUsage::CALL_ASSISTANT;
case AUDIO_USAGE_EMERGENCY:
- return media::AudioUsage::EMERGENCY;
+ return AudioUsage::EMERGENCY;
case AUDIO_USAGE_SAFETY:
- return media::AudioUsage::SAFETY;
+ return AudioUsage::SAFETY;
case AUDIO_USAGE_VEHICLE_STATUS:
- return media::AudioUsage::VEHICLE_STATUS;
+ return AudioUsage::VEHICLE_STATUS;
case AUDIO_USAGE_ANNOUNCEMENT:
- return media::AudioUsage::ANNOUNCEMENT;
+ return AudioUsage::ANNOUNCEMENT;
}
return unexpected(BAD_VALUE);
}
@@ -2153,6 +2177,10 @@
return AUDIO_FLAG_NO_SYSTEM_CAPTURE;
case media::AudioFlag::CAPTURE_PRIVATE:
return AUDIO_FLAG_CAPTURE_PRIVATE;
+ case media::AudioFlag::CONTENT_SPATIALIZED:
+ return AUDIO_FLAG_CONTENT_SPATIALIZED;
+ case media::AudioFlag::NEVER_SPATIALIZE:
+ return AUDIO_FLAG_NEVER_SPATIALIZE;
}
return unexpected(BAD_VALUE);
}
@@ -2190,6 +2218,10 @@
return media::AudioFlag::NO_SYSTEM_CAPTURE;
case AUDIO_FLAG_CAPTURE_PRIVATE:
return media::AudioFlag::CAPTURE_PRIVATE;
+ case AUDIO_FLAG_CONTENT_SPATIALIZED:
+ return media::AudioFlag::CONTENT_SPATIALIZED;
+ case AUDIO_FLAG_NEVER_SPATIALIZE:
+ return media::AudioFlag::NEVER_SPATIALIZE;
}
return unexpected(BAD_VALUE);
}
@@ -2215,7 +2247,7 @@
legacy.content_type = VALUE_OR_RETURN(
aidl2legacy_AudioContentType_audio_content_type_t(aidl.contentType));
legacy.usage = VALUE_OR_RETURN(aidl2legacy_AudioUsage_audio_usage_t(aidl.usage));
- legacy.source = VALUE_OR_RETURN(aidl2legacy_AudioSourceType_audio_source_t(aidl.source));
+ legacy.source = VALUE_OR_RETURN(aidl2legacy_AudioSource_audio_source_t(aidl.source));
legacy.flags = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_flags_mask_t_mask(aidl.flags));
RETURN_IF_ERROR(aidl2legacy_string(aidl.tags, legacy.tags, sizeof(legacy.tags)));
return legacy;
@@ -2227,51 +2259,51 @@
aidl.contentType = VALUE_OR_RETURN(
legacy2aidl_audio_content_type_t_AudioContentType(legacy.content_type));
aidl.usage = VALUE_OR_RETURN(legacy2aidl_audio_usage_t_AudioUsage(legacy.usage));
- aidl.source = VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSourceType(legacy.source));
+ aidl.source = VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSource(legacy.source));
aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_flags_mask_t_int32_t_mask(legacy.flags));
aidl.tags = VALUE_OR_RETURN(legacy2aidl_string(legacy.tags, sizeof(legacy.tags)));
return aidl;
}
ConversionResult<audio_encapsulation_mode_t>
-aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t(media::AudioEncapsulationMode aidl) {
+aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t(AudioEncapsulationMode aidl) {
switch (aidl) {
- case media::AudioEncapsulationMode::NONE:
+ case AudioEncapsulationMode::INVALID:
+ break; // return error
+ case AudioEncapsulationMode::NONE:
return AUDIO_ENCAPSULATION_MODE_NONE;
- case media::AudioEncapsulationMode::ELEMENTARY_STREAM:
+ case AudioEncapsulationMode::ELEMENTARY_STREAM:
return AUDIO_ENCAPSULATION_MODE_ELEMENTARY_STREAM;
- case media::AudioEncapsulationMode::HANDLE:
+ case AudioEncapsulationMode::HANDLE:
return AUDIO_ENCAPSULATION_MODE_HANDLE;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioEncapsulationMode>
+ConversionResult<AudioEncapsulationMode>
legacy2aidl_audio_encapsulation_mode_t_AudioEncapsulationMode(audio_encapsulation_mode_t legacy) {
switch (legacy) {
case AUDIO_ENCAPSULATION_MODE_NONE:
- return media::AudioEncapsulationMode::NONE;
+ return AudioEncapsulationMode::NONE;
case AUDIO_ENCAPSULATION_MODE_ELEMENTARY_STREAM:
- return media::AudioEncapsulationMode::ELEMENTARY_STREAM;
+ return AudioEncapsulationMode::ELEMENTARY_STREAM;
case AUDIO_ENCAPSULATION_MODE_HANDLE:
- return media::AudioEncapsulationMode::HANDLE;
+ return AudioEncapsulationMode::HANDLE;
}
return unexpected(BAD_VALUE);
}
ConversionResult<audio_offload_info_t>
-aidl2legacy_AudioOffloadInfo_audio_offload_info_t(const media::AudioOffloadInfo& aidl) {
- audio_offload_info_t legacy;
- legacy.version = VALUE_OR_RETURN(convertIntegral<uint16_t>(aidl.version));
- legacy.size = sizeof(audio_offload_info_t);
- audio_config_base_t config = VALUE_OR_RETURN(
- aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.config, false /*isInput*/));
- legacy.sample_rate = config.sample_rate;
- legacy.channel_mask = config.channel_mask;
- legacy.format = config.format;
+aidl2legacy_AudioOffloadInfo_audio_offload_info_t(const AudioOffloadInfo& aidl) {
+ audio_offload_info_t legacy = AUDIO_INFO_INITIALIZER;
+ audio_config_base_t base = VALUE_OR_RETURN(
+ aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.base, false /*isInput*/));
+ legacy.sample_rate = base.sample_rate;
+ legacy.channel_mask = base.channel_mask;
+ legacy.format = base.format;
legacy.stream_type = VALUE_OR_RETURN(
aidl2legacy_AudioStreamType_audio_stream_type_t(aidl.streamType));
- legacy.bit_rate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.bitRate));
+ legacy.bit_rate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.bitRatePerSecond));
legacy.duration_us = VALUE_OR_RETURN(convertIntegral<int64_t>(aidl.durationUs));
legacy.has_video = aidl.hasVideo;
legacy.is_streaming = aidl.isStreaming;
@@ -2285,22 +2317,20 @@
return legacy;
}
-ConversionResult<media::AudioOffloadInfo>
+ConversionResult<AudioOffloadInfo>
legacy2aidl_audio_offload_info_t_AudioOffloadInfo(const audio_offload_info_t& legacy) {
- media::AudioOffloadInfo aidl;
+ AudioOffloadInfo aidl;
// Version 0.1 fields.
if (legacy.size < offsetof(audio_offload_info_t, usage) + sizeof(audio_offload_info_t::usage)) {
return unexpected(BAD_VALUE);
}
- aidl.version = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.version));
- aidl.config.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
- aidl.config.channelMask = VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
- legacy.channel_mask, false /*isInput*/));
- aidl.config.format = VALUE_OR_RETURN(
- legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
+ const audio_config_base_t base = { .sample_rate = legacy.sample_rate,
+ .channel_mask = legacy.channel_mask, .format = legacy.format };
+ aidl.base = VALUE_OR_RETURN(legacy2aidl_audio_config_base_t_AudioConfigBase(
+ base, false /*isInput*/));
aidl.streamType = VALUE_OR_RETURN(
legacy2aidl_audio_stream_type_t_AudioStreamType(legacy.stream_type));
- aidl.bitRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.bit_rate));
+ aidl.bitRatePerSecond = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.bit_rate));
aidl.durationUs = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy.duration_us));
aidl.hasVideo = legacy.has_video;
aidl.isStreaming = legacy.is_streaming;
@@ -2324,25 +2354,25 @@
}
ConversionResult<audio_config_t>
-aidl2legacy_AudioConfig_audio_config_t(const media::AudioConfig& aidl, bool isInput) {
- audio_config_t legacy;
- legacy.sample_rate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.sampleRate));
- legacy.channel_mask = VALUE_OR_RETURN(
- aidl2legacy_AudioChannelLayout_audio_channel_mask_t(aidl.channelMask, isInput));
- legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormatDescription_audio_format_t(aidl.format));
+aidl2legacy_AudioConfig_audio_config_t(const AudioConfig& aidl, bool isInput) {
+ const audio_config_base_t legacyBase = VALUE_OR_RETURN(
+ aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.base, isInput));
+ audio_config_t legacy = AUDIO_CONFIG_INITIALIZER;
+ legacy.sample_rate = legacyBase.sample_rate;
+ legacy.channel_mask = legacyBase.channel_mask;
+ legacy.format = legacyBase.format;
legacy.offload_info = VALUE_OR_RETURN(
aidl2legacy_AudioOffloadInfo_audio_offload_info_t(aidl.offloadInfo));
legacy.frame_count = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.frameCount));
return legacy;
}
-ConversionResult<media::AudioConfig>
+ConversionResult<AudioConfig>
legacy2aidl_audio_config_t_AudioConfig(const audio_config_t& legacy, bool isInput) {
- media::AudioConfig aidl;
- aidl.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
- aidl.channelMask = VALUE_OR_RETURN(
- legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
- aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
+ const audio_config_base_t base = { .sample_rate = legacy.sample_rate,
+ .channel_mask = legacy.channel_mask, .format = legacy.format };
+ AudioConfig aidl;
+ aidl.base = VALUE_OR_RETURN(legacy2aidl_audio_config_base_t_AudioConfigBase(base, isInput));
aidl.offloadInfo = VALUE_OR_RETURN(
legacy2aidl_audio_offload_info_t_AudioOffloadInfo(legacy.offload_info));
aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy.frame_count));
@@ -2350,7 +2380,7 @@
}
ConversionResult<audio_config_base_t>
-aidl2legacy_AudioConfigBase_audio_config_base_t(const media::AudioConfigBase& aidl, bool isInput) {
+aidl2legacy_AudioConfigBase_audio_config_base_t(const AudioConfigBase& aidl, bool isInput) {
audio_config_base_t legacy;
legacy.sample_rate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.sampleRate));
legacy.channel_mask = VALUE_OR_RETURN(
@@ -2359,9 +2389,9 @@
return legacy;
}
-ConversionResult<media::AudioConfigBase>
+ConversionResult<AudioConfigBase>
legacy2aidl_audio_config_base_t_AudioConfigBase(const audio_config_base_t& legacy, bool isInput) {
- media::AudioConfigBase aidl;
+ AudioConfigBase aidl;
aidl.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
aidl.channelMask = VALUE_OR_RETURN(
legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
@@ -2424,7 +2454,7 @@
}
ConversionResult<audio_uuid_t>
-aidl2legacy_AudioUuid_audio_uuid_t(const media::AudioUuid& aidl) {
+aidl2legacy_AudioUuid_audio_uuid_t(const AudioUuid& aidl) {
audio_uuid_t legacy;
legacy.timeLow = VALUE_OR_RETURN(convertReinterpret<uint32_t>(aidl.timeLow));
legacy.timeMid = VALUE_OR_RETURN(convertIntegral<uint16_t>(aidl.timeMid));
@@ -2437,9 +2467,9 @@
return legacy;
}
-ConversionResult<media::AudioUuid>
+ConversionResult<AudioUuid>
legacy2aidl_audio_uuid_t_AudioUuid(const audio_uuid_t& legacy) {
- media::AudioUuid aidl;
+ AudioUuid aidl;
aidl.timeLow = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.timeLow));
aidl.timeMid = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.timeMid));
aidl.timeHiAndVersion = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.timeHiAndVersion));
@@ -2480,28 +2510,28 @@
ConversionResult<audio_encapsulation_metadata_type_t>
aidl2legacy_AudioEncapsulationMetadataType_audio_encapsulation_metadata_type_t(
- media::AudioEncapsulationMetadataType aidl) {
+ AudioEncapsulationMetadataType aidl) {
switch (aidl) {
- case media::AudioEncapsulationMetadataType::NONE:
+ case AudioEncapsulationMetadataType::NONE:
return AUDIO_ENCAPSULATION_METADATA_TYPE_NONE;
- case media::AudioEncapsulationMetadataType::FRAMEWORK_TUNER:
+ case AudioEncapsulationMetadataType::FRAMEWORK_TUNER:
return AUDIO_ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER;
- case media::AudioEncapsulationMetadataType::DVB_AD_DESCRIPTOR:
+ case AudioEncapsulationMetadataType::DVB_AD_DESCRIPTOR:
return AUDIO_ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioEncapsulationMetadataType>
+ConversionResult<AudioEncapsulationMetadataType>
legacy2aidl_audio_encapsulation_metadata_type_t_AudioEncapsulationMetadataType(
audio_encapsulation_metadata_type_t legacy) {
switch (legacy) {
case AUDIO_ENCAPSULATION_METADATA_TYPE_NONE:
- return media::AudioEncapsulationMetadataType::NONE;
+ return AudioEncapsulationMetadataType::NONE;
case AUDIO_ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER:
- return media::AudioEncapsulationMetadataType::FRAMEWORK_TUNER;
+ return AudioEncapsulationMetadataType::FRAMEWORK_TUNER;
case AUDIO_ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR:
- return media::AudioEncapsulationMetadataType::DVB_AD_DESCRIPTOR;
+ return AudioEncapsulationMetadataType::DVB_AD_DESCRIPTOR;
}
return unexpected(BAD_VALUE);
}
@@ -2511,9 +2541,9 @@
return convertBitmask<uint32_t,
int32_t,
audio_encapsulation_mode_t,
- media::AudioEncapsulationMode>(
+ AudioEncapsulationMode>(
aidl, aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t,
- indexToEnum_index<media::AudioEncapsulationMode>,
+ indexToEnum_index<AudioEncapsulationMode>,
enumToMask_index<uint32_t, audio_encapsulation_mode_t>);
}
@@ -2521,11 +2551,11 @@
legacy2aidl_AudioEncapsulationMode_mask(uint32_t legacy) {
return convertBitmask<int32_t,
uint32_t,
- media::AudioEncapsulationMode,
+ AudioEncapsulationMode,
audio_encapsulation_mode_t>(
legacy, legacy2aidl_audio_encapsulation_mode_t_AudioEncapsulationMode,
indexToEnum_index<audio_encapsulation_mode_t>,
- enumToMask_index<int32_t, media::AudioEncapsulationMode>);
+ enumToMask_index<int32_t, AudioEncapsulationMode>);
}
ConversionResult<uint32_t>
@@ -2533,9 +2563,9 @@
return convertBitmask<uint32_t,
int32_t,
audio_encapsulation_metadata_type_t,
- media::AudioEncapsulationMetadataType>(
+ AudioEncapsulationMetadataType>(
aidl, aidl2legacy_AudioEncapsulationMetadataType_audio_encapsulation_metadata_type_t,
- indexToEnum_index<media::AudioEncapsulationMetadataType>,
+ indexToEnum_index<AudioEncapsulationMetadataType>,
enumToMask_index<uint32_t, audio_encapsulation_metadata_type_t>);
}
@@ -2543,106 +2573,78 @@
legacy2aidl_AudioEncapsulationMetadataType_mask(uint32_t legacy) {
return convertBitmask<int32_t,
uint32_t,
- media::AudioEncapsulationMetadataType,
+ AudioEncapsulationMetadataType,
audio_encapsulation_metadata_type_t>(
legacy, legacy2aidl_audio_encapsulation_metadata_type_t_AudioEncapsulationMetadataType,
indexToEnum_index<audio_encapsulation_metadata_type_t>,
- enumToMask_index<int32_t, media::AudioEncapsulationMetadataType>);
-}
-
-ConversionResult<audio_mix_latency_class_t>
-aidl2legacy_AudioMixLatencyClass_audio_mix_latency_class_t(
- media::AudioMixLatencyClass aidl) {
- switch (aidl) {
- case media::AudioMixLatencyClass::LOW:
- return AUDIO_LATENCY_LOW;
- case media::AudioMixLatencyClass::NORMAL:
- return AUDIO_LATENCY_NORMAL;
- }
- return unexpected(BAD_VALUE);
-}
-
-ConversionResult<media::AudioMixLatencyClass>
-legacy2aidl_audio_mix_latency_class_t_AudioMixLatencyClass(
- audio_mix_latency_class_t legacy) {
- switch (legacy) {
- case AUDIO_LATENCY_LOW:
- return media::AudioMixLatencyClass::LOW;
- case AUDIO_LATENCY_NORMAL:
- return media::AudioMixLatencyClass::NORMAL;
- }
- return unexpected(BAD_VALUE);
+ enumToMask_index<int32_t, AudioEncapsulationMetadataType>);
}
ConversionResult<audio_port_device_ext>
-aidl2legacy_AudioPortDeviceExt_audio_port_device_ext(const media::AudioPortDeviceExt& aidl) {
+aidl2legacy_AudioDevice_audio_port_device_ext(
+ const AudioDevice& aidl, const media::AudioPortDeviceExtSys& aidlSys) {
audio_port_device_ext legacy;
- legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
- legacy.type = VALUE_OR_RETURN(
- aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.device.type));
- RETURN_IF_ERROR(
- aidl2legacy_string(aidl.device.address, legacy.address, sizeof(legacy.address)));
+ legacy.hw_module = VALUE_OR_RETURN(
+ aidl2legacy_int32_t_audio_module_handle_t(aidlSys.hwModule));
+ RETURN_IF_ERROR(aidl2legacy_AudioDevice_audio_device(aidl, &legacy.type, legacy.address));
legacy.encapsulation_modes = VALUE_OR_RETURN(
- aidl2legacy_AudioEncapsulationMode_mask(aidl.encapsulationModes));
+ aidl2legacy_AudioEncapsulationMode_mask(aidlSys.encapsulationModes));
legacy.encapsulation_metadata_types = VALUE_OR_RETURN(
- aidl2legacy_AudioEncapsulationMetadataType_mask(aidl.encapsulationMetadataTypes));
+ aidl2legacy_AudioEncapsulationMetadataType_mask(
+ aidlSys.encapsulationMetadataTypes));
return legacy;
}
-ConversionResult<media::AudioPortDeviceExt>
-legacy2aidl_audio_port_device_ext_AudioPortDeviceExt(const audio_port_device_ext& legacy) {
- media::AudioPortDeviceExt aidl;
- aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
- aidl.device.type = VALUE_OR_RETURN(
- legacy2aidl_audio_devices_t_AudioDeviceDescription(legacy.type));
- aidl.device.address = VALUE_OR_RETURN(
- legacy2aidl_string(legacy.address, sizeof(legacy.address)));
- aidl.encapsulationModes = VALUE_OR_RETURN(
+status_t legacy2aidl_audio_port_device_ext_AudioDevice(
+ const audio_port_device_ext& legacy,
+ AudioDevice* aidl, media::AudioPortDeviceExtSys* aidlDeviceExt) {
+ aidlDeviceExt->hwModule = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
+ *aidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_device_AudioDevice(legacy.type, legacy.address));
+ aidlDeviceExt->encapsulationModes = VALUE_OR_RETURN_STATUS(
legacy2aidl_AudioEncapsulationMode_mask(legacy.encapsulation_modes));
- aidl.encapsulationMetadataTypes = VALUE_OR_RETURN(
+ aidlDeviceExt->encapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
legacy2aidl_AudioEncapsulationMetadataType_mask(legacy.encapsulation_metadata_types));
- return aidl;
+ return OK;
}
ConversionResult<audio_port_mix_ext>
-aidl2legacy_AudioPortMixExt_audio_port_mix_ext(const media::AudioPortMixExt& aidl) {
- audio_port_mix_ext legacy;
- legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
+aidl2legacy_AudioPortMixExt_audio_port_mix_ext(
+ const AudioPortMixExt& aidl, const media::AudioPortMixExtSys& aidlSys) {
+ audio_port_mix_ext legacy{};
+ legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidlSys.hwModule));
legacy.handle = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_io_handle_t(aidl.handle));
- legacy.latency_class = VALUE_OR_RETURN(
- aidl2legacy_AudioMixLatencyClass_audio_mix_latency_class_t(aidl.latencyClass));
return legacy;
}
-ConversionResult<media::AudioPortMixExt>
-legacy2aidl_audio_port_mix_ext_AudioPortMixExt(const audio_port_mix_ext& legacy) {
- media::AudioPortMixExt aidl;
- aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
- aidl.handle = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(legacy.handle));
- aidl.latencyClass = VALUE_OR_RETURN(
- legacy2aidl_audio_mix_latency_class_t_AudioMixLatencyClass(legacy.latency_class));
- return aidl;
+status_t
+legacy2aidl_audio_port_mix_ext_AudioPortMixExt(const audio_port_mix_ext& legacy,
+ AudioPortMixExt* aidl, media::AudioPortMixExtSys* aidlMixExt) {
+ aidlMixExt->hwModule = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
+ aidl->handle = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(legacy.handle));
+ return OK;
}
ConversionResult<audio_port_session_ext>
-aidl2legacy_AudioPortSessionExt_audio_port_session_ext(const media::AudioPortSessionExt& aidl) {
+aidl2legacy_int32_t_audio_port_session_ext(int32_t aidl) {
audio_port_session_ext legacy;
- legacy.session = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_session_t(aidl.session));
+ legacy.session = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_session_t(aidl));
return legacy;
}
-ConversionResult<media::AudioPortSessionExt>
-legacy2aidl_audio_port_session_ext_AudioPortSessionExt(const audio_port_session_ext& legacy) {
- media::AudioPortSessionExt aidl;
- aidl.session = VALUE_OR_RETURN(legacy2aidl_audio_session_t_int32_t(legacy.session));
- return aidl;
+ConversionResult<int32_t>
+legacy2aidl_audio_port_session_ext_int32_t(const audio_port_session_ext& legacy) {
+ return legacy2aidl_audio_session_t_int32_t(legacy.session);
}
// This type is unnamed in the original definition, thus we name it here.
using audio_port_v7_ext = decltype(audio_port_v7::ext);
-ConversionResult<audio_port_v7_ext> aidl2legacy_AudioPortExt(
- const media::AudioPortExt& aidl, media::AudioPortType type) {
+ConversionResult<audio_port_v7_ext> aidl2legacy_AudioPortExt_audio_port_v7_ext(
+ const AudioPortExt& aidl, media::AudioPortType type,
+ const media::AudioPortExtSys& aidlSys) {
audio_port_v7_ext legacy;
switch (type) {
case media::AudioPortType::NONE:
@@ -2651,67 +2653,82 @@
return legacy;
case media::AudioPortType::DEVICE:
legacy.device = VALUE_OR_RETURN(
- aidl2legacy_AudioPortDeviceExt_audio_port_device_ext(
- VALUE_OR_RETURN(UNION_GET(aidl, device))));
+ aidl2legacy_AudioDevice_audio_port_device_ext(
+ VALUE_OR_RETURN(UNION_GET(aidl, device)),
+ VALUE_OR_RETURN(UNION_GET(aidlSys, device))));
return legacy;
case media::AudioPortType::MIX:
legacy.mix = VALUE_OR_RETURN(
aidl2legacy_AudioPortMixExt_audio_port_mix_ext(
- VALUE_OR_RETURN(UNION_GET(aidl, mix))));
+ VALUE_OR_RETURN(UNION_GET(aidl, mix)),
+ VALUE_OR_RETURN(UNION_GET(aidlSys, mix))));
return legacy;
case media::AudioPortType::SESSION:
- legacy.session = VALUE_OR_RETURN(aidl2legacy_AudioPortSessionExt_audio_port_session_ext(
- VALUE_OR_RETURN(UNION_GET(aidl, session))));
+ legacy.session = VALUE_OR_RETURN(
+ aidl2legacy_int32_t_audio_port_session_ext(
+ VALUE_OR_RETURN(UNION_GET(aidl, session))));
return legacy;
}
LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
-ConversionResult<media::AudioPortExt> legacy2aidl_AudioPortExt(
- const audio_port_v7_ext& legacy, audio_port_type_t type) {
- media::AudioPortExt aidl;
+status_t legacy2aidl_AudioPortExt(
+ const audio_port_v7_ext& legacy, audio_port_type_t type,
+ AudioPortExt* aidl, media::AudioPortExtSys* aidlSys) {
switch (type) {
case AUDIO_PORT_TYPE_NONE:
- UNION_SET(aidl, unspecified, false);
- return aidl;
- case AUDIO_PORT_TYPE_DEVICE:
- UNION_SET(aidl, device,
- VALUE_OR_RETURN(
- legacy2aidl_audio_port_device_ext_AudioPortDeviceExt(legacy.device)));
- return aidl;
- case AUDIO_PORT_TYPE_MIX:
- UNION_SET(aidl, mix,
- VALUE_OR_RETURN(legacy2aidl_audio_port_mix_ext_AudioPortMixExt(legacy.mix)));
- return aidl;
+ UNION_SET(*aidl, unspecified, false);
+ UNION_SET(*aidlSys, unspecified, false);
+ return OK;
+ case AUDIO_PORT_TYPE_DEVICE: {
+ AudioDevice device;
+ media::AudioPortDeviceExtSys deviceSys;
+ RETURN_STATUS_IF_ERROR(
+ legacy2aidl_audio_port_device_ext_AudioDevice(
+ legacy.device, &device, &deviceSys));
+ UNION_SET(*aidl, device, device);
+ UNION_SET(*aidlSys, device, deviceSys);
+ return OK;
+ }
+ case AUDIO_PORT_TYPE_MIX: {
+ AudioPortMixExt mix;
+ media::AudioPortMixExtSys mixSys;
+ RETURN_STATUS_IF_ERROR(
+ legacy2aidl_audio_port_mix_ext_AudioPortMixExt(
+ legacy.mix, &mix, &mixSys));
+ UNION_SET(*aidl, mix, mix);
+ UNION_SET(*aidlSys, mix, mixSys);
+ return OK;
+ }
case AUDIO_PORT_TYPE_SESSION:
- UNION_SET(aidl, session,
- VALUE_OR_RETURN(legacy2aidl_audio_port_session_ext_AudioPortSessionExt(
- legacy.session)));
- return aidl;
+ UNION_SET(*aidl, session, VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_port_session_ext_int32_t(legacy.session)));
+ UNION_SET(*aidlSys, unspecified, false);
+ return OK;
}
LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
ConversionResult<audio_profile>
-aidl2legacy_AudioProfile_audio_profile(const media::AudioProfile& aidl, bool isInput) {
+aidl2legacy_AudioProfile_audio_profile(const AudioProfile& aidl, bool isInput) {
audio_profile legacy;
legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormatDescription_audio_format_t(aidl.format));
- if (aidl.samplingRates.size() > std::size(legacy.sample_rates)) {
+ if (aidl.sampleRates.size() > std::size(legacy.sample_rates)) {
return unexpected(BAD_VALUE);
}
RETURN_IF_ERROR(
- convertRange(aidl.samplingRates.begin(), aidl.samplingRates.end(), legacy.sample_rates,
+ convertRange(aidl.sampleRates.begin(), aidl.sampleRates.end(), legacy.sample_rates,
convertIntegral<int32_t, unsigned int>));
- legacy.num_sample_rates = aidl.samplingRates.size();
+ legacy.num_sample_rates = aidl.sampleRates.size();
if (aidl.channelMasks.size() > std::size(legacy.channel_masks)) {
return unexpected(BAD_VALUE);
}
RETURN_IF_ERROR(
convertRange(aidl.channelMasks.begin(), aidl.channelMasks.end(), legacy.channel_masks,
- [isInput](const media::AudioChannelLayout& l) {
+ [isInput](const AudioChannelLayout& l) {
return aidl2legacy_AudioChannelLayout_audio_channel_mask_t(l, isInput);
}));
legacy.num_channel_masks = aidl.channelMasks.size();
@@ -2721,9 +2738,9 @@
return legacy;
}
-ConversionResult<media::AudioProfile>
+ConversionResult<AudioProfile>
legacy2aidl_audio_profile_AudioProfile(const audio_profile& legacy, bool isInput) {
- media::AudioProfile aidl;
+ AudioProfile aidl;
aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
if (legacy.num_sample_rates > std::size(legacy.sample_rates)) {
@@ -2731,7 +2748,7 @@
}
RETURN_IF_ERROR(
convertRange(legacy.sample_rates, legacy.sample_rates + legacy.num_sample_rates,
- std::back_inserter(aidl.samplingRates),
+ std::back_inserter(aidl.sampleRates),
convertIntegral<unsigned int, int32_t>));
if (legacy.num_channel_masks > std::size(legacy.channel_masks)) {
@@ -2751,11 +2768,11 @@
}
ConversionResult<audio_gain>
-aidl2legacy_AudioGain_audio_gain(const media::AudioGain& aidl) {
+aidl2legacy_AudioGain_audio_gain(const AudioGain& aidl, bool isInput) {
audio_gain legacy;
legacy.mode = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_gain_mode_t_mask(aidl.mode));
legacy.channel_mask = VALUE_OR_RETURN(aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
- aidl.channelMask, aidl.isInput));
+ aidl.channelMask, isInput));
legacy.min_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.minValue));
legacy.max_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.maxValue));
legacy.default_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.defaultValue));
@@ -2765,11 +2782,10 @@
return legacy;
}
-ConversionResult<media::AudioGain>
+ConversionResult<AudioGain>
legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy, bool isInput) {
- media::AudioGain aidl;
+ AudioGain aidl;
aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_int32_t_mask(legacy.mode));
- aidl.isInput = isInput;
aidl.channelMask = VALUE_OR_RETURN(
legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
aidl.minValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.min_value));
@@ -2784,50 +2800,59 @@
ConversionResult<audio_port_v7>
aidl2legacy_AudioPort_audio_port_v7(const media::AudioPort& aidl) {
audio_port_v7 legacy;
- legacy.id = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.id));
- legacy.role = VALUE_OR_RETURN(aidl2legacy_AudioPortRole_audio_port_role_t(aidl.role));
- legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioPortType_audio_port_type_t(aidl.type));
- RETURN_IF_ERROR(aidl2legacy_string(aidl.name, legacy.name, sizeof(legacy.name)));
+ legacy.id = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.hal.id));
+ legacy.role = VALUE_OR_RETURN(aidl2legacy_AudioPortRole_audio_port_role_t(aidl.sys.role));
+ legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioPortType_audio_port_type_t(aidl.sys.type));
+ RETURN_IF_ERROR(aidl2legacy_string(aidl.hal.name, legacy.name, sizeof(legacy.name)));
- if (aidl.profiles.size() > std::size(legacy.audio_profiles)) {
+ if (aidl.hal.profiles.size() > std::size(legacy.audio_profiles)) {
return unexpected(BAD_VALUE);
}
- const bool isInput = VALUE_OR_RETURN(direction(aidl.role, aidl.type)) == Direction::INPUT;
- RETURN_IF_ERROR(convertRange(aidl.profiles.begin(), aidl.profiles.end(), legacy.audio_profiles,
- [isInput](const media::AudioProfile& p) {
- return aidl2legacy_AudioProfile_audio_profile(p, isInput);
- }));
- legacy.num_audio_profiles = aidl.profiles.size();
+ const bool isInput =
+ VALUE_OR_RETURN(direction(aidl.sys.role, aidl.sys.type)) == Direction::INPUT;
+ RETURN_IF_ERROR(convertRange(
+ aidl.hal.profiles.begin(), aidl.hal.profiles.end(), legacy.audio_profiles,
+ [isInput](const AudioProfile& p) {
+ return aidl2legacy_AudioProfile_audio_profile(p, isInput);
+ }));
+ legacy.num_audio_profiles = aidl.hal.profiles.size();
- if (aidl.extraAudioDescriptors.size() > std::size(legacy.extra_audio_descriptors)) {
+ if (aidl.hal.extraAudioDescriptors.size() > std::size(legacy.extra_audio_descriptors)) {
return unexpected(BAD_VALUE);
}
RETURN_IF_ERROR(
- convertRange(aidl.extraAudioDescriptors.begin(), aidl.extraAudioDescriptors.end(),
- legacy.extra_audio_descriptors,
- aidl2legacy_ExtraAudioDescriptor_audio_extra_audio_descriptor));
- legacy.num_extra_audio_descriptors = aidl.extraAudioDescriptors.size();
+ convertRange(
+ aidl.hal.extraAudioDescriptors.begin(), aidl.hal.extraAudioDescriptors.end(),
+ legacy.extra_audio_descriptors,
+ aidl2legacy_ExtraAudioDescriptor_audio_extra_audio_descriptor));
+ legacy.num_extra_audio_descriptors = aidl.hal.extraAudioDescriptors.size();
- if (aidl.gains.size() > std::size(legacy.gains)) {
+ if (aidl.hal.gains.size() > std::size(legacy.gains)) {
return unexpected(BAD_VALUE);
}
- RETURN_IF_ERROR(convertRange(aidl.gains.begin(), aidl.gains.end(), legacy.gains,
- aidl2legacy_AudioGain_audio_gain));
- legacy.num_gains = aidl.gains.size();
+ RETURN_IF_ERROR(convertRange(aidl.hal.gains.begin(), aidl.hal.gains.end(), legacy.gains,
+ [isInput](const AudioGain& g) {
+ return aidl2legacy_AudioGain_audio_gain(g, isInput);
+ }));
+ legacy.num_gains = aidl.hal.gains.size();
+ media::AudioPortConfig aidlPortConfig;
+ aidlPortConfig.hal = aidl.hal.activeConfig;
+ aidlPortConfig.sys = aidl.sys.activeConfig;
legacy.active_config = VALUE_OR_RETURN(
- aidl2legacy_AudioPortConfig_audio_port_config(aidl.activeConfig));
- legacy.ext = VALUE_OR_RETURN(aidl2legacy_AudioPortExt(aidl.ext, aidl.type));
+ aidl2legacy_AudioPortConfig_audio_port_config(aidlPortConfig));
+ legacy.ext = VALUE_OR_RETURN(
+ aidl2legacy_AudioPortExt_audio_port_v7_ext(aidl.hal.ext, aidl.sys.type, aidl.sys.ext));
return legacy;
}
ConversionResult<media::AudioPort>
legacy2aidl_audio_port_v7_AudioPort(const audio_port_v7& legacy) {
media::AudioPort aidl;
- aidl.id = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy.id));
- aidl.role = VALUE_OR_RETURN(legacy2aidl_audio_port_role_t_AudioPortRole(legacy.role));
- aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_port_type_t_AudioPortType(legacy.type));
- aidl.name = VALUE_OR_RETURN(legacy2aidl_string(legacy.name, sizeof(legacy.name)));
+ aidl.hal.id = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy.id));
+ aidl.sys.role = VALUE_OR_RETURN(legacy2aidl_audio_port_role_t_AudioPortRole(legacy.role));
+ aidl.sys.type = VALUE_OR_RETURN(legacy2aidl_audio_port_type_t_AudioPortType(legacy.type));
+ aidl.hal.name = VALUE_OR_RETURN(legacy2aidl_string(legacy.name, sizeof(legacy.name)));
if (legacy.num_audio_profiles > std::size(legacy.audio_profiles)) {
return unexpected(BAD_VALUE);
@@ -2835,7 +2860,7 @@
const bool isInput = VALUE_OR_RETURN(direction(legacy.role, legacy.type)) == Direction::INPUT;
RETURN_IF_ERROR(
convertRange(legacy.audio_profiles, legacy.audio_profiles + legacy.num_audio_profiles,
- std::back_inserter(aidl.profiles),
+ std::back_inserter(aidl.hal.profiles),
[isInput](const audio_profile& p) {
return legacy2aidl_audio_profile_AudioProfile(p, isInput);
}));
@@ -2843,10 +2868,11 @@
if (legacy.num_extra_audio_descriptors > std::size(legacy.extra_audio_descriptors)) {
return unexpected(BAD_VALUE);
}
+ aidl.sys.profiles.resize(legacy.num_audio_profiles);
RETURN_IF_ERROR(
convertRange(legacy.extra_audio_descriptors,
legacy.extra_audio_descriptors + legacy.num_extra_audio_descriptors,
- std::back_inserter(aidl.extraAudioDescriptors),
+ std::back_inserter(aidl.hal.extraAudioDescriptors),
legacy2aidl_audio_extra_audio_descriptor_ExtraAudioDescriptor));
if (legacy.num_gains > std::size(legacy.gains)) {
@@ -2854,55 +2880,59 @@
}
RETURN_IF_ERROR(
convertRange(legacy.gains, legacy.gains + legacy.num_gains,
- std::back_inserter(aidl.gains),
+ std::back_inserter(aidl.hal.gains),
[isInput](const audio_gain& g) {
return legacy2aidl_audio_gain_AudioGain(g, isInput);
}));
+ aidl.sys.gains.resize(legacy.num_gains);
- aidl.activeConfig = VALUE_OR_RETURN(
+ media::AudioPortConfig aidlPortConfig = VALUE_OR_RETURN(
legacy2aidl_audio_port_config_AudioPortConfig(legacy.active_config));
- aidl.ext = VALUE_OR_RETURN(legacy2aidl_AudioPortExt(legacy.ext, legacy.type));
+ aidl.hal.activeConfig = aidlPortConfig.hal;
+ aidl.sys.activeConfig = aidlPortConfig.sys;
+ RETURN_IF_ERROR(
+ legacy2aidl_AudioPortExt(legacy.ext, legacy.type, &aidl.hal.ext, &aidl.sys.ext));
return aidl;
}
ConversionResult<audio_mode_t>
-aidl2legacy_AudioMode_audio_mode_t(media::AudioMode aidl) {
+aidl2legacy_AudioMode_audio_mode_t(AudioMode aidl) {
switch (aidl) {
- case media::AudioMode::INVALID:
+ case AudioMode::SYS_RESERVED_INVALID:
return AUDIO_MODE_INVALID;
- case media::AudioMode::CURRENT:
+ case AudioMode::SYS_RESERVED_CURRENT:
return AUDIO_MODE_CURRENT;
- case media::AudioMode::NORMAL:
+ case AudioMode::NORMAL:
return AUDIO_MODE_NORMAL;
- case media::AudioMode::RINGTONE:
+ case AudioMode::RINGTONE:
return AUDIO_MODE_RINGTONE;
- case media::AudioMode::IN_CALL:
+ case AudioMode::IN_CALL:
return AUDIO_MODE_IN_CALL;
- case media::AudioMode::IN_COMMUNICATION:
+ case AudioMode::IN_COMMUNICATION:
return AUDIO_MODE_IN_COMMUNICATION;
- case media::AudioMode::CALL_SCREEN:
+ case AudioMode::CALL_SCREEN:
return AUDIO_MODE_CALL_SCREEN;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioMode>
+ConversionResult<AudioMode>
legacy2aidl_audio_mode_t_AudioMode(audio_mode_t legacy) {
switch (legacy) {
case AUDIO_MODE_INVALID:
- return media::AudioMode::INVALID;
+ return AudioMode::SYS_RESERVED_INVALID;
case AUDIO_MODE_CURRENT:
- return media::AudioMode::CURRENT;
+ return AudioMode::SYS_RESERVED_CURRENT;
case AUDIO_MODE_NORMAL:
- return media::AudioMode::NORMAL;
+ return AudioMode::NORMAL;
case AUDIO_MODE_RINGTONE:
- return media::AudioMode::RINGTONE;
+ return AudioMode::RINGTONE;
case AUDIO_MODE_IN_CALL:
- return media::AudioMode::IN_CALL;
+ return AudioMode::IN_CALL;
case AUDIO_MODE_IN_COMMUNICATION:
- return media::AudioMode::IN_COMMUNICATION;
+ return AudioMode::IN_COMMUNICATION;
case AUDIO_MODE_CALL_SCREEN:
- return media::AudioMode::CALL_SCREEN;
+ return AudioMode::CALL_SCREEN;
case AUDIO_MODE_CNT:
break;
}
@@ -3052,30 +3082,30 @@
}
ConversionResult<audio_standard_t>
-aidl2legacy_AudioStandard_audio_standard_t(media::AudioStandard aidl) {
+aidl2legacy_AudioStandard_audio_standard_t(AudioStandard aidl) {
switch (aidl) {
- case media::AudioStandard::NONE:
+ case AudioStandard::NONE:
return AUDIO_STANDARD_NONE;
- case media::AudioStandard::EDID:
+ case AudioStandard::EDID:
return AUDIO_STANDARD_EDID;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioStandard>
+ConversionResult<AudioStandard>
legacy2aidl_audio_standard_t_AudioStandard(audio_standard_t legacy) {
switch (legacy) {
case AUDIO_STANDARD_NONE:
- return media::AudioStandard::NONE;
+ return AudioStandard::NONE;
case AUDIO_STANDARD_EDID:
- return media::AudioStandard::EDID;
+ return AudioStandard::EDID;
}
return unexpected(BAD_VALUE);
}
ConversionResult<audio_extra_audio_descriptor>
aidl2legacy_ExtraAudioDescriptor_audio_extra_audio_descriptor(
- const media::ExtraAudioDescriptor& aidl) {
+ const ExtraAudioDescriptor& aidl) {
audio_extra_audio_descriptor legacy;
legacy.standard = VALUE_OR_RETURN(aidl2legacy_AudioStandard_audio_standard_t(aidl.standard));
if (aidl.audioDescriptor.size() > EXTRA_AUDIO_DESCRIPTOR_SIZE) {
@@ -3090,10 +3120,10 @@
return legacy;
}
-ConversionResult<media::ExtraAudioDescriptor>
+ConversionResult<ExtraAudioDescriptor>
legacy2aidl_audio_extra_audio_descriptor_ExtraAudioDescriptor(
const audio_extra_audio_descriptor& legacy) {
- media::ExtraAudioDescriptor aidl;
+ ExtraAudioDescriptor aidl;
aidl.standard = VALUE_OR_RETURN(legacy2aidl_audio_standard_t_AudioStandard(legacy.standard));
if (legacy.descriptor_length > EXTRA_AUDIO_DESCRIPTOR_SIZE) {
return unexpected(BAD_VALUE);
@@ -3109,24 +3139,24 @@
ConversionResult<audio_encapsulation_type_t>
aidl2legacy_AudioEncapsulationType_audio_encapsulation_type_t(
- const media::AudioEncapsulationType& aidl) {
+ const AudioEncapsulationType& aidl) {
switch (aidl) {
- case media::AudioEncapsulationType::NONE:
+ case AudioEncapsulationType::NONE:
return AUDIO_ENCAPSULATION_TYPE_NONE;
- case media::AudioEncapsulationType::IEC61937:
+ case AudioEncapsulationType::IEC61937:
return AUDIO_ENCAPSULATION_TYPE_IEC61937;
}
return unexpected(BAD_VALUE);
}
-ConversionResult<media::AudioEncapsulationType>
+ConversionResult<AudioEncapsulationType>
legacy2aidl_audio_encapsulation_type_t_AudioEncapsulationType(
const audio_encapsulation_type_t & legacy) {
switch (legacy) {
case AUDIO_ENCAPSULATION_TYPE_NONE:
- return media::AudioEncapsulationType::NONE;
+ return AudioEncapsulationType::NONE;
case AUDIO_ENCAPSULATION_TYPE_IEC61937:
- return media::AudioEncapsulationType::IEC61937;
+ return AudioEncapsulationType::IEC61937;
}
return unexpected(BAD_VALUE);
}
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 9b9ceea..f6446bb 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -25,11 +25,13 @@
static_libs: [
"audioflinger-aidl-cpp",
"audiopolicy-aidl-cpp",
+ "spatializer-aidl-cpp",
"av-types-aidl-cpp",
],
export_static_lib_headers: [
"audioflinger-aidl-cpp",
"audiopolicy-aidl-cpp",
+ "spatializer-aidl-cpp",
"av-types-aidl-cpp",
],
target: {
@@ -49,6 +51,7 @@
"PolicyAidlConversion.cpp"
],
shared_libs: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"audioflinger-aidl-cpp",
"audiopolicy-aidl-cpp",
@@ -69,6 +72,7 @@
include_dirs: ["system/media/audio_utils/include"],
export_include_dirs: ["include"],
export_shared_lib_headers: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"audioflinger-aidl-cpp",
"audiopolicy-aidl-cpp",
@@ -109,9 +113,11 @@
"TrackPlayerBase.cpp",
],
shared_libs: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"audioflinger-aidl-cpp",
"audiopolicy-aidl-cpp",
+ "spatializer-aidl-cpp",
"audiopolicy-types-aidl-cpp",
"av-types-aidl-cpp",
"capture_state_listener-aidl-cpp",
@@ -131,12 +137,12 @@
"libprocessgroup",
"libshmemcompat",
"libutils",
- "libvibrator",
"framework-permission-aidl-cpp",
],
export_shared_lib_headers: [
"audioflinger-aidl-cpp",
"audiopolicy-aidl-cpp",
+ "spatializer-aidl-cpp",
"framework-permission-aidl-cpp",
"libbinder",
],
@@ -224,6 +230,7 @@
"libaudioclient_aidl_conversion_util",
],
shared_libs: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"libbase",
"libbinder",
@@ -235,6 +242,7 @@
"framework-permission-aidl-cpp",
],
export_shared_lib_headers: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"libbase",
"shared-file-region-aidl-cpp",
@@ -303,60 +311,31 @@
local_include_dir: "aidl",
srcs: [
"aidl/android/media/AudioAttributesInternal.aidl",
- "aidl/android/media/AudioChannelLayout.aidl",
"aidl/android/media/AudioClient.aidl",
- "aidl/android/media/AudioConfig.aidl",
- "aidl/android/media/AudioConfigBase.aidl",
- "aidl/android/media/AudioContentType.aidl",
- "aidl/android/media/AudioDevice.aidl",
- "aidl/android/media/AudioDeviceDescription.aidl",
- "aidl/android/media/AudioDeviceType.aidl",
"aidl/android/media/AudioDualMonoMode.aidl",
- "aidl/android/media/AudioEncapsulationMode.aidl",
- "aidl/android/media/AudioEncapsulationMetadataType.aidl",
- "aidl/android/media/AudioEncapsulationType.aidl",
- "aidl/android/media/AudioFormatDescription.aidl",
- "aidl/android/media/AudioFormatType.aidl",
"aidl/android/media/AudioFlag.aidl",
- "aidl/android/media/AudioGain.aidl",
- "aidl/android/media/AudioGainConfig.aidl",
- "aidl/android/media/AudioGainMode.aidl",
+ "aidl/android/media/AudioGainSys.aidl",
"aidl/android/media/AudioInputFlags.aidl",
"aidl/android/media/AudioIoConfigEvent.aidl",
"aidl/android/media/AudioIoDescriptor.aidl",
"aidl/android/media/AudioIoFlags.aidl",
- "aidl/android/media/AudioMixLatencyClass.aidl",
- "aidl/android/media/AudioMode.aidl",
- "aidl/android/media/AudioOffloadInfo.aidl",
"aidl/android/media/AudioOutputFlags.aidl",
"aidl/android/media/AudioPatch.aidl",
"aidl/android/media/AudioPlaybackRate.aidl",
"aidl/android/media/AudioPort.aidl",
+ "aidl/android/media/AudioPortSys.aidl",
"aidl/android/media/AudioPortConfig.aidl",
- "aidl/android/media/AudioPortConfigType.aidl",
- "aidl/android/media/AudioPortConfigDeviceExt.aidl",
- "aidl/android/media/AudioPortConfigExt.aidl",
- "aidl/android/media/AudioPortConfigMixExt.aidl",
- "aidl/android/media/AudioPortConfigMixExtUseCase.aidl",
- "aidl/android/media/AudioPortConfigSessionExt.aidl",
- "aidl/android/media/AudioPortDeviceExt.aidl",
- "aidl/android/media/AudioPortExt.aidl",
- "aidl/android/media/AudioPortMixExt.aidl",
+ "aidl/android/media/AudioPortConfigSys.aidl",
+ "aidl/android/media/AudioPortDeviceExtSys.aidl",
+ "aidl/android/media/AudioPortExtSys.aidl",
+ "aidl/android/media/AudioPortMixExtSys.aidl",
"aidl/android/media/AudioPortRole.aidl",
- "aidl/android/media/AudioPortSessionExt.aidl",
"aidl/android/media/AudioPortType.aidl",
- "aidl/android/media/AudioProfile.aidl",
- "aidl/android/media/AudioSourceType.aidl",
- "aidl/android/media/AudioStandard.aidl",
- "aidl/android/media/AudioStreamType.aidl",
+ "aidl/android/media/AudioProfileSys.aidl",
"aidl/android/media/AudioTimestampInternal.aidl",
"aidl/android/media/AudioUniqueIdUse.aidl",
- "aidl/android/media/AudioUsage.aidl",
- "aidl/android/media/AudioUuid.aidl",
"aidl/android/media/AudioVibratorInfo.aidl",
"aidl/android/media/EffectDescriptor.aidl",
- "aidl/android/media/ExtraAudioDescriptor.aidl",
- "aidl/android/media/PcmType.aidl",
"aidl/android/media/TrackSecondaryOutputInfo.aidl",
],
imports: [
@@ -398,11 +377,13 @@
"aidl/android/media/AudioProductStrategy.aidl",
"aidl/android/media/AudioVolumeGroup.aidl",
"aidl/android/media/DeviceRole.aidl",
- "aidl/android/media/HeadTrackingMode.aidl",
"aidl/android/media/SoundTriggerSession.aidl",
"aidl/android/media/SpatializationLevel.aidl",
+ "aidl/android/media/SpatializationMode.aidl",
+ "aidl/android/media/SpatializerHeadTrackingMode.aidl",
],
imports: [
+ "android.media.audio.common.types",
"audioclient-types-aidl",
],
backend: {
@@ -445,6 +426,7 @@
"aidl/android/media/IAudioTrackCallback.aidl",
],
imports: [
+ "android.media.audio.common.types",
"audioclient-types-aidl",
"av-types-aidl",
"effect-aidl",
@@ -476,18 +458,47 @@
"aidl/android/media/GetInputForAttrResponse.aidl",
"aidl/android/media/GetOutputForAttrResponse.aidl",
"aidl/android/media/GetSpatializerResponse.aidl",
- "aidl/android/media/Int.aidl",
"aidl/android/media/RecordClientInfo.aidl",
"aidl/android/media/IAudioPolicyService.aidl",
"aidl/android/media/IAudioPolicyServiceClient.aidl",
- "aidl/android/media/INativeSpatializerCallback.aidl",
- "aidl/android/media/ISpatializer.aidl",
],
imports: [
+ "android.media.audio.common.types",
"audioclient-types-aidl",
"audiopolicy-types-aidl",
"capture_state_listener-aidl",
"framework-permission-aidl",
+ "spatializer-aidl",
+ ],
+
+ double_loadable: true,
+ backend: {
+ cpp: {
+ min_sdk_version: "29",
+ apex_available: [
+ "//apex_available:platform",
+ "com.android.media",
+ ],
+ },
+ java: {
+ sdk_version: "module_current",
+ },
+ },
+}
+
+aidl_interface {
+ name: "spatializer-aidl",
+ unstable: true,
+ local_include_dir: "aidl",
+ host_supported: true,
+ vendor_available: true,
+ srcs: [
+ "aidl/android/media/INativeSpatializerCallback.aidl",
+ "aidl/android/media/ISpatializer.aidl",
+ "aidl/android/media/ISpatializerHeadTrackingCallback.aidl",
+ ],
+ imports: [
+ "audiopolicy-types-aidl",
],
double_loadable: true,
diff --git a/media/libaudioclient/AudioAttributes.cpp b/media/libaudioclient/AudioAttributes.cpp
index 83bf5a7..260c06c 100644
--- a/media/libaudioclient/AudioAttributes.cpp
+++ b/media/libaudioclient/AudioAttributes.cpp
@@ -24,9 +24,6 @@
#include <media/AudioAttributes.h>
#include <media/PolicyAidlConversion.h>
-#define RETURN_STATUS_IF_ERROR(x) \
- { auto _tmp = (x); if (_tmp != OK) return _tmp; }
-
namespace android {
status_t AudioAttributes::readFromParcel(const Parcel* parcel) {
diff --git a/media/libaudioclient/AudioEffect.cpp b/media/libaudioclient/AudioEffect.cpp
index 9091599..62f863d 100644
--- a/media/libaudioclient/AudioEffect.cpp
+++ b/media/libaudioclient/AudioEffect.cpp
@@ -32,16 +32,12 @@
#include <private/media/AudioEffectShared.h>
#include <utils/Log.h>
-#define RETURN_STATUS_IF_ERROR(x) \
- { \
- auto _tmp = (x); \
- if (_tmp != OK) return _tmp; \
- }
-
namespace android {
using aidl_utils::statusTFromBinderStatus;
using binder::Status;
using media::IAudioPolicyService;
+using media::audio::common::AudioSource;
+using media::audio::common::AudioUuid;
namespace {
@@ -571,7 +567,7 @@
int32_t audioSessionAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_session_t_int32_t(audioSession));
- media::Int countAidl;
+ media::audio::common::Int countAidl;
countAidl.value = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*count));
std::vector<media::EffectDescriptor> retAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
@@ -619,12 +615,12 @@
uuid = *EFFECT_UUID_NULL;
}
- media::AudioUuid typeAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_uuid_t_AudioUuid(type));
- media::AudioUuid uuidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_uuid_t_AudioUuid(uuid));
+ AudioUuid typeAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_uuid_t_AudioUuid(type));
+ AudioUuid uuidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_uuid_t_AudioUuid(uuid));
std::string opPackageNameAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_String16_string(opPackageName));
- media::AudioSourceType sourceAidl = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_source_t_AudioSourceType(source));
+ AudioSource sourceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_source_t_AudioSource(source));
int32_t retAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->addSourceDefaultEffect(typeAidl, opPackageNameAidl, uuidAidl, priority, sourceAidl,
@@ -662,11 +658,11 @@
uuid = *EFFECT_UUID_NULL;
}
- media::AudioUuid typeAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_uuid_t_AudioUuid(type));
- media::AudioUuid uuidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_uuid_t_AudioUuid(uuid));
+ AudioUuid typeAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_uuid_t_AudioUuid(type));
+ AudioUuid uuidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_uuid_t_AudioUuid(uuid));
std::string opPackageNameAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_String16_string(opPackageName));
- media::AudioUsage usageAidl = VALUE_OR_RETURN_STATUS(
+ media::audio::common::AudioUsage usageAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_usage_t_AudioUsage(usage));
int32_t retAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
diff --git a/media/libaudioclient/AudioProductStrategy.cpp b/media/libaudioclient/AudioProductStrategy.cpp
index f98027a..ecd423a 100644
--- a/media/libaudioclient/AudioProductStrategy.cpp
+++ b/media/libaudioclient/AudioProductStrategy.cpp
@@ -21,9 +21,6 @@
#include <media/AudioAttributes.h>
#include <media/PolicyAidlConversion.h>
-#define RETURN_STATUS_IF_ERROR(x) \
- { auto _tmp = (x); if (_tmp != OK) return _tmp; }
-
namespace android {
status_t AudioProductStrategy::readFromParcel(const Parcel* parcel) {
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 65bf97d..edad6d1 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -40,19 +40,25 @@
if (!_tmp.ok()) return aidl_utils::binderStatusFromStatusT(_tmp.error()); \
std::move(_tmp.value()); })
-#define RETURN_STATUS_IF_ERROR(x) \
- { \
- auto _tmp = (x); \
- if (_tmp != OK) return _tmp; \
- }
-
// ----------------------------------------------------------------------------
namespace android {
using aidl_utils::statusTFromBinderStatus;
using binder::Status;
+using content::AttributionSourceState;
using media::IAudioPolicyService;
-using android::content::AttributionSourceState;
+using media::audio::common::AudioConfig;
+using media::audio::common::AudioConfigBase;
+using media::audio::common::AudioDevice;
+using media::audio::common::AudioDeviceDescription;
+using media::audio::common::AudioFormatDescription;
+using media::audio::common::AudioMMapPolicyInfo;
+using media::audio::common::AudioMMapPolicyType;
+using media::audio::common::AudioOffloadInfo;
+using media::audio::common::AudioSource;
+using media::audio::common::AudioStreamType;
+using media::audio::common::AudioUsage;
+using media::audio::common::Int;
// client singleton for AudioFlinger binder interface
Mutex AudioSystem::gLock;
@@ -846,10 +852,8 @@
name = device_name;
}
- media::AudioDevice deviceAidl;
- deviceAidl.type = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
- deviceAidl.address = address;
+ AudioDevice deviceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_device_AudioDevice(device, address));
return statusTFromBinderStatus(
aps->setDeviceConnectionState(
@@ -867,10 +871,8 @@
if (aps == 0) return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
auto result = [&]() -> ConversionResult<audio_policy_dev_state_t> {
- media::AudioDevice deviceAidl;
- deviceAidl.type = VALUE_OR_RETURN(
- legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
- deviceAidl.address = device_address;
+ AudioDevice deviceAidl = VALUE_OR_RETURN(
+ legacy2aidl_audio_device_AudioDevice(device, device_address));
media::AudioPolicyDeviceState result;
RETURN_IF_ERROR(statusTFromBinderStatus(
@@ -898,10 +900,8 @@
name = device_name;
}
- media::AudioDevice deviceAidl;
- deviceAidl.type = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
- deviceAidl.address = address;
+ AudioDevice deviceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_device_AudioDevice(device, address));
return statusTFromBinderStatus(
aps->handleDeviceConfigChange(deviceAidl, name, VALUE_OR_RETURN_STATUS(
@@ -954,7 +954,7 @@
if (aps == 0) return AUDIO_IO_HANDLE_NONE;
auto result = [&]() -> ConversionResult<audio_io_handle_t> {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN(
+ AudioStreamType streamAidl = VALUE_OR_RETURN(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t outputAidl;
RETURN_IF_ERROR(
@@ -1002,7 +1002,7 @@
media::AudioAttributesInternal attrAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_attributes_t_AudioAttributesInternal(*attr));
int32_t sessionAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_session_t_int32_t(session));
- media::AudioConfig configAidl = VALUE_OR_RETURN_STATUS(
+ AudioConfig configAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_t_AudioConfig(*config, false /*isInput*/));
int32_t flagsAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
@@ -1096,7 +1096,7 @@
int32_t inputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(*input));
int32_t riidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_unique_id_t_int32_t(riid));
int32_t sessionAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_session_t_int32_t(session));
- media::AudioConfigBase configAidl = VALUE_OR_RETURN_STATUS(
+ AudioConfigBase configAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_base_t_AudioConfigBase(*config, true /*isInput*/));
int32_t flagsAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
int32_t selectedDeviceIdAidl = VALUE_OR_RETURN_STATUS(
@@ -1153,7 +1153,7 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t indexMinAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(indexMin));
int32_t indexMaxAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(indexMax));
@@ -1167,10 +1167,10 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t indexAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(index));
- media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
+ AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
return statusTFromBinderStatus(
aps->setStreamVolumeIndex(streamAidl, deviceAidl, indexAidl));
@@ -1182,9 +1182,9 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
- media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
+ AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
int32_t indexAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
@@ -1204,7 +1204,7 @@
media::AudioAttributesInternal attrAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_attributes_t_AudioAttributesInternal(attr));
int32_t indexAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(index));
- media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
+ AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
return statusTFromBinderStatus(
aps->setVolumeIndexForAttributes(attrAidl, deviceAidl, indexAidl));
@@ -1218,7 +1218,7 @@
media::AudioAttributesInternal attrAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_attributes_t_AudioAttributesInternal(attr));
- media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
+ AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
int32_t indexAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
@@ -1258,7 +1258,7 @@
if (aps == 0) return PRODUCT_STRATEGY_NONE;
auto result = [&]() -> ConversionResult<product_strategy_t> {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN(
+ AudioStreamType streamAidl = VALUE_OR_RETURN(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t resultAidl;
RETURN_IF_ERROR(statusTFromBinderStatus(
@@ -1273,9 +1273,9 @@
if (aps == 0) return DeviceTypeSet{};
auto result = [&]() -> ConversionResult<DeviceTypeSet> {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN(
+ AudioStreamType streamAidl = VALUE_OR_RETURN(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
- std::vector<media::AudioDeviceDescription> resultAidl;
+ std::vector<AudioDeviceDescription> resultAidl;
RETURN_IF_ERROR(statusTFromBinderStatus(
aps->getDevicesForStream(streamAidl, &resultAidl)));
return convertContainer<DeviceTypeSet>(resultAidl,
@@ -1294,7 +1294,7 @@
media::AudioAttributesEx aaAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_AudioAttributes_AudioAttributesEx(aa));
- std::vector<media::AudioDevice> retAidl;
+ std::vector<AudioDevice> retAidl;
RETURN_STATUS_IF_ERROR(
statusTFromBinderStatus(aps->getDevicesForAttributes(aaAidl, &retAidl)));
*devices = VALUE_OR_RETURN_STATUS(
@@ -1372,7 +1372,7 @@
if (aps == 0) return PERMISSION_DENIED;
if (state == NULL) return BAD_VALUE;
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t inPastMsAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(inPastMs));
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
@@ -1386,7 +1386,7 @@
if (aps == 0) return PERMISSION_DENIED;
if (state == NULL) return BAD_VALUE;
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t inPastMsAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(inPastMs));
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
@@ -1399,8 +1399,8 @@
if (aps == 0) return PERMISSION_DENIED;
if (state == NULL) return BAD_VALUE;
- media::AudioSourceType streamAidl = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_source_t_AudioSourceType(stream));
+ AudioSource streamAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_source_t_AudioSource(stream));
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->isSourceActive(streamAidl, state)));
return OK;
@@ -1444,9 +1444,9 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == nullptr) return PERMISSION_DENIED;
- std::vector<media::AudioUsage> systemUsagesAidl = VALUE_OR_RETURN_STATUS(
- convertContainer<std::vector<media::AudioUsage>>(systemUsages,
- legacy2aidl_audio_usage_t_AudioUsage));
+ std::vector<AudioUsage> systemUsagesAidl = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<AudioUsage>>(systemUsages,
+ legacy2aidl_audio_usage_t_AudioUsage));
return statusTFromBinderStatus(aps->setSupportedSystemUsages(systemUsagesAidl));
}
@@ -1466,7 +1466,7 @@
if (aps == 0) return AUDIO_OFFLOAD_NOT_SUPPORTED;
auto result = [&]() -> ConversionResult<audio_offload_mode_t> {
- media::AudioOffloadInfo infoAidl = VALUE_OR_RETURN(
+ AudioOffloadInfo infoAidl = VALUE_OR_RETURN(
legacy2aidl_audio_offload_info_t_AudioOffloadInfo(info));
media::AudioOffloadMode retAidl;
RETURN_IF_ERROR(
@@ -1494,7 +1494,7 @@
legacy2aidl_audio_port_role_t_AudioPortRole(role));
media::AudioPortType typeAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_port_type_t_AudioPortType(type));
- media::Int numPortsAidl;
+ Int numPortsAidl;
numPortsAidl.value = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*num_ports));
std::vector<media::AudioPort> portsAidl;
int32_t generationAidl;
@@ -1561,7 +1561,7 @@
if (aps == 0) return PERMISSION_DENIED;
- media::Int numPatchesAidl;
+ Int numPatchesAidl;
numPatchesAidl.value = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*num_patches));
std::vector<media::AudioPatch> patchesAidl;
int32_t generationAidl;
@@ -1718,7 +1718,7 @@
if (aps == 0) return AUDIO_MODE_INVALID;
auto result = [&]() -> ConversionResult<audio_mode_t> {
- media::AudioMode retAidl;
+ media::audio::common::AudioMode retAidl;
RETURN_IF_ERROR(statusTFromBinderStatus(aps->getPhoneState(&retAidl)));
return aidl2legacy_AudioMode_audio_mode_t(retAidl);
}();
@@ -1743,8 +1743,8 @@
if (aps == 0) return PERMISSION_DENIED;
int32_t uidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(uid));
- std::vector<media::AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
+ std::vector<AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<AudioDevice>>(devices,
legacy2aidl_AudioDeviceTypeAddress));
return statusTFromBinderStatus(aps->setUidDeviceAffinities(uidAidl, devicesAidl));
}
@@ -1763,9 +1763,9 @@
if (aps == 0) return PERMISSION_DENIED;
int32_t userIdAidl = VALUE_OR_RETURN_STATUS(convertReinterpret<int32_t>(userId));
- std::vector<media::AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ std::vector<AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
return statusTFromBinderStatus(
aps->setUserIdDeviceAffinities(userIdAidl, devicesAidl));
}
@@ -1838,10 +1838,10 @@
if (aps == 0) return NAN;
auto result = [&]() -> ConversionResult<float> {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN(
+ AudioStreamType streamAidl = VALUE_OR_RETURN(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t indexAidl = VALUE_OR_RETURN(convertIntegral<int32_t>(index));
- media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN(
+ AudioDeviceDescription deviceAidl = VALUE_OR_RETURN(
legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
float retAidl;
RETURN_IF_ERROR(statusTFromBinderStatus(
@@ -1874,10 +1874,10 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- media::Int numSurroundFormatsAidl;
+ Int numSurroundFormatsAidl;
numSurroundFormatsAidl.value =
VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*numSurroundFormats));
- std::vector<media::AudioFormatDescription> surroundFormatsAidl;
+ std::vector<AudioFormatDescription> surroundFormatsAidl;
std::vector<bool> surroundFormatsEnabledAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->getSurroundFormats(&numSurroundFormatsAidl, &surroundFormatsAidl,
@@ -1901,10 +1901,10 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- media::Int numSurroundFormatsAidl;
+ Int numSurroundFormatsAidl;
numSurroundFormatsAidl.value =
VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*numSurroundFormats));
- std::vector<media::AudioFormatDescription> surroundFormatsAidl;
+ std::vector<AudioFormatDescription> surroundFormatsAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->getReportedSurroundFormats(&numSurroundFormatsAidl, &surroundFormatsAidl)));
@@ -1920,7 +1920,7 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- media::AudioFormatDescription audioFormatAidl = VALUE_OR_RETURN_STATUS(
+ AudioFormatDescription audioFormatAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_format_t_AudioFormatDescription(audioFormat));
return statusTFromBinderStatus(
aps->setSurroundFormatEnabled(audioFormatAidl, enabled));
@@ -1982,7 +1982,7 @@
& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- std::vector<media::AudioFormatDescription> formatsAidl;
+ std::vector<AudioFormatDescription> formatsAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->getHwOffloadEncodingFormatsSupportedForA2DP(&formatsAidl)));
*formats = VALUE_OR_RETURN_STATUS(
@@ -2127,9 +2127,9 @@
int32_t strategyAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_product_strategy_t_int32_t(strategy));
media::DeviceRole roleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_device_role_t_DeviceRole(role));
- std::vector<media::AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ std::vector<AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
return statusTFromBinderStatus(
aps->setDevicesRoleForStrategy(strategyAidl, roleAidl, devicesAidl));
}
@@ -2155,7 +2155,7 @@
}
int32_t strategyAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_product_strategy_t_int32_t(strategy));
media::DeviceRole roleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_device_role_t_DeviceRole(role));
- std::vector<media::AudioDevice> devicesAidl;
+ std::vector<AudioDevice> devicesAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->getDevicesForRoleAndStrategy(strategyAidl, roleAidl, &devicesAidl)));
devices = VALUE_OR_RETURN_STATUS(
@@ -2172,12 +2172,12 @@
return PERMISSION_DENIED;
}
- media::AudioSourceType audioSourceAidl = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_source_t_AudioSourceType(audioSource));
+ AudioSource audioSourceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_source_t_AudioSource(audioSource));
media::DeviceRole roleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_device_role_t_DeviceRole(role));
- std::vector<media::AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ std::vector<AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
return statusTFromBinderStatus(
aps->setDevicesRoleForCapturePreset(audioSourceAidl, roleAidl, devicesAidl));
}
@@ -2189,12 +2189,12 @@
if (aps == 0) {
return PERMISSION_DENIED;
}
- media::AudioSourceType audioSourceAidl = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_source_t_AudioSourceType(audioSource));
+ AudioSource audioSourceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_source_t_AudioSource(audioSource));
media::DeviceRole roleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_device_role_t_DeviceRole(role));
- std::vector<media::AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ std::vector<AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
return statusTFromBinderStatus(
aps->addDevicesRoleForCapturePreset(audioSourceAidl, roleAidl, devicesAidl));
}
@@ -2205,12 +2205,12 @@
if (aps == 0) {
return PERMISSION_DENIED;
}
- media::AudioSourceType audioSourceAidl = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_source_t_AudioSourceType(audioSource));
+ AudioSource audioSourceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_source_t_AudioSource(audioSource));
media::DeviceRole roleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_device_role_t_DeviceRole(role));
- std::vector<media::AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ std::vector<AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
return statusTFromBinderStatus(
aps->removeDevicesRoleForCapturePreset(audioSourceAidl, roleAidl, devicesAidl));
}
@@ -2221,8 +2221,8 @@
if (aps == 0) {
return PERMISSION_DENIED;
}
- media::AudioSourceType audioSourceAidl = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_source_t_AudioSourceType(audioSource));
+ AudioSource audioSourceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_source_t_AudioSource(audioSource));
media::DeviceRole roleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_device_role_t_DeviceRole(role));
return statusTFromBinderStatus(
aps->clearDevicesRoleForCapturePreset(audioSourceAidl, roleAidl));
@@ -2235,10 +2235,10 @@
if (aps == 0) {
return PERMISSION_DENIED;
}
- media::AudioSourceType audioSourceAidl = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_source_t_AudioSourceType(audioSource));
+ AudioSource audioSourceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_source_t_AudioSource(audioSource));
media::DeviceRole roleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_device_role_t_DeviceRole(role));
- std::vector<media::AudioDevice> devicesAidl;
+ std::vector<AudioDevice> devicesAidl;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->getDevicesForRoleAndCapturePreset(audioSourceAidl, roleAidl, &devicesAidl)));
devices = VALUE_OR_RETURN_STATUS(
@@ -2277,11 +2277,11 @@
std::optional<media::AudioAttributesInternal> attrAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_attributes_t_AudioAttributesInternal(attributes));
- std::optional<media::AudioConfig> configAidl = VALUE_OR_RETURN_STATUS(
+ std::optional<AudioConfig> configAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_t_AudioConfig(configuration, false /*isInput*/));
- std::vector<media::AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ std::vector<AudioDevice> devicesAidl = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
aps->canBeSpatialized(attrAidl, configAidl, devicesAidl, canBeSpatialized)));
return OK;
@@ -2352,6 +2352,31 @@
return af->setVibratorInfos(vibratorInfos);
}
+status_t AudioSystem::getMmapPolicyInfo(
+ AudioMMapPolicyType policyType, std::vector<AudioMMapPolicyInfo> *policyInfos) {
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == nullptr) {
+ return PERMISSION_DENIED;
+ }
+ return af->getMmapPolicyInfos(policyType, policyInfos);
+}
+
+int32_t AudioSystem::getAAudioMixerBurstCount() {
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == nullptr) {
+ return PERMISSION_DENIED;
+ }
+ return af->getAAudioMixerBurstCount();
+}
+
+int32_t AudioSystem::getAAudioHardwareBurstMinUsec() {
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == nullptr) {
+ return PERMISSION_DENIED;
+ }
+ return af->getAAudioHardwareBurstMinUsec();
+}
+
// ---------------------------------------------------------------------------
int AudioSystem::AudioPolicyServiceClient::addAudioPortCallback(
@@ -2463,12 +2488,12 @@
Status AudioSystem::AudioPolicyServiceClient::onRecordingConfigurationUpdate(
int32_t event,
const media::RecordClientInfo& clientInfo,
- const media::AudioConfigBase& clientConfig,
+ const AudioConfigBase& clientConfig,
const std::vector<media::EffectDescriptor>& clientEffects,
- const media::AudioConfigBase& deviceConfig,
+ const AudioConfigBase& deviceConfig,
const std::vector<media::EffectDescriptor>& effects,
int32_t patchHandle,
- media::AudioSourceType source) {
+ AudioSource source) {
record_config_callback cb = NULL;
{
Mutex::Autolock _l(AudioSystem::gLock);
@@ -2494,7 +2519,7 @@
audio_patch_handle_t patchHandleLegacy = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_int32_t_audio_patch_handle_t(patchHandle));
audio_source_t sourceLegacy = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioSourceType_audio_source_t(source));
+ aidl2legacy_AudioSource_audio_source_t(source));
cb(eventLegacy, &clientInfoLegacy, &clientConfigLegacy, clientEffectsLegacy,
&deviceConfigLegacy, effectsLegacy, patchHandleLegacy, sourceLegacy);
}
@@ -2538,7 +2563,7 @@
legacy.riid = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_unique_id_t(aidl.riid));
legacy.uid = VALUE_OR_RETURN(aidl2legacy_int32_t_uid_t(aidl.uid));
legacy.session = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_session_t(aidl.session));
- legacy.source = VALUE_OR_RETURN(aidl2legacy_AudioSourceType_audio_source_t(aidl.source));
+ legacy.source = VALUE_OR_RETURN(aidl2legacy_AudioSource_audio_source_t(aidl.source));
legacy.port_id = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.portId));
legacy.silenced = aidl.silenced;
return legacy;
@@ -2550,7 +2575,7 @@
aidl.riid = VALUE_OR_RETURN(legacy2aidl_audio_unique_id_t_int32_t(legacy.riid));
aidl.uid = VALUE_OR_RETURN(legacy2aidl_uid_t_int32_t(legacy.uid));
aidl.session = VALUE_OR_RETURN(legacy2aidl_audio_session_t_int32_t(legacy.session));
- aidl.source = VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSourceType(legacy.source));
+ aidl.source = VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSource(legacy.source));
aidl.portId = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy.port_id));
aidl.silenced = legacy.silenced;
return aidl;
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 5b7760b..8cf22c4 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -21,6 +21,7 @@
#include <inttypes.h>
#include <math.h>
#include <sys/resource.h>
+#include <thread>
#include <android/media/IAudioPolicyService.h>
#include <android-base/macros.h>
@@ -170,7 +171,7 @@
if (aps == 0) return false;
auto result = [&]() -> ConversionResult<bool> {
- media::AudioConfigBase configAidl = VALUE_OR_RETURN(
+ media::audio::common::AudioConfigBase configAidl = VALUE_OR_RETURN(
legacy2aidl_audio_config_base_t_AudioConfigBase(config, false /*isInput*/));
media::AudioAttributesInternal attributesAidl = VALUE_OR_RETURN(
legacy2aidl_audio_attributes_t_AudioAttributesInternal(attributes));
@@ -947,6 +948,44 @@
mAudioTrack->flush();
}
+bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
+{
+ using namespace std::chrono_literals;
+
+ pause();
+
+ AutoMutex lock(mLock);
+ // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
+ if (isOffloadedOrDirect_l()) return true;
+
+ // Wait for the track state to be anything besides pausing.
+ // This ensures that the volume has ramped down.
+ constexpr auto SLEEP_INTERVAL_MS = 10ms;
+ auto begin = std::chrono::steady_clock::now();
+ while (true) {
+ // wait for state to change
+ const int state = mProxy->getState();
+
+ mLock.unlock(); // only local variables accessed until lock.
+ auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
+ std::chrono::steady_clock::now() - begin);
+ if (state != CBLK_STATE_PAUSING) {
+ ALOGV("%s: success state:%d after %lld ms", __func__, state, elapsed.count());
+ return true;
+ }
+ std::chrono::milliseconds remaining = timeout - elapsed;
+ if (remaining.count() <= 0) {
+ ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
+ __func__, state, CBLK_STATE_PAUSING, elapsed.count());
+ return false;
+ }
+ // It is conceivable that the track is restored while sleeping;
+ // as this logic is advisory, we allow that.
+ std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
+ mLock.lock();
+ }
+}
+
void AudioTrack::pause()
{
const int64_t beginNs = systemTime();
@@ -1797,7 +1836,7 @@
mAwaitBoost = true;
}
} else {
- ALOGD("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
+ ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
__func__, mPortId, mReqFrameCount, mFrameCount);
}
}
diff --git a/media/libaudioclient/AudioVolumeGroup.cpp b/media/libaudioclient/AudioVolumeGroup.cpp
index 361f7b8..ab95246 100644
--- a/media/libaudioclient/AudioVolumeGroup.cpp
+++ b/media/libaudioclient/AudioVolumeGroup.cpp
@@ -26,11 +26,10 @@
#include <media/AudioAttributes.h>
#include <media/PolicyAidlConversion.h>
-#define RETURN_STATUS_IF_ERROR(x) \
- { auto _tmp = (x); if (_tmp != OK) return _tmp; }
-
namespace android {
+using media::audio::common::AudioStreamType;
+
status_t AudioVolumeGroup::readFromParcel(const Parcel *parcel)
{
media::AudioVolumeGroup aidl;
@@ -55,7 +54,7 @@
legacy.getAudioAttributes(),
legacy2aidl_audio_attributes_t_AudioAttributesInternal));
aidl.streams = VALUE_OR_RETURN(
- convertContainer<std::vector<media::AudioStreamType>>(legacy.getStreamTypes(),
+ convertContainer<std::vector<AudioStreamType>>(legacy.getStreamTypes(),
legacy2aidl_audio_stream_type_t_AudioStreamType));
return aidl;
}
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 6d2ec93..88e7396 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -17,6 +17,7 @@
#define LOG_TAG "IAudioFlinger"
//#define LOG_NDEBUG 0
+
#include <utils/Log.h>
#include <stdint.h>
@@ -30,6 +31,13 @@
using aidl_utils::statusTFromBinderStatus;
using binder::Status;
+using media::audio::common::AudioChannelLayout;
+using media::audio::common::AudioFormatDescription;
+using media::audio::common::AudioMMapPolicyInfo;
+using media::audio::common::AudioMMapPolicyType;
+using media::audio::common::AudioMode;
+using media::audio::common::AudioStreamType;
+using media::audio::common::AudioUuid;
#define MAX_ITEMS_PER_LIST 1024
@@ -40,12 +48,6 @@
std::move(_tmp.value()); \
})
-#define RETURN_STATUS_IF_ERROR(x) \
- { \
- auto _tmp = (x); \
- if (_tmp != OK) return _tmp; \
- }
-
#define RETURN_BINDER_IF_ERROR(x) \
{ \
auto _tmp = (x); \
@@ -252,7 +254,7 @@
audio_format_t AudioFlingerClientAdapter::format(audio_io_handle_t output) const {
auto result = [&]() -> ConversionResult<audio_format_t> {
int32_t outputAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(output));
- media::AudioFormatDescription aidlRet;
+ AudioFormatDescription aidlRet;
RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->format(outputAidl, &aidlRet)));
return aidl2legacy_AudioFormatDescription_audio_format_t(aidlRet);
}();
@@ -319,14 +321,14 @@
status_t AudioFlingerClientAdapter::setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output) {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
return statusTFromBinderStatus(mDelegate->setStreamVolume(streamAidl, value, outputAidl));
}
status_t AudioFlingerClientAdapter::setStreamMute(audio_stream_type_t stream, bool muted) {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
return statusTFromBinderStatus(mDelegate->setStreamMute(streamAidl, muted));
}
@@ -334,7 +336,7 @@
float AudioFlingerClientAdapter::streamVolume(audio_stream_type_t stream,
audio_io_handle_t output) const {
auto result = [&]() -> ConversionResult<float> {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
float aidlRet;
@@ -348,7 +350,7 @@
bool AudioFlingerClientAdapter::streamMute(audio_stream_type_t stream) const {
auto result = [&]() -> ConversionResult<bool> {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
bool aidlRet;
RETURN_IF_ERROR(statusTFromBinderStatus(
@@ -360,7 +362,7 @@
}
status_t AudioFlingerClientAdapter::setMode(audio_mode_t mode) {
- media::AudioMode modeAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_mode_t_AudioMode(mode));
+ AudioMode modeAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_mode_t_AudioMode(mode));
return statusTFromBinderStatus(mDelegate->setMode(modeAidl));
}
@@ -420,9 +422,9 @@
audio_channel_mask_t channelMask) const {
auto result = [&]() -> ConversionResult<size_t> {
int32_t sampleRateAidl = VALUE_OR_RETURN(convertIntegral<int32_t>(sampleRate));
- media::AudioFormatDescription formatAidl = VALUE_OR_RETURN(
+ AudioFormatDescription formatAidl = VALUE_OR_RETURN(
legacy2aidl_audio_format_t_AudioFormatDescription(format));
- media::AudioChannelLayout channelMaskAidl = VALUE_OR_RETURN(
+ AudioChannelLayout channelMaskAidl = VALUE_OR_RETURN(
legacy2aidl_audio_channel_mask_t_AudioChannelLayout(channelMask, true /*isInput*/));
int64_t aidlRet;
RETURN_IF_ERROR(statusTFromBinderStatus(
@@ -479,7 +481,7 @@
}
status_t AudioFlingerClientAdapter::invalidateStream(audio_stream_type_t stream) {
- media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
+ AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
return statusTFromBinderStatus(mDelegate->invalidateStream(streamAidl));
}
@@ -578,9 +580,9 @@
const effect_uuid_t* pTypeUUID,
uint32_t preferredTypeFlag,
effect_descriptor_t* pDescriptor) const {
- media::AudioUuid effectUuidAidl = VALUE_OR_RETURN_STATUS(
+ AudioUuid effectUuidAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_uuid_t_AudioUuid(*pEffectUUID));
- media::AudioUuid typeUuidAidl = VALUE_OR_RETURN_STATUS(
+ AudioUuid typeUuidAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_uuid_t_AudioUuid(*pTypeUUID));
int32_t preferredTypeFlagAidl = VALUE_OR_RETURN_STATUS(
convertReinterpret<int32_t>(preferredTypeFlag));
@@ -775,6 +777,32 @@
return statusTFromBinderStatus(mDelegate->updateSecondaryOutputs(trackSecondaryOutputInfos));
}
+status_t AudioFlingerClientAdapter::getMmapPolicyInfos(
+ AudioMMapPolicyType policyType, std::vector<AudioMMapPolicyInfo> *policyInfos) {
+ return statusTFromBinderStatus(mDelegate->getMmapPolicyInfos(policyType, policyInfos));
+}
+
+int32_t AudioFlingerClientAdapter::getAAudioMixerBurstCount() {
+ auto result = [&]() -> ConversionResult<int32_t> {
+ int32_t aidlRet;
+ RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->getAAudioMixerBurstCount(&aidlRet)));
+ return convertIntegral<int32_t>(aidlRet);
+ }();
+ // Failure is ignored.
+ return result.value_or(0);
+}
+
+int32_t AudioFlingerClientAdapter::getAAudioHardwareBurstMinUsec() {
+ auto result = [&]() -> ConversionResult<int32_t> {
+ int32_t aidlRet;
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getAAudioHardwareBurstMinUsec(&aidlRet)));
+ return convertIntegral<int32_t>(aidlRet);
+ }();
+ // Failure is ignored.
+ return result.value_or(0);
+}
+
////////////////////////////////////////////////////////////////////////////////////////////////////
// AudioFlingerServerAdapter
@@ -820,7 +848,7 @@
}
Status AudioFlingerServerAdapter::format(int32_t output,
- media::AudioFormatDescription* _aidl_return) {
+ AudioFormatDescription* _aidl_return) {
audio_io_handle_t outputLegacy = VALUE_OR_RETURN_BINDER(
aidl2legacy_int32_t_audio_io_handle_t(output));
*_aidl_return = VALUE_OR_RETURN_BINDER(
@@ -870,7 +898,7 @@
return Status::fromStatusT(mDelegate->getMasterBalance(_aidl_return));
}
-Status AudioFlingerServerAdapter::setStreamVolume(media::AudioStreamType stream, float value,
+Status AudioFlingerServerAdapter::setStreamVolume(AudioStreamType stream, float value,
int32_t output) {
audio_stream_type_t streamLegacy = VALUE_OR_RETURN_BINDER(
aidl2legacy_AudioStreamType_audio_stream_type_t(stream));
@@ -879,13 +907,13 @@
return Status::fromStatusT(mDelegate->setStreamVolume(streamLegacy, value, outputLegacy));
}
-Status AudioFlingerServerAdapter::setStreamMute(media::AudioStreamType stream, bool muted) {
+Status AudioFlingerServerAdapter::setStreamMute(AudioStreamType stream, bool muted) {
audio_stream_type_t streamLegacy = VALUE_OR_RETURN_BINDER(
aidl2legacy_AudioStreamType_audio_stream_type_t(stream));
return Status::fromStatusT(mDelegate->setStreamMute(streamLegacy, muted));
}
-Status AudioFlingerServerAdapter::streamVolume(media::AudioStreamType stream, int32_t output,
+Status AudioFlingerServerAdapter::streamVolume(AudioStreamType stream, int32_t output,
float* _aidl_return) {
audio_stream_type_t streamLegacy = VALUE_OR_RETURN_BINDER(
aidl2legacy_AudioStreamType_audio_stream_type_t(stream));
@@ -895,14 +923,14 @@
return Status::ok();
}
-Status AudioFlingerServerAdapter::streamMute(media::AudioStreamType stream, bool* _aidl_return) {
+Status AudioFlingerServerAdapter::streamMute(AudioStreamType stream, bool* _aidl_return) {
audio_stream_type_t streamLegacy = VALUE_OR_RETURN_BINDER(
aidl2legacy_AudioStreamType_audio_stream_type_t(stream));
*_aidl_return = mDelegate->streamMute(streamLegacy);
return Status::ok();
}
-Status AudioFlingerServerAdapter::setMode(media::AudioMode mode) {
+Status AudioFlingerServerAdapter::setMode(AudioMode mode) {
audio_mode_t modeLegacy = VALUE_OR_RETURN_BINDER(aidl2legacy_AudioMode_audio_mode_t(mode));
return Status::fromStatusT(mDelegate->setMode(modeLegacy));
}
@@ -948,8 +976,8 @@
}
Status AudioFlingerServerAdapter::getInputBufferSize(int32_t sampleRate,
- const media::AudioFormatDescription& format,
- const media::AudioChannelLayout& channelMask,
+ const AudioFormatDescription& format,
+ const AudioChannelLayout& channelMask,
int64_t* _aidl_return) {
uint32_t sampleRateLegacy = VALUE_OR_RETURN_BINDER(convertIntegral<uint32_t>(sampleRate));
audio_format_t formatLegacy = VALUE_OR_RETURN_BINDER(
@@ -1006,7 +1034,7 @@
return Status::fromStatusT(mDelegate->closeInput(inputLegacy));
}
-Status AudioFlingerServerAdapter::invalidateStream(media::AudioStreamType stream) {
+Status AudioFlingerServerAdapter::invalidateStream(AudioStreamType stream) {
audio_stream_type_t streamLegacy = VALUE_OR_RETURN_BINDER(
aidl2legacy_AudioStreamType_audio_stream_type_t(stream));
return Status::fromStatusT(mDelegate->invalidateStream(streamLegacy));
@@ -1081,8 +1109,8 @@
return Status::ok();
}
-Status AudioFlingerServerAdapter::getEffectDescriptor(const media::AudioUuid& effectUUID,
- const media::AudioUuid& typeUUID,
+Status AudioFlingerServerAdapter::getEffectDescriptor(const AudioUuid& effectUUID,
+ const AudioUuid& typeUUID,
int32_t preferredTypeFlag,
media::EffectDescriptor* _aidl_return) {
effect_uuid_t effectUuidLegacy = VALUE_OR_RETURN_BINDER(
@@ -1247,4 +1275,21 @@
return Status::fromStatusT(mDelegate->updateSecondaryOutputs(trackSecondaryOutputs));
}
+Status AudioFlingerServerAdapter::getMmapPolicyInfos(
+ AudioMMapPolicyType policyType, std::vector<AudioMMapPolicyInfo> *_aidl_return) {
+ return Status::fromStatusT(mDelegate->getMmapPolicyInfos(policyType, _aidl_return));
+}
+
+Status AudioFlingerServerAdapter::getAAudioMixerBurstCount(int32_t* _aidl_return) {
+ *_aidl_return = VALUE_OR_RETURN_BINDER(
+ convertIntegral<int32_t>(mDelegate->getAAudioMixerBurstCount()));
+ return Status::ok();
+}
+
+Status AudioFlingerServerAdapter::getAAudioHardwareBurstMinUsec(int32_t* _aidl_return) {
+ *_aidl_return = VALUE_OR_RETURN_BINDER(
+ convertIntegral<int32_t>(mDelegate->getAAudioHardwareBurstMinUsec()));
+ return Status::ok();
+}
+
} // namespace android
diff --git a/media/libaudioclient/PolicyAidlConversion.cpp b/media/libaudioclient/PolicyAidlConversion.cpp
index 676bb37..fd94568 100644
--- a/media/libaudioclient/PolicyAidlConversion.cpp
+++ b/media/libaudioclient/PolicyAidlConversion.cpp
@@ -25,6 +25,7 @@
namespace android {
using base::unexpected;
+using media::audio::common::AudioDeviceAddress;
ConversionResult<volume_group_t>
aidl2legacy_int32_t_volume_group_t(int32_t aidl) {
@@ -152,7 +153,7 @@
case media::AudioMixMatchCriterionValue::source:
legacy.mSource = VALUE_OR_RETURN(
- aidl2legacy_AudioSourceType_audio_source_t(UNION_GET(aidl, source).value()));
+ aidl2legacy_AudioSource_audio_source_t(UNION_GET(aidl, source).value()));
*rule |= RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET;
return legacy;
@@ -184,7 +185,7 @@
case RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET:
UNION_SET(aidl, source,
- VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSourceType(legacy.mSource)));
+ VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSource(legacy.mSource)));
break;
case RULE_MATCH_UID:
@@ -238,9 +239,8 @@
aidl.format, false /*isInput*/));
legacy.mRouteFlags = VALUE_OR_RETURN(
aidl2legacy_AudioMixRouteFlag_uint32_t_mask(aidl.routeFlags));
- legacy.mDeviceType = VALUE_OR_RETURN(
- aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.device.type));
- legacy.mDeviceAddress = VALUE_OR_RETURN(aidl2legacy_string_view_String8(aidl.device.address));
+ RETURN_IF_ERROR(aidl2legacy_AudioDevice_audio_device(
+ aidl.device, &legacy.mDeviceType, &legacy.mDeviceAddress));
legacy.mCbFlags = VALUE_OR_RETURN(aidl2legacy_AudioMixCallbackFlag_uint32_t_mask(aidl.cbFlags));
legacy.mAllowPrivilegedMediaPlaybackCapture = aidl.allowPrivilegedMediaPlaybackCapture;
legacy.mVoiceCommunicationCaptureAllowed = aidl.voiceCommunicationCaptureAllowed;
@@ -261,9 +261,9 @@
legacy.mFormat, false /*isInput*/));
aidl.routeFlags = VALUE_OR_RETURN(
legacy2aidl_uint32_t_AudioMixRouteFlag_mask(legacy.mRouteFlags));
- aidl.device.type = VALUE_OR_RETURN(
- legacy2aidl_audio_devices_t_AudioDeviceDescription(legacy.mDeviceType));
- aidl.device.address = VALUE_OR_RETURN(legacy2aidl_String8_string(legacy.mDeviceAddress));
+ aidl.device = VALUE_OR_RETURN(
+ legacy2aidl_audio_device_AudioDevice(
+ legacy.mDeviceType, legacy.mDeviceAddress));
aidl.cbFlags = VALUE_OR_RETURN(legacy2aidl_uint32_t_AudioMixCallbackFlag_mask(legacy.mCbFlags));
aidl.allowPrivilegedMediaPlaybackCapture = legacy.mAllowPrivilegedMediaPlaybackCapture;
aidl.voiceCommunicationCaptureAllowed = legacy.mVoiceCommunicationCaptureAllowed;
diff --git a/media/libaudioclient/aidl/android/media/AudioAttributesEx.aidl b/media/libaudioclient/aidl/android/media/AudioAttributesEx.aidl
index 04a02c7..335866f 100644
--- a/media/libaudioclient/aidl/android/media/AudioAttributesEx.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioAttributesEx.aidl
@@ -17,7 +17,7 @@
package android.media;
import android.media.AudioAttributesInternal;
-import android.media.AudioStreamType;
+import android.media.audio.common.AudioStreamType;
/**
* This is the equivalent of the android::AudioAttributes C++ type.
diff --git a/media/libaudioclient/aidl/android/media/AudioAttributesInternal.aidl b/media/libaudioclient/aidl/android/media/AudioAttributesInternal.aidl
index 699df0a..2e74206 100644
--- a/media/libaudioclient/aidl/android/media/AudioAttributesInternal.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioAttributesInternal.aidl
@@ -16,9 +16,9 @@
package android.media;
-import android.media.AudioContentType;
-import android.media.AudioSourceType;
-import android.media.AudioUsage;
+import android.media.audio.common.AudioContentType;
+import android.media.audio.common.AudioSource;
+import android.media.audio.common.AudioUsage;
/**
* The "Internal" suffix of this type name is to disambiguate it from the
@@ -28,7 +28,7 @@
parcelable AudioAttributesInternal {
AudioContentType contentType;
AudioUsage usage;
- AudioSourceType source;
+ AudioSource source;
// Bitmask, indexed by AudioFlag.
int flags;
@utf8InCpp String tags; /* UTF8 */
diff --git a/media/libaudioclient/aidl/android/media/AudioChannelLayout.aidl b/media/libaudioclient/aidl/android/media/AudioChannelLayout.aidl
deleted file mode 100644
index 3259105..0000000
--- a/media/libaudioclient/aidl/android/media/AudioChannelLayout.aidl
+++ /dev/null
@@ -1,274 +0,0 @@
-/*
- * Copyright (C) 2021 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-/**
- * This structure describes a layout of a multi-channel stream.
- * There are two possible ways for representing a layout:
- *
- * - indexed mask, which tells what channels of an audio frame are used, but
- * doesn't label them in any way, thus a correspondence between channels in
- * the same position of frames originating from different streams must be
- * established externally;
- *
- * - layout mask, which gives a label to each channel, thus allowing to
- * match channels between streams of different layouts.
- *
- * Both representations are agnostic of the direction of audio transfer. Also,
- * by construction, the number of bits set to '1' in the mask indicates the
- * number of channels in the audio frame. A channel mask per se only defines the
- * presence or absence of a channel, not the order. Please see 'INTERLEAVE_*'
- * constants for the platform convention of order.
- *
- * The structure also defines a "voice mask" which is a special case of
- * layout mask, intended for processing voice audio from telecommunication
- * use cases.
- */
-union AudioChannelLayout {
- /**
- * This variant is used for representing the "null" ("none") value
- * for the channel layout. The field value must always be '0'.
- */
- int none = 0;
- /**
- * This variant is used for indicating an "invalid" layout for use by the
- * framework only. HAL implementations must not accept or emit
- * AudioChannelLayout values for this variant. The field value must always
- * be '0'.
- */
- int invalid = 0;
- /**
- * This variant is used for representing indexed masks. The mask indicates
- * what channels are used. For example, the mask that specifies to use only
- * channels 1 and 3 when interacting with a multi-channel device is defined
- * as a combination of the 1st and the 3rd bits and thus is equal to 5. See
- * also the 'INDEX_MASK_*' constants. The 'indexMask' field must have at
- * least one bit set.
- */
- int indexMask;
- /**
- * This variant is used for representing layout masks.
- * It is recommended to use one of 'LAYOUT_*' values. The 'layoutMask' field
- * must have at least one bit set.
- */
- int layoutMask;
- /**
- * This variant is used for processing of voice audio input and output.
- * It is recommended to use one of 'VOICE_*' values. The 'voiceMask' field
- * must have at least one bit set.
- */
- int voiceMask;
-
- /**
- * 'INDEX_MASK_*' constants define how many channels are used.
- * The mask constants below are 'canonical' masks. Each 'INDEX_MASK_N'
- * constant declares that all N channels are used and arranges
- * them starting from the LSB.
- */
- const int INDEX_MASK_1 = (1 << 1) - 1;
- const int INDEX_MASK_2 = (1 << 2) - 1;
- const int INDEX_MASK_3 = (1 << 3) - 1;
- const int INDEX_MASK_4 = (1 << 4) - 1;
- const int INDEX_MASK_5 = (1 << 5) - 1;
- const int INDEX_MASK_6 = (1 << 6) - 1;
- const int INDEX_MASK_7 = (1 << 7) - 1;
- const int INDEX_MASK_8 = (1 << 8) - 1;
- const int INDEX_MASK_9 = (1 << 9) - 1;
- const int INDEX_MASK_10 = (1 << 10) - 1;
- const int INDEX_MASK_11 = (1 << 11) - 1;
- const int INDEX_MASK_12 = (1 << 12) - 1;
- const int INDEX_MASK_13 = (1 << 13) - 1;
- const int INDEX_MASK_14 = (1 << 14) - 1;
- const int INDEX_MASK_15 = (1 << 15) - 1;
- const int INDEX_MASK_16 = (1 << 16) - 1;
- const int INDEX_MASK_17 = (1 << 17) - 1;
- const int INDEX_MASK_18 = (1 << 18) - 1;
- const int INDEX_MASK_19 = (1 << 19) - 1;
- const int INDEX_MASK_20 = (1 << 20) - 1;
- const int INDEX_MASK_21 = (1 << 21) - 1;
- const int INDEX_MASK_22 = (1 << 22) - 1;
- const int INDEX_MASK_23 = (1 << 23) - 1;
- const int INDEX_MASK_24 = (1 << 24) - 1;
-
- /**
- * 'LAYOUT_*' constants define channel layouts recognized by
- * the audio system. The order of the channels in the frame is assumed
- * to be from the LSB to MSB for all the bits set to '1'.
- */
- const int LAYOUT_MONO = CHANNEL_FRONT_LEFT;
- const int LAYOUT_STEREO =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT;
- const int LAYOUT_2POINT1 =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_LOW_FREQUENCY;
- const int LAYOUT_TRI =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_FRONT_CENTER;
- const int LAYOUT_TRI_BACK =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_BACK_CENTER;
- const int LAYOUT_3POINT1 =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_FRONT_CENTER |
- CHANNEL_LOW_FREQUENCY;
- const int LAYOUT_2POINT0POINT2 =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
- const int LAYOUT_2POINT1POINT2 =
- LAYOUT_2POINT0POINT2 | CHANNEL_LOW_FREQUENCY;
- const int LAYOUT_3POINT0POINT2 =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_FRONT_CENTER |
- CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
- const int LAYOUT_3POINT1POINT2 =
- LAYOUT_3POINT0POINT2 | CHANNEL_LOW_FREQUENCY;
- const int LAYOUT_QUAD =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT;
- const int LAYOUT_QUAD_SIDE =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT;
- const int LAYOUT_SURROUND =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_FRONT_CENTER | CHANNEL_BACK_CENTER;
- const int LAYOUT_PENTA = LAYOUT_QUAD | CHANNEL_FRONT_CENTER;
- const int LAYOUT_5POINT1 =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY |
- CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT;
- const int LAYOUT_5POINT1_SIDE =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY |
- CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT;
- const int LAYOUT_5POINT1POINT2 = LAYOUT_5POINT1 |
- CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
- const int LAYOUT_5POINT1POINT4 = LAYOUT_5POINT1 |
- CHANNEL_TOP_FRONT_LEFT | CHANNEL_TOP_FRONT_RIGHT |
- CHANNEL_TOP_BACK_LEFT | CHANNEL_TOP_BACK_RIGHT;
- const int LAYOUT_6POINT1 =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY |
- CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT | CHANNEL_BACK_CENTER;
- const int LAYOUT_7POINT1 = LAYOUT_5POINT1 |
- CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT;
- const int LAYOUT_7POINT1POINT2 = LAYOUT_7POINT1 |
- CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
- const int LAYOUT_7POINT1POINT4 = LAYOUT_7POINT1 |
- CHANNEL_TOP_FRONT_LEFT | CHANNEL_TOP_FRONT_RIGHT |
- CHANNEL_TOP_BACK_LEFT | CHANNEL_TOP_BACK_RIGHT;
- const int LAYOUT_9POINT1POINT4 = LAYOUT_7POINT1POINT4 |
- CHANNEL_FRONT_WIDE_LEFT | CHANNEL_FRONT_WIDE_RIGHT;
- const int LAYOUT_9POINT1POINT6 = LAYOUT_9POINT1POINT4 |
- CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
- const int LAYOUT_13POINT_360RA =
- CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
- CHANNEL_FRONT_CENTER |
- CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT |
- CHANNEL_TOP_FRONT_LEFT | CHANNEL_TOP_FRONT_RIGHT |
- CHANNEL_TOP_FRONT_CENTER |
- CHANNEL_TOP_BACK_LEFT | CHANNEL_TOP_BACK_RIGHT |
- CHANNEL_BOTTOM_FRONT_LEFT | CHANNEL_BOTTOM_FRONT_RIGHT |
- CHANNEL_BOTTOM_FRONT_CENTER;
- const int LAYOUT_22POINT2 = LAYOUT_7POINT1POINT4 |
- CHANNEL_FRONT_LEFT_OF_CENTER | CHANNEL_FRONT_RIGHT_OF_CENTER |
- CHANNEL_BACK_CENTER | CHANNEL_TOP_CENTER |
- CHANNEL_TOP_FRONT_CENTER | CHANNEL_TOP_BACK_CENTER |
- CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT |
- CHANNEL_BOTTOM_FRONT_LEFT | CHANNEL_BOTTOM_FRONT_RIGHT |
- CHANNEL_BOTTOM_FRONT_CENTER |
- CHANNEL_LOW_FREQUENCY_2;
- const int LAYOUT_MONO_HAPTIC_A =
- LAYOUT_MONO | CHANNEL_HAPTIC_A;
- const int LAYOUT_STEREO_HAPTIC_A =
- LAYOUT_STEREO | CHANNEL_HAPTIC_A;
- const int LAYOUT_HAPTIC_AB =
- CHANNEL_HAPTIC_A | CHANNEL_HAPTIC_B;
- const int LAYOUT_MONO_HAPTIC_AB =
- LAYOUT_MONO | LAYOUT_HAPTIC_AB;
- const int LAYOUT_STEREO_HAPTIC_AB =
- LAYOUT_STEREO | LAYOUT_HAPTIC_AB;
- const int LAYOUT_FRONT_BACK =
- CHANNEL_FRONT_CENTER | CHANNEL_BACK_CENTER;
-
- /**
- * Expresses the convention when stereo audio samples are stored interleaved
- * in an array. This should improve readability by allowing code to use
- * symbolic indices instead of hard-coded [0] and [1].
- *
- * For multi-channel beyond stereo, the platform convention is that channels
- * are interleaved in order from least significant channel mask bit to most
- * significant channel mask bit, with unused bits skipped. Any exceptions
- * to this convention will be noted at the appropriate API.
- */
- const int INTERLEAVE_LEFT = 0;
- const int INTERLEAVE_RIGHT = 1;
-
- /**
- * 'CHANNEL_*' constants are used to build 'LAYOUT_*' masks. Each constant
- * must have exactly one bit set. The values do not match
- * 'android.media.AudioFormat.CHANNEL_OUT_*' constants from the SDK
- * for better efficiency in masks processing.
- */
- const int CHANNEL_FRONT_LEFT = 1 << 0;
- const int CHANNEL_FRONT_RIGHT = 1 << 1;
- const int CHANNEL_FRONT_CENTER = 1 << 2;
- const int CHANNEL_LOW_FREQUENCY = 1 << 3;
- const int CHANNEL_BACK_LEFT = 1 << 4;
- const int CHANNEL_BACK_RIGHT = 1 << 5;
- const int CHANNEL_FRONT_LEFT_OF_CENTER = 1 << 6;
- const int CHANNEL_FRONT_RIGHT_OF_CENTER = 1 << 7;
- const int CHANNEL_BACK_CENTER = 1 << 8;
- const int CHANNEL_SIDE_LEFT = 1 << 9;
- const int CHANNEL_SIDE_RIGHT = 1 << 10;
- const int CHANNEL_TOP_CENTER = 1 << 11;
- const int CHANNEL_TOP_FRONT_LEFT = 1 << 12;
- const int CHANNEL_TOP_FRONT_CENTER = 1 << 13;
- const int CHANNEL_TOP_FRONT_RIGHT = 1 << 14;
- const int CHANNEL_TOP_BACK_LEFT = 1 << 15;
- const int CHANNEL_TOP_BACK_CENTER = 1 << 16;
- const int CHANNEL_TOP_BACK_RIGHT = 1 << 17;
- const int CHANNEL_TOP_SIDE_LEFT = 1 << 18;
- const int CHANNEL_TOP_SIDE_RIGHT = 1 << 19;
- const int CHANNEL_BOTTOM_FRONT_LEFT = 1 << 20;
- const int CHANNEL_BOTTOM_FRONT_CENTER = 1 << 21;
- const int CHANNEL_BOTTOM_FRONT_RIGHT = 1 << 22;
- const int CHANNEL_LOW_FREQUENCY_2 = 1 << 23;
- const int CHANNEL_FRONT_WIDE_LEFT = 1 << 24;
- const int CHANNEL_FRONT_WIDE_RIGHT = 1 << 25;
- /**
- * Haptic channels are not part of multichannel standards, however they
- * enhance user experience when playing so they are packed together with the
- * channels of the program. To avoid collision with positional channels the
- * values for haptic channels start at the MSB of an integer (after the sign
- * bit) and move down to LSB.
- */
- const int CHANNEL_HAPTIC_B = 1 << 29;
- const int CHANNEL_HAPTIC_A = 1 << 30;
-
- /**
- * 'VOICE_*' constants define layouts for voice audio. The order of the
- * channels in the frame is assumed to be from the LSB to MSB for all the
- * bits set to '1'.
- */
- const int VOICE_UPLINK_MONO = CHANNEL_VOICE_UPLINK;
- const int VOICE_DNLINK_MONO = CHANNEL_VOICE_DNLINK;
- const int VOICE_CALL_MONO = CHANNEL_VOICE_UPLINK | CHANNEL_VOICE_DNLINK;
-
- /**
- * 'CHANNEL_VOICE_*' constants are used to build 'VOICE_*' masks. Each
- * constant must have exactly one bit set. Use the same values as
- * 'android.media.AudioFormat.CHANNEL_IN_VOICE_*' constants from the SDK.
- */
- const int CHANNEL_VOICE_UPLINK = 0x4000;
- const int CHANNEL_VOICE_DNLINK = 0x8000;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioConfig.aidl b/media/libaudioclient/aidl/android/media/AudioConfig.aidl
deleted file mode 100644
index 6996d42..0000000
--- a/media/libaudioclient/aidl/android/media/AudioConfig.aidl
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioChannelLayout;
-import android.media.AudioFormatDescription;
-import android.media.AudioOffloadInfo;
-
-/**
- * {@hide}
- */
-parcelable AudioConfig {
- int sampleRate;
- AudioChannelLayout channelMask;
- AudioFormatDescription format;
- AudioOffloadInfo offloadInfo;
- long frameCount;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioConfigBase.aidl b/media/libaudioclient/aidl/android/media/AudioConfigBase.aidl
deleted file mode 100644
index e84161b..0000000
--- a/media/libaudioclient/aidl/android/media/AudioConfigBase.aidl
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioChannelLayout;
-import android.media.AudioFormatDescription;
-
-/**
- * {@hide}
- */
-parcelable AudioConfigBase {
- int sampleRate;
- AudioChannelLayout channelMask;
- AudioFormatDescription format;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioContentType.aidl b/media/libaudioclient/aidl/android/media/AudioContentType.aidl
deleted file mode 100644
index f734fba..0000000
--- a/media/libaudioclient/aidl/android/media/AudioContentType.aidl
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-@Backing(type="int")
-enum AudioContentType {
- UNKNOWN = 0,
- SPEECH = 1,
- MUSIC = 2,
- MOVIE = 3,
- SONIFICATION = 4,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioDevice.aidl b/media/libaudioclient/aidl/android/media/AudioDevice.aidl
deleted file mode 100644
index a815874..0000000
--- a/media/libaudioclient/aidl/android/media/AudioDevice.aidl
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioDeviceDescription;
-
-/**
- * {@hide}
- */
-parcelable AudioDevice {
- AudioDeviceDescription type;
- @utf8InCpp String address;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioDeviceDescription.aidl b/media/libaudioclient/aidl/android/media/AudioDeviceDescription.aidl
deleted file mode 100644
index f7548b9..0000000
--- a/media/libaudioclient/aidl/android/media/AudioDeviceDescription.aidl
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- * Copyright (C) 2021 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioDeviceType;
-
-parcelable AudioDeviceDescription {
- /**
- * Type and directionality of the device. For bidirectional audio devices
- * two descriptions need to be created, having the same value for
- * the 'connection' field.
- *
- * See 'AudioDeviceType' for the list of supported values.
- */
- AudioDeviceType type = AudioDeviceType.NONE;
- /**
- * Specifies the type of the connection of the device to the audio system.
- * Usually it's some kind of a communication protocol, e.g. Bluetooth SCO or
- * USB. There is a list of connection types recognized by the framework,
- * defined using 'CONNECTION_' constants. Vendors can add their own
- * connection types with "vx.<vendor>." prefix.
- *
- * When the 'connection' field is left empty and 'type != NONE | DEFAULT',
- * it is assumed that the device is permanently attached to the audio
- * system, e.g. a built-in speaker or microphone.
- *
- * The 'connection' field must be left empty if 'type' is 'NONE' or
- * '{IN|OUT}_DEFAULT'.
- */
- @utf8InCpp String connection;
- /**
- * Analog connection, for example, via 3.5 mm analog jack.
- */
- const @utf8InCpp String CONNECTION_ANALOG = "analog";
- /**
- * Low-End (Analog) Desk Dock.
- */
- const @utf8InCpp String CONNECTION_ANALOG_DOCK = "analog-dock";
- /**
- * Bluetooth A2DP connection.
- */
- const @utf8InCpp String CONNECTION_BT_A2DP = "bt-a2dp";
- /**
- * Bluetooth Low Energy (LE) connection.
- */
- const @utf8InCpp String CONNECTION_BT_LE = "bt-le";
- /**
- * Bluetooth SCO connection.
- */
- const @utf8InCpp String CONNECTION_BT_SCO = "bt-sco";
- /**
- * Bus connection. Mostly used in automotive scenarios.
- */
- const @utf8InCpp String CONNECTION_BUS = "bus";
- /**
- * High-End (Digital) Desk Dock.
- */
- const @utf8InCpp String CONNECTION_DIGITAL_DOCK = "digital-dock";
- /**
- * HDMI connection.
- */
- const @utf8InCpp String CONNECTION_HDMI = "hdmi";
- /**
- * HDMI ARC connection.
- */
- const @utf8InCpp String CONNECTION_HDMI_ARC = "hdmi-arc";
- /**
- * HDMI eARC connection.
- */
- const @utf8InCpp String CONNECTION_HDMI_EARC = "hdmi-earc";
- /**
- * IP v4 connection.
- */
- const @utf8InCpp String CONNECTION_IP_V4 = "ip-v4";
- /**
- * SPDIF connection.
- */
- const @utf8InCpp String CONNECTION_SPDIF = "spdif";
- /**
- * A wireless connection when the actual protocol is unspecified.
- */
- const @utf8InCpp String CONNECTION_WIRELESS = "wireless";
- /**
- * USB connection.
- */
- const @utf8InCpp String CONNECTION_USB = "usb";
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioDeviceType.aidl b/media/libaudioclient/aidl/android/media/AudioDeviceType.aidl
deleted file mode 100644
index 4da9fd6..0000000
--- a/media/libaudioclient/aidl/android/media/AudioDeviceType.aidl
+++ /dev/null
@@ -1,158 +0,0 @@
-/*
- * Copyright (C) 2021 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-/**
- * The type of the audio device. Only used as part of 'AudioDeviceDescription'
- * structure.
- *
- * Types are divided into "input" and "output" categories. Audio devices that
- * have both audio input and output, for example, headsets, are represented by a
- * pair of input and output device types.
- *
- * The 'AudioDeviceType' intentionally binds together directionality and 'kind'
- * of the device to avoid making them fully orthogonal. This is because not all
- * types of devices are bidirectional, for example, speakers can only be used
- * for output and microphones can only be used for input (at least, in the
- * context of the audio framework).
- */
-@Backing(type="int")
-enum AudioDeviceType {
- /**
- * "None" type is a "null" value. All fields of 'AudioDeviceDescription'
- * must have default / empty / null values.
- */
- NONE = 0,
- /**
- * The "default" device is used when the client does not have any
- * preference for a particular device.
- */
- IN_DEFAULT = 1,
- /**
- * A device implementing Android Open Accessory protocol.
- */
- IN_ACCESSORY = 2,
- /**
- * Input from a DSP front-end proxy device.
- */
- IN_AFE_PROXY = 3,
- /**
- * Used when only the connection protocol is known, e.g. a "HDMI Device."
- */
- IN_DEVICE = 4,
- /**
- * A device providing reference input for echo canceller.
- */
- IN_ECHO_REFERENCE = 5,
- /**
- * FM Tuner input.
- */
- IN_FM_TUNER = 6,
- /**
- * A microphone of a headset.
- */
- IN_HEADSET = 7,
- /**
- * Loopback input.
- */
- IN_LOOPBACK = 8,
- /**
- * The main microphone (the frontal mic on mobile devices).
- */
- IN_MICROPHONE = 9,
- /**
- * The secondary microphone (the back mic on mobile devices).
- */
- IN_MICROPHONE_BACK = 10,
- /**
- * Input from a submix of other streams.
- */
- IN_SUBMIX = 11,
- /**
- * Audio received via the telephone line.
- */
- IN_TELEPHONY_RX = 12,
- /**
- * TV Tuner audio input.
- */
- IN_TV_TUNER = 13,
- /**
- * The "default" device is used when the client does not have any
- * preference for a particular device.
- */
- OUT_DEFAULT = 129,
- /**
- * A device implementing Android Open Accessory protocol.
- */
- OUT_ACCESSORY = 130,
- /**
- * Output from a DSP front-end proxy device.
- */
- OUT_AFE_PROXY = 131,
- /**
- * Car audio system.
- */
- OUT_CARKIT = 132,
- /**
- * Used when only the connection protocol is known, e.g. a "HDMI Device."
- */
- OUT_DEVICE = 133,
- /**
- * The echo canceller device.
- */
- OUT_ECHO_CANCELLER = 134,
- /**
- * The FM Tuner device.
- */
- OUT_FM = 135,
- /**
- * Headphones.
- */
- OUT_HEADPHONE = 136,
- /**
- * Headphones of a headset.
- */
- OUT_HEADSET = 137,
- /**
- * Hearing aid.
- */
- OUT_HEARING_AID = 138,
- /**
- * Secondary line level output.
- */
- OUT_LINE_AUX = 139,
- /**
- * The main speaker.
- */
- OUT_SPEAKER = 140,
- /**
- * The speaker of a mobile device in the case when it is close to the ear.
- */
- OUT_SPEAKER_EARPIECE = 141,
- /**
- * The main speaker with overload / overheating protection.
- */
- OUT_SPEAKER_SAFE = 142,
- /**
- * Output into a submix.
- */
- OUT_SUBMIX = 143,
- /**
- * Output into a telephone line.
- */
- OUT_TELEPHONY_TX = 144,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioEncapsulationMetadataType.aidl b/media/libaudioclient/aidl/android/media/AudioEncapsulationMetadataType.aidl
deleted file mode 100644
index b03adfe..0000000
--- a/media/libaudioclient/aidl/android/media/AudioEncapsulationMetadataType.aidl
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-@Backing(type="int")
-enum AudioEncapsulationMetadataType {
- NONE = 0,
- FRAMEWORK_TUNER = 1,
- DVB_AD_DESCRIPTOR = 2,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioEncapsulationMode.aidl b/media/libaudioclient/aidl/android/media/AudioEncapsulationMode.aidl
deleted file mode 100644
index 9e04e82..0000000
--- a/media/libaudioclient/aidl/android/media/AudioEncapsulationMode.aidl
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-@Backing(type="int")
-enum AudioEncapsulationMode {
- NONE = 0,
- ELEMENTARY_STREAM = 1,
- HANDLE = 2,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioFlag.aidl b/media/libaudioclient/aidl/android/media/AudioFlag.aidl
index 58b493b..91361fb 100644
--- a/media/libaudioclient/aidl/android/media/AudioFlag.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioFlag.aidl
@@ -34,4 +34,6 @@
MUTE_HAPTIC = 11,
NO_SYSTEM_CAPTURE = 12,
CAPTURE_PRIVATE = 13,
+ CONTENT_SPATIALIZED = 14,
+ NEVER_SPATIALIZE = 15,
}
diff --git a/media/libaudioclient/aidl/android/media/AudioFormatDescription.aidl b/media/libaudioclient/aidl/android/media/AudioFormatDescription.aidl
deleted file mode 100644
index a656348..0000000
--- a/media/libaudioclient/aidl/android/media/AudioFormatDescription.aidl
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (C) 2021 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioFormatType;
-import android.media.PcmType;
-
-/**
- * An extensible type for specifying audio formats. All formats are largely
- * divided into two classes: PCM and non-PCM (bitstreams). Bitstreams can
- * be encapsulated into PCM streams.
- *
- * The type defined in a way to make each format uniquely identifiable, so
- * that if the framework and the HAL construct a value for the same type
- * (e.g. PCM 16 bit), they will produce identical parcelables which will have
- * identical hashes. This makes possible deduplicating type descriptions
- * by the framework when they are received from different HAL modules without
- * relying on having some centralized registry of enumeration values.
- *
- * {@hide}
- */
-parcelable AudioFormatDescription {
- /**
- * The type of the audio format. See the 'AudioFormatType' for the
- * list of supported values.
- */
- AudioFormatType type = AudioFormatType.DEFAULT;
- /**
- * The type of the PCM stream or the transport stream for PCM
- * encapsulations. See 'PcmType' for the list of supported values.
- */
- PcmType pcm = PcmType.DEFAULT;
- /**
- * Optional encoding specification. Must be left empty when:
- *
- * - 'type == DEFAULT && pcm == DEFAULT' -- that means "default" type;
- * - 'type == PCM' -- that means a regular PCM stream (not an encapsulation
- * of an encoded bitstream).
- *
- * For PCM encapsulations of encoded bitstreams (e.g. an encapsulation
- * according to IEC-61937 standard), the value of the 'pcm' field must
- * be set accordingly, as an example, PCM_INT_16_BIT must be used for
- * IEC-61937. Note that 'type == NON_PCM' in this case.
- *
- * Encoding names mostly follow IANA standards for media types (MIME), and
- * frameworks/av/media/libstagefright/foundation/MediaDefs.cpp with the
- * latter having priority. Since there are still many audio types not found
- * in any of these lists, the following rules are applied:
- *
- * - If there is a direct MIME type for the encoding, the MIME type name
- * is used as is, e.g. "audio/eac3" for the EAC-3 format.
- * - If the encoding is a "subformat" of a MIME-registered format,
- * the latter is augmented with a suffix, e.g. "audio/eac3-joc" for the
- * JOC extension of EAC-3.
- * - If it's a proprietary format, a "vnd." prefix is added, similar to
- * IANA rules, e.g. "audio/vnd.dolby.truehd".
- * - Otherwise, "x-" prefix is added, e.g. "audio/x-iec61937".
- * - All MIME types not found in the IANA formats list have an associated
- * comment.
- *
- * For PCM encapsulations with a known bitstream format, the latter
- * is added to the encapsulation encoding as a suffix, after a "+" char.
- * For example, an IEC61937 encapsulation of AC3 has the following
- * representation:
- * type = NON_PCM,
- * pcm = PcmType.INT_16_BIT,
- * encoding = "audio/x-iec61937+audio/ac3"
- */
- @utf8InCpp String encoding;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioFormatType.aidl b/media/libaudioclient/aidl/android/media/AudioFormatType.aidl
deleted file mode 100644
index 31ed2be..0000000
--- a/media/libaudioclient/aidl/android/media/AudioFormatType.aidl
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (C) 2021 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-/**
- * The type of the audio format. Only used as part of 'AudioFormatDescription'
- * structure.
- */
-@Backing(type="byte")
-enum AudioFormatType {
- /**
- * "Default" type is used when the client does not care about the actual
- * format. All fields of 'AudioFormatDescription' must have default / empty
- * / null values.
- */
- DEFAULT = 0,
- /**
- * When the 'encoding' field of 'AudioFormatDescription' is not empty, it
- * specifies the codec used for bitstream (non-PCM) data. It is also used
- * in the case when the bitstream data is encapsulated into a PCM stream,
- * see the documentation for 'AudioFormatDescription'.
- */
- NON_PCM = DEFAULT,
- /**
- * PCM type. The 'pcm' field of 'AudioFormatDescription' is used to specify
- * the actual sample size and representation.
- */
- PCM = 1,
- /**
- * Value reserved for system use only. HALs must never return this value to
- * the system or accept it from the system.
- */
- SYS_RESERVED_INVALID = -1,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioGain.aidl b/media/libaudioclient/aidl/android/media/AudioGain.aidl
deleted file mode 100644
index 4cfa96e..0000000
--- a/media/libaudioclient/aidl/android/media/AudioGain.aidl
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioChannelLayout;
-
-/**
- * {@hide}
- */
-parcelable AudioGain {
- int index;
- boolean isInput;
- boolean useForVolume;
- /** Bitmask, indexed by AudioGainMode. */
- int mode;
- AudioChannelLayout channelMask;
- int minValue;
- int maxValue;
- int defaultValue;
- int stepValue;
- int minRampMs;
- int maxRampMs;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl b/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl
deleted file mode 100644
index afa3aca..0000000
--- a/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioChannelLayout;
-
-/**
- * {@hide}
- */
-parcelable AudioGainConfig {
- /** Index of the corresponding audio_gain in the audio_port gains[] table. */
- int index;
-
- /** Mode requested for this command. Bitfield indexed by AudioGainMode. */
- int mode;
-
- /**
- * Channels which gain value follows. N/A in joint mode.
- */
- AudioChannelLayout channelMask;
-
- /**
- * Gain values in millibels.
- * For each channel ordered from LSb to MSb in channel mask. The number of values is 1 in joint
- * mode, otherwise equals the number of bits implied by channelMask.
- */
- int[] values;
-
- /** Ramp duration in ms. */
- int rampDurationMs;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioGainMode.aidl b/media/libaudioclient/aidl/android/media/AudioGainSys.aidl
similarity index 79%
rename from media/libaudioclient/aidl/android/media/AudioGainMode.aidl
rename to media/libaudioclient/aidl/android/media/AudioGainSys.aidl
index e1b9f0b..9ec8390 100644
--- a/media/libaudioclient/aidl/android/media/AudioGainMode.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioGainSys.aidl
@@ -13,14 +13,16 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
+
package android.media;
/**
+ * Provides additional runtime information for AudioGain, used by the framework.
+ *
* {@hide}
*/
-@Backing(type="int")
-enum AudioGainMode {
- JOINT = 0,
- CHANNELS = 1,
- RAMP = 2,
+parcelable AudioGainSys {
+ int index;
+ boolean isInput;
+ boolean useForVolume;
}
diff --git a/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl b/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl
index efdf99b..b01f902 100644
--- a/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl
@@ -16,9 +16,9 @@
package android.media;
-import android.media.AudioChannelLayout;
-import android.media.AudioFormatDescription;
import android.media.AudioPatch;
+import android.media.audio.common.AudioChannelLayout;
+import android.media.audio.common.AudioFormatDescription;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/AudioMix.aidl b/media/libaudioclient/aidl/android/media/AudioMix.aidl
index 7473372..88b0450 100644
--- a/media/libaudioclient/aidl/android/media/AudioMix.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioMix.aidl
@@ -16,12 +16,12 @@
package android.media;
-import android.media.AudioConfig;
-import android.media.AudioDevice;
import android.media.AudioMixCallbackFlag;
import android.media.AudioMixMatchCriterion;
import android.media.AudioMixRouteFlag;
import android.media.AudioMixType;
+import android.media.audio.common.AudioConfig;
+import android.media.audio.common.AudioDevice;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/AudioMixLatencyClass.aidl b/media/libaudioclient/aidl/android/media/AudioMixLatencyClass.aidl
deleted file mode 100644
index d70b364..0000000
--- a/media/libaudioclient/aidl/android/media/AudioMixLatencyClass.aidl
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-@Backing(type="int")
-enum AudioMixLatencyClass {
- LOW = 0,
- NORMAL = 1,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioMixMatchCriterionValue.aidl b/media/libaudioclient/aidl/android/media/AudioMixMatchCriterionValue.aidl
index e26a9e1..921a93a 100644
--- a/media/libaudioclient/aidl/android/media/AudioMixMatchCriterionValue.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioMixMatchCriterionValue.aidl
@@ -16,15 +16,15 @@
package android.media;
-import android.media.AudioSourceType;
-import android.media.AudioUsage;
+import android.media.audio.common.AudioSource;
+import android.media.audio.common.AudioUsage;
/**
* {@hide}
*/
union AudioMixMatchCriterionValue {
AudioUsage usage = AudioUsage.UNKNOWN;
- AudioSourceType source;
+ AudioSource source;
/** Interpreted as uid_t. */
int uid;
int userId;
diff --git a/media/libaudioclient/aidl/android/media/AudioMode.aidl b/media/libaudioclient/aidl/android/media/AudioMode.aidl
deleted file mode 100644
index 7067dd3..0000000
--- a/media/libaudioclient/aidl/android/media/AudioMode.aidl
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-@Backing(type="int")
-enum AudioMode {
- INVALID = -2,
- CURRENT = -1,
- NORMAL = 0,
- RINGTONE = 1,
- IN_CALL = 2,
- IN_COMMUNICATION = 3,
- CALL_SCREEN = 4,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioOffloadInfo.aidl b/media/libaudioclient/aidl/android/media/AudioOffloadInfo.aidl
deleted file mode 100644
index f326305..0000000
--- a/media/libaudioclient/aidl/android/media/AudioOffloadInfo.aidl
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioConfigBase;
-import android.media.AudioEncapsulationMode;
-import android.media.AudioStreamType;
-import android.media.AudioUsage;
-
-/**
- * {@hide}
- */
-parcelable AudioOffloadInfo {
- /** Version of the info structure. Interpreted as a uint16_t version constant. */
- int version;
- /** Audio configuration. */
- AudioConfigBase config;
- /** Stream type. */
- AudioStreamType streamType;
- /** Bit rate in bits per second. */
- int bitRate;
- /** Duration in microseconds, -1 if unknown. */
- long durationUs;
- /** true if stream is tied to a video stream. */
- boolean hasVideo;
- /** true if streaming, false if local playback. */
- boolean isStreaming;
- int bitWidth;
- /** Offload fragment size. */
- int offloadBufferSize;
- AudioUsage usage;
- AudioEncapsulationMode encapsulationMode;
- /** Content id from tuner HAL (0 if none). */
- int contentId;
- /** Sync id from tuner HAL (0 if none). */
- int syncId;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioPort.aidl b/media/libaudioclient/aidl/android/media/AudioPort.aidl
index bf0e5b7..ff177c0 100644
--- a/media/libaudioclient/aidl/android/media/AudioPort.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPort.aidl
@@ -16,35 +16,13 @@
package android.media;
-import android.media.AudioGain;
-import android.media.AudioPortConfig;
-import android.media.AudioPortExt;
-import android.media.AudioPortRole;
-import android.media.AudioPortType;
-import android.media.AudioProfile;
-import android.media.ExtraAudioDescriptor;
+import android.media.AudioPortSys;
+import android.media.audio.common.AudioPort;
/**
* {@hide}
*/
parcelable AudioPort {
- /** Port unique ID. Interpreted as audio_port_handle_t. */
- int id;
- /** Sink or source. */
- AudioPortRole role;
- /** Device, mix ... */
- AudioPortType type;
- @utf8InCpp String name;
- /** AudioProfiles supported by this port (format, Rates, Channels). */
- AudioProfile[] profiles;
- /**
- * ExtraAudioDescriptors supported by this port. The format is not unrecognized to the
- * platform. The audio capability is described by a hardware descriptor.
- */
- ExtraAudioDescriptor[] extraAudioDescriptors;
- /** Gain controllers. */
- AudioGain[] gains;
- /** Current audio port configuration. */
- AudioPortConfig activeConfig;
- AudioPortExt ext;
+ AudioPort hal;
+ AudioPortSys sys;
}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl
index be32a69..3a4ca31 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl
@@ -16,43 +16,13 @@
package android.media;
-import android.media.AudioChannelLayout;
-import android.media.AudioGainConfig;
-import android.media.AudioIoFlags;
-import android.media.AudioPortConfigExt;
-import android.media.AudioPortConfigType;
-import android.media.AudioPortRole;
-import android.media.AudioPortType;
-import android.media.AudioFormatDescription;
+import android.media.AudioPortConfigSys;
+import android.media.audio.common.AudioPortConfig;
/**
* {@hide}
*/
parcelable AudioPortConfig {
- /**
- * Port unique ID.
- * Interpreted as audio_port_handle_t.
- */
- int id;
- /** Sink or source. */
- AudioPortRole role;
- /** Device, mix ... */
- AudioPortType type;
- /** Bitmask, indexed by AudioPortConfigType. */
- int configMask;
- /** Sampling rate in Hz. */
- int sampleRate;
- /**
- * Channel mask, if applicable.
- */
- AudioChannelLayout channelMask;
- /**
- * Format, if applicable.
- */
- AudioFormatDescription format;
- /** Gain to apply, if applicable. */
- AudioGainConfig gain;
- /** Framework only: HW_AV_SYNC, DIRECT, ... */
- AudioIoFlags flags;
- AudioPortConfigExt ext;
+ AudioPortConfig hal;
+ AudioPortConfigSys sys;
}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl
deleted file mode 100644
index 31e5330..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioDeviceDescription;
-
-/**
- * {@hide}
- */
-parcelable AudioPortConfigDeviceExt {
- /**
- * Module the device is attached to.
- * Interpreted as audio_module_handle_t.
- */
- int hwModule;
- /**
- * Device type.
- */
- AudioDeviceDescription type;
- /** Device address. "" if N/A. */
- @utf8InCpp String address;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl
deleted file mode 100644
index 5d635b6..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigExt.aidl
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioPortConfigDeviceExt;
-import android.media.AudioPortConfigMixExt;
-import android.media.AudioPortConfigSessionExt;
-
-/**
- * {@hide}
- */
-union AudioPortConfigExt {
- /**
- * This represents an empty union. Value is ignored.
- * TODO(ytai): replace with the canonical representation for an empty union, as soon as it is
- * established.
- */
- boolean unspecified;
- /** Device specific info. */
- AudioPortConfigDeviceExt device;
- /** Mix specific info. */
- AudioPortConfigMixExt mix;
- /** Session specific info. */
- AudioPortConfigSessionExt session;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigMixExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigMixExt.aidl
deleted file mode 100644
index d3226f2..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigMixExt.aidl
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioPortConfigMixExtUseCase;
-
-/**
- * {@hide}
- */
-parcelable AudioPortConfigMixExt {
- /**
- * Module the stream is attached to.
- * Interpreted as audio_module_handle_t.
- */
- int hwModule;
- /**
- * I/O handle of the input/output stream.
- * Interpreted as audio_io_handle_t.
- */
- int handle;
- AudioPortConfigMixExtUseCase usecase;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl
deleted file mode 100644
index c61f044..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigMixExtUseCase.aidl
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioSourceType;
-import android.media.AudioStreamType;
-
-/**
- * {@hide}
- */
-union AudioPortConfigMixExtUseCase {
- /**
- * This to be set if the containing config has the AudioPortRole::NONE role.
- * This represents an empty value (value is ignored).
- * TODO(ytai): replace with the canonical representation for an empty union, as soon as it is
- * established.
- */
- boolean unspecified;
- /** This to be set if the containing config has the AudioPortRole::SOURCE role. */
- AudioStreamType stream;
- /** This to be set if the containing config has the AudioPortRole::SINK role. */
- AudioSourceType source;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigSessionExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigSessionExt.aidl
deleted file mode 100644
index a2cbf62..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigSessionExt.aidl
+++ /dev/null
@@ -1,24 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-/**
- * {@hide}
- */
-parcelable AudioPortConfigSessionExt {
- int session;
-}
diff --git a/media/libaudioclient/aidl/android/media/Int.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigSys.aidl
similarity index 62%
rename from media/libaudioclient/aidl/android/media/Int.aidl
rename to media/libaudioclient/aidl/android/media/AudioPortConfigSys.aidl
index 24f4d62..6a615cd 100644
--- a/media/libaudioclient/aidl/android/media/Int.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigSys.aidl
@@ -16,12 +16,20 @@
package android.media;
+import android.media.AudioIoFlags;
+import android.media.AudioPortExtSys;
+import android.media.AudioPortRole;
+import android.media.AudioPortType;
+
/**
- * This is a simple wrapper around an 'int', putting it in a parcelable, so it can be used as an
- * inout parameter, be made @nullable, etc.
- *
* {@hide}
*/
-parcelable Int {
- int value;
+parcelable AudioPortConfigSys {
+ /** Sink or source. */
+ AudioPortRole role;
+ /** Device, mix ... */
+ AudioPortType type;
+ /** Flags: HW_AV_SYNC, DIRECT, ... Can be left unspecified. */
+ @nullable AudioIoFlags flags;
+ AudioPortExtSys ext;
}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl
deleted file mode 100644
index 6e22b8d..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-@Backing(type="int")
-enum AudioPortConfigType {
- SAMPLE_RATE = 0,
- CHANNEL_MASK = 1,
- FORMAT = 2,
- GAIN = 3,
- FLAGS = 4,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortDeviceExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortDeviceExtSys.aidl
similarity index 85%
rename from media/libaudioclient/aidl/android/media/AudioPortDeviceExt.aidl
rename to media/libaudioclient/aidl/android/media/AudioPortDeviceExtSys.aidl
index b758f23..0f5a9b6 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortDeviceExt.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortDeviceExtSys.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2020 The Android Open Source Project
+ * Copyright (C) 2021 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -16,15 +16,12 @@
package android.media;
-import android.media.AudioDevice;
-
/**
* {@hide}
*/
-parcelable AudioPortDeviceExt {
+parcelable AudioPortDeviceExtSys {
/** Module the device is attached to. Interpreted as audio_module_handle_t. */
int hwModule;
- AudioDevice device;
/** Bitmask, indexed by AudioEncapsulationMode. */
int encapsulationModes;
/** Bitmask, indexed by AudioEncapsulationMetadataType. */
diff --git a/media/libaudioclient/aidl/android/media/AudioPortExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortExt.aidl
deleted file mode 100644
index 453784b..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortExt.aidl
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioPortDeviceExt;
-import android.media.AudioPortMixExt;
-import android.media.AudioPortSessionExt;
-
-/**
- * {@hide}
- */
-union AudioPortExt {
- /**
- * This represents an empty union. Value is ignored.
- * TODO(ytai): replace with the canonical representation for an empty union, as soon as it is
- * established.
- */
- boolean unspecified;
- /** Device specific info. */
- AudioPortDeviceExt device;
- /** Mix specific info. */
- AudioPortMixExt mix;
- /** Session specific info. */
- AudioPortSessionExt session;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioEncapsulationType.aidl b/media/libaudioclient/aidl/android/media/AudioPortExtSys.aidl
similarity index 62%
rename from media/libaudioclient/aidl/android/media/AudioEncapsulationType.aidl
rename to media/libaudioclient/aidl/android/media/AudioPortExtSys.aidl
index b08a604..2cdf4f6 100644
--- a/media/libaudioclient/aidl/android/media/AudioEncapsulationType.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortExtSys.aidl
@@ -16,14 +16,19 @@
package android.media;
+import android.media.AudioPortDeviceExtSys;
+import android.media.AudioPortMixExtSys;
+
/**
- * Audio encapsulation type is used to describe if the audio data should be sent with a particular
- * encapsulation type or not.
- *
* {@hide}
*/
-@Backing(type="int")
-enum AudioEncapsulationType {
- NONE = 0,
- IEC61937 = 1,
-}
\ No newline at end of file
+union AudioPortExtSys {
+ /**
+ * This represents an empty union. Value is ignored.
+ */
+ boolean unspecified;
+ /** System-only parameters when the port is an audio device. */
+ AudioPortDeviceExtSys device;
+ /** System-only parameters when the port is an audio mix. */
+ AudioPortMixExtSys mix;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortMixExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortMixExt.aidl
deleted file mode 100644
index 62cdb8e..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortMixExt.aidl
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioMixLatencyClass;
-
-/**
- * {@hide}
- */
-parcelable AudioPortMixExt {
- /** Module the stream is attached to. Interpreted as audio_module_handle_t. */
- int hwModule;
- /** I/O handle of the input/output stream. Interpreted as audio_io_handle_t. */
- int handle;
- /** Latency class */
- AudioMixLatencyClass latencyClass;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioStandard.aidl b/media/libaudioclient/aidl/android/media/AudioPortMixExtSys.aidl
similarity index 81%
rename from media/libaudioclient/aidl/android/media/AudioStandard.aidl
rename to media/libaudioclient/aidl/android/media/AudioPortMixExtSys.aidl
index e131d0d..5999885 100644
--- a/media/libaudioclient/aidl/android/media/AudioStandard.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortMixExtSys.aidl
@@ -13,15 +13,13 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
+
package android.media;
/**
- * The audio standard that describe audio playback/capture capabilites.
- *
* {@hide}
*/
-@Backing(type="int")
-enum AudioStandard {
- NONE = 0,
- EDID = 1,
+parcelable AudioPortMixExtSys {
+ /** Module the stream is attached to. Interpreted as audio_module_handle_t. */
+ int hwModule;
}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortSessionExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortSessionExt.aidl
deleted file mode 100644
index dbca168..0000000
--- a/media/libaudioclient/aidl/android/media/AudioPortSessionExt.aidl
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-/**
- * {@hide}
- */
-parcelable AudioPortSessionExt {
- /** Audio session. Interpreted as audio_session_t. */
- int session;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortSys.aidl b/media/libaudioclient/aidl/android/media/AudioPortSys.aidl
new file mode 100644
index 0000000..27c0fe5
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPortSys.aidl
@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+import android.media.AudioGainSys;
+import android.media.AudioPortConfigSys;
+import android.media.AudioPortExtSys;
+import android.media.AudioPortRole;
+import android.media.AudioPortType;
+import android.media.AudioProfileSys;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPortSys {
+ /** Sink or source. */
+ AudioPortRole role;
+ /** Device, mix ... */
+ AudioPortType type;
+ /** System-only parameters for each AudioProfile from 'port.profiles'. */
+ AudioProfileSys[] profiles;
+ /** System-only parameters for each AudioGain from 'port.gains'. */
+ AudioGainSys[] gains;
+ /** System-only parameters for 'port.activeConfig'. */
+ AudioPortConfigSys activeConfig;
+ /** System-only extra parameters for 'port.ext'. */
+ AudioPortExtSys ext;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioProfile.aidl b/media/libaudioclient/aidl/android/media/AudioProfile.aidl
deleted file mode 100644
index 9fb8d49..0000000
--- a/media/libaudioclient/aidl/android/media/AudioProfile.aidl
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioChannelLayout;
-import android.media.AudioEncapsulationType;
-import android.media.AudioFormatDescription;
-
-/**
- * {@hide}
- */
-parcelable AudioProfile {
- @utf8InCpp String name;
- /** The format for an audio profile should only be set when initialized. */
- AudioFormatDescription format;
- AudioChannelLayout[] channelMasks;
- int[] samplingRates;
- boolean isDynamicFormat;
- boolean isDynamicChannels;
- boolean isDynamicRate;
- AudioEncapsulationType encapsulationType;
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioGainMode.aidl b/media/libaudioclient/aidl/android/media/AudioProfileSys.aidl
similarity index 68%
copy from media/libaudioclient/aidl/android/media/AudioGainMode.aidl
copy to media/libaudioclient/aidl/android/media/AudioProfileSys.aidl
index e1b9f0b..329c9d5 100644
--- a/media/libaudioclient/aidl/android/media/AudioGainMode.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioProfileSys.aidl
@@ -13,14 +13,18 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
+
package android.media;
/**
+ * Provides indication whether the parameters of the AudioProfiles in the
+ * AudioPort are dynamic. Each instance of AudioProfileSys corresponds
+ * to an instance of AudioProfile.
+ *
* {@hide}
*/
-@Backing(type="int")
-enum AudioGainMode {
- JOINT = 0,
- CHANNELS = 1,
- RAMP = 2,
+parcelable AudioProfileSys {
+ boolean isDynamicFormat;
+ boolean isDynamicChannels;
+ boolean isDynamicRate;
}
diff --git a/media/libaudioclient/aidl/android/media/AudioSourceType.aidl b/media/libaudioclient/aidl/android/media/AudioSourceType.aidl
deleted file mode 100644
index 8673b92..0000000
--- a/media/libaudioclient/aidl/android/media/AudioSourceType.aidl
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-@Backing(type="int")
-enum AudioSourceType {
- INVALID = -1,
- DEFAULT = 0,
- MIC = 1,
- VOICE_UPLINK = 2,
- VOICE_DOWNLINK = 3,
- VOICE_CALL = 4,
- CAMCORDER = 5,
- VOICE_RECOGNITION = 6,
- VOICE_COMMUNICATION = 7,
- REMOTE_SUBMIX = 8,
- UNPROCESSED = 9,
- VOICE_PERFORMANCE = 10,
- ECHO_REFERENCE = 1997,
- FM_TUNER = 1998,
- /**
- * A low-priority, preemptible audio source for for background software
- * hotword detection. Same tuning as VOICE_RECOGNITION.
- * Used only internally by the framework.
- */
- HOTWORD = 1999,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioStreamType.aidl b/media/libaudioclient/aidl/android/media/AudioStreamType.aidl
deleted file mode 100644
index d777882..0000000
--- a/media/libaudioclient/aidl/android/media/AudioStreamType.aidl
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-@Backing(type="int")
-enum AudioStreamType {
- DEFAULT = -1,
- VOICE_CALL = 0,
- SYSTEM = 1,
- RING = 2,
- MUSIC = 3,
- ALARM = 4,
- NOTIFICATION = 5,
- BLUETOOTH_SCO = 6,
- ENFORCED_AUDIBLE = 7,
- DTMF = 8,
- TTS = 9,
- ACCESSIBILITY = 10,
- ASSISTANT = 11,
- /** For dynamic policy output mixes. Only used by the audio policy */
- REROUTING = 12,
- /** For audio flinger tracks volume. Only used by the audioflinger */
- PATCH = 13,
- /** stream for corresponding to AUDIO_USAGE_CALL_ASSISTANT */
- CALL_ASSISTANT = 14,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioUsage.aidl b/media/libaudioclient/aidl/android/media/AudioUsage.aidl
deleted file mode 100644
index 66c5c30..0000000
--- a/media/libaudioclient/aidl/android/media/AudioUsage.aidl
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-@Backing(type="int")
-enum AudioUsage {
- UNKNOWN = 0,
- MEDIA = 1,
- VOICE_COMMUNICATION = 2,
- VOICE_COMMUNICATION_SIGNALLING = 3,
- ALARM = 4,
- NOTIFICATION = 5,
- NOTIFICATION_TELEPHONY_RINGTONE = 6,
- NOTIFICATION_COMMUNICATION_REQUEST = 7,
- NOTIFICATION_COMMUNICATION_INSTANT = 8,
- NOTIFICATION_COMMUNICATION_DELAYED = 9,
- NOTIFICATION_EVENT = 10,
- ASSISTANCE_ACCESSIBILITY = 11,
- ASSISTANCE_NAVIGATION_GUIDANCE = 12,
- ASSISTANCE_SONIFICATION = 13,
- GAME = 14,
- VIRTUAL_SOURCE = 15,
- ASSISTANT = 16,
- CALL_ASSISTANT = 17,
- EMERGENCY = 1000,
- SAFETY = 1001,
- VEHICLE_STATUS = 1002,
- ANNOUNCEMENT = 1003,
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioUuid.aidl b/media/libaudioclient/aidl/android/media/AudioUuid.aidl
deleted file mode 100644
index bba9039..0000000
--- a/media/libaudioclient/aidl/android/media/AudioUuid.aidl
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * Copyright (C) 2020 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-package android.media;
-
-/**
- * {@hide}
- */
-parcelable AudioUuid {
- int timeLow;
- int timeMid;
- int timeHiAndVersion;
- int clockSeq;
- byte[] node; // Length = 6
-}
diff --git a/media/libaudioclient/aidl/android/media/AudioVolumeGroup.aidl b/media/libaudioclient/aidl/android/media/AudioVolumeGroup.aidl
index 3a29a08..b95a1d3 100644
--- a/media/libaudioclient/aidl/android/media/AudioVolumeGroup.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioVolumeGroup.aidl
@@ -17,7 +17,7 @@
package android.media;
import android.media.AudioAttributesInternal;
-import android.media.AudioStreamType;
+import android.media.audio.common.AudioStreamType;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl b/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl
index 35a56eb..bcca04a 100644
--- a/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl
@@ -16,10 +16,10 @@
package android.media;
-import android.media.AudioDevice;
+import android.content.AttributionSourceState;
import android.media.EffectDescriptor;
import android.media.IEffectClient;
-import android.content.AttributionSourceState;
+import android.media.audio.common.AudioDevice;
/**
* Input arguments of the createEffect() method.
diff --git a/media/libaudioclient/aidl/android/media/CreateRecordRequest.aidl b/media/libaudioclient/aidl/android/media/CreateRecordRequest.aidl
index 7e3c240..b938a3e 100644
--- a/media/libaudioclient/aidl/android/media/CreateRecordRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateRecordRequest.aidl
@@ -18,7 +18,7 @@
import android.media.AudioAttributesInternal;
import android.media.AudioClient;
-import android.media.AudioConfigBase;
+import android.media.audio.common.AudioConfigBase;
/**
* CreateRecordRequest contains all input arguments sent by AudioRecord to AudioFlinger
diff --git a/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl b/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
index 61195d0..7d159d0 100644
--- a/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
@@ -16,9 +16,9 @@
package android.media;
-import android.media.AudioConfigBase;
import android.media.IAudioRecord;
import android.media.SharedFileRegion;
+import android.media.audio.common.AudioConfigBase;
/**
* CreateRecordResponse contains all output arguments returned by AudioFlinger to AudioRecord
diff --git a/media/libaudioclient/aidl/android/media/CreateTrackRequest.aidl b/media/libaudioclient/aidl/android/media/CreateTrackRequest.aidl
index 014b3ca..212221e 100644
--- a/media/libaudioclient/aidl/android/media/CreateTrackRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateTrackRequest.aidl
@@ -18,9 +18,9 @@
import android.media.AudioAttributesInternal;
import android.media.AudioClient;
-import android.media.AudioConfig;
import android.media.IAudioTrackCallback;
import android.media.SharedFileRegion;
+import android.media.audio.common.AudioConfig;
/**
* CreateTrackInput contains all input arguments sent by AudioTrack to AudioFlinger
diff --git a/media/libaudioclient/aidl/android/media/CreateTrackResponse.aidl b/media/libaudioclient/aidl/android/media/CreateTrackResponse.aidl
index 40473fa..da6f454 100644
--- a/media/libaudioclient/aidl/android/media/CreateTrackResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateTrackResponse.aidl
@@ -16,7 +16,7 @@
package android.media;
-import android.media.AudioStreamType;
+import android.media.audio.common.AudioStreamType;
import android.media.IAudioTrack;
/**
diff --git a/media/libaudioclient/aidl/android/media/EffectDescriptor.aidl b/media/libaudioclient/aidl/android/media/EffectDescriptor.aidl
index 35a3d74..e5b5158 100644
--- a/media/libaudioclient/aidl/android/media/EffectDescriptor.aidl
+++ b/media/libaudioclient/aidl/android/media/EffectDescriptor.aidl
@@ -16,7 +16,7 @@
package android.media;
-import android.media.AudioUuid;
+import android.media.audio.common.AudioUuid;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/ExtraAudioDescriptor.aidl b/media/libaudioclient/aidl/android/media/ExtraAudioDescriptor.aidl
deleted file mode 100644
index ec5b67a..0000000
--- a/media/libaudioclient/aidl/android/media/ExtraAudioDescriptor.aidl
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (C) 2021 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-import android.media.AudioEncapsulationType;
-import android.media.AudioStandard;
-
-/**
- * The audio descriptor that descibes playback/capture capabilities according to
- * a particular standard.
- *
- * {@hide}
- */
-parcelable ExtraAudioDescriptor {
- AudioStandard standard;
- byte[] audioDescriptor;
- AudioEncapsulationType encapsulationType;
-}
diff --git a/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl b/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
index 164fb9d..963877a 100644
--- a/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
@@ -16,7 +16,7 @@
package android.media;
-import android.media.AudioStreamType;
+import android.media.audio.common.AudioStreamType;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
index 16f70c1..c55c66e 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
@@ -16,14 +16,10 @@
package android.media;
-import android.media.AudioChannelLayout;
-import android.media.AudioMode;
import android.media.AudioPatch;
import android.media.AudioPort;
import android.media.AudioPortConfig;
-import android.media.AudioStreamType;
import android.media.AudioUniqueIdUse;
-import android.media.AudioUuid;
import android.media.AudioVibratorInfo;
import android.media.CreateEffectRequest;
import android.media.CreateEffectResponse;
@@ -42,7 +38,13 @@
import android.media.MicrophoneInfoData;
import android.media.RenderPosition;
import android.media.TrackSecondaryOutputInfo;
-import android.media.AudioFormatDescription;
+import android.media.audio.common.AudioChannelLayout;
+import android.media.audio.common.AudioFormatDescription;
+import android.media.audio.common.AudioMMapPolicyInfo;
+import android.media.audio.common.AudioMMapPolicyType;
+import android.media.audio.common.AudioMode;
+import android.media.audio.common.AudioStreamType;
+import android.media.audio.common.AudioUuid;
/**
* {@hide}
@@ -217,4 +219,10 @@
// This usually happens when there is a dynamic policy registered.
void updateSecondaryOutputs(
in TrackSecondaryOutputInfo[] trackSecondaryOutputInfos);
+
+ AudioMMapPolicyInfo[] getMmapPolicyInfos(AudioMMapPolicyType policyType);
+
+ int getAAudioMixerBurstCount();
+
+ int getAAudioHardwareBurstMinUsec();
}
diff --git a/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl b/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
index 7022b9d..c59655d 100644
--- a/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
@@ -16,18 +16,11 @@
package android.media;
-import android.media.AudioFormatDescription;
import android.content.AttributionSourceState;
import android.media.AudioAttributesEx;
import android.media.AudioAttributesInternal;
-import android.media.AudioConfig;
-import android.media.AudioConfigBase;
-import android.media.AudioDevice;
-import android.media.AudioDeviceDescription;
import android.media.AudioMix;
-import android.media.AudioMode;
-import android.media.AudioOffloadInfo;
import android.media.AudioOffloadMode;
import android.media.AudioPatch;
import android.media.AudioPolicyDeviceState;
@@ -38,10 +31,6 @@
import android.media.AudioPortRole;
import android.media.AudioPortType;
import android.media.AudioProductStrategy;
-import android.media.AudioSourceType;
-import android.media.AudioStreamType;
-import android.media.AudioUsage;
-import android.media.AudioUuid;
import android.media.AudioVolumeGroup;
import android.media.DeviceRole;
import android.media.EffectDescriptor;
@@ -51,8 +40,19 @@
import android.media.IAudioPolicyServiceClient;
import android.media.ICaptureStateListener;
import android.media.INativeSpatializerCallback;
-import android.media.Int;
import android.media.SoundTriggerSession;
+import android.media.audio.common.AudioConfig;
+import android.media.audio.common.AudioConfigBase;
+import android.media.audio.common.AudioDevice;
+import android.media.audio.common.AudioDeviceDescription;
+import android.media.audio.common.AudioFormatDescription;
+import android.media.audio.common.AudioMode;
+import android.media.audio.common.AudioOffloadInfo;
+import android.media.audio.common.AudioSource;
+import android.media.audio.common.AudioStreamType;
+import android.media.audio.common.AudioUsage;
+import android.media.audio.common.AudioUuid;
+import android.media.audio.common.Int;
/**
* IAudioPolicyService interface (see AudioPolicyInterface for method descriptions).
@@ -157,7 +157,7 @@
boolean isStreamActiveRemotely(AudioStreamType stream, int inPastMs);
- boolean isSourceActive(AudioSourceType source);
+ boolean isSourceActive(AudioSource source);
/**
* On input, count represents the maximum length of the returned array.
@@ -172,7 +172,7 @@
@utf8InCpp String opPackageName,
in AudioUuid uuid,
int priority,
- AudioSourceType source);
+ AudioSource source);
int /* audio_unique_id_t */ addStreamDefaultEffect(in AudioUuid type,
@utf8InCpp String opPackageName,
@@ -331,22 +331,22 @@
AudioDevice[] getDevicesForRoleAndStrategy(int /* product_strategy_t */ strategy,
DeviceRole role);
- void setDevicesRoleForCapturePreset(AudioSourceType audioSource,
+ void setDevicesRoleForCapturePreset(AudioSource audioSource,
DeviceRole role,
in AudioDevice[] devices);
- void addDevicesRoleForCapturePreset(AudioSourceType audioSource,
+ void addDevicesRoleForCapturePreset(AudioSource audioSource,
DeviceRole role,
in AudioDevice[] devices);
- void removeDevicesRoleForCapturePreset(AudioSourceType audioSource,
+ void removeDevicesRoleForCapturePreset(AudioSource audioSource,
DeviceRole role,
in AudioDevice[] devices);
- void clearDevicesRoleForCapturePreset(AudioSourceType audioSource,
+ void clearDevicesRoleForCapturePreset(AudioSource audioSource,
DeviceRole role);
- AudioDevice[] getDevicesForRoleAndCapturePreset(AudioSourceType audioSource,
+ AudioDevice[] getDevicesForRoleAndCapturePreset(AudioSource audioSource,
DeviceRole role);
boolean registerSoundTriggerCaptureStateListener(ICaptureStateListener listener);
diff --git a/media/libaudioclient/aidl/android/media/IAudioPolicyServiceClient.aidl b/media/libaudioclient/aidl/android/media/IAudioPolicyServiceClient.aidl
index a7782b8..d93a59d 100644
--- a/media/libaudioclient/aidl/android/media/IAudioPolicyServiceClient.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioPolicyServiceClient.aidl
@@ -16,10 +16,10 @@
package android.media;
-import android.media.AudioConfigBase;
-import android.media.AudioSourceType;
import android.media.EffectDescriptor;
import android.media.RecordClientInfo;
+import android.media.audio.common.AudioConfigBase;
+import android.media.audio.common.AudioSource;
/**
* {@hide}
@@ -43,7 +43,7 @@
in AudioConfigBase deviceConfig,
in EffectDescriptor[] effects,
int /* audio_patch_handle_t */ patchHandle,
- AudioSourceType source);
+ AudioSource source);
/** Notifies a change of audio routing */
void onRoutingUpdated();
}
diff --git a/media/libaudioclient/aidl/android/media/INativeSpatializerCallback.aidl b/media/libaudioclient/aidl/android/media/INativeSpatializerCallback.aidl
index f34df05..0e9634c 100644
--- a/media/libaudioclient/aidl/android/media/INativeSpatializerCallback.aidl
+++ b/media/libaudioclient/aidl/android/media/INativeSpatializerCallback.aidl
@@ -17,17 +17,18 @@
package android.media;
import android.media.SpatializationLevel;
+import android.media.SpatializerHeadTrackingMode;
/**
* The INativeSpatializerCallback interface is a callback associated to the
- * ISpatializer interface. The callback is used by the spatializer stage
- * implementation in native audio server to communicate stage changes to the
+ * ISpatializer interface. The callback is used by the spatializer
+ * implementation in native audio server to communicate state changes to the
* client controlling the spatializer with the ISpatializer interface.
* {@hide}
*/
-interface INativeSpatializerCallback {
- /** Called when the spatialization level applied by the vitualizer stage changes
+oneway interface INativeSpatializerCallback {
+ /** Called when the spatialization level applied by the spatializer changes
* (e.g. when the spatializer is enabled or disabled)
*/
- oneway void onLevelChanged(SpatializationLevel level);
+ void onLevelChanged(SpatializationLevel level);
}
diff --git a/media/libaudioclient/aidl/android/media/ISpatializer.aidl b/media/libaudioclient/aidl/android/media/ISpatializer.aidl
index 212d8fe..b871238 100644
--- a/media/libaudioclient/aidl/android/media/ISpatializer.aidl
+++ b/media/libaudioclient/aidl/android/media/ISpatializer.aidl
@@ -16,8 +16,11 @@
package android.media;
-import android.media.HeadTrackingMode;
+import android.media.ISpatializerHeadTrackingCallback;
import android.media.SpatializationLevel;
+import android.media.SpatializationMode;
+import android.media.SpatializerHeadTrackingMode;
+
/**
* The ISpatializer interface is used to control the native audio service implementation
@@ -47,22 +50,89 @@
/** Gets the selected spatialization level (see SpatializationLevel.aidl) */
SpatializationLevel getLevel();
- /** Reports the list of supported head tracking modes (see HeadTrackingMode.aidl). The list can
- * be empty if the spatializer implementation does not support head tracking or if no
- * head tracking device is connected.
+ /** Reports if the spatializer engine supports head tracking or not.
+ * This is a pre condition independent of the fact that a head tracking sensor is
+ * registered or not.
*/
- HeadTrackingMode[] getSupportedHeadTrackingModes();
+ boolean isHeadTrackingSupported();
- /** Selects the desired head tracking mode (see HeadTrackingMode.aidl) */
- void setDesiredHeadTrackingMode(HeadTrackingMode mode);
+ /** Reports the list of supported head tracking modes (see SpatializerHeadTrackingMode.aidl).
+ * The list can be empty if the spatializer implementation does not support head tracking or if
+ * no head tracking sensor is registered (see setHeadSensor() and setScreenSensor()).
+ */
+ SpatializerHeadTrackingMode[] getSupportedHeadTrackingModes();
+
+ /** Selects the desired head tracking mode (see SpatializerHeadTrackingMode.aidl) */
+ void setDesiredHeadTrackingMode(SpatializerHeadTrackingMode mode);
/** Gets the actual head tracking mode. Can be different from the desired mode if conditions to
* enable the desired mode are not met (e.g if the head tracking device was removed)
*/
- HeadTrackingMode getActualHeadTrackingMode();
+ SpatializerHeadTrackingMode getActualHeadTrackingMode();
/** Reset the head tracking algorithm to consider current head pose as neutral */
- void recenterHeadtracker();
- /** Set the screen to stage transform to use by the head tracking algorithm */
+ void recenterHeadTracker();
+
+ /** Set the screen to stage transform to use by the head tracking algorithm
+ * The screen to stage transform is conveyed as a vector of 6 elements,
+ * where the first three are a translation vector and
+ * the last three are a rotation vector.
+ */
void setGlobalTransform(in float[] screenToStage);
+
+ /**
+ * Set the sensor that is to be used for head-tracking.
+ * -1 can be used to disable head-tracking.
+ */
+ void setHeadSensor(int sensorHandle);
+
+ /**
+ * Set the sensor that is to be used for screen-tracking.
+ * -1 can be used to disable screen-tracking.
+ */
+ void setScreenSensor(int sensorHandle);
+
+ /**
+ * Sets the display orientation.
+ * Orientation is expressed in the angle of rotation from the physical "up" side of the screen
+ * to the logical "up" side of the content displayed the screen. Counterclockwise angles, as
+ * viewed while facing the screen are positive.
+ */
+ void setDisplayOrientation(float physicalToLogicalAngle);
+
+ /**
+ * Sets the hinge angle for foldable devices.
+ */
+ void setHingeAngle(float hingeAngle);
+
+ /** Reports the list of supported spatialization modess (see SpatializationMode.aidl).
+ * The list should never be empty if an ISpatializer interface was successfully
+ * retrieved with IAudioPolicyService.getSpatializer().
+ */
+ SpatializationMode[] getSupportedModes();
+
+ /**
+ * Registers a callback to monitor head tracking functions.
+ * Only one callback can be registered on a Spatializer.
+ * The last callback registered wins and passing a nullptr unregisters
+ * last registered callback.
+ */
+ void registerHeadTrackingCallback(@nullable ISpatializerHeadTrackingCallback callback);
+
+ /**
+ * Sets a parameter to the spatializer engine. Used by effect implementor for vendor
+ * specific configuration.
+ */
+ void setParameter(int key, in byte[] value);
+
+ /**
+ * Gets a parameter from the spatializer engine. Used by effect implementor for vendor
+ * specific configuration.
+ */
+ void getParameter(int key, inout byte[] value);
+
+ /**
+ * Gets the io handle of the output stream the spatializer is connected to.
+ */
+ int getOutput();
}
diff --git a/media/libaudioclient/aidl/android/media/ISpatializerHeadTrackingCallback.aidl b/media/libaudioclient/aidl/android/media/ISpatializerHeadTrackingCallback.aidl
new file mode 100644
index 0000000..23d5e13
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/ISpatializerHeadTrackingCallback.aidl
@@ -0,0 +1,39 @@
+/*
+ * Copyright 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+import android.media.SpatializationLevel;
+import android.media.SpatializerHeadTrackingMode;
+
+/**
+ * The ISpatializerHeadTrackingCallback interface is a callback associated to the
+ * Spatializer head tracking function. It can be registered via the ISpatializer
+ * interface to monitor head tracking related states.
+ * {@hide}
+ */
+oneway interface ISpatializerHeadTrackingCallback {
+ /** Called when the head tracking mode has changed
+ */
+ void onHeadTrackingModeChanged(SpatializerHeadTrackingMode mode);
+
+ /** Called when the head to stage pose hase been updated
+ * The head to stage pose is conveyed as a vector of 6 elements,
+ * where the first three are a translation vector and
+ * the last three are a rotation vector.
+ */
+ void onHeadToSoundStagePoseUpdated(in float[] headToStage);
+}
diff --git a/media/libaudioclient/aidl/android/media/OpenInputRequest.aidl b/media/libaudioclient/aidl/android/media/OpenInputRequest.aidl
index 2e55526..75ff8e9 100644
--- a/media/libaudioclient/aidl/android/media/OpenInputRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenInputRequest.aidl
@@ -16,9 +16,9 @@
package android.media;
-import android.media.AudioConfig;
-import android.media.AudioDevice;
-import android.media.AudioSourceType;
+import android.media.audio.common.AudioConfig;
+import android.media.audio.common.AudioDevice;
+import android.media.audio.common.AudioSource;
/**
* {@hide}
@@ -30,7 +30,7 @@
int input;
AudioConfig config;
AudioDevice device;
- AudioSourceType source;
+ AudioSource source;
/** Bitmask, indexed by AudioInputFlag. */
int flags;
}
diff --git a/media/libaudioclient/aidl/android/media/OpenInputResponse.aidl b/media/libaudioclient/aidl/android/media/OpenInputResponse.aidl
index b613ba5..41bc38a 100644
--- a/media/libaudioclient/aidl/android/media/OpenInputResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenInputResponse.aidl
@@ -16,8 +16,8 @@
package android.media;
-import android.media.AudioConfig;
-import android.media.AudioDevice;
+import android.media.audio.common.AudioConfig;
+import android.media.audio.common.AudioDevice;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
index 1541948..90e7ea6 100644
--- a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
@@ -16,9 +16,9 @@
package android.media;
-import android.media.AudioConfig;
-import android.media.AudioConfigBase;
import android.media.AudioPort;
+import android.media.audio.common.AudioConfig;
+import android.media.audio.common.AudioConfigBase;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/OpenOutputResponse.aidl b/media/libaudioclient/aidl/android/media/OpenOutputResponse.aidl
index a051969..451a0bf 100644
--- a/media/libaudioclient/aidl/android/media/OpenOutputResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputResponse.aidl
@@ -16,7 +16,7 @@
package android.media;
-import android.media.AudioConfig;
+import android.media.audio.common.AudioConfig;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/PcmType.aidl b/media/libaudioclient/aidl/android/media/PcmType.aidl
deleted file mode 100644
index c9e327c..0000000
--- a/media/libaudioclient/aidl/android/media/PcmType.aidl
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (C) 2021 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-package android.media;
-
-/**
- * The type of the encoding used for representing PCM samples. Only used as
- * part of 'AudioFormatDescription' structure.
- */
-@Backing(type="byte")
-enum PcmType {
- /**
- * "Default" value used when the type 'AudioFormatDescription' is "default".
- */
- DEFAULT = 0,
- /**
- * Unsigned 8-bit integer.
- */
- UINT_8_BIT = DEFAULT,
- /**
- * Signed 16-bit integer.
- */
- INT_16_BIT = 1,
- /**
- * Signed 32-bit integer.
- */
- INT_32_BIT = 2,
- /**
- * Q8.24 fixed point format.
- */
- FIXED_Q_8_24 = 3,
- /**
- * IEEE 754 32-bit floating point format.
- */
- FLOAT_32_BIT = 4,
- /**
- * Signed 24-bit integer.
- */
- INT_24_BIT = 5,
-}
diff --git a/media/libaudioclient/aidl/android/media/RecordClientInfo.aidl b/media/libaudioclient/aidl/android/media/RecordClientInfo.aidl
index 3280460..7dad58d 100644
--- a/media/libaudioclient/aidl/android/media/RecordClientInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/RecordClientInfo.aidl
@@ -16,7 +16,7 @@
package android.media;
-import android.media.AudioSourceType;
+import android.media.audio.common.AudioSource;
/**
* {@hide}
@@ -28,7 +28,7 @@
int uid;
/** Interpreted as audio_session_t. */
int session;
- AudioSourceType source;
+ AudioSource source;
/** Interpreted as audio_port_handle_t. */
int portId;
boolean silenced;
diff --git a/media/libaudioclient/aidl/android/media/SoundTriggerSession.aidl b/media/libaudioclient/aidl/android/media/SoundTriggerSession.aidl
index b9708b2..4b540a9 100644
--- a/media/libaudioclient/aidl/android/media/SoundTriggerSession.aidl
+++ b/media/libaudioclient/aidl/android/media/SoundTriggerSession.aidl
@@ -16,7 +16,7 @@
package android.media;
-import android.media.AudioDeviceDescription;
+import android.media.audio.common.AudioDeviceDescription;
/**
* {@hide}
diff --git a/media/libaudioclient/aidl/android/media/SpatializationLevel.aidl b/media/libaudioclient/aidl/android/media/SpatializationLevel.aidl
index cef42bb..961c5a1 100644
--- a/media/libaudioclient/aidl/android/media/SpatializationLevel.aidl
+++ b/media/libaudioclient/aidl/android/media/SpatializationLevel.aidl
@@ -17,7 +17,7 @@
package android.media;
/**
- * The spatialization level or mode supported by the spatializer stage effect implementation.
+ * The spatialization level supported by the spatializer stage effect implementation.
* Used by methods of the ISpatializer interface.
* {@hide}
*/
diff --git a/media/libaudioclient/aidl/android/media/HeadTrackingMode.aidl b/media/libaudioclient/aidl/android/media/SpatializationMode.aidl
similarity index 61%
copy from media/libaudioclient/aidl/android/media/HeadTrackingMode.aidl
copy to media/libaudioclient/aidl/android/media/SpatializationMode.aidl
index d6cf410..5d8fd93 100644
--- a/media/libaudioclient/aidl/android/media/HeadTrackingMode.aidl
+++ b/media/libaudioclient/aidl/android/media/SpatializationMode.aidl
@@ -16,20 +16,15 @@
package android.media;
-
/**
- * The head tracking mode supported by the spatializer stage effect implementation.
+ * The spatialization mode supported by the spatializer stage effect implementation.
* Used by methods of the ISpatializer interface.
* {@hide}
*/
@Backing(type="byte")
-enum HeadTrackingMode {
- /** Head tracking is active in a mode not listed below (forward compatibility) */
- OTHER = 0,
- /** Head tracking is disabled */
- DISABLED = 1,
- /** Head tracking is performed relative to the real work environment */
- RELATIVE_WORLD = 2,
- /** Head tracking is performed relative to the device's screen */
- RELATIVE_SCREEN = 3,
+enum SpatializationMode {
+ /** The spatializer supports binaural mode (over headphones type devices). */
+ SPATIALIZATER_BINAURAL = 0,
+ /** The spatializer supports transaural mode (over speaker type devices). */
+ SPATIALIZATER_TRANSAURAL = 1,
}
diff --git a/media/libaudioclient/aidl/android/media/HeadTrackingMode.aidl b/media/libaudioclient/aidl/android/media/SpatializerHeadTrackingMode.aidl
similarity index 90%
rename from media/libaudioclient/aidl/android/media/HeadTrackingMode.aidl
rename to media/libaudioclient/aidl/android/media/SpatializerHeadTrackingMode.aidl
index d6cf410..58e0f61 100644
--- a/media/libaudioclient/aidl/android/media/HeadTrackingMode.aidl
+++ b/media/libaudioclient/aidl/android/media/SpatializerHeadTrackingMode.aidl
@@ -18,12 +18,12 @@
/**
- * The head tracking mode supported by the spatializer stage effect implementation.
+ * The head tracking mode supported by the spatializer effect implementation.
* Used by methods of the ISpatializer interface.
* {@hide}
*/
@Backing(type="byte")
-enum HeadTrackingMode {
+enum SpatializerHeadTrackingMode {
/** Head tracking is active in a mode not listed below (forward compatibility) */
OTHER = 0,
/** Head tracking is disabled */
diff --git a/media/libaudioclient/fuzzer/Android.bp b/media/libaudioclient/fuzzer/Android.bp
index b290aa8..969e3e6 100644
--- a/media/libaudioclient/fuzzer/Android.bp
+++ b/media/libaudioclient/fuzzer/Android.bp
@@ -46,6 +46,7 @@
],
shared_libs: [
"android.hardware.audio.common-util",
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"audioflinger-aidl-cpp",
"audiopolicy-aidl-cpp",
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index 52868d9..80fe1ba 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -604,7 +604,7 @@
request.config = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_t_AudioConfig(config, true /*isInput*/));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(deviceTypeAddr));
- request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSourceType(source));
+ request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSource(source));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
media::OpenInputResponse response{};
diff --git a/media/libaudioclient/include/media/AidlConversion.h b/media/libaudioclient/include/media/AidlConversion.h
index 2cf127c..52ea91b 100644
--- a/media/libaudioclient/include/media/AidlConversion.h
+++ b/media/libaudioclient/include/media/AidlConversion.h
@@ -22,38 +22,43 @@
#include <system/audio.h>
#include <android/media/AudioAttributesInternal.h>
-#include <android/media/AudioChannelLayout.h>
#include <android/media/AudioClient.h>
-#include <android/media/AudioConfig.h>
-#include <android/media/AudioConfigBase.h>
-#include <android/media/AudioDeviceDescription.h>
#include <android/media/AudioDualMonoMode.h>
-#include <android/media/AudioEncapsulationMode.h>
-#include <android/media/AudioEncapsulationMetadataType.h>
-#include <android/media/AudioEncapsulationType.h>
#include <android/media/AudioFlag.h>
-#include <android/media/AudioFormatDescription.h>
-#include <android/media/AudioGain.h>
-#include <android/media/AudioGainMode.h>
#include <android/media/AudioInputFlags.h>
#include <android/media/AudioIoConfigEvent.h>
#include <android/media/AudioIoDescriptor.h>
-#include <android/media/AudioMixLatencyClass.h>
-#include <android/media/AudioMode.h>
#include <android/media/AudioOutputFlags.h>
#include <android/media/AudioPlaybackRate.h>
#include <android/media/AudioPort.h>
-#include <android/media/AudioPortConfigType.h>
-#include <android/media/AudioPortDeviceExt.h>
-#include <android/media/AudioPortExt.h>
-#include <android/media/AudioPortMixExt.h>
-#include <android/media/AudioPortSessionExt.h>
-#include <android/media/AudioProfile.h>
+#include <android/media/AudioPortConfig.h>
+#include <android/media/AudioPortDeviceExtSys.h>
#include <android/media/AudioTimestampInternal.h>
#include <android/media/AudioUniqueIdUse.h>
#include <android/media/EffectDescriptor.h>
-#include <android/media/ExtraAudioDescriptor.h>
#include <android/media/TrackSecondaryOutputInfo.h>
+#include <android/media/audio/common/AudioChannelLayout.h>
+#include <android/media/audio/common/AudioConfig.h>
+#include <android/media/audio/common/AudioConfigBase.h>
+#include <android/media/audio/common/AudioContentType.h>
+#include <android/media/audio/common/AudioDeviceDescription.h>
+#include <android/media/audio/common/AudioEncapsulationMetadataType.h>
+#include <android/media/audio/common/AudioEncapsulationMode.h>
+#include <android/media/audio/common/AudioEncapsulationType.h>
+#include <android/media/audio/common/AudioFormatDescription.h>
+#include <android/media/audio/common/AudioGain.h>
+#include <android/media/audio/common/AudioGainConfig.h>
+#include <android/media/audio/common/AudioGainMode.h>
+#include <android/media/audio/common/AudioMode.h>
+#include <android/media/audio/common/AudioOffloadInfo.h>
+#include <android/media/audio/common/AudioPortExt.h>
+#include <android/media/audio/common/AudioPortMixExt.h>
+#include <android/media/audio/common/AudioProfile.h>
+#include <android/media/audio/common/AudioSource.h>
+#include <android/media/audio/common/AudioStandard.h>
+#include <android/media/audio/common/AudioUsage.h>
+#include <android/media/audio/common/AudioUuid.h>
+#include <android/media/audio/common/ExtraAudioDescriptor.h>
#include <android/media/SharedFileRegion.h>
#include <binder/IMemory.h>
@@ -89,13 +94,6 @@
ConversionResult<audio_hw_sync_t> aidl2legacy_int32_t_audio_hw_sync_t(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_audio_hw_sync_t_int32_t(audio_hw_sync_t legacy);
-// The legacy enum is unnamed. Thus, we use int32_t.
-ConversionResult<int32_t> aidl2legacy_AudioPortConfigType_int32_t(
- media::AudioPortConfigType aidl);
-// The legacy enum is unnamed. Thus, we use int32_t.
-ConversionResult<media::AudioPortConfigType> legacy2aidl_int32_t_AudioPortConfigType(
- int32_t legacy);
-
ConversionResult<unsigned int> aidl2legacy_int32_t_config_mask(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_config_mask_int32_t(unsigned int legacy);
@@ -132,32 +130,48 @@
audio_port_type_t legacy);
ConversionResult<audio_channel_mask_t> aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
- const media::AudioChannelLayout& aidl, bool isInput);
-ConversionResult<media::AudioChannelLayout> legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
- audio_channel_mask_t legacy, bool isInput);
+ const media::audio::common::AudioChannelLayout& aidl, bool isInput);
+ConversionResult<media::audio::common::AudioChannelLayout>
+legacy2aidl_audio_channel_mask_t_AudioChannelLayout(audio_channel_mask_t legacy, bool isInput);
ConversionResult<audio_devices_t> aidl2legacy_AudioDeviceDescription_audio_devices_t(
- const media::AudioDeviceDescription& aidl);
-ConversionResult<media::AudioDeviceDescription> legacy2aidl_audio_devices_t_AudioDeviceDescription(
- audio_devices_t legacy);
+ const media::audio::common::AudioDeviceDescription& aidl);
+ConversionResult<media::audio::common::AudioDeviceDescription>
+legacy2aidl_audio_devices_t_AudioDeviceDescription(audio_devices_t legacy);
+
+status_t aidl2legacy_AudioDevice_audio_device(
+ const media::audio::common::AudioDevice& aidl,
+ audio_devices_t* legacyType, char* legacyAddress);
+status_t aidl2legacy_AudioDevice_audio_device(
+ const media::audio::common::AudioDevice& aidl,
+ audio_devices_t* legacyType, String8* legacyAddress);
+status_t aidl2legacy_AudioDevice_audio_device(
+ const media::audio::common::AudioDevice& aidl,
+ audio_devices_t* legacyType, std::string* legacyAddress);
+ConversionResult<media::audio::common::AudioDevice>
+legacy2aidl_audio_device_AudioDevice(
+ audio_devices_t legacyType, const char* legacyAddress);
+ConversionResult<media::audio::common::AudioDevice>
+legacy2aidl_audio_device_AudioDevice(
+ audio_devices_t legacyType, const String8& legacyAddress);
ConversionResult<audio_format_t> aidl2legacy_AudioFormatDescription_audio_format_t(
- const media::AudioFormatDescription& aidl);
-ConversionResult<media::AudioFormatDescription> legacy2aidl_audio_format_t_AudioFormatDescription(
- audio_format_t legacy);
+ const media::audio::common::AudioFormatDescription& aidl);
+ConversionResult<media::audio::common::AudioFormatDescription>
+legacy2aidl_audio_format_t_AudioFormatDescription(audio_format_t legacy);
ConversionResult<audio_gain_mode_t>
-aidl2legacy_AudioGainMode_audio_gain_mode_t(media::AudioGainMode aidl);
-ConversionResult<media::AudioGainMode>
+aidl2legacy_AudioGainMode_audio_gain_mode_t(media::audio::common::AudioGainMode aidl);
+ConversionResult<media::audio::common::AudioGainMode>
legacy2aidl_audio_gain_mode_t_AudioGainMode(audio_gain_mode_t legacy);
ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t_mask(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t_mask(audio_gain_mode_t legacy);
ConversionResult<audio_gain_config> aidl2legacy_AudioGainConfig_audio_gain_config(
- const media::AudioGainConfig& aidl, media::AudioPortRole role, media::AudioPortType type);
-ConversionResult<media::AudioGainConfig> legacy2aidl_audio_gain_config_AudioGainConfig(
- const audio_gain_config& legacy, audio_port_role_t role, audio_port_type_t type);
+ const media::audio::common::AudioGainConfig& aidl, bool isInput);
+ConversionResult<media::audio::common::AudioGainConfig>
+legacy2aidl_audio_gain_config_AudioGainConfig(const audio_gain_config& legacy, bool isInput);
ConversionResult<audio_input_flags_t> aidl2legacy_AudioInputFlags_audio_input_flags_t(
media::AudioInputFlags aidl);
@@ -185,35 +199,37 @@
const audio_io_flags& legacy, audio_port_role_t role, audio_port_type_t type);
ConversionResult<audio_port_config_device_ext>
-aidl2legacy_AudioPortConfigDeviceExt_audio_port_config_device_ext(
- const media::AudioPortConfigDeviceExt& aidl);
-ConversionResult<media::AudioPortConfigDeviceExt>
-legacy2aidl_audio_port_config_device_ext_AudioPortConfigDeviceExt(
- const audio_port_config_device_ext& legacy);
+aidl2legacy_AudioDevice_audio_port_config_device_ext(
+ const media::audio::common::AudioDevice& aidl,
+ const media::AudioPortDeviceExtSys& aidlDeviceExt);
+status_t legacy2aidl_audio_port_config_device_ext_AudioDevice(
+ const audio_port_config_device_ext& legacy,
+ media::audio::common::AudioDevice* aidl, media::AudioPortDeviceExtSys* aidlDeviceExt);
ConversionResult<audio_stream_type_t> aidl2legacy_AudioStreamType_audio_stream_type_t(
- media::AudioStreamType aidl);
-ConversionResult<media::AudioStreamType> legacy2aidl_audio_stream_type_t_AudioStreamType(
- audio_stream_type_t legacy);
+ media::audio::common::AudioStreamType aidl);
+ConversionResult<media::audio::common::AudioStreamType>
+legacy2aidl_audio_stream_type_t_AudioStreamType(audio_stream_type_t legacy);
-ConversionResult<audio_source_t> aidl2legacy_AudioSourceType_audio_source_t(
- media::AudioSourceType aidl);
-ConversionResult<media::AudioSourceType> legacy2aidl_audio_source_t_AudioSourceType(
+ConversionResult<audio_source_t> aidl2legacy_AudioSource_audio_source_t(
+ media::audio::common::AudioSource aidl);
+ConversionResult<media::audio::common::AudioSource>
+ legacy2aidl_audio_source_t_AudioSource(
audio_source_t legacy);
ConversionResult<audio_session_t> aidl2legacy_int32_t_audio_session_t(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_audio_session_t_int32_t(audio_session_t legacy);
-ConversionResult<audio_port_config_mix_ext> aidl2legacy_AudioPortConfigMixExt(
- const media::AudioPortConfigMixExt& aidl, media::AudioPortRole role);
-ConversionResult<media::AudioPortConfigMixExt> legacy2aidl_AudioPortConfigMixExt(
- const audio_port_config_mix_ext& legacy, audio_port_role_t role);
+ConversionResult<audio_port_config_mix_ext> aidl2legacy_AudioPortMixExt(
+ const media::audio::common::AudioPortMixExt& aidl, media::AudioPortRole role,
+ const media::AudioPortMixExtSys& aidlMixExt);
+status_t legacy2aidl_AudioPortMixExt(
+ const audio_port_config_mix_ext& legacy, audio_port_role_t role,
+ media::audio::common::AudioPortMixExt* aidl, media::AudioPortMixExtSys* aidlMixExt);
ConversionResult<audio_port_config_session_ext>
-aidl2legacy_AudioPortConfigSessionExt_audio_port_config_session_ext(
- const media::AudioPortConfigSessionExt& aidl);
-ConversionResult<media::AudioPortConfigSessionExt>
-legacy2aidl_audio_port_config_session_ext_AudioPortConfigSessionExt(
+aidl2legacy_int32_t_audio_port_config_session_ext(int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_port_config_session_ext_AudioPortConfigSessionExt(
const audio_port_config_session_ext& legacy);
ConversionResult<audio_port_config> aidl2legacy_AudioPortConfig_audio_port_config(
@@ -237,13 +253,14 @@
const AudioClient& legacy);
ConversionResult<audio_content_type_t>
-aidl2legacy_AudioContentType_audio_content_type_t(media::AudioContentType aidl);
-ConversionResult<media::AudioContentType>
+aidl2legacy_AudioContentType_audio_content_type_t(
+ media::audio::common::AudioContentType aidl);
+ConversionResult<media::audio::common::AudioContentType>
legacy2aidl_audio_content_type_t_AudioContentType(audio_content_type_t legacy);
ConversionResult<audio_usage_t>
-aidl2legacy_AudioUsage_audio_usage_t(media::AudioUsage aidl);
-ConversionResult<media::AudioUsage>
+aidl2legacy_AudioUsage_audio_usage_t(media::audio::common::AudioUsage aidl);
+ConversionResult<media::audio::common::AudioUsage>
legacy2aidl_audio_usage_t_AudioUsage(audio_usage_t legacy);
ConversionResult<audio_flags_mask_t>
@@ -262,23 +279,26 @@
legacy2aidl_audio_attributes_t_AudioAttributesInternal(const audio_attributes_t& legacy);
ConversionResult<audio_encapsulation_mode_t>
-aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t(media::AudioEncapsulationMode aidl);
-ConversionResult<media::AudioEncapsulationMode>
+aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t(
+ media::audio::common::AudioEncapsulationMode aidl);
+ConversionResult<media::audio::common::AudioEncapsulationMode>
legacy2aidl_audio_encapsulation_mode_t_AudioEncapsulationMode(audio_encapsulation_mode_t legacy);
ConversionResult<audio_offload_info_t>
-aidl2legacy_AudioOffloadInfo_audio_offload_info_t(const media::AudioOffloadInfo& aidl);
-ConversionResult<media::AudioOffloadInfo>
+aidl2legacy_AudioOffloadInfo_audio_offload_info_t(
+ const media::audio::common::AudioOffloadInfo& aidl);
+ConversionResult<media::audio::common::AudioOffloadInfo>
legacy2aidl_audio_offload_info_t_AudioOffloadInfo(const audio_offload_info_t& legacy);
ConversionResult<audio_config_t>
-aidl2legacy_AudioConfig_audio_config_t(const media::AudioConfig& aidl, bool isInput);
-ConversionResult<media::AudioConfig>
+aidl2legacy_AudioConfig_audio_config_t(const media::audio::common::AudioConfig& aidl, bool isInput);
+ConversionResult<media::audio::common::AudioConfig>
legacy2aidl_audio_config_t_AudioConfig(const audio_config_t& legacy, bool isInput);
ConversionResult<audio_config_base_t>
-aidl2legacy_AudioConfigBase_audio_config_base_t(const media::AudioConfigBase& aidl, bool isInput);
-ConversionResult<media::AudioConfigBase>
+aidl2legacy_AudioConfigBase_audio_config_base_t(
+ const media::audio::common::AudioConfigBase& aidl, bool isInput);
+ConversionResult<media::audio::common::AudioConfigBase>
legacy2aidl_audio_config_base_t_AudioConfigBase(const audio_config_base_t& legacy, bool isInput);
ConversionResult<sp<IMemory>>
@@ -297,8 +317,8 @@
legacy2aidl_AudioTimestamp_AudioTimestampInternal(const AudioTimestamp& legacy);
ConversionResult<audio_uuid_t>
-aidl2legacy_AudioUuid_audio_uuid_t(const media::AudioUuid& aidl);
-ConversionResult<media::AudioUuid>
+aidl2legacy_AudioUuid_audio_uuid_t(const media::audio::common::AudioUuid& aidl);
+ConversionResult<media::audio::common::AudioUuid>
legacy2aidl_audio_uuid_t_AudioUuid(const audio_uuid_t& legacy);
ConversionResult<effect_descriptor_t>
@@ -308,8 +328,8 @@
ConversionResult<audio_encapsulation_metadata_type_t>
aidl2legacy_AudioEncapsulationMetadataType_audio_encapsulation_metadata_type_t(
- media::AudioEncapsulationMetadataType aidl);
-ConversionResult<media::AudioEncapsulationMetadataType>
+ media::audio::common::AudioEncapsulationMetadataType aidl);
+ConversionResult<media::audio::common::AudioEncapsulationMetadataType>
legacy2aidl_audio_encapsulation_metadata_type_t_AudioEncapsulationMetadataType(
audio_encapsulation_metadata_type_t legacy);
@@ -323,37 +343,38 @@
ConversionResult<int32_t>
legacy2aidl_AudioEncapsulationMetadataType_mask(uint32_t legacy);
-ConversionResult<audio_mix_latency_class_t>
-aidl2legacy_AudioMixLatencyClass_audio_mix_latency_class_t(
- media::AudioMixLatencyClass aidl);
-ConversionResult<media::AudioMixLatencyClass>
-legacy2aidl_audio_mix_latency_class_t_AudioMixLatencyClass(
- audio_mix_latency_class_t legacy);
-
ConversionResult<audio_port_device_ext>
-aidl2legacy_AudioPortDeviceExt_audio_port_device_ext(const media::AudioPortDeviceExt& aidl);
-ConversionResult<media::AudioPortDeviceExt>
-legacy2aidl_audio_port_device_ext_AudioPortDeviceExt(const audio_port_device_ext& legacy);
+aidl2legacy_AudioDevice_audio_port_device_ext(
+ const media::audio::common::AudioDevice& aidl,
+ const media::AudioPortDeviceExtSys& aidlDeviceExt);
+status_t legacy2aidl_audio_port_device_ext_AudioDevice(
+ const audio_port_device_ext& legacy,
+ media::audio::common::AudioDevice* aidl,
+ media::AudioPortDeviceExtSys* aidlDeviceExt);
ConversionResult<audio_port_mix_ext>
-aidl2legacy_AudioPortMixExt_audio_port_mix_ext(const media::AudioPortMixExt& aidl);
-ConversionResult<media::AudioPortMixExt>
-legacy2aidl_audio_port_mix_ext_AudioPortMixExt(const audio_port_mix_ext& legacy);
+aidl2legacy_AudioPortMixExt_audio_port_mix_ext(
+ const media::audio::common::AudioPortMixExt& aidl,
+ const media::AudioPortMixExtSys& aidlMixExt);
+status_t legacy2aidl_audio_port_mix_ext_AudioPortMixExt(
+ const audio_port_mix_ext& legacy,
+ media::audio::common::AudioPortMixExt* aidl,
+ media::AudioPortMixExtSys* aidlMixExt);
ConversionResult<audio_port_session_ext>
-aidl2legacy_AudioPortSessionExt_audio_port_session_ext(const media::AudioPortSessionExt& aidl);
-ConversionResult<media::AudioPortSessionExt>
-legacy2aidl_audio_port_session_ext_AudioPortSessionExt(const audio_port_session_ext& legacy);
+aidl2legacy_int32_t_audio_port_session_ext(int32_t aidl);
+ConversionResult<int32_t>
+legacy2aidl_audio_port_session_ext_int32_t(const audio_port_session_ext& legacy);
ConversionResult<audio_profile>
-aidl2legacy_AudioProfile_audio_profile(const media::AudioProfile& aidl, bool isInput);
-ConversionResult<media::AudioProfile>
+aidl2legacy_AudioProfile_audio_profile(
+ const media::audio::common::AudioProfile& aidl, bool isInput);
+ConversionResult<media::audio::common::AudioProfile>
legacy2aidl_audio_profile_AudioProfile(const audio_profile& legacy, bool isInput);
ConversionResult<audio_gain>
-aidl2legacy_AudioGain_audio_gain(const media::AudioGain& aidl);
-// The AIDL structure provides a flag for direction indication while the legacy type doesn't.
-ConversionResult<media::AudioGain>
+aidl2legacy_AudioGain_audio_gain(const media::audio::common::AudioGain& aidl, bool isInput);
+ConversionResult<media::audio::common::AudioGain>
legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy, bool isInput);
ConversionResult<audio_port_v7>
@@ -362,8 +383,8 @@
legacy2aidl_audio_port_v7_AudioPort(const audio_port_v7& legacy);
ConversionResult<audio_mode_t>
-aidl2legacy_AudioMode_audio_mode_t(media::AudioMode aidl);
-ConversionResult<media::AudioMode>
+aidl2legacy_AudioMode_audio_mode_t(media::audio::common::AudioMode aidl);
+ConversionResult<media::audio::common::AudioMode>
legacy2aidl_audio_mode_t_AudioMode(audio_mode_t legacy);
ConversionResult<audio_unique_id_use_t>
@@ -397,21 +418,21 @@
legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(const audio_playback_rate_t& legacy);
ConversionResult<audio_standard_t>
-aidl2legacy_AudioStandard_audio_standard_t(media::AudioStandard aidl);
-ConversionResult<media::AudioStandard>
+aidl2legacy_AudioStandard_audio_standard_t(media::audio::common::AudioStandard aidl);
+ConversionResult<media::audio::common::AudioStandard>
legacy2aidl_audio_standard_t_AudioStandard(audio_standard_t legacy);
ConversionResult<audio_extra_audio_descriptor>
aidl2legacy_ExtraAudioDescriptor_audio_extra_audio_descriptor(
- const media::ExtraAudioDescriptor& aidl);
-ConversionResult<media::ExtraAudioDescriptor>
+ const media::audio::common::ExtraAudioDescriptor& aidl);
+ConversionResult<media::audio::common::ExtraAudioDescriptor>
legacy2aidl_audio_extra_audio_descriptor_ExtraAudioDescriptor(
const audio_extra_audio_descriptor& legacy);
ConversionResult<audio_encapsulation_type_t>
aidl2legacy_AudioEncapsulationType_audio_encapsulation_type_t(
- const media::AudioEncapsulationType& aidl);
-ConversionResult<media::AudioEncapsulationType>
+ const media::audio::common::AudioEncapsulationType& aidl);
+ConversionResult<media::audio::common::AudioEncapsulationType>
legacy2aidl_audio_encapsulation_type_t_AudioEncapsulationType(
const audio_encapsulation_type_t & legacy);
diff --git a/media/libaudioclient/include/media/AidlConversionUtil.h b/media/libaudioclient/include/media/AidlConversionUtil.h
index c1a2be3..dfabd55 100644
--- a/media/libaudioclient/include/media/AidlConversionUtil.h
+++ b/media/libaudioclient/include/media/AidlConversionUtil.h
@@ -41,6 +41,9 @@
#define RETURN_IF_ERROR(result) \
if (status_t _tmp = (result); _tmp != OK) return base::unexpected(_tmp);
+#define RETURN_STATUS_IF_ERROR(result) \
+ if (status_t _tmp = (result); _tmp != OK) return _tmp;
+
#define VALUE_OR_RETURN_STATUS(x) \
({ \
auto _tmp = (x); \
@@ -119,6 +122,47 @@
return output;
}
+/**
+ * A generic template that helps to "zip" two input containers of the same size
+ * into a single vector of converted types. The conversion function must
+ * thus accept two arguments.
+ */
+template<typename OutputContainer, typename InputContainer1,
+ typename InputContainer2, typename Func>
+ConversionResult<OutputContainer>
+convertContainers(const InputContainer1& input1, const InputContainer2& input2,
+ const Func& itemConversion) {
+ auto iter2 = input2.begin();
+ OutputContainer output;
+ auto ins = std::inserter(output, output.begin());
+ for (const auto& item1 : input1) {
+ RETURN_IF_ERROR(iter2 != input2.end() ? OK : BAD_VALUE);
+ *ins = VALUE_OR_RETURN(itemConversion(item1, *iter2++));
+ }
+ return output;
+}
+
+/**
+ * A generic template that helps to "unzip" a per-element conversion into
+ * a pair of elements into a pair of containers. The conversion function
+ * must emit a pair of elements.
+ */
+template<typename OutputContainer1, typename OutputContainer2,
+ typename InputContainer, typename Func>
+ConversionResult<std::pair<OutputContainer1, OutputContainer2>>
+convertContainerSplit(const InputContainer& input, const Func& itemConversion) {
+ OutputContainer1 output1;
+ OutputContainer2 output2;
+ auto ins1 = std::inserter(output1, output1.begin());
+ auto ins2 = std::inserter(output2, output2.begin());
+ for (const auto& item : input) {
+ auto out_pair = VALUE_OR_RETURN(itemConversion(item));
+ *ins1 = out_pair.first;
+ *ins2 = out_pair.second;
+ }
+ return std::make_pair(output1, output2);
+}
+
////////////////////////////////////////////////////////////////////////////////////////////////////
// The code below establishes:
// IntegralTypeOf<T>, which works for either integral types (in which case it evaluates to T), or
diff --git a/media/libaudioclient/include/media/AudioCommonTypes.h b/media/libaudioclient/include/media/AudioCommonTypes.h
index bd341e3..862a0f9 100644
--- a/media/libaudioclient/include/media/AudioCommonTypes.h
+++ b/media/libaudioclient/include/media/AudioCommonTypes.h
@@ -19,9 +19,9 @@
#include <functional>
-#include <android/media/AudioChannelLayout.h>
-#include <android/media/AudioDeviceDescription.h>
-#include <android/media/AudioFormatDescription.h>
+#include <android/media/audio/common/AudioChannelLayout.h>
+#include <android/media/audio/common/AudioDeviceDescription.h>
+#include <android/media/audio/common/AudioFormatDescription.h>
#include <binder/Parcelable.h>
#include <system/audio.h>
#include <system/audio_policy.h>
@@ -42,10 +42,11 @@
// possibility of processing types belonging to different versions of the type,
// e.g. a HAL may be using a previous version of the AIDL interface.
-template<> struct hash<android::media::AudioChannelLayout>
+template<> struct hash<android::media::audio::common::AudioChannelLayout>
{
- std::size_t operator()(const android::media::AudioChannelLayout& acl) const noexcept {
- using Tag = android::media::AudioChannelLayout::Tag;
+ std::size_t operator()(
+ const android::media::audio::common::AudioChannelLayout& acl) const noexcept {
+ using Tag = android::media::audio::common::AudioChannelLayout::Tag;
const size_t seed = std::hash<Tag>{}(acl.getTag());
switch (acl.getTag()) {
case Tag::none:
@@ -63,22 +64,24 @@
}
};
-template<> struct hash<android::media::AudioDeviceDescription>
+template<> struct hash<android::media::audio::common::AudioDeviceDescription>
{
- std::size_t operator()(const android::media::AudioDeviceDescription& add) const noexcept {
+ std::size_t operator()(
+ const android::media::audio::common::AudioDeviceDescription& add) const noexcept {
return hash_combine(
- std::hash<android::media::AudioDeviceType>{}(add.type),
+ std::hash<android::media::audio::common::AudioDeviceType>{}(add.type),
std::hash<std::string>{}(add.connection));
}
};
-template<> struct hash<android::media::AudioFormatDescription>
+template<> struct hash<android::media::audio::common::AudioFormatDescription>
{
- std::size_t operator()(const android::media::AudioFormatDescription& afd) const noexcept {
+ std::size_t operator()(
+ const android::media::audio::common::AudioFormatDescription& afd) const noexcept {
return hash_combine(
- std::hash<android::media::AudioFormatType>{}(afd.type),
+ std::hash<android::media::audio::common::AudioFormatType>{}(afd.type),
hash_combine(
- std::hash<android::media::PcmType>{}(afd.pcm),
+ std::hash<android::media::audio::common::PcmType>{}(afd.pcm),
std::hash<std::string>{}(afd.encoding)));
}
};
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 50ab42a..45ee8bd 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -19,12 +19,17 @@
#include <sys/types.h>
+#include <set>
+#include <vector>
+
+#include <android/content/AttributionSourceState.h>
#include <android/media/AudioVibratorInfo.h>
#include <android/media/BnAudioFlingerClient.h>
#include <android/media/BnAudioPolicyServiceClient.h>
#include <android/media/INativeSpatializerCallback.h>
#include <android/media/ISpatializer.h>
-#include <android/content/AttributionSourceState.h>
+#include <android/media/audio/common/AudioMMapPolicyInfo.h>
+#include <android/media/audio/common/AudioMMapPolicyType.h>
#include <media/AidlConversionUtil.h>
#include <media/AudioContainers.h>
#include <media/AudioDeviceTypeAddr.h>
@@ -33,13 +38,11 @@
#include <media/AudioVolumeGroup.h>
#include <media/AudioIoDescriptor.h>
#include <media/MicrophoneInfo.h>
-#include <set>
#include <system/audio.h>
#include <system/audio_effect.h>
#include <system/audio_policy.h>
#include <utils/Errors.h>
#include <utils/Mutex.h>
-#include <vector>
using android::content::AttributionSourceState;
@@ -613,6 +616,14 @@
static status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos);
+ static status_t getMmapPolicyInfo(
+ media::audio::common::AudioMMapPolicyType policyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos);
+
+ static int32_t getAAudioMixerBurstCount();
+
+ static int32_t getAAudioHardwareBurstMinUsec();
+
private:
class AudioFlingerClient: public IBinder::DeathRecipient, public media::BnAudioFlingerClient
@@ -689,12 +700,12 @@
binder::Status onRecordingConfigurationUpdate(
int32_t event,
const media::RecordClientInfo& clientInfo,
- const media::AudioConfigBase& clientConfig,
+ const media::audio::common::AudioConfigBase& clientConfig,
const std::vector<media::EffectDescriptor>& clientEffects,
- const media::AudioConfigBase& deviceConfig,
+ const media::audio::common::AudioConfigBase& deviceConfig,
const std::vector<media::EffectDescriptor>& effects,
int32_t patchHandle,
- media::AudioSourceType source) override;
+ media::audio::common::AudioSource source) override;
binder::Status onRoutingUpdated();
private:
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index a2cd12f..5ba9f91 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -28,6 +28,7 @@
#include <utils/threads.h>
#include <android/content/AttributionSourceState.h>
+#include <chrono>
#include <string>
#include "android/media/BnAudioTrackCallback.h"
@@ -510,6 +511,14 @@
*/
void pause();
+ /* Pause and wait (with timeout) for the audio track to ramp to silence.
+ *
+ * \param timeout is the time limit to wait before returning.
+ * A negative number is treated as 0.
+ * \return true if the track is ramped to silence, false if the timeout occurred.
+ */
+ bool pauseAndWait(const std::chrono::milliseconds& timeout);
+
/* Set volume for this track, mostly used for games' sound effects
* left and right volumes. Levels must be >= 0.0 and <= 1.0.
* This is the older API. New applications should use setVolume(float) when possible.
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index a2ce145..485648d 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -37,10 +37,12 @@
#include <string>
#include <vector>
+#include <android/content/AttributionSourceState.h>
#include <android/media/AudioVibratorInfo.h>
#include <android/media/BnAudioFlingerService.h>
#include <android/media/BpAudioFlingerService.h>
-#include <android/content/AttributionSourceState.h>
+#include <android/media/audio/common/AudioMMapPolicyInfo.h>
+#include <android/media/audio/common/AudioMMapPolicyType.h>
#include "android/media/CreateEffectRequest.h"
#include "android/media/CreateEffectResponse.h"
#include "android/media/CreateRecordRequest.h"
@@ -348,6 +350,14 @@
virtual status_t updateSecondaryOutputs(
const TrackSecondaryOutputsMap& trackSecondaryOutputs) = 0;
+
+ virtual status_t getMmapPolicyInfos(
+ media::audio::common::AudioMMapPolicyType policyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos) = 0;
+
+ virtual int32_t getAAudioMixerBurstCount() = 0;
+
+ virtual int32_t getAAudioHardwareBurstMinUsec() = 0;
};
/**
@@ -445,6 +455,14 @@
status_t updateSecondaryOutputs(
const TrackSecondaryOutputsMap& trackSecondaryOutputs) override;
+ status_t getMmapPolicyInfos(
+ media::audio::common::AudioMMapPolicyType policyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos) override;
+
+ int32_t getAAudioMixerBurstCount() override;
+
+ int32_t getAAudioHardwareBurstMinUsec() override;
+
private:
const sp<media::IAudioFlingerService> mDelegate;
};
@@ -529,6 +547,9 @@
SET_AUDIO_HAL_PIDS = media::BnAudioFlingerService::TRANSACTION_setAudioHalPids,
SET_VIBRATOR_INFOS = media::BnAudioFlingerService::TRANSACTION_setVibratorInfos,
UPDATE_SECONDARY_OUTPUTS = media::BnAudioFlingerService::TRANSACTION_updateSecondaryOutputs,
+ GET_MMAP_POLICY_INFOS = media::BnAudioFlingerService::TRANSACTION_getMmapPolicyInfos,
+ GET_AAUDIO_MIXER_BURST_COUNT = media::BnAudioFlingerService::TRANSACTION_getAAudioMixerBurstCount,
+ GET_AAUDIO_HARDWARE_BURST_MIN_USEC = media::BnAudioFlingerService::TRANSACTION_getAAudioHardwareBurstMinUsec,
};
/**
@@ -570,7 +591,8 @@
Status createRecord(const media::CreateRecordRequest& request,
media::CreateRecordResponse* _aidl_return) override;
Status sampleRate(int32_t ioHandle, int32_t* _aidl_return) override;
- Status format(int32_t output, media::AudioFormatDescription* _aidl_return) override;
+ Status format(int32_t output,
+ media::audio::common::AudioFormatDescription* _aidl_return) override;
Status frameCount(int32_t ioHandle, int64_t* _aidl_return) override;
Status latency(int32_t output, int32_t* _aidl_return) override;
Status setMasterVolume(float value) override;
@@ -579,12 +601,13 @@
Status masterMute(bool* _aidl_return) override;
Status setMasterBalance(float balance) override;
Status getMasterBalance(float* _aidl_return) override;
- Status setStreamVolume(media::AudioStreamType stream, float value, int32_t output) override;
- Status setStreamMute(media::AudioStreamType stream, bool muted) override;
- Status
- streamVolume(media::AudioStreamType stream, int32_t output, float* _aidl_return) override;
- Status streamMute(media::AudioStreamType stream, bool* _aidl_return) override;
- Status setMode(media::AudioMode mode) override;
+ Status setStreamVolume(media::audio::common::AudioStreamType stream,
+ float value, int32_t output) override;
+ Status setStreamMute(media::audio::common::AudioStreamType stream, bool muted) override;
+ Status streamVolume(media::audio::common::AudioStreamType stream,
+ int32_t output, float* _aidl_return) override;
+ Status streamMute(media::audio::common::AudioStreamType stream, bool* _aidl_return) override;
+ Status setMode(media::audio::common::AudioMode mode) override;
Status setMicMute(bool state) override;
Status getMicMute(bool* _aidl_return) override;
Status setRecordSilenced(int32_t portId, bool silenced) override;
@@ -592,8 +615,9 @@
Status
getParameters(int32_t ioHandle, const std::string& keys, std::string* _aidl_return) override;
Status registerClient(const sp<media::IAudioFlingerClient>& client) override;
- Status getInputBufferSize(int32_t sampleRate, const media::AudioFormatDescription& format,
- const media::AudioChannelLayout& channelMask,
+ Status getInputBufferSize(int32_t sampleRate,
+ const media::audio::common::AudioFormatDescription& format,
+ const media::audio::common::AudioChannelLayout& channelMask,
int64_t* _aidl_return) override;
Status openOutput(const media::OpenOutputRequest& request,
media::OpenOutputResponse* _aidl_return) override;
@@ -604,7 +628,7 @@
Status openInput(const media::OpenInputRequest& request,
media::OpenInputResponse* _aidl_return) override;
Status closeInput(int32_t input) override;
- Status invalidateStream(media::AudioStreamType stream) override;
+ Status invalidateStream(media::audio::common::AudioStreamType stream) override;
Status setVoiceVolume(float volume) override;
Status getRenderPosition(int32_t output, media::RenderPosition* _aidl_return) override;
Status getInputFramesLost(int32_t ioHandle, int32_t* _aidl_return) override;
@@ -613,7 +637,8 @@
Status releaseAudioSessionId(int32_t audioSession, int32_t pid) override;
Status queryNumberEffects(int32_t* _aidl_return) override;
Status queryEffect(int32_t index, media::EffectDescriptor* _aidl_return) override;
- Status getEffectDescriptor(const media::AudioUuid& effectUUID, const media::AudioUuid& typeUUID,
+ Status getEffectDescriptor(const media::audio::common::AudioUuid& effectUUID,
+ const media::audio::common::AudioUuid& typeUUID,
int32_t preferredTypeFlag,
media::EffectDescriptor* _aidl_return) override;
Status createEffect(const media::CreateEffectRequest& request,
@@ -639,6 +664,11 @@
Status setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos) override;
Status updateSecondaryOutputs(
const std::vector<media::TrackSecondaryOutputInfo>& trackSecondaryOutputInfos) override;
+ Status getMmapPolicyInfos(
+ media::audio::common::AudioMMapPolicyType policyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *_aidl_return) override;
+ Status getAAudioMixerBurstCount(int32_t* _aidl_return) override;
+ Status getAAudioHardwareBurstMinUsec(int32_t* _aidl_return) override;
private:
const sp<AudioFlingerServerAdapter::Delegate> mDelegate;
diff --git a/media/libaudioclient/include/media/PolicyAidlConversion.h b/media/libaudioclient/include/media/PolicyAidlConversion.h
index 873f27a..2296fdb 100644
--- a/media/libaudioclient/include/media/PolicyAidlConversion.h
+++ b/media/libaudioclient/include/media/PolicyAidlConversion.h
@@ -23,10 +23,8 @@
#include <android/media/AudioMix.h>
#include <android/media/AudioMixCallbackFlag.h>
-#include <android/media/AudioMixLatencyClass.h>
#include <android/media/AudioMixRouteFlag.h>
#include <android/media/AudioMixType.h>
-#include <android/media/AudioMode.h>
#include <android/media/AudioOffloadMode.h>
#include <android/media/AudioPolicyForceUse.h>
#include <android/media/AudioPolicyForcedConfig.h>
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 2279244..891293e 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -33,6 +33,7 @@
"libutils",
],
static_libs: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"libaudioclient_aidl_conversion",
"libstagefright_foundation",
diff --git a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
index 424b387..f963e68 100644
--- a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
+++ b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
@@ -22,116 +22,123 @@
using namespace android;
using namespace android::aidl_utils;
+using media::audio::common::AudioChannelLayout;
+using media::audio::common::AudioDeviceDescription;
+using media::audio::common::AudioDeviceType;
+using media::audio::common::AudioFormatDescription;
+using media::audio::common::AudioFormatType;
+using media::audio::common::PcmType;
+
namespace {
template<typename T> size_t hash(const T& t) {
return std::hash<T>{}(t);
}
-media::AudioChannelLayout make_ACL_None() {
- return media::AudioChannelLayout{};
+AudioChannelLayout make_ACL_None() {
+ return AudioChannelLayout{};
}
-media::AudioChannelLayout make_ACL_Invalid() {
- return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::invalid>(0);
+AudioChannelLayout make_ACL_Invalid() {
+ return AudioChannelLayout::make<AudioChannelLayout::Tag::invalid>(0);
}
-media::AudioChannelLayout make_ACL_Stereo() {
- return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::layoutMask>(
- media::AudioChannelLayout::LAYOUT_STEREO);
+AudioChannelLayout make_ACL_Stereo() {
+ return AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::LAYOUT_STEREO);
}
-media::AudioChannelLayout make_ACL_ChannelIndex2() {
- return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::indexMask>(
- media::AudioChannelLayout::INDEX_MASK_2);
+AudioChannelLayout make_ACL_ChannelIndex2() {
+ return AudioChannelLayout::make<AudioChannelLayout::Tag::indexMask>(
+ AudioChannelLayout::INDEX_MASK_2);
}
-media::AudioChannelLayout make_ACL_ChannelIndexArbitrary() {
+AudioChannelLayout make_ACL_ChannelIndexArbitrary() {
// Use channels 1 and 3.
- return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::indexMask>(5);
+ return AudioChannelLayout::make<AudioChannelLayout::Tag::indexMask>(5);
}
-media::AudioChannelLayout make_ACL_VoiceCall() {
- return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::voiceMask>(
- media::AudioChannelLayout::VOICE_CALL_MONO);
+AudioChannelLayout make_ACL_VoiceCall() {
+ return AudioChannelLayout::make<AudioChannelLayout::Tag::voiceMask>(
+ AudioChannelLayout::VOICE_CALL_MONO);
}
-media::AudioDeviceDescription make_AudioDeviceDescription(media::AudioDeviceType type,
+AudioDeviceDescription make_AudioDeviceDescription(AudioDeviceType type,
const std::string& connection = "") {
- media::AudioDeviceDescription result;
+ AudioDeviceDescription result;
result.type = type;
result.connection = connection;
return result;
}
-media::AudioDeviceDescription make_ADD_None() {
- return media::AudioDeviceDescription{};
+AudioDeviceDescription make_ADD_None() {
+ return AudioDeviceDescription{};
}
-media::AudioDeviceDescription make_ADD_DefaultIn() {
- return make_AudioDeviceDescription(media::AudioDeviceType::IN_DEFAULT);
+AudioDeviceDescription make_ADD_DefaultIn() {
+ return make_AudioDeviceDescription(AudioDeviceType::IN_DEFAULT);
}
-media::AudioDeviceDescription make_ADD_DefaultOut() {
- return make_AudioDeviceDescription(media::AudioDeviceType::OUT_DEFAULT);
+AudioDeviceDescription make_ADD_DefaultOut() {
+ return make_AudioDeviceDescription(AudioDeviceType::OUT_DEFAULT);
}
-media::AudioDeviceDescription make_ADD_WiredHeadset() {
- return make_AudioDeviceDescription(media::AudioDeviceType::OUT_HEADSET,
- media::AudioDeviceDescription::CONNECTION_ANALOG());
+AudioDeviceDescription make_ADD_WiredHeadset() {
+ return make_AudioDeviceDescription(AudioDeviceType::OUT_HEADSET,
+ AudioDeviceDescription::CONNECTION_ANALOG());
}
-media::AudioDeviceDescription make_ADD_BtScoHeadset() {
- return make_AudioDeviceDescription(media::AudioDeviceType::OUT_HEADSET,
- media::AudioDeviceDescription::CONNECTION_BT_SCO());
+AudioDeviceDescription make_ADD_BtScoHeadset() {
+ return make_AudioDeviceDescription(AudioDeviceType::OUT_HEADSET,
+ AudioDeviceDescription::CONNECTION_BT_SCO());
}
-media::AudioFormatDescription make_AudioFormatDescription(media::AudioFormatType type) {
- media::AudioFormatDescription result;
+AudioFormatDescription make_AudioFormatDescription(AudioFormatType type) {
+ AudioFormatDescription result;
result.type = type;
return result;
}
-media::AudioFormatDescription make_AudioFormatDescription(media::PcmType pcm) {
- auto result = make_AudioFormatDescription(media::AudioFormatType::PCM);
+AudioFormatDescription make_AudioFormatDescription(PcmType pcm) {
+ auto result = make_AudioFormatDescription(AudioFormatType::PCM);
result.pcm = pcm;
return result;
}
-media::AudioFormatDescription make_AudioFormatDescription(const std::string& encoding) {
- media::AudioFormatDescription result;
+AudioFormatDescription make_AudioFormatDescription(const std::string& encoding) {
+ AudioFormatDescription result;
result.encoding = encoding;
return result;
}
-media::AudioFormatDescription make_AudioFormatDescription(media::PcmType transport,
+AudioFormatDescription make_AudioFormatDescription(PcmType transport,
const std::string& encoding) {
auto result = make_AudioFormatDescription(encoding);
result.pcm = transport;
return result;
}
-media::AudioFormatDescription make_AFD_Default() {
- return media::AudioFormatDescription{};
+AudioFormatDescription make_AFD_Default() {
+ return AudioFormatDescription{};
}
-media::AudioFormatDescription make_AFD_Invalid() {
- return make_AudioFormatDescription(media::AudioFormatType::SYS_RESERVED_INVALID);
+AudioFormatDescription make_AFD_Invalid() {
+ return make_AudioFormatDescription(AudioFormatType::SYS_RESERVED_INVALID);
}
-media::AudioFormatDescription make_AFD_Pcm16Bit() {
- return make_AudioFormatDescription(media::PcmType::INT_16_BIT);
+AudioFormatDescription make_AFD_Pcm16Bit() {
+ return make_AudioFormatDescription(PcmType::INT_16_BIT);
}
-media::AudioFormatDescription make_AFD_Bitstream() {
+AudioFormatDescription make_AFD_Bitstream() {
return make_AudioFormatDescription("example");
}
-media::AudioFormatDescription make_AFD_Encap() {
- return make_AudioFormatDescription(media::PcmType::INT_16_BIT, "example.encap");
+AudioFormatDescription make_AFD_Encap() {
+ return make_AudioFormatDescription(PcmType::INT_16_BIT, "example.encap");
}
-media::AudioFormatDescription make_AFD_Encap_with_Enc() {
+AudioFormatDescription make_AFD_Encap_with_Enc() {
auto afd = make_AFD_Encap();
afd.encoding += "+example";
return afd;
@@ -160,24 +167,24 @@
};
TEST_F(HashIdentityTest, AudioChannelLayoutHashIdentity) {
- verifyHashIdentity<media::AudioChannelLayout>({
+ verifyHashIdentity<AudioChannelLayout>({
make_ACL_None, make_ACL_Invalid, make_ACL_Stereo, make_ACL_ChannelIndex2,
make_ACL_ChannelIndexArbitrary, make_ACL_VoiceCall});
}
TEST_F(HashIdentityTest, AudioDeviceDescriptionHashIdentity) {
- verifyHashIdentity<media::AudioDeviceDescription>({
+ verifyHashIdentity<AudioDeviceDescription>({
make_ADD_None, make_ADD_DefaultIn, make_ADD_DefaultOut, make_ADD_WiredHeadset,
make_ADD_BtScoHeadset});
}
TEST_F(HashIdentityTest, AudioFormatDescriptionHashIdentity) {
- verifyHashIdentity<media::AudioFormatDescription>({
+ verifyHashIdentity<AudioFormatDescription>({
make_AFD_Default, make_AFD_Invalid, make_AFD_Pcm16Bit, make_AFD_Bitstream,
make_AFD_Encap, make_AFD_Encap_with_Enc});
}
-using ChannelLayoutParam = std::tuple<media::AudioChannelLayout, bool /*isInput*/>;
+using ChannelLayoutParam = std::tuple<AudioChannelLayout, bool /*isInput*/>;
class AudioChannelLayoutRoundTripTest :
public testing::TestWithParam<ChannelLayoutParam> {};
TEST_P(AudioChannelLayoutRoundTripTest, Aidl2Legacy2Aidl) {
@@ -192,7 +199,7 @@
INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutRoundTrip,
AudioChannelLayoutRoundTripTest,
testing::Combine(
- testing::Values(media::AudioChannelLayout{}, make_ACL_Invalid(), make_ACL_Stereo(),
+ testing::Values(AudioChannelLayout{}, make_ACL_Invalid(), make_ACL_Stereo(),
make_ACL_ChannelIndex2(), make_ACL_ChannelIndexArbitrary()),
testing::Values(false, true)));
INSTANTIATE_TEST_SUITE_P(AudioChannelVoiceRoundTrip,
@@ -201,7 +208,7 @@
testing::Combine(testing::Values(make_ACL_VoiceCall()), testing::Values(true)));
class AudioDeviceDescriptionRoundTripTest :
- public testing::TestWithParam<media::AudioDeviceDescription> {};
+ public testing::TestWithParam<AudioDeviceDescription> {};
TEST_P(AudioDeviceDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
const auto initial = GetParam();
auto conv = aidl2legacy_AudioDeviceDescription_audio_devices_t(initial);
@@ -212,11 +219,11 @@
}
INSTANTIATE_TEST_SUITE_P(AudioDeviceDescriptionRoundTrip,
AudioDeviceDescriptionRoundTripTest,
- testing::Values(media::AudioDeviceDescription{}, make_ADD_DefaultIn(),
+ testing::Values(AudioDeviceDescription{}, make_ADD_DefaultIn(),
make_ADD_DefaultOut(), make_ADD_WiredHeadset(), make_ADD_BtScoHeadset()));
class AudioFormatDescriptionRoundTripTest :
- public testing::TestWithParam<media::AudioFormatDescription> {};
+ public testing::TestWithParam<AudioFormatDescription> {};
TEST_P(AudioFormatDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
const auto initial = GetParam();
auto conv = aidl2legacy_AudioFormatDescription_audio_format_t(initial);
@@ -227,4 +234,4 @@
}
INSTANTIATE_TEST_SUITE_P(AudioFormatDescriptionRoundTrip,
AudioFormatDescriptionRoundTripTest,
- testing::Values(make_AFD_Invalid(), media::AudioFormatDescription{}, make_AFD_Pcm16Bit()));
+ testing::Values(make_AFD_Invalid(), AudioFormatDescription{}, make_AFD_Pcm16Bit()));
diff --git a/media/libaudiofoundation/Android.bp b/media/libaudiofoundation/Android.bp
index 3bef55b..727b86f 100644
--- a/media/libaudiofoundation/Android.bp
+++ b/media/libaudiofoundation/Android.bp
@@ -24,9 +24,11 @@
"libmedia_helper_headers",
],
static_libs: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
],
export_static_lib_headers: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
],
host_supported: true,
@@ -52,6 +54,7 @@
],
shared_libs: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"libaudioclient_aidl_conversion",
"libaudioutils",
@@ -63,6 +66,7 @@
],
export_shared_lib_headers: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"libaudioclient_aidl_conversion",
],
diff --git a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
index 7f54474..26eea87 100644
--- a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
+++ b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
@@ -25,6 +25,9 @@
namespace android {
+using media::audio::common::AudioDevice;
+using media::audio::common::AudioDeviceAddress;
+
namespace {
static const std::string SUPPRESSED = "SUPPRESSED";
@@ -157,18 +160,16 @@
}
ConversionResult<AudioDeviceTypeAddr>
-aidl2legacy_AudioDeviceTypeAddress(const media::AudioDevice& aidl) {
- audio_devices_t type = VALUE_OR_RETURN(
- aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.type));
- return AudioDeviceTypeAddr(type, aidl.address);
+aidl2legacy_AudioDeviceTypeAddress(const AudioDevice& aidl) {
+ audio_devices_t type;
+ std::string address;
+ RETURN_IF_ERROR(aidl2legacy_AudioDevice_audio_device(aidl, &type, &address));
+ return AudioDeviceTypeAddr(type, address);
}
-ConversionResult<media::AudioDevice>
+ConversionResult<AudioDevice>
legacy2aidl_AudioDeviceTypeAddress(const AudioDeviceTypeAddr& legacy) {
- media::AudioDevice aidl;
- aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_AudioDeviceDescription(legacy.mType));
- aidl.address = legacy.getAddress();
- return aidl;
+ return legacy2aidl_audio_device_AudioDevice(legacy.mType, legacy.getAddress());
}
} // namespace android
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
index ea0258a..1a8fbf0 100644
--- a/media/libaudiofoundation/AudioGain.cpp
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -24,22 +24,18 @@
#define ALOGVV(a...) do { } while(0)
#endif
+#include <math.h>
+
#include <algorithm>
#include <android-base/stringprintf.h>
#include <media/AudioGain.h>
#include <utils/Log.h>
-#include <math.h>
-
namespace android {
AudioGain::AudioGain(int index, bool isInput)
-{
- mIndex = index;
- mIsInput = isInput;
- memset(&mGain, 0, sizeof(struct audio_gain));
-}
+ : mIndex(index), mIsInput(isInput) {}
void AudioGain::getDefaultConfig(struct audio_gain_config *config)
{
@@ -123,53 +119,24 @@
mGain.max_ramp_ms == other->mGain.max_ramp_ms;
}
-status_t AudioGain::writeToParcel(android::Parcel *parcel) const {
- media::AudioGain parcelable;
- return writeToParcelable(&parcelable)
- ?: parcelable.writeToParcel(parcel);
+ConversionResult<AudioGain::Aidl> AudioGain::toParcelable() const {
+ media::audio::common::AudioGain aidl = VALUE_OR_RETURN(
+ legacy2aidl_audio_gain_AudioGain(mGain, mIsInput));
+ media::AudioGainSys aidlSys;
+ aidlSys.index = VALUE_OR_RETURN(convertIntegral<int32_t>(mIndex));
+ aidlSys.isInput = mIsInput;
+ aidlSys.useForVolume = mUseForVolume;
+ return std::make_pair(aidl, aidlSys);
}
-status_t AudioGain::writeToParcelable(media::AudioGain* parcelable) const {
- parcelable->index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mIndex));
- parcelable->isInput = mIsInput;
- parcelable->useForVolume = mUseForVolume;
- parcelable->mode = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_gain_mode_t_int32_t_mask(mGain.mode));
- parcelable->channelMask = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
- mGain.channel_mask, mIsInput));
- parcelable->minValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.min_value));
- parcelable->maxValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.max_value));
- parcelable->defaultValue = VALUE_OR_RETURN_STATUS(
- convertIntegral<int32_t>(mGain.default_value));
- parcelable->stepValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.step_value));
- parcelable->minRampMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.min_ramp_ms));
- parcelable->maxRampMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.max_ramp_ms));
- return OK;
-}
-
-status_t AudioGain::readFromParcel(const android::Parcel *parcel) {
- media::AudioGain parcelable;
- return parcelable.readFromParcel(parcel)
- ?: readFromParcelable(parcelable);
-}
-
-status_t AudioGain::readFromParcelable(const media::AudioGain& parcelable) {
- mIndex = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.index));
- mIsInput = parcelable.isInput;
- mUseForVolume = parcelable.useForVolume;
- mGain.mode = VALUE_OR_RETURN_STATUS(
- aidl2legacy_int32_t_audio_gain_mode_t_mask(parcelable.mode));
- mGain.channel_mask = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
- parcelable.channelMask, parcelable.isInput));
- mGain.min_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.minValue));
- mGain.max_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.maxValue));
- mGain.default_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.defaultValue));
- mGain.step_value = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.stepValue));
- mGain.min_ramp_ms = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.minRampMs));
- mGain.max_ramp_ms = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.maxRampMs));
- return OK;
+ConversionResult<sp<AudioGain>> AudioGain::fromParcelable(const AudioGain::Aidl& aidl) {
+ const media::AudioGainSys& sys = aidl.second;
+ auto index = VALUE_OR_RETURN(convertIntegral<int>(sys.index));
+ sp<AudioGain> legacy = sp<AudioGain>::make(index, sys.isInput);
+ legacy->mGain = VALUE_OR_RETURN(
+ aidl2legacy_AudioGain_audio_gain(aidl.first, sys.isInput));
+ legacy->mUseForVolume = sys.useForVolume;
+ return legacy;
}
bool AudioGains::equals(const AudioGains &other) const
@@ -180,59 +147,30 @@
});
}
-status_t AudioGains::writeToParcel(android::Parcel *parcel) const {
- status_t status = NO_ERROR;
- if ((status = parcel->writeVectorSize(*this)) != NO_ERROR) return status;
- for (const auto &audioGain : *this) {
- if ((status = parcel->writeParcelable(*audioGain)) != NO_ERROR) {
- break;
- }
- }
- return status;
-}
-
-status_t AudioGains::readFromParcel(const android::Parcel *parcel) {
- status_t status = NO_ERROR;
- this->clear();
- if ((status = parcel->resizeOutVector(this)) != NO_ERROR) return status;
- for (size_t i = 0; i < this->size(); i++) {
- this->at(i) = new AudioGain(0, false);
- if ((status = parcel->readParcelable(this->at(i).get())) != NO_ERROR) {
- this->clear();
- break;
- }
- }
- return status;
-}
-
ConversionResult<sp<AudioGain>>
-aidl2legacy_AudioGain(const media::AudioGain& aidl) {
- sp<AudioGain> legacy = new AudioGain(0, false);
- status_t status = legacy->readFromParcelable(aidl);
- if (status != OK) {
- return base::unexpected(status);
- }
- return legacy;
+aidl2legacy_AudioGain(const AudioGain::Aidl& aidl) {
+ return AudioGain::fromParcelable(aidl);
}
-ConversionResult<media::AudioGain>
+ConversionResult<AudioGain::Aidl>
legacy2aidl_AudioGain(const sp<AudioGain>& legacy) {
- media::AudioGain aidl;
- status_t status = legacy->writeToParcelable(&aidl);
- if (status != OK) {
- return base::unexpected(status);
- }
- return aidl;
+ return legacy->toParcelable();
}
ConversionResult<AudioGains>
-aidl2legacy_AudioGains(const std::vector<media::AudioGain>& aidl) {
- return convertContainer<AudioGains>(aidl, aidl2legacy_AudioGain);
+aidl2legacy_AudioGains(const AudioGains::Aidl& aidl) {
+ return convertContainers<AudioGains>(aidl.first, aidl.second,
+ [](const media::audio::common::AudioGain& g,
+ const media::AudioGainSys& gs) {
+ return aidl2legacy_AudioGain(std::make_pair(g, gs));
+ });
}
-ConversionResult<std::vector<media::AudioGain>>
+ConversionResult<AudioGains::Aidl>
legacy2aidl_AudioGains(const AudioGains& legacy) {
- return convertContainer<std::vector<media::AudioGain>>(legacy, legacy2aidl_AudioGain);
+ return convertContainerSplit<
+ std::vector<media::audio::common::AudioGain>,
+ std::vector<media::AudioGainSys>>(legacy, legacy2aidl_AudioGain);
}
} // namespace android
diff --git a/media/libaudiofoundation/AudioPort.cpp b/media/libaudiofoundation/AudioPort.cpp
index c70a6c2..099aff4 100644
--- a/media/libaudiofoundation/AudioPort.cpp
+++ b/media/libaudiofoundation/AudioPort.cpp
@@ -18,7 +18,6 @@
#include <algorithm>
#include <utility>
-#include <android/media/ExtraAudioDescriptor.h>
#include <android-base/stringprintf.h>
#include <media/AudioPort.h>
#include <utils/Log.h>
@@ -207,13 +206,19 @@
}
status_t AudioPort::writeToParcelable(media::AudioPort* parcelable) const {
- parcelable->name = mName;
- parcelable->type = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_type_t_AudioPortType(mType));
- parcelable->role = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_role_t_AudioPortRole(mRole));
- parcelable->profiles = VALUE_OR_RETURN_STATUS(
+ parcelable->hal.name = mName;
+ parcelable->sys.type = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_port_type_t_AudioPortType(mType));
+ parcelable->sys.role = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_port_role_t_AudioPortRole(mRole));
+ auto aidlProfiles = VALUE_OR_RETURN_STATUS(
legacy2aidl_AudioProfileVector(mProfiles, useInputChannelMask()));
- parcelable->extraAudioDescriptors = mExtraAudioDescriptors;
- parcelable->gains = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioGains(mGains));
+ parcelable->hal.profiles = aidlProfiles.first;
+ parcelable->sys.profiles = aidlProfiles.second;
+ parcelable->hal.extraAudioDescriptors = mExtraAudioDescriptors;
+ auto aidlGains = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioGains(mGains));
+ parcelable->hal.gains = aidlGains.first;
+ parcelable->sys.gains = aidlGains.second;
return OK;
}
@@ -224,13 +229,18 @@
}
status_t AudioPort::readFromParcelable(const media::AudioPort& parcelable) {
- mName = parcelable.name;
- mType = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioPortType_audio_port_type_t(parcelable.type));
- mRole = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioPortRole_audio_port_role_t(parcelable.role));
+ mName = parcelable.hal.name;
+ mType = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioPortType_audio_port_type_t(parcelable.sys.type));
+ mRole = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioPortRole_audio_port_role_t(parcelable.sys.role));
mProfiles = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioProfileVector(parcelable.profiles, useInputChannelMask()));
- mExtraAudioDescriptors = parcelable.extraAudioDescriptors;
- mGains = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioGains(parcelable.gains));
+ aidl2legacy_AudioProfileVector(
+ std::make_pair(parcelable.hal.profiles, parcelable.sys.profiles),
+ useInputChannelMask()));
+ mExtraAudioDescriptors = parcelable.hal.extraAudioDescriptors;
+ mGains = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioGains(std::make_pair(parcelable.hal.gains, parcelable.sys.gains)));
return OK;
}
@@ -333,46 +343,40 @@
}
status_t AudioPortConfig::writeToParcelable(
- media::AudioPortConfig* parcelable, bool isInput) const {
- parcelable->sampleRate = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mSamplingRate));
+ media::audio::common::AudioPortConfig* parcelable, bool isInput) const {
+ media::audio::common::Int aidl_sampleRate;
+ aidl_sampleRate.value = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mSamplingRate));
+ parcelable->sampleRate = aidl_sampleRate;
parcelable->format = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_format_t_AudioFormatDescription(mFormat));
parcelable->channelMask = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_channel_mask_t_AudioChannelLayout(mChannelMask, isInput));
parcelable->id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
- parcelable->gain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.index));
- parcelable->gain.mode = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_gain_mode_t_int32_t_mask(mGain.mode));
- parcelable->gain.channelMask = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_channel_mask_t_AudioChannelLayout(mGain.channel_mask, isInput));
- parcelable->gain.rampDurationMs = VALUE_OR_RETURN_STATUS(
- convertIntegral<int32_t>(mGain.ramp_duration_ms));
- parcelable->gain.values = VALUE_OR_RETURN_STATUS(convertContainer<std::vector<int32_t>>(
- mGain.values, convertIntegral<int32_t, int>));
+ media::audio::common::AudioGainConfig aidl_gain = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_gain_config_AudioGainConfig(mGain, isInput));
+ parcelable->gain = aidl_gain;
return OK;
}
status_t AudioPortConfig::readFromParcelable(
- const media::AudioPortConfig& parcelable, bool isInput) {
- mSamplingRate = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.sampleRate));
- mFormat = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioFormatDescription_audio_format_t(parcelable.format));
- mChannelMask = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioChannelLayout_audio_channel_mask_t(parcelable.channelMask, isInput));
- mId = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_port_handle_t(parcelable.id));
- mGain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.gain.index));
- mGain.mode = VALUE_OR_RETURN_STATUS(
- aidl2legacy_int32_t_audio_gain_mode_t_mask(parcelable.gain.mode));
- mGain.channel_mask = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
- parcelable.gain.channelMask, isInput));
- mGain.ramp_duration_ms = VALUE_OR_RETURN_STATUS(
- convertIntegral<unsigned int>(parcelable.gain.rampDurationMs));
- if (parcelable.gain.values.size() > std::size(mGain.values)) {
- return BAD_VALUE;
+ const media::audio::common::AudioPortConfig& parcelable, bool isInput) {
+ if (parcelable.sampleRate.has_value()) {
+ mSamplingRate = VALUE_OR_RETURN_STATUS(
+ convertIntegral<unsigned int>(parcelable.sampleRate.value().value));
}
- for (size_t i = 0; i < parcelable.gain.values.size(); ++i) {
- mGain.values[i] = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.gain.values[i]));
+ if (parcelable.format.has_value()) {
+ mFormat = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioFormatDescription_audio_format_t(parcelable.format.value()));
+ }
+ if (parcelable.channelMask.has_value()) {
+ mChannelMask = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
+ parcelable.channelMask.value(), isInput));
+ }
+ mId = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_port_handle_t(parcelable.id));
+ if (parcelable.gain.has_value()) {
+ mGain = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioGainConfig_audio_gain_config(parcelable.gain.value(), isInput));
}
return OK;
}
diff --git a/media/libaudiofoundation/AudioProfile.cpp b/media/libaudiofoundation/AudioProfile.cpp
index 47b2d54..15f2d68 100644
--- a/media/libaudiofoundation/AudioProfile.cpp
+++ b/media/libaudiofoundation/AudioProfile.cpp
@@ -27,6 +27,8 @@
namespace android {
+using media::audio::common::AudioChannelLayout;
+
bool operator == (const AudioProfile &left, const AudioProfile &right)
{
return (left.getFormat() == right.getFormat()) &&
@@ -154,57 +156,65 @@
return *this;
}
-ConversionResult<media::AudioProfile>
+ConversionResult<AudioProfile::Aidl>
AudioProfile::toParcelable(bool isInput) const {
- media::AudioProfile parcelable;
+ media::audio::common::AudioProfile parcelable;
parcelable.name = mName;
- parcelable.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormatDescription(mFormat));
+ parcelable.format = VALUE_OR_RETURN(
+ legacy2aidl_audio_format_t_AudioFormatDescription(mFormat));
+ // Note: legacy 'audio_profile' imposes a limit on the number of
+ // channel masks and sampling rates. That's why it's not used here
+ // and conversions are performed directly on the fields instead
+ // of using 'legacy2aidl_audio_profile_AudioProfile' from AidlConversion.
parcelable.channelMasks = VALUE_OR_RETURN(
- convertContainer<std::vector<media::AudioChannelLayout>>(
+ convertContainer<std::vector<AudioChannelLayout>>(
mChannelMasks,
[isInput](audio_channel_mask_t m) {
return legacy2aidl_audio_channel_mask_t_AudioChannelLayout(m, isInput);
}));
- parcelable.samplingRates = VALUE_OR_RETURN(
+ parcelable.sampleRates = VALUE_OR_RETURN(
convertContainer<std::vector<int32_t>>(mSamplingRates,
convertIntegral<int32_t, uint32_t>));
- parcelable.isDynamicFormat = mIsDynamicFormat;
- parcelable.isDynamicChannels = mIsDynamicChannels;
- parcelable.isDynamicRate = mIsDynamicRate;
parcelable.encapsulationType = VALUE_OR_RETURN(
legacy2aidl_audio_encapsulation_type_t_AudioEncapsulationType(mEncapsulationType));
- return parcelable;
+ media::AudioProfileSys parcelableSys;
+ parcelableSys.isDynamicFormat = mIsDynamicFormat;
+ parcelableSys.isDynamicChannels = mIsDynamicChannels;
+ parcelableSys.isDynamicRate = mIsDynamicRate;
+ return std::make_pair(parcelable, parcelableSys);
}
-ConversionResult<sp<AudioProfile>>
-AudioProfile::fromParcelable(const media::AudioProfile& parcelable, bool isInput) {
+ConversionResult<sp<AudioProfile>> AudioProfile::fromParcelable(
+ const AudioProfile::Aidl& aidl, bool isInput) {
sp<AudioProfile> legacy = new AudioProfile();
+ const auto& parcelable = aidl.first;
legacy->mName = parcelable.name;
legacy->mFormat = VALUE_OR_RETURN(
aidl2legacy_AudioFormatDescription_audio_format_t(parcelable.format));
legacy->mChannelMasks = VALUE_OR_RETURN(
convertContainer<ChannelMaskSet>(parcelable.channelMasks,
- [isInput](const media::AudioChannelLayout& l) {
+ [isInput](const AudioChannelLayout& l) {
return aidl2legacy_AudioChannelLayout_audio_channel_mask_t(l, isInput);
}));
legacy->mSamplingRates = VALUE_OR_RETURN(
- convertContainer<SampleRateSet>(parcelable.samplingRates,
+ convertContainer<SampleRateSet>(parcelable.sampleRates,
convertIntegral<uint32_t, int32_t>));
- legacy->mIsDynamicFormat = parcelable.isDynamicFormat;
- legacy->mIsDynamicChannels = parcelable.isDynamicChannels;
- legacy->mIsDynamicRate = parcelable.isDynamicRate;
legacy->mEncapsulationType = VALUE_OR_RETURN(
aidl2legacy_AudioEncapsulationType_audio_encapsulation_type_t(
parcelable.encapsulationType));
+ const auto& parcelableSys = aidl.second;
+ legacy->mIsDynamicFormat = parcelableSys.isDynamicFormat;
+ legacy->mIsDynamicChannels = parcelableSys.isDynamicChannels;
+ legacy->mIsDynamicRate = parcelableSys.isDynamicRate;
return legacy;
}
ConversionResult<sp<AudioProfile>>
-aidl2legacy_AudioProfile(const media::AudioProfile& aidl, bool isInput) {
+aidl2legacy_AudioProfile(const AudioProfile::Aidl& aidl, bool isInput) {
return AudioProfile::fromParcelable(aidl, isInput);
}
-ConversionResult<media::AudioProfile>
+ConversionResult<AudioProfile::Aidl>
legacy2aidl_AudioProfile(const sp<AudioProfile>& legacy, bool isInput) {
return legacy->toParcelable(isInput);
}
@@ -329,16 +339,19 @@
}
ConversionResult<AudioProfileVector>
-aidl2legacy_AudioProfileVector(const std::vector<media::AudioProfile>& aidl, bool isInput) {
- return convertContainer<AudioProfileVector>(aidl,
- [isInput](const media::AudioProfile& p) {
- return aidl2legacy_AudioProfile(p, isInput);
+aidl2legacy_AudioProfileVector(const AudioProfileVector::Aidl& aidl, bool isInput) {
+ return convertContainers<AudioProfileVector>(aidl.first, aidl.second,
+ [isInput](const media::audio::common::AudioProfile& p,
+ const media::AudioProfileSys& ps) {
+ return aidl2legacy_AudioProfile(std::make_pair(p, ps), isInput);
});
}
-ConversionResult<std::vector<media::AudioProfile>>
+ConversionResult<AudioProfileVector::Aidl>
legacy2aidl_AudioProfileVector(const AudioProfileVector& legacy, bool isInput) {
- return convertContainer<std::vector<media::AudioProfile>>(legacy,
+ return convertContainerSplit<
+ std::vector<media::audio::common::AudioProfile>,
+ std::vector<media::AudioProfileSys>>(legacy,
[isInput](const sp<AudioProfile>& p) {
return legacy2aidl_AudioProfile(p, isInput);
});
diff --git a/media/libaudiofoundation/DeviceDescriptorBase.cpp b/media/libaudiofoundation/DeviceDescriptorBase.cpp
index 3cce722..cf829b5 100644
--- a/media/libaudiofoundation/DeviceDescriptorBase.cpp
+++ b/media/libaudiofoundation/DeviceDescriptorBase.cpp
@@ -166,16 +166,18 @@
status_t DeviceDescriptorBase::writeToParcelable(media::AudioPort* parcelable) const {
AudioPort::writeToParcelable(parcelable);
- AudioPortConfig::writeToParcelable(&parcelable->activeConfig, useInputChannelMask());
- parcelable->id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
+ AudioPortConfig::writeToParcelable(&parcelable->hal.activeConfig, useInputChannelMask());
+ parcelable->hal.id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
- media::AudioPortDeviceExt ext;
- ext.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(mDeviceTypeAddr));
- ext.encapsulationModes = VALUE_OR_RETURN_STATUS(
+ media::audio::common::AudioDevice device = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_AudioDeviceTypeAddress(mDeviceTypeAddr));
+ UNION_SET(parcelable->hal.ext, device, device);
+ media::AudioPortDeviceExtSys deviceSys;
+ deviceSys.encapsulationModes = VALUE_OR_RETURN_STATUS(
legacy2aidl_AudioEncapsulationMode_mask(mEncapsulationModes));
- ext.encapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
+ deviceSys.encapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
legacy2aidl_AudioEncapsulationMetadataType_mask(mEncapsulationMetadataTypes));
- UNION_SET(parcelable->ext, device, std::move(ext));
+ UNION_SET(parcelable->sys.ext, device, deviceSys);
return OK;
}
@@ -186,22 +188,26 @@
}
status_t DeviceDescriptorBase::readFromParcelable(const media::AudioPort& parcelable) {
- if (parcelable.type != media::AudioPortType::DEVICE) {
+ if (parcelable.sys.type != media::AudioPortType::DEVICE) {
return BAD_VALUE;
}
status_t status = AudioPort::readFromParcelable(parcelable)
- ?: AudioPortConfig::readFromParcelable(parcelable.activeConfig, useInputChannelMask());
+ ?: AudioPortConfig::readFromParcelable(
+ parcelable.hal.activeConfig, useInputChannelMask());
if (status != OK) {
return status;
}
- media::AudioPortDeviceExt ext = VALUE_OR_RETURN_STATUS(UNION_GET(parcelable.ext, device));
+ media::audio::common::AudioDevice device = VALUE_OR_RETURN_STATUS(
+ UNION_GET(parcelable.hal.ext, device));
mDeviceTypeAddr = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioDeviceTypeAddress(ext.device));
+ aidl2legacy_AudioDeviceTypeAddress(device));
+ media::AudioPortDeviceExtSys deviceSys = VALUE_OR_RETURN_STATUS(
+ UNION_GET(parcelable.sys.ext, device));
mEncapsulationModes = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioEncapsulationMode_mask(ext.encapsulationModes));
+ aidl2legacy_AudioEncapsulationMode_mask(deviceSys.encapsulationModes));
mEncapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioEncapsulationMetadataType_mask(ext.encapsulationMetadataTypes));
+ aidl2legacy_AudioEncapsulationMetadataType_mask(deviceSys.encapsulationMetadataTypes));
return OK;
}
diff --git a/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h b/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
index 8edcc58..11aa222 100644
--- a/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
+++ b/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
@@ -19,7 +19,7 @@
#include <string>
#include <vector>
-#include <android/media/AudioDevice.h>
+#include <android/media/audio/common/AudioDevice.h>
#include <binder/Parcelable.h>
#include <binder/Parcel.h>
#include <media/AudioContainers.h>
@@ -32,6 +32,7 @@
class AudioDeviceTypeAddr : public Parcelable {
public:
AudioDeviceTypeAddr() = default;
+ AudioDeviceTypeAddr(const AudioDeviceTypeAddr&) = default;
AudioDeviceTypeAddr(audio_devices_t type, const std::string& address);
@@ -88,8 +89,8 @@
// Conversion routines, according to AidlConversion.h conventions.
ConversionResult<AudioDeviceTypeAddr>
-aidl2legacy_AudioDeviceTypeAddress(const media::AudioDevice& aidl);
-ConversionResult<media::AudioDevice>
+aidl2legacy_AudioDeviceTypeAddress(const media::audio::common::AudioDevice& aidl);
+ConversionResult<media::audio::common::AudioDevice>
legacy2aidl_AudioDeviceTypeAddress(const AudioDeviceTypeAddr& legacy);
} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioGain.h b/media/libaudiofoundation/include/media/AudioGain.h
index 28769d2..10088f2 100644
--- a/media/libaudiofoundation/include/media/AudioGain.h
+++ b/media/libaudiofoundation/include/media/AudioGain.h
@@ -16,23 +16,23 @@
#pragma once
-#include <android/media/AudioGain.h>
-#include <binder/Parcel.h>
-#include <binder/Parcelable.h>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include <android/media/AudioGainSys.h>
#include <media/AidlConversion.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
#include <system/audio.h>
-#include <string>
-#include <vector>
namespace android {
-class AudioGain: public RefBase, public Parcelable
+class AudioGain: public RefBase
{
public:
AudioGain(int index, bool isInput);
- virtual ~AudioGain() {}
+ virtual ~AudioGain() = default;
void setMode(audio_gain_mode_t mode) { mGain.mode = mode; }
const audio_gain_mode_t &getMode() const { return mGain.mode; }
@@ -71,26 +71,24 @@
bool equals(const sp<AudioGain>& other) const;
- status_t writeToParcel(Parcel* parcel) const override;
- status_t readFromParcel(const Parcel* parcel) override;
-
- status_t writeToParcelable(media::AudioGain* parcelable) const;
- status_t readFromParcelable(const media::AudioGain& parcelable);
+ using Aidl = std::pair<media::audio::common::AudioGain, media::AudioGainSys>;
+ ConversionResult<Aidl> toParcelable() const;
+ static ConversionResult<sp<AudioGain>> fromParcelable(const Aidl& aidl);
private:
int mIndex;
- struct audio_gain mGain;
bool mIsInput;
+ struct audio_gain mGain = {};
bool mUseForVolume = false;
};
// Conversion routines, according to AidlConversion.h conventions.
ConversionResult<sp<AudioGain>>
-aidl2legacy_AudioGain(const media::AudioGain& aidl);
-ConversionResult<media::AudioGain>
+aidl2legacy_AudioGain(const AudioGain::Aidl& aidl);
+ConversionResult<AudioGain::Aidl>
legacy2aidl_AudioGain(const sp<AudioGain>& legacy);
-class AudioGains : public std::vector<sp<AudioGain> >, public Parcelable
+class AudioGains : public std::vector<sp<AudioGain>>
{
public:
bool canUseForVolume() const
@@ -103,7 +101,7 @@
return false;
}
- int32_t add(const sp<AudioGain> gain)
+ int32_t add(const sp<AudioGain>& gain)
{
push_back(gain);
return 0;
@@ -111,14 +109,15 @@
bool equals(const AudioGains& other) const;
- status_t writeToParcel(Parcel* parcel) const override;
- status_t readFromParcel(const Parcel* parcel) override;
+ using Aidl = std::pair<
+ std::vector<media::audio::common::AudioGain>,
+ std::vector<media::AudioGainSys>>;
};
// Conversion routines, according to AidlConversion.h conventions.
ConversionResult<AudioGains>
-aidl2legacy_AudioGains(const std::vector<media::AudioGain>& aidl);
-ConversionResult<std::vector<media::AudioGain>>
+aidl2legacy_AudioGains(const AudioGains::Aidl& aidl);
+ConversionResult<AudioGains::Aidl>
legacy2aidl_AudioGains(const AudioGains& legacy);
} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioPort.h b/media/libaudiofoundation/include/media/AudioPort.h
index 6e1d032..aa9b5f6 100644
--- a/media/libaudiofoundation/include/media/AudioPort.h
+++ b/media/libaudiofoundation/include/media/AudioPort.h
@@ -21,7 +21,7 @@
#include <android/media/AudioPort.h>
#include <android/media/AudioPortConfig.h>
-#include <android/media/ExtraAudioDescriptor.h>
+#include <android/media/audio/common/ExtraAudioDescriptor.h>
#include <binder/Parcel.h>
#include <binder/Parcelable.h>
#include <media/AudioGain.h>
@@ -69,10 +69,10 @@
AudioProfileVector &getAudioProfiles() { return mProfiles; }
void setExtraAudioDescriptors(
- const std::vector<media::ExtraAudioDescriptor> extraAudioDescriptors) {
+ const std::vector<media::audio::common::ExtraAudioDescriptor> extraAudioDescriptors) {
mExtraAudioDescriptors = extraAudioDescriptors;
}
- std::vector<media::ExtraAudioDescriptor> &getExtraAudioDescriptors() {
+ std::vector<media::audio::common::ExtraAudioDescriptor> &getExtraAudioDescriptors() {
return mExtraAudioDescriptors;
}
@@ -114,7 +114,7 @@
// Audio capabilities that are defined by hardware descriptors when the format is unrecognized
// by the platform, e.g. short audio descriptor in EDID for HDMI.
- std::vector<media::ExtraAudioDescriptor> mExtraAudioDescriptors;
+ std::vector<media::audio::common::ExtraAudioDescriptor> mExtraAudioDescriptors;
private:
template <typename T, std::enable_if_t<std::is_same<T, struct audio_port>::value
|| std::is_same<T, struct audio_port_v7>::value, int> = 0>
@@ -152,8 +152,10 @@
bool equals(const sp<AudioPortConfig>& other) const;
- status_t writeToParcelable(media::AudioPortConfig* parcelable, bool isInput) const;
- status_t readFromParcelable(const media::AudioPortConfig& parcelable, bool isInput);
+ status_t writeToParcelable(
+ media::audio::common::AudioPortConfig* parcelable, bool isInput) const;
+ status_t readFromParcelable(
+ const media::audio::common::AudioPortConfig& parcelable, bool isInput);
protected:
unsigned int mSamplingRate = 0u;
diff --git a/media/libaudiofoundation/include/media/AudioProfile.h b/media/libaudiofoundation/include/media/AudioProfile.h
index e34a49f..62670e4 100644
--- a/media/libaudiofoundation/include/media/AudioProfile.h
+++ b/media/libaudiofoundation/include/media/AudioProfile.h
@@ -17,11 +17,10 @@
#pragma once
#include <string>
+#include <utility>
#include <vector>
-#include <android/media/AudioProfile.h>
-#include <binder/Parcel.h>
-#include <binder/Parcelable.h>
+#include <android/media/AudioProfileSys.h>
#include <media/AidlConversion.h>
#include <media/AudioContainers.h>
#include <system/audio.h>
@@ -81,9 +80,10 @@
bool equals(const sp<AudioProfile>& other) const;
- ConversionResult<media::AudioProfile> toParcelable(bool isInput) const;
+ using Aidl = std::pair<media::audio::common::AudioProfile, media::AudioProfileSys>;
+ ConversionResult<Aidl> toParcelable(bool isInput) const;
static ConversionResult<sp<AudioProfile>> fromParcelable(
- const media::AudioProfile& parcelable, bool isInput);
+ const Aidl& aidl, bool isInput);
private:
@@ -104,8 +104,8 @@
// Conversion routines, according to AidlConversion.h conventions.
ConversionResult<sp<AudioProfile>>
-aidl2legacy_AudioProfile(const media::AudioProfile& aidl, bool isInput);
-ConversionResult<media::AudioProfile>
+aidl2legacy_AudioProfile(const AudioProfile::Aidl& aidl, bool isInput);
+ConversionResult<AudioProfile::Aidl>
legacy2aidl_AudioProfile(const sp<AudioProfile>& legacy, bool isInput);
class AudioProfileVector : public std::vector<sp<AudioProfile>>
@@ -134,14 +134,18 @@
virtual void dump(std::string *dst, int spaces) const;
bool equals(const AudioProfileVector& other) const;
+
+ using Aidl = std::pair<
+ std::vector<media::audio::common::AudioProfile>,
+ std::vector<media::AudioProfileSys>>;
};
bool operator == (const AudioProfile &left, const AudioProfile &right);
// Conversion routines, according to AidlConversion.h conventions.
ConversionResult<AudioProfileVector>
-aidl2legacy_AudioProfileVector(const std::vector<media::AudioProfile>& aidl, bool isInput);
-ConversionResult<std::vector<media::AudioProfile>>
+aidl2legacy_AudioProfileVector(const AudioProfileVector::Aidl& aidl, bool isInput);
+ConversionResult<AudioProfileVector::Aidl>
legacy2aidl_AudioProfileVector(const AudioProfileVector& legacy, bool isInput);
AudioProfileVector intersectAudioProfiles(const AudioProfileVector& profiles1,
diff --git a/media/libaudiofoundation/tests/Android.bp b/media/libaudiofoundation/tests/Android.bp
index f3cd446..3f1fbea 100644
--- a/media/libaudiofoundation/tests/Android.bp
+++ b/media/libaudiofoundation/tests/Android.bp
@@ -18,6 +18,7 @@
],
static_libs: [
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"libaudioclient_aidl_conversion",
"libaudiofoundation",
diff --git a/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp b/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp
index 1696980..0315a59 100644
--- a/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp
+++ b/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp
@@ -75,15 +75,40 @@
return audioProfiles;
}
-TEST(AudioFoundationParcelableTest, ParcelingAudioGain) {
- Parcel data;
- AudioGains audioGains = getAudioGainsForTest();
+TEST(AudioFoundationParcelableTest, ParcelingAudioProfile) {
+ sp<AudioProfile> profile = getAudioProfileVectorForTest()[0];
+ auto conv = legacy2aidl_AudioProfile(profile, false /*isInput*/);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = aidl2legacy_AudioProfile(conv.value(), false /*isInput*/);
+ ASSERT_TRUE(convBack.ok());
+ ASSERT_TRUE(profile->equals(convBack.value()));
+}
- ASSERT_EQ(data.writeParcelable(audioGains), NO_ERROR);
- data.setDataPosition(0);
- AudioGains audioGainsFromParcel;
- ASSERT_EQ(data.readParcelable(&audioGainsFromParcel), NO_ERROR);
- ASSERT_TRUE(audioGainsFromParcel.equals(audioGains));
+TEST(AudioFoundationParcelableTest, ParcelingAudioProfileVector) {
+ AudioProfileVector profiles = getAudioProfileVectorForTest();
+ auto conv = legacy2aidl_AudioProfileVector(profiles, false /*isInput*/);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = aidl2legacy_AudioProfileVector(conv.value(), false /*isInput*/);
+ ASSERT_TRUE(convBack.ok());
+ ASSERT_TRUE(profiles.equals(convBack.value()));
+}
+
+TEST(AudioFoundationParcelableTest, ParcelingAudioGain) {
+ sp<AudioGain> audioGain = getAudioGainsForTest()[0];
+ auto conv = legacy2aidl_AudioGain(audioGain);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = aidl2legacy_AudioGain(conv.value());
+ ASSERT_TRUE(convBack.ok());
+ ASSERT_TRUE(audioGain->equals(convBack.value()));
+}
+
+TEST(AudioFoundationParcelableTest, ParcelingAudioGains) {
+ AudioGains audioGains = getAudioGainsForTest();
+ auto conv = legacy2aidl_AudioGains(audioGains);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = aidl2legacy_AudioGains(conv.value());
+ ASSERT_TRUE(convBack.ok());
+ ASSERT_TRUE(audioGains.equals(convBack.value()));
}
TEST(AudioFoundationParcelableTest, ParcelingAudioPort) {
@@ -105,11 +130,11 @@
Parcel data;
sp<AudioPortConfig> audioPortConfig = new AudioPortConfigTestStub();
audioPortConfig->applyAudioPortConfig(&TEST_AUDIO_PORT_CONFIG);
- media::AudioPortConfig parcelable{};
+ media::audio::common::AudioPortConfig parcelable{};
ASSERT_EQ(NO_ERROR, audioPortConfig->writeToParcelable(&parcelable, false /*isInput*/));
ASSERT_EQ(NO_ERROR, data.writeParcelable(parcelable));
data.setDataPosition(0);
- media::AudioPortConfig parcelableFromParcel{};
+ media::audio::common::AudioPortConfig parcelableFromParcel{};
ASSERT_EQ(NO_ERROR, data.readParcelable(&parcelableFromParcel));
sp<AudioPortConfig> audioPortConfigFromParcel = new AudioPortConfigTestStub();
ASSERT_EQ(NO_ERROR, audioPortConfigFromParcel->readFromParcelable(
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index bd24c84..5fe74f9 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -30,6 +30,7 @@
],
shared_libs: [
+ "audioclient-types-aidl-cpp",
"libdl",
"libhidlbase",
"liblog",
diff --git a/media/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index a2c6e8a..d6576f5 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -50,6 +50,7 @@
"libmedia_helper",
"libmediautils",
"libutils",
+ "audioclient-types-aidl-cpp",
],
header_libs: [
"android.hardware.audio.common.util@all-versions",
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index aa94eea..47acb19 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -20,6 +20,7 @@
//#define LOG_NDEBUG 0
#include <cutils/native_handle.h>
+#include <cutils/properties.h>
#include <hwbinder/IPCThreadState.h>
#include <media/AudioContainers.h>
#include <utils/Log.h>
diff --git a/media/libaudiohal/impl/DeviceHalHidl.h b/media/libaudiohal/impl/DeviceHalHidl.h
index 2694ab3..9fd0ac0 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.h
+++ b/media/libaudiohal/impl/DeviceHalHidl.h
@@ -119,6 +119,23 @@
status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
+ status_t getMmapPolicyInfos(
+ media::audio::common::AudioMMapPolicyType policyType __unused,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos __unused) override {
+ // TODO: Implement the HAL query when moving to AIDL HAL.
+ return INVALID_OPERATION;
+ }
+
+ int32_t getAAudioMixerBurstCount() override {
+ // TODO: Implement the HAL query when moving to AIDL HAL.
+ return INVALID_OPERATION;
+ }
+
+ int32_t getAAudioHardwareBurstMinUsec() override {
+ // TODO: Implement the HAL query when moving to AIDL HAL.
+ return INVALID_OPERATION;
+ }
+
status_t dump(int fd, const Vector<String16>& args) override;
private:
diff --git a/media/libaudiohal/impl/DeviceHalLocal.h b/media/libaudiohal/impl/DeviceHalLocal.h
index 2fde936..ee1d2c5 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.h
+++ b/media/libaudiohal/impl/DeviceHalLocal.h
@@ -112,6 +112,23 @@
status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
+ status_t getMmapPolicyInfos(
+ media::audio::common::AudioMMapPolicyType policyType __unused,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos __unused) override {
+ // This function will only be available on AIDL HAL.
+ return INVALID_OPERATION;
+ }
+
+ int32_t getAAudioMixerBurstCount() override {
+ // This function will only be available on AIDL HAL.
+ return INVALID_OPERATION;
+ }
+
+ int32_t getAAudioHardwareBurstMinUsec() override {
+ // This function will only be available on AIDL HAL.
+ return INVALID_OPERATION;
+ }
+
status_t dump(int fd, const Vector<String16>& args) override;
void closeOutputStream(struct audio_stream_out *stream_out);
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHidl.h b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
index 6f84efe..b46259b 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHidl.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
@@ -46,6 +46,8 @@
status_t setCallbackOnce(sp<DevicesFactoryHalCallback> callback) override;
+ float getHalVersion() const override { return MAJOR_VERSION + (float)MINOR_VERSION / 10; }
+
private:
friend class ServiceNotificationListener;
void addDeviceFactory(sp<IDevicesFactory> factory, bool needToNotify);
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHybrid.h b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
index 568a1fb..5baefa4b 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
@@ -40,6 +40,10 @@
status_t setCallbackOnce(sp<DevicesFactoryHalCallback> callback) override;
+ float getHalVersion() const override {
+ return MAJOR_VERSION + (float)MINOR_VERSION / 10;
+ }
+
private:
sp<DevicesFactoryHalInterface> mLocalFactory;
sp<DevicesFactoryHalInterface> mHidlFactory;
diff --git a/media/libaudiohal/impl/DevicesFactoryHalLocal.h b/media/libaudiohal/impl/DevicesFactoryHalLocal.h
index 32bf362..d2b9104 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalLocal.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalLocal.h
@@ -41,6 +41,10 @@
return INVALID_OPERATION;
}
+ float getHalVersion() const override {
+ return MAJOR_VERSION + (float)MINOR_VERSION / 10;
+ }
+
private:
friend class DevicesFactoryHalHybrid;
diff --git a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
index 69cbcec..70c3199 100644
--- a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
@@ -17,6 +17,8 @@
#ifndef ANDROID_HARDWARE_DEVICE_HAL_INTERFACE_H
#define ANDROID_HARDWARE_DEVICE_HAL_INTERFACE_H
+#include <android/media/audio/common/AudioMMapPolicyInfo.h>
+#include <android/media/audio/common/AudioMMapPolicyType.h>
#include <media/audiohal/EffectHalInterface.h>
#include <media/MicrophoneInfo.h>
#include <system/audio.h>
@@ -120,6 +122,12 @@
virtual status_t removeDeviceEffect(
audio_port_handle_t device, sp<EffectHalInterface> effect) = 0;
+ virtual status_t getMmapPolicyInfos(
+ media::audio::common::AudioMMapPolicyType policyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos) = 0;
+ virtual int32_t getAAudioMixerBurstCount() = 0;
+ virtual int32_t getAAudioHardwareBurstMinUsec() = 0;
+
virtual status_t dump(int fd, const Vector<String16>& args) = 0;
protected:
diff --git a/media/libaudiohal/include/media/audiohal/DevicesFactoryHalInterface.h b/media/libaudiohal/include/media/audiohal/DevicesFactoryHalInterface.h
index 5091558..17010e6 100644
--- a/media/libaudiohal/include/media/audiohal/DevicesFactoryHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DevicesFactoryHalInterface.h
@@ -43,6 +43,8 @@
// The callback can be only set once.
virtual status_t setCallbackOnce(sp<DevicesFactoryHalCallback> callback) = 0;
+ virtual float getHalVersion() const = 0;
+
static sp<DevicesFactoryHalInterface> create();
protected:
diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index 2988c67..ab6a8b6 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h
@@ -328,7 +328,9 @@
DO_CHANNEL_POSITION(21);
DO_CHANNEL_POSITION(22);
DO_CHANNEL_POSITION(23);
- static_assert(FCC_LIMIT <= FCC_24); // Note: this may need to change.
+ DO_CHANNEL_POSITION(24);
+ DO_CHANNEL_POSITION(25);
+ static_assert(FCC_LIMIT <= FCC_26); // Note: this may need to change.
#pragma pop_macro("DO_CHANNEL_POSITION")
}
@@ -346,6 +348,8 @@
[7] = AUDIO_CHANNEL_OUT_6POINT1,
[8] = AUDIO_CHANNEL_OUT_7POINT1,
[12] = AUDIO_CHANNEL_OUT_7POINT1POINT4,
+ [14] = AUDIO_CHANNEL_OUT_9POINT1POINT4,
+ [16] = AUDIO_CHANNEL_OUT_9POINT1POINT6,
[24] = AUDIO_CHANNEL_OUT_22POINT2,
};
return channelCount < std::size(canonical) ? canonical[channelCount] : AUDIO_CHANNEL_NONE;
diff --git a/media/libeffects/downmix/EffectDownmix.cpp b/media/libeffects/downmix/EffectDownmix.cpp
index f500bc3..d8f5787 100644
--- a/media/libeffects/downmix/EffectDownmix.cpp
+++ b/media/libeffects/downmix/EffectDownmix.cpp
@@ -19,7 +19,7 @@
#include <log/log.h>
#include "EffectDownmix.h"
-#include <math.h>
+#include <audio_utils/ChannelMix.h>
// Do not submit with DOWNMIX_TEST_CHANNEL_INDEX defined, strictly for testing
//#define DOWNMIX_TEST_CHANNEL_INDEX 0
@@ -35,12 +35,13 @@
} downmix_state_t;
/* parameters for each downmixer */
-typedef struct {
+struct downmix_object_t {
downmix_state_t state;
downmix_type_t type;
bool apply_volume_correction;
uint8_t input_channel_count;
-} downmix_object_t;
+ android::audio_utils::channels::ChannelMix channelMix;
+};
typedef struct downmix_module_s {
const struct effect_interface_s *itfe;
@@ -77,11 +78,6 @@
downmix_object_t *pDownmixer, int32_t param, uint32_t size, void *pValue);
static int Downmix_getParameter(
downmix_object_t *pDownmixer, int32_t param, uint32_t *pSize, void *pValue);
-static void Downmix_foldFromQuad(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
-static void Downmix_foldFrom5Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
-static void Downmix_foldFrom7Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
-static bool Downmix_foldGeneric(
- uint32_t mask, float *pSrc, float *pDst, size_t numFrames, bool accumulate);
// effect_handle_t interface implementation for downmix effect
const struct effect_interface_s gDownmixInterface = {
@@ -192,9 +188,11 @@
if (!mask) {
return false;
}
- // check against unsupported channels
- if (mask & ~AUDIO_CHANNEL_OUT_22POINT2) {
- ALOGE("Unsupported channels in %u", mask & ~AUDIO_CHANNEL_OUT_22POINT2);
+ // check against unsupported channels (up to FCC_26)
+ constexpr uint32_t MAXIMUM_CHANNEL_MASK = AUDIO_CHANNEL_OUT_22POINT2
+ | AUDIO_CHANNEL_OUT_FRONT_WIDE_LEFT | AUDIO_CHANNEL_OUT_FRONT_WIDE_RIGHT;
+ if (mask & ~MAXIMUM_CHANNEL_MASK) {
+ ALOGE("Unsupported channels in %#x", mask & ~MAXIMUM_CHANNEL_MASK);
return false;
}
return true;
@@ -315,7 +313,8 @@
audio_buffer_t *inBuffer, audio_buffer_t *outBuffer) {
downmix_object_t *pDownmixer;
- float *pSrc, *pDst;
+ const float *pSrc;
+ float *pDst;
downmix_module_t *pDwmModule = (downmix_module_t *)self;
if (pDwmModule == NULL) {
@@ -344,7 +343,8 @@
const bool accumulate =
(pDwmModule->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE);
- const uint32_t downmixInputChannelMask = pDwmModule->config.inputCfg.channels;
+ const audio_channel_mask_t downmixInputChannelMask =
+ (audio_channel_mask_t)pDwmModule->config.inputCfg.channels;
switch(pDownmixer->type) {
@@ -368,38 +368,13 @@
}
break;
- case DOWNMIX_TYPE_FOLD:
-#ifdef DOWNMIX_ALWAYS_USE_GENERIC_DOWNMIXER
- // bypass the optimized downmix routines for the common formats
- if (!Downmix_foldGeneric(
- downmixInputChannelMask, pSrc, pDst, numFrames, accumulate)) {
- ALOGE("Multichannel configuration %#x is not supported",
- downmixInputChannelMask);
- return -EINVAL;
- }
- break;
-#endif
- // optimize for the common formats
- switch (downmixInputChannelMask) {
- case AUDIO_CHANNEL_OUT_QUAD_BACK:
- case AUDIO_CHANNEL_OUT_QUAD_SIDE:
- Downmix_foldFromQuad(pSrc, pDst, numFrames, accumulate);
- break;
- case AUDIO_CHANNEL_OUT_5POINT1_BACK:
- case AUDIO_CHANNEL_OUT_5POINT1_SIDE:
- Downmix_foldFrom5Point1(pSrc, pDst, numFrames, accumulate);
- break;
- case AUDIO_CHANNEL_OUT_7POINT1:
- Downmix_foldFrom7Point1(pSrc, pDst, numFrames, accumulate);
- break;
- default:
- if (!Downmix_foldGeneric(
- downmixInputChannelMask, pSrc, pDst, numFrames, accumulate)) {
+ case DOWNMIX_TYPE_FOLD: {
+ if (!pDownmixer->channelMix.process(
+ pSrc, pDst, numFrames, accumulate, downmixInputChannelMask)) {
ALOGE("Multichannel configuration %#x is not supported",
downmixInputChannelMask);
return -EINVAL;
}
- break;
}
break;
@@ -674,6 +649,12 @@
ALOGE("Downmix_Configure error: invalid config");
return -EINVAL;
}
+ // when configuring the effect, do not allow a blank or unsupported channel mask
+ if (!Downmix_validChannelMask(pConfig->inputCfg.channels)) {
+ ALOGE("Downmix_Configure error: input channel mask(0x%x) not supported",
+ pConfig->inputCfg.channels);
+ return -EINVAL;
+ }
if (&pDwmModule->config != pConfig) {
memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
@@ -684,12 +665,6 @@
pDownmixer->apply_volume_correction = false;
pDownmixer->input_channel_count = 8; // matches default input of AUDIO_CHANNEL_OUT_7POINT1
} else {
- // when configuring the effect, do not allow a blank or unsupported channel mask
- if (!Downmix_validChannelMask(pConfig->inputCfg.channels)) {
- ALOGE("Downmix_Configure error: input channel mask(0x%x) not supported",
- pConfig->inputCfg.channels);
- return -EINVAL;
- }
pDownmixer->input_channel_count =
audio_channel_count_from_out_mask(pConfig->inputCfg.channels);
}
@@ -780,7 +755,6 @@
return 0;
} /* end Downmix_setParameter */
-
/*----------------------------------------------------------------------------
* Downmix_getParameter()
*----------------------------------------------------------------------------
@@ -829,299 +803,3 @@
return 0;
} /* end Downmix_getParameter */
-
-/*----------------------------------------------------------------------------
- * Downmix_foldFromQuad()
- *----------------------------------------------------------------------------
- * Purpose:
- * downmix a quad signal to stereo
- *
- * Inputs:
- * pSrc quad audio samples to downmix
- * numFrames the number of quad frames to downmix
- * accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
- * or overwrite pDst (when false)
- *
- * Outputs:
- * pDst downmixed stereo audio samples
- *
- *----------------------------------------------------------------------------
- */
-void Downmix_foldFromQuad(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
- // sample at index 0 is FL
- // sample at index 1 is FR
- // sample at index 2 is RL
- // sample at index 3 is RR
- if (accumulate) {
- while (numFrames) {
- // FL + RL
- pDst[0] = clamp_float(pDst[0] + ((pSrc[0] + pSrc[2]) / 2.0f));
- // FR + RR
- pDst[1] = clamp_float(pDst[1] + ((pSrc[1] + pSrc[3]) / 2.0f));
- pSrc += 4;
- pDst += 2;
- numFrames--;
- }
- } else { // same code as above but without adding and clamping pDst[i] to itself
- while (numFrames) {
- // FL + RL
- pDst[0] = clamp_float((pSrc[0] + pSrc[2]) / 2.0f);
- // FR + RR
- pDst[1] = clamp_float((pSrc[1] + pSrc[3]) / 2.0f);
- pSrc += 4;
- pDst += 2;
- numFrames--;
- }
- }
-}
-
-/*----------------------------------------------------------------------------
- * Downmix_foldFrom5Point1()
- *----------------------------------------------------------------------------
- * Purpose:
- * downmix a 5.1 signal to stereo
- *
- * Inputs:
- * pSrc 5.1 audio samples to downmix
- * numFrames the number of 5.1 frames to downmix
- * accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
- * or overwrite pDst (when false)
- *
- * Outputs:
- * pDst downmixed stereo audio samples
- *
- *----------------------------------------------------------------------------
- */
-void Downmix_foldFrom5Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
- float lt, rt, centerPlusLfeContrib; // samples in Q19.12 format
- // sample at index 0 is FL
- // sample at index 1 is FR
- // sample at index 2 is FC
- // sample at index 3 is LFE
- // sample at index 4 is RL
- // sample at index 5 is RR
- // code is mostly duplicated between the two values of accumulate to avoid repeating the test
- // for every sample
- if (accumulate) {
- while (numFrames) {
- // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
- centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
- + (pSrc[3] * MINUS_3_DB_IN_FLOAT);
- // FL + centerPlusLfeContrib + RL
- lt = pSrc[0] + centerPlusLfeContrib + pSrc[4];
- // FR + centerPlusLfeContrib + RR
- rt = pSrc[1] + centerPlusLfeContrib + pSrc[5];
- // accumulate in destination
- pDst[0] = clamp_float(pDst[0] + (lt / 2.0f));
- pDst[1] = clamp_float(pDst[1] + (rt / 2.0f));
- pSrc += 6;
- pDst += 2;
- numFrames--;
- }
- } else { // same code as above but without adding and clamping pDst[i] to itself
- while (numFrames) {
- // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
- centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
- + (pSrc[3] * MINUS_3_DB_IN_FLOAT);
- // FL + centerPlusLfeContrib + RL
- lt = pSrc[0] + centerPlusLfeContrib + pSrc[4];
- // FR + centerPlusLfeContrib + RR
- rt = pSrc[1] + centerPlusLfeContrib + pSrc[5];
- // store in destination
- pDst[0] = clamp_float(lt / 2.0f); // differs from when accumulate is true above
- pDst[1] = clamp_float(rt / 2.0f); // differs from when accumulate is true above
- pSrc += 6;
- pDst += 2;
- numFrames--;
- }
- }
-}
-
-/*----------------------------------------------------------------------------
- * Downmix_foldFrom7Point1()
- *----------------------------------------------------------------------------
- * Purpose:
- * downmix a 7.1 signal to stereo
- *
- * Inputs:
- * pSrc 7.1 audio samples to downmix
- * numFrames the number of 7.1 frames to downmix
- * accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
- * or overwrite pDst (when false)
- *
- * Outputs:
- * pDst downmixed stereo audio samples
- *
- *----------------------------------------------------------------------------
- */
-void Downmix_foldFrom7Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
- float lt, rt, centerPlusLfeContrib; // samples in Q19.12 format
- // sample at index 0 is FL
- // sample at index 1 is FR
- // sample at index 2 is FC
- // sample at index 3 is LFE
- // sample at index 4 is RL
- // sample at index 5 is RR
- // sample at index 6 is SL
- // sample at index 7 is SR
- // code is mostly duplicated between the two values of accumulate to avoid repeating the test
- // for every sample
- if (accumulate) {
- while (numFrames) {
- // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
- centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
- + (pSrc[3] * MINUS_3_DB_IN_FLOAT);
- // FL + centerPlusLfeContrib + SL + RL
- lt = pSrc[0] + centerPlusLfeContrib + pSrc[6] + pSrc[4];
- // FR + centerPlusLfeContrib + SR + RR
- rt = pSrc[1] + centerPlusLfeContrib + pSrc[7] + pSrc[5];
- //accumulate in destination
- pDst[0] = clamp_float(pDst[0] + (lt / 2.0f));
- pDst[1] = clamp_float(pDst[1] + (rt / 2.0f));
- pSrc += 8;
- pDst += 2;
- numFrames--;
- }
- } else { // same code as above but without adding and clamping pDst[i] to itself
- while (numFrames) {
- // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
- centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
- + (pSrc[3] * MINUS_3_DB_IN_FLOAT);
- // FL + centerPlusLfeContrib + SL + RL
- lt = pSrc[0] + centerPlusLfeContrib + pSrc[6] + pSrc[4];
- // FR + centerPlusLfeContrib + SR + RR
- rt = pSrc[1] + centerPlusLfeContrib + pSrc[7] + pSrc[5];
- // store in destination
- pDst[0] = clamp_float(lt / 2.0f); // differs from when accumulate is true above
- pDst[1] = clamp_float(rt / 2.0f); // differs from when accumulate is true above
- pSrc += 8;
- pDst += 2;
- numFrames--;
- }
- }
-}
-
-/*----------------------------------------------------------------------------
- * Downmix_foldGeneric()
- *----------------------------------------------------------------------------
- * Purpose:
- * downmix to stereo a multichannel signal of arbitrary channel position mask.
- *
- * Inputs:
- * mask the channel mask of pSrc
- * pSrc multichannel audio buffer to downmix
- * numFrames the number of multichannel frames to downmix
- * accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
- * or overwrite pDst (when false)
- *
- * Outputs:
- * pDst downmixed stereo audio samples
- *
- * Returns: false if multichannel format is not supported
- *
- *----------------------------------------------------------------------------
- */
-bool Downmix_foldGeneric(
- uint32_t mask, float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
-
- if (!Downmix_validChannelMask(mask)) {
- return false;
- }
- const int numChan = audio_channel_count_from_out_mask(mask);
-
- // compute at what index each channel is: samples will be in the following order:
- // FL FR FC LFE BL BR BC SL SR
- //
- // (transfer matrix)
- // FL FR FC LFE BL BR BC SL SR
- // 0.5 0.353 0.353 0.5 0.353 0.5
- // 0.5 0.353 0.353 0.5 0.353 0.5
-
- // derive the indices for the transfer matrix columns that have non-zero values.
- int indexFL = -1;
- int indexFR = -1;
- int indexFC = -1;
- int indexLFE = -1;
- int indexBL = -1;
- int indexBR = -1;
- int indexBC = -1;
- int indexSL = -1;
- int indexSR = -1;
- int index = 0;
- for (unsigned tmp = mask;
- (tmp & (AUDIO_CHANNEL_OUT_7POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER)) != 0;
- ++index) {
- const unsigned lowestBit = tmp & -(signed)tmp;
- switch (lowestBit) {
- case AUDIO_CHANNEL_OUT_FRONT_LEFT:
- indexFL = index;
- break;
- case AUDIO_CHANNEL_OUT_FRONT_RIGHT:
- indexFR = index;
- break;
- case AUDIO_CHANNEL_OUT_FRONT_CENTER:
- indexFC = index;
- break;
- case AUDIO_CHANNEL_OUT_LOW_FREQUENCY:
- indexLFE = index;
- break;
- case AUDIO_CHANNEL_OUT_BACK_LEFT:
- indexBL = index;
- break;
- case AUDIO_CHANNEL_OUT_BACK_RIGHT:
- indexBR = index;
- break;
- case AUDIO_CHANNEL_OUT_BACK_CENTER:
- indexBC = index;
- break;
- case AUDIO_CHANNEL_OUT_SIDE_LEFT:
- indexSL = index;
- break;
- case AUDIO_CHANNEL_OUT_SIDE_RIGHT:
- indexSR = index;
- break;
- }
- tmp ^= lowestBit;
- }
-
- // With good branch prediction, this should run reasonably fast.
- // Also consider using a transfer matrix form.
- while (numFrames) {
- // compute contribution of FC, BC and LFE
- float centersLfeContrib = 0;
- if (indexFC >= 0) centersLfeContrib = pSrc[indexFC];
- if (indexLFE >= 0) centersLfeContrib += pSrc[indexLFE];
- if (indexBC >= 0) centersLfeContrib += pSrc[indexBC];
- centersLfeContrib *= MINUS_3_DB_IN_FLOAT;
-
- float ch[2];
- ch[0] = centersLfeContrib;
- ch[1] = centersLfeContrib;
-
- // mix in left / right channels
- if (indexFL >= 0) ch[0] += pSrc[indexFL];
- if (indexFR >= 0) ch[1] += pSrc[indexFR];
-
- if (indexSL >= 0) ch[0] += pSrc[indexSL];
- if (indexSR >= 0) ch[1] += pSrc[indexSR]; // note pair checks enforce this if indexSL != 0
-
- if (indexBL >= 0) ch[0] += pSrc[indexBL];
- if (indexBR >= 0) ch[1] += pSrc[indexBR]; // note pair checks enforce this if indexBL != 0
-
- // scale to prevent overflow.
- ch[0] *= 0.5f;
- ch[1] *= 0.5f;
-
- if (accumulate) {
- ch[0] += pDst[0];
- ch[1] += pDst[1];
- }
-
- pDst[0] = clamp_float(ch[0]);
- pDst[1] = clamp_float(ch[1]);
- pSrc += numChan;
- pDst += 2;
- numFrames--;
- }
- return true;
-}
diff --git a/media/libeffects/downmix/benchmark/downmix_benchmark.cpp b/media/libeffects/downmix/benchmark/downmix_benchmark.cpp
index ee169c2..d9d40ed 100644
--- a/media/libeffects/downmix/benchmark/downmix_benchmark.cpp
+++ b/media/libeffects/downmix/benchmark/downmix_benchmark.cpp
@@ -35,16 +35,14 @@
AUDIO_CHANNEL_OUT_STEREO,
AUDIO_CHANNEL_OUT_2POINT1,
AUDIO_CHANNEL_OUT_2POINT0POINT2,
- AUDIO_CHANNEL_OUT_QUAD,
- AUDIO_CHANNEL_OUT_QUAD_BACK,
+ AUDIO_CHANNEL_OUT_QUAD, // AUDIO_CHANNEL_OUT_QUAD_BACK
AUDIO_CHANNEL_OUT_QUAD_SIDE,
AUDIO_CHANNEL_OUT_SURROUND,
AUDIO_CHANNEL_OUT_2POINT1POINT2,
AUDIO_CHANNEL_OUT_3POINT0POINT2,
AUDIO_CHANNEL_OUT_PENTA,
AUDIO_CHANNEL_OUT_3POINT1POINT2,
- AUDIO_CHANNEL_OUT_5POINT1,
- AUDIO_CHANNEL_OUT_5POINT1_BACK,
+ AUDIO_CHANNEL_OUT_5POINT1, // AUDIO_CHANNEL_OUT_5POINT1_BACK
AUDIO_CHANNEL_OUT_5POINT1_SIDE,
AUDIO_CHANNEL_OUT_6POINT1,
AUDIO_CHANNEL_OUT_5POINT1POINT2,
@@ -62,58 +60,34 @@
static constexpr size_t kFrameCount = 1000;
/*
-Pixel 3XL
-downmix_benchmark:
- #BM_Downmix/0 4723 ns 4708 ns 148694
- #BM_Downmix/1 4717 ns 4702 ns 148873
- #BM_Downmix/2 4803 ns 4788 ns 145893
- #BM_Downmix/3 5056 ns 5041 ns 139110
- #BM_Downmix/4 4710 ns 4696 ns 149625
- #BM_Downmix/5 1514 ns 1509 ns 463694
- #BM_Downmix/6 1513 ns 1509 ns 463451
- #BM_Downmix/7 1516 ns 1511 ns 463899
- #BM_Downmix/8 4445 ns 4431 ns 157831
- #BM_Downmix/9 5081 ns 5065 ns 138412
- #BM_Downmix/10 4354 ns 4341 ns 161247
- #BM_Downmix/11 4411 ns 4397 ns 158893
- #BM_Downmix/12 4434 ns 4420 ns 157992
- #BM_Downmix/13 4845 ns 4830 ns 144873
- #BM_Downmix/14 4851 ns 4835 ns 144954
- #BM_Downmix/15 4884 ns 4870 ns 144233
- #BM_Downmix/16 5832 ns 5813 ns 120565
- #BM_Downmix/17 5241 ns 5224 ns 133927
- #BM_Downmix/18 5044 ns 5028 ns 139131
- #BM_Downmix/19 5244 ns 5227 ns 132315
- #BM_Downmix/20 5943 ns 5923 ns 117759
- #BM_Downmix/21 5990 ns 5971 ns 117263
- #BM_Downmix/22 4468 ns 4454 ns 156689
- #BM_Downmix/23 7306 ns 7286 ns 95911
---
-downmix_benchmark: (generic fold)
- #BM_Downmix/0 4722 ns 4707 ns 149847
- #BM_Downmix/1 4714 ns 4698 ns 148748
- #BM_Downmix/2 4794 ns 4779 ns 145661
- #BM_Downmix/3 5053 ns 5035 ns 139172
- #BM_Downmix/4 4695 ns 4678 ns 149762
- #BM_Downmix/5 4381 ns 4368 ns 159675
- #BM_Downmix/6 4387 ns 4373 ns 160267
- #BM_Downmix/7 4732 ns 4717 ns 148514
- #BM_Downmix/8 4430 ns 4415 ns 158133
- #BM_Downmix/9 5101 ns 5084 ns 138353
- #BM_Downmix/10 4356 ns 4343 ns 160821
- #BM_Downmix/11 4397 ns 4383 ns 159995
- #BM_Downmix/12 4438 ns 4424 ns 158117
- #BM_Downmix/13 5243 ns 5226 ns 133863
- #BM_Downmix/14 5259 ns 5242 ns 131855
- #BM_Downmix/15 5245 ns 5228 ns 133686
- #BM_Downmix/16 5829 ns 5809 ns 120543
- #BM_Downmix/17 5245 ns 5228 ns 133533
- #BM_Downmix/18 5935 ns 5916 ns 118282
- #BM_Downmix/19 5263 ns 5245 ns 133657
- #BM_Downmix/20 5998 ns 5978 ns 114693
- #BM_Downmix/21 5989 ns 5969 ns 117450
- #BM_Downmix/22 4442 ns 4431 ns 157913
- #BM_Downmix/23 7309 ns 7290 ns 95797
+Pixel 4XL
+$ adb shell /data/benchmarktest/downmix_benchmark/vendor/downmix_benchmark
+
+--------------------------------------------------------
+Benchmark Time CPU Iterations
+--------------------------------------------------------
+BM_Downmix/0 3638 ns 3624 ns 197517 AUDIO_CHANNEL_OUT_MONO
+BM_Downmix/1 4040 ns 4024 ns 178766
+BM_Downmix/2 4759 ns 4740 ns 134741 AUDIO_CHANNEL_OUT_STEREO
+BM_Downmix/3 6042 ns 6017 ns 129546 AUDIO_CHANNEL_OUT_2POINT1
+BM_Downmix/4 6897 ns 6868 ns 96316 AUDIO_CHANNEL_OUT_2POINT0POINT2
+BM_Downmix/5 2117 ns 2109 ns 331705 AUDIO_CHANNEL_OUT_QUAD
+BM_Downmix/6 2097 ns 2088 ns 335421 AUDIO_CHANNEL_OUT_QUAD_SIDE
+BM_Downmix/7 7291 ns 7263 ns 96256 AUDIO_CHANNEL_OUT_SURROUND
+BM_Downmix/8 8246 ns 8206 ns 84318 AUDIO_CHANNEL_OUT_2POINT1POINT2
+BM_Downmix/9 8341 ns 8303 ns 84298 AUDIO_CHANNEL_OUT_3POINT0POINT2
+BM_Downmix/10 7549 ns 7517 ns 84293 AUDIO_CHANNEL_OUT_PENTA
+BM_Downmix/11 9395 ns 9354 ns 75209 AUDIO_CHANNEL_OUT_3POINT1POINT2
+BM_Downmix/12 3267 ns 3253 ns 215596 AUDIO_CHANNEL_OUT_5POINT1
+BM_Downmix/13 3178 ns 3163 ns 220132 AUDIO_CHANNEL_OUT_5POINT1_SIDE
+BM_Downmix/14 10245 ns 10199 ns 67486 AUDIO_CHANNEL_OUT_6POINT1
+BM_Downmix/15 10975 ns 10929 ns 61359 AUDIO_CHANNEL_OUT_5POINT1POINT2
+BM_Downmix/16 3796 ns 3780 ns 184728 AUDIO_CHANNEL_OUT_7POINT1
+BM_Downmix/17 13562 ns 13503 ns 51823 AUDIO_CHANNEL_OUT_5POINT1POINT4
+BM_Downmix/18 13573 ns 13516 ns 51800 AUDIO_CHANNEL_OUT_7POINT1POINT2
+BM_Downmix/19 15502 ns 15435 ns 47147 AUDIO_CHANNEL_OUT_7POINT1POINT4
+BM_Downmix/20 16693 ns 16624 ns 42109 AUDIO_CHANNEL_OUT_13POINT_360RA
+BM_Downmix/21 28267 ns 28116 ns 24982 AUDIO_CHANNEL_OUT_22POINT2
*/
static void BM_Downmix(benchmark::State& state) {
@@ -125,7 +99,7 @@
std::minstd_rand gen(channelMask);
std::uniform_real_distribution<> dis(-1.0f, 1.0f);
std::vector<float> input(kFrameCount * channelCount);
- std::vector<float> output(kFrameCount * 2);
+ std::vector<float> output(kFrameCount * FCC_2);
for (auto& in : input) {
in = dis(gen);
}
@@ -187,7 +161,8 @@
benchmark::ClobberMemory();
}
- state.SetComplexityN(state.range(0));
+ state.SetComplexityN(channelCount);
+ state.SetLabel(audio_channel_out_mask_to_string(channelMask));
if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle); status != 0) {
ALOGE("release_effect returned an error = %d\n", status);
diff --git a/media/libeffects/downmix/tests/downmix_tests.cpp b/media/libeffects/downmix/tests/downmix_tests.cpp
index d4b7a3a..20e19a3 100644
--- a/media/libeffects/downmix/tests/downmix_tests.cpp
+++ b/media/libeffects/downmix/tests/downmix_tests.cpp
@@ -33,16 +33,14 @@
AUDIO_CHANNEL_OUT_STEREO,
AUDIO_CHANNEL_OUT_2POINT1,
AUDIO_CHANNEL_OUT_2POINT0POINT2,
- AUDIO_CHANNEL_OUT_QUAD,
- AUDIO_CHANNEL_OUT_QUAD_BACK,
+ AUDIO_CHANNEL_OUT_QUAD, // AUDIO_CHANNEL_OUT_QUAD_BACK
AUDIO_CHANNEL_OUT_QUAD_SIDE,
AUDIO_CHANNEL_OUT_SURROUND,
AUDIO_CHANNEL_OUT_2POINT1POINT2,
AUDIO_CHANNEL_OUT_3POINT0POINT2,
AUDIO_CHANNEL_OUT_PENTA,
AUDIO_CHANNEL_OUT_3POINT1POINT2,
- AUDIO_CHANNEL_OUT_5POINT1,
- AUDIO_CHANNEL_OUT_5POINT1_BACK,
+ AUDIO_CHANNEL_OUT_5POINT1, // AUDIO_CHANNEL_OUT_5POINT1_BACK
AUDIO_CHANNEL_OUT_5POINT1_SIDE,
AUDIO_CHANNEL_OUT_6POINT1,
AUDIO_CHANNEL_OUT_5POINT1POINT2,
@@ -52,10 +50,72 @@
AUDIO_CHANNEL_OUT_7POINT1POINT4,
AUDIO_CHANNEL_OUT_13POINT_360RA,
AUDIO_CHANNEL_OUT_22POINT2,
+ audio_channel_mask_t(AUDIO_CHANNEL_OUT_22POINT2
+ | AUDIO_CHANNEL_OUT_FRONT_WIDE_LEFT | AUDIO_CHANNEL_OUT_FRONT_WIDE_RIGHT),
};
-static constexpr audio_channel_mask_t kConsideredChannels =
- (audio_channel_mask_t)(AUDIO_CHANNEL_OUT_7POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER);
+constexpr float COEF_25 = 0.2508909536f;
+constexpr float COEF_35 = 0.3543928915f;
+constexpr float COEF_36 = 0.3552343859f;
+constexpr float COEF_61 = 0.6057043428f;
+
+constexpr inline float kScaleFromChannelIdxLeft[] = {
+ 1.f, // AUDIO_CHANNEL_OUT_FRONT_LEFT = 0x1u,
+ 0.f, // AUDIO_CHANNEL_OUT_FRONT_RIGHT = 0x2u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_CENTER = 0x4u,
+ 0.5f, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 0x8u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_BACK_LEFT = 0x10u,
+ 0.f, // AUDIO_CHANNEL_OUT_BACK_RIGHT = 0x20u,
+ COEF_61, // AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40u,
+ COEF_25, // AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80u,
+ 0.5f, // AUDIO_CHANNEL_OUT_BACK_CENTER = 0x100u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_SIDE_LEFT = 0x200u,
+ 0.f, // AUDIO_CHANNEL_OUT_SIDE_RIGHT = 0x400u,
+ COEF_36, // AUDIO_CHANNEL_OUT_TOP_CENTER = 0x800u,
+ 1.f, // AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 0x1000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 0x2000u,
+ 0.f, // AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 0x4000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 0x8000u,
+ COEF_35, // AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 0x10000u,
+ 0.f, // AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 0x20000u,
+ COEF_61, // AUDIO_CHANNEL_OUT_TOP_SIDE_LEFT = 0x40000u,
+ 0.f, // AUDIO_CHANNEL_OUT_TOP_SIDE_RIGHT = 0x80000u,
+ 1.f, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_LEFT = 0x100000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_CENTER = 0x200000u,
+ 0.f, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_RIGHT = 0x400000u,
+ 0.f, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2 = 0x800000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_WIDE_LEFT = 0x1000000u,
+ 0.f, // AUDIO_CHANNEL_OUT_FRONT_WIDE_RIGHT = 0x2000000u,
+};
+
+constexpr inline float kScaleFromChannelIdxRight[] = {
+ 0.f, // AUDIO_CHANNEL_OUT_FRONT_LEFT = 0x1u,
+ 1.f, // AUDIO_CHANNEL_OUT_FRONT_RIGHT = 0x2u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_CENTER = 0x4u,
+ 0.5f, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 0x8u,
+ 0.f, // AUDIO_CHANNEL_OUT_BACK_LEFT = 0x10u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_BACK_RIGHT = 0x20u,
+ COEF_25, // AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40u,
+ COEF_61, // AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80u,
+ 0.5f, // AUDIO_CHANNEL_OUT_BACK_CENTER = 0x100u,
+ 0.f, // AUDIO_CHANNEL_OUT_SIDE_LEFT = 0x200u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_SIDE_RIGHT = 0x400u,
+ COEF_36, // AUDIO_CHANNEL_OUT_TOP_CENTER = 0x800u,
+ 0.f, // AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 0x1000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 0x2000u,
+ 1.f, // AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 0x4000u,
+ 0.f, // AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 0x8000u,
+ COEF_35, // AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 0x10000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 0x20000u,
+ 0.f, // AUDIO_CHANNEL_OUT_TOP_SIDE_LEFT = 0x40000u,
+ COEF_61, // AUDIO_CHANNEL_OUT_TOP_SIDE_RIGHT = 0x80000u,
+ 0.f, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_LEFT = 0x100000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_CENTER = 0x200000u,
+ 1.f, // AUDIO_CHANNEL_OUT_BOTTOM_FRONT_RIGHT = 0x400000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2 = 0x800000u,
+ 0.f, // AUDIO_CHANNEL_OUT_FRONT_WIDE_LEFT = 0x1000000u,
+ M_SQRT1_2, // AUDIO_CHANNEL_OUT_FRONT_WIDE_RIGHT = 0x2000000u,
+};
// Downmix doesn't change with sample rate
static constexpr size_t kSampleRates[] = {
@@ -93,8 +153,8 @@
void testBalance(int sampleRate, audio_channel_mask_t channelMask) {
using namespace ::android::audio_utils::channels;
- size_t frames = 100;
- unsigned outChannels = 2;
+ size_t frames = 100; // set to an even number (2, 4, 6 ... ) stream alternates +1, -1.
+ constexpr unsigned outChannels = 2;
unsigned inChannels = audio_channel_count_from_out_mask(channelMask);
std::vector<float> input(frames * inChannels);
std::vector<float> output(frames * outChannels);
@@ -102,7 +162,7 @@
double savedPower[32][2]{};
for (unsigned i = 0, channel = channelMask; channel != 0; ++i) {
const int index = __builtin_ctz(channel);
- ASSERT_LT(index, FCC_24);
+ ASSERT_LT(index, FCC_26);
const int pairIndex = pairIdxFromChannelIdx(index);
const AUDIO_GEOMETRY_SIDE side = sideFromChannelIdx(index);
const int channelBit = 1 << index;
@@ -119,7 +179,7 @@
auto stats = channelStatistics(output, 2 /* channels */);
// printf("power: %s %s\n", stats[0].toString().c_str(), stats[1].toString().c_str());
- double power[2] = { stats[0].getVariance(), stats[1].getVariance() };
+ double power[2] = { stats[0].getPopVariance(), stats[1].getPopVariance() };
// Check symmetric power for pair channels on exchange of left/right position.
// to do this, we save previous power measurements.
@@ -130,28 +190,39 @@
savedPower[index][0] = power[0];
savedPower[index][1] = power[1];
- // Confirm exactly the mix amount prescribed by the existing downmix effect.
- // For future changes to the downmix effect, the nearness needs to be relaxed
- // to compare behavior S or earlier.
- if ((channelBit & kConsideredChannels) == 0) {
- // for channels not considered, expect 0 power for legacy downmix
- EXPECT_EQ(0.f, power[0]);
- EXPECT_EQ(0.f, power[1]);
- continue;
- }
- constexpr float POWER_TOLERANCE = 0.01; // for variance sum error.
+ constexpr float POWER_TOLERANCE = 0.001;
+ const float expectedPower =
+ kScaleFromChannelIdxLeft[index] * kScaleFromChannelIdxLeft[index]
+ + kScaleFromChannelIdxRight[index] * kScaleFromChannelIdxRight[index];
+ EXPECT_NEAR(expectedPower, power[0] + power[1], POWER_TOLERANCE);
switch (side) {
case AUDIO_GEOMETRY_SIDE_LEFT:
- EXPECT_NEAR(0.25f, power[0], POWER_TOLERANCE);
+ if (channelBit == AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) {
+ break;
+ }
EXPECT_EQ(0.f, power[1]);
break;
case AUDIO_GEOMETRY_SIDE_RIGHT:
+ if (channelBit == AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) {
+ break;
+ }
EXPECT_EQ(0.f, power[0]);
- EXPECT_NEAR(0.25f, power[1], POWER_TOLERANCE);
break;
case AUDIO_GEOMETRY_SIDE_CENTER:
- EXPECT_NEAR(0.125f, power[0], POWER_TOLERANCE);
- EXPECT_NEAR(0.125f, power[1], POWER_TOLERANCE);
+ if (channelBit == AUDIO_CHANNEL_OUT_LOW_FREQUENCY) {
+ if (channelMask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) {
+ EXPECT_EQ(0.f, power[1]);
+ break;
+ } else {
+ EXPECT_NEAR_EPSILON(power[0], power[1]); // always true
+ EXPECT_NEAR(expectedPower, power[0] + power[1], POWER_TOLERANCE);
+ break;
+ }
+ } else if (channelBit == AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) {
+ EXPECT_EQ(0.f, power[0]);
+ EXPECT_NEAR(expectedPower, power[1], POWER_TOLERANCE);
+ break;
+ }
EXPECT_NEAR_EPSILON(power[0], power[1]);
break;
}
@@ -178,6 +249,7 @@
handle_, EFFECT_CMD_SET_CONFIG,
sizeof(effect_config_t), &config_, &replySize, &reply);
ASSERT_EQ(0, err);
+ ASSERT_EQ(0, reply);
err = (downmixApi->command)(
handle_, EFFECT_CMD_ENABLE,
0, nullptr, &replySize, &reply);
@@ -188,6 +260,27 @@
ASSERT_EQ(0, err);
}
+ // This test assumes the channel mask is invalid.
+ void testInvalidChannelMask(audio_channel_mask_t invalidChannelMask) {
+ reconfig(48000 /* sampleRate */, invalidChannelMask);
+ const int32_t sessionId = 0;
+ const int32_t ioId = 0;
+ int32_t err = AUDIO_EFFECT_LIBRARY_INFO_SYM.create_effect(
+ &downmix_uuid_, sessionId, ioId, &handle_);
+ ASSERT_EQ(0, err);
+
+ const struct effect_interface_s * const downmixApi = *handle_;
+ int32_t reply = 0;
+ uint32_t replySize = (uint32_t)sizeof(reply);
+ err = (downmixApi->command)(
+ handle_, EFFECT_CMD_SET_CONFIG,
+ sizeof(effect_config_t), &config_, &replySize, &reply);
+ ASSERT_EQ(0, err);
+ ASSERT_NE(0, reply); // error has occurred.
+ err = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(handle_);
+ ASSERT_EQ(0, err);
+ }
+
private:
void reconfig(int sampleRate, audio_channel_mask_t channelMask) {
config_.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
@@ -234,6 +327,16 @@
int inputChannelCount_{};
};
+TEST(DownmixTestSimple, invalidChannelMask) {
+ // Fill in a dummy test method to use DownmixTest outside of a parameterized test.
+ class DownmixTestComplete : public DownmixTest {
+ void TestBody() override {}
+ } downmixtest;
+
+ constexpr auto INVALID_CHANNEL_MASK = audio_channel_mask_t(1 << 31);
+ downmixtest.testInvalidChannelMask(INVALID_CHANNEL_MASK);
+}
+
TEST_P(DownmixTest, basic) {
testBalance(kSampleRates[std::get<0>(GetParam())],
kChannelPositionMasks[std::get<1>(GetParam())]);
@@ -244,10 +347,11 @@
::testing::Combine(
::testing::Range(0, (int)std::size(kSampleRates)),
::testing::Range(0, (int)std::size(kChannelPositionMasks))
- ));
-
-int main(int argc, /* const */ char** argv) {
- ::testing::InitGoogleTest(&argc, argv);
- int status = RUN_ALL_TESTS();
- return status;
-}
+ ),
+ [](const testing::TestParamInfo<DownmixTest::ParamType>& info) {
+ const int index = std::get<1>(info.param);
+ const audio_channel_mask_t channelMask = kChannelPositionMasks[index];
+ const std::string name = std::string(audio_channel_out_mask_to_string(channelMask))
+ + "_" + std::to_string(std::get<0>(info.param)) + "_" + std::to_string(index);
+ return name;
+ });
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.cpp b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.cpp
index 7e5caed..ccef5ab 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.cpp
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.cpp
@@ -135,7 +135,6 @@
LVM_UINT32 fs =
(LVM_UINT32)LVEQNB_SampleRateTab[(LVM_UINT16)pParams->SampleRate]; /* Sample rate */
LVM_UINT32 fc; /* Filter centre frequency */
- LVM_INT16 QFactor; /* Filter Q factor */
pInstance->NBands = pParams->NBands;
@@ -144,7 +143,6 @@
* Get the filter settings
*/
fc = (LVM_UINT32)pParams->pBandDefinition[i].Frequency; /* Get the band centre frequency */
- QFactor = (LVM_INT16)pParams->pBandDefinition[i].QFactor; /* Get the band Q factor */
pInstance->pBiquadType[i] = LVEQNB_SinglePrecision_Float; /* Default to single precision */
@@ -313,9 +311,9 @@
*/
pInstance->eqBiquad.resize(pParams->NBands,
android::audio_utils::BiquadFilter<LVM_FLOAT>(pParams->NrChannels));
- LVEQNB_ClearFilterHistory(pInstance);
if (bChange || modeChange) {
+ LVEQNB_ClearFilterHistory(pInstance);
/*
* If the sample rate has changed clear the history
*/
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.cpp b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.cpp
index 8e63502..ffed6d4 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.cpp
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.cpp
@@ -421,7 +421,6 @@
* Intermediate variables and temporary values
*/
LVM_FLOAT T0;
- LVM_FLOAT D;
LVM_FLOAT A0;
LVM_FLOAT B1;
LVM_FLOAT B2;
@@ -444,9 +443,6 @@
* Calculating the intermediate values
*/
T0 = Frequency * LVPSA_Float_TwoPiOnFsTable[Fs]; /* T0 = 2 * Pi * Fc / Fs */
- D = 3200; /* Floating point value 1.000000 (1*100*2^5) */
- /* Force D = 1 : the function was originally used for a peaking filter.
- The D parameter do not exist for a BandPass filter coefficients */
/*
* Calculate the B2 coefficient
@@ -535,7 +531,6 @@
* Intermediate variables and temporary values
*/
LVM_FLOAT T0;
- LVM_FLOAT D;
LVM_FLOAT A0;
LVM_FLOAT B1;
LVM_FLOAT B2;
@@ -558,9 +553,6 @@
* Calculating the intermediate values
*/
T0 = Frequency * LVPSA_Float_TwoPiOnFsTable[Fs]; /* T0 = 2 * Pi * Fc / Fs */
- D = 3200; /* Floating point value 1.000000 (1*100*2^5) */
- /* Force D = 1 : the function was originally used for a peaking filter.
- The D parameter do not exist for a BandPass filter coefficients */
/*
* Calculate the B2 coefficient
diff --git a/media/libeffects/lvm/tests/EffectBundleTest.cpp b/media/libeffects/lvm/tests/EffectBundleTest.cpp
index 881ffb1..018cb7c 100644
--- a/media/libeffects/lvm/tests/EffectBundleTest.cpp
+++ b/media/libeffects/lvm/tests/EffectBundleTest.cpp
@@ -14,29 +14,39 @@
* limitations under the License.
*/
+#include <system/audio_effects/effect_bassboost.h>
+#include <system/audio_effects/effect_equalizer.h>
+#include <system/audio_effects/effect_virtualizer.h>
#include "EffectTestHelper.h"
-using namespace android;
-// Update isBassBoost, if the order of effects is updated
-constexpr effect_uuid_t kEffectUuids[] = {
+using namespace android;
+typedef enum {
+ EFFECT_BASS_BOOST,
+ EFFECT_EQUALIZER,
+ EFFECT_VIRTUALIZER,
+ EFFECT_VOLUME
+} effect_type_t;
+
+const std::map<effect_type_t, effect_uuid_t> kEffectUuids = {
// NXP SW BassBoost
- {0x8631f300, 0x72e2, 0x11df, 0xb57e, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
- // NXP SW Virtualizer
- {0x1d4033c0, 0x8557, 0x11df, 0x9f2d, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
+ {EFFECT_BASS_BOOST,
+ {0x8631f300, 0x72e2, 0x11df, 0xb57e, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}},
// NXP SW Equalizer
- {0xce772f20, 0x847d, 0x11df, 0xbb17, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
+ {EFFECT_EQUALIZER,
+ {0xce772f20, 0x847d, 0x11df, 0xbb17, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}},
+ // NXP SW Virtualizer
+ {EFFECT_VIRTUALIZER,
+ {0x1d4033c0, 0x8557, 0x11df, 0x9f2d, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}},
// NXP SW Volume
- {0x119341a0, 0x8469, 0x11df, 0x81f9, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
+ {EFFECT_VOLUME, {0x119341a0, 0x8469, 0x11df, 0x81f9, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}},
};
-static bool isBassBoost(const effect_uuid_t* uuid) {
- // Update this, if the order of effects in kEffectUuids is updated
- return uuid == &kEffectUuids[0];
-}
+const size_t kNumEffectUuids = std::size(kEffectUuids);
-constexpr size_t kNumEffectUuids = std::size(kEffectUuids);
+constexpr float kMinAmplitude = -1.0f;
+constexpr float kMaxAmplitude = 1.0f;
-typedef std::tuple<int, int, int, int, int> SingleEffectTestParam;
+using SingleEffectTestParam = std::tuple<int, int, int, int, int>;
class SingleEffectTest : public ::testing::TestWithParam<SingleEffectTestParam> {
public:
SingleEffectTest()
@@ -46,7 +56,8 @@
mFrameCount(EffectTestHelper::kFrameCounts[std::get<2>(GetParam())]),
mLoopCount(EffectTestHelper::kLoopCounts[std::get<3>(GetParam())]),
mTotalFrameCount(mFrameCount * mLoopCount),
- mUuid(&kEffectUuids[std::get<4>(GetParam())]) {}
+ mEffectType((effect_type_t)std::get<4>(GetParam())),
+ mUuid(kEffectUuids.at(mEffectType)) {}
const size_t mChMask;
const size_t mChannelCount;
@@ -54,7 +65,8 @@
const size_t mFrameCount;
const size_t mLoopCount;
const size_t mTotalFrameCount;
- const effect_uuid_t* mUuid;
+ const effect_type_t mEffectType;
+ const effect_uuid_t mUuid;
};
// Tests applying a single effect
@@ -63,7 +75,7 @@
<< "chMask: " << mChMask << " sampleRate: " << mSampleRate
<< " frameCount: " << mFrameCount << " loopCount: " << mLoopCount);
- EffectTestHelper effect(mUuid, mChMask, mChMask, mSampleRate, mFrameCount, mLoopCount);
+ EffectTestHelper effect(&mUuid, mChMask, mChMask, mSampleRate, mFrameCount, mLoopCount);
ASSERT_NO_FATAL_FAILURE(effect.createEffect());
ASSERT_NO_FATAL_FAILURE(effect.setConfig());
@@ -72,7 +84,7 @@
std::vector<float> input(mTotalFrameCount * mChannelCount);
std::vector<float> output(mTotalFrameCount * mChannelCount);
std::minstd_rand gen(mChMask);
- std::uniform_real_distribution<> dis(-1.0f, 1.0f);
+ std::uniform_real_distribution<> dis(kMinAmplitude, kMaxAmplitude);
for (auto& in : input) {
in = dis(gen);
}
@@ -88,7 +100,7 @@
::testing::Range(0, (int)EffectTestHelper::kNumLoopCounts),
::testing::Range(0, (int)kNumEffectUuids)));
-typedef std::tuple<int, int, int, int> SingleEffectComparisonTestParam;
+using SingleEffectComparisonTestParam = std::tuple<int, int, int, int>;
class SingleEffectComparisonTest
: public ::testing::TestWithParam<SingleEffectComparisonTestParam> {
public:
@@ -97,13 +109,15 @@
mFrameCount(EffectTestHelper::kFrameCounts[std::get<1>(GetParam())]),
mLoopCount(EffectTestHelper::kLoopCounts[std::get<2>(GetParam())]),
mTotalFrameCount(mFrameCount * mLoopCount),
- mUuid(&kEffectUuids[std::get<3>(GetParam())]) {}
+ mEffectType((effect_type_t)std::get<3>(GetParam())),
+ mUuid(kEffectUuids.at(mEffectType)) {}
const size_t mSampleRate;
const size_t mFrameCount;
const size_t mLoopCount;
const size_t mTotalFrameCount;
- const effect_uuid_t* mUuid;
+ const effect_type_t mEffectType;
+ const effect_uuid_t mUuid;
};
// Compares first two channels in multi-channel output to stereo output when same effect is applied
@@ -115,7 +129,7 @@
std::vector<float> monoInput(mTotalFrameCount);
std::minstd_rand gen(mSampleRate);
- std::uniform_real_distribution<> dis(-1.0f, 1.0f);
+ std::uniform_real_distribution<> dis(kMinAmplitude, kMaxAmplitude);
for (auto& in : monoInput) {
in = dis(gen);
}
@@ -126,7 +140,7 @@
mTotalFrameCount * sizeof(float) * FCC_1);
// Apply effect on stereo channels
- EffectTestHelper stereoEffect(mUuid, AUDIO_CHANNEL_OUT_STEREO, AUDIO_CHANNEL_OUT_STEREO,
+ EffectTestHelper stereoEffect(&mUuid, AUDIO_CHANNEL_OUT_STEREO, AUDIO_CHANNEL_OUT_STEREO,
mSampleRate, mFrameCount, mLoopCount);
ASSERT_NO_FATAL_FAILURE(stereoEffect.createEffect());
@@ -142,7 +156,7 @@
for (size_t chMask : EffectTestHelper::kChMasks) {
size_t channelCount = audio_channel_count_from_out_mask(chMask);
- EffectTestHelper testEffect(mUuid, chMask, chMask, mSampleRate, mFrameCount, mLoopCount);
+ EffectTestHelper testEffect(&mUuid, chMask, chMask, mSampleRate, mFrameCount, mLoopCount);
ASSERT_NO_FATAL_FAILURE(testEffect.createEffect());
ASSERT_NO_FATAL_FAILURE(testEffect.setConfig());
@@ -170,7 +184,7 @@
memcpy_to_i16_from_float(stereoTestI16.data(), stereoTestOutput.data(),
mTotalFrameCount * FCC_2);
- if (isBassBoost(mUuid)) {
+ if (EFFECT_BASS_BOOST == mEffectType) {
// SNR must be above the threshold
float snr = computeSnr<int16_t>(stereoRefI16.data(), stereoTestI16.data(),
mTotalFrameCount * FCC_2);
@@ -191,6 +205,135 @@
::testing::Range(0, (int)EffectTestHelper::kNumLoopCounts),
::testing::Range(0, (int)kNumEffectUuids)));
+using SingleEffectDefaultSetParamTestParam = std::tuple<int, int, int>;
+class SingleEffectDefaultSetParamTest
+ : public ::testing::TestWithParam<SingleEffectDefaultSetParamTestParam> {
+ public:
+ SingleEffectDefaultSetParamTest()
+ : mChMask(EffectTestHelper::kChMasks[std::get<0>(GetParam())]),
+ mChannelCount(audio_channel_count_from_out_mask(mChMask)),
+ mSampleRate(16000),
+ mFrameCount(EffectTestHelper::kFrameCounts[std::get<1>(GetParam())]),
+ mLoopCount(1),
+ mTotalFrameCount(mFrameCount * mLoopCount),
+ mEffectType((effect_type_t)std::get<2>(GetParam())),
+ mUuid(kEffectUuids.at(mEffectType)) {}
+
+ const size_t mChMask;
+ const size_t mChannelCount;
+ const size_t mSampleRate;
+ const size_t mFrameCount;
+ const size_t mLoopCount;
+ const size_t mTotalFrameCount;
+ const effect_type_t mEffectType;
+ const effect_uuid_t mUuid;
+};
+
+// Tests verifying that redundant setParam calls do not alter output
+TEST_P(SingleEffectDefaultSetParamTest, SimpleProcess) {
+ SCOPED_TRACE(testing::Message()
+ << "chMask: " << mChMask << " sampleRate: " << mSampleRate
+ << " frameCount: " << mFrameCount << " loopCount: " << mLoopCount);
+ // effect.process() handles mTotalFrameCount * mChannelCount samples in each call.
+ // This test calls process() twice per effect, hence total samples when allocating
+ // input and output vectors is twice the number of samples processed in one call.
+ size_t totalNumSamples = 2 * mTotalFrameCount * mChannelCount;
+ // Initialize input buffer with deterministic pseudo-random values
+ std::vector<float> input(totalNumSamples);
+ std::minstd_rand gen(mChMask);
+ std::uniform_real_distribution<> dis(kMinAmplitude, kMaxAmplitude);
+ for (auto& in : input) {
+ in = dis(gen);
+ }
+
+ uint32_t key;
+ int32_t value1, value2;
+ switch (mEffectType) {
+ case EFFECT_BASS_BOOST:
+ key = BASSBOOST_PARAM_STRENGTH;
+ value1 = 1;
+ value2 = 14;
+ break;
+ case EFFECT_VIRTUALIZER:
+ key = VIRTUALIZER_PARAM_STRENGTH;
+ value1 = 0;
+ value2 = 100;
+ break;
+ case EFFECT_EQUALIZER:
+ key = EQ_PARAM_CUR_PRESET;
+ value1 = 0;
+ value2 = 1;
+ break;
+ case EFFECT_VOLUME:
+ key = 0 /* VOLUME_PARAM_LEVEL */;
+ value1 = 0;
+ value2 = -100;
+ break;
+ default:
+ FAIL() << "Unsupported effect type : " << mEffectType;
+ }
+
+ EffectTestHelper refEffect(&mUuid, mChMask, mChMask, mSampleRate, mFrameCount, mLoopCount);
+
+ ASSERT_NO_FATAL_FAILURE(refEffect.createEffect());
+ ASSERT_NO_FATAL_FAILURE(refEffect.setConfig());
+
+ if (EFFECT_BASS_BOOST == mEffectType) {
+ ASSERT_NO_FATAL_FAILURE(refEffect.setParam<int16_t>(key, value1));
+ } else {
+ ASSERT_NO_FATAL_FAILURE(refEffect.setParam<int32_t>(key, value1));
+ }
+ std::vector<float> refOutput(totalNumSamples);
+ float* pInput = input.data();
+ float* pOutput = refOutput.data();
+ ASSERT_NO_FATAL_FAILURE(refEffect.process(pInput, pOutput));
+
+ pInput += totalNumSamples / 2;
+ pOutput += totalNumSamples / 2;
+ ASSERT_NO_FATAL_FAILURE(refEffect.process(pInput, pOutput));
+ ASSERT_NO_FATAL_FAILURE(refEffect.releaseEffect());
+
+ EffectTestHelper testEffect(&mUuid, mChMask, mChMask, mSampleRate, mFrameCount, mLoopCount);
+
+ ASSERT_NO_FATAL_FAILURE(testEffect.createEffect());
+ ASSERT_NO_FATAL_FAILURE(testEffect.setConfig());
+
+ if (EFFECT_BASS_BOOST == mEffectType) {
+ ASSERT_NO_FATAL_FAILURE(testEffect.setParam<int16_t>(key, value1));
+ } else {
+ ASSERT_NO_FATAL_FAILURE(testEffect.setParam<int32_t>(key, value1));
+ }
+
+ std::vector<float> testOutput(totalNumSamples);
+ pInput = input.data();
+ pOutput = testOutput.data();
+ ASSERT_NO_FATAL_FAILURE(testEffect.process(pInput, pOutput));
+
+ // Call setParam once to change the parameters, and then call setParam again
+ // to restore the parameters to the initial state, making the first setParam
+ // call redundant
+ if (EFFECT_BASS_BOOST == mEffectType) {
+ ASSERT_NO_FATAL_FAILURE(testEffect.setParam<int16_t>(key, value2));
+ ASSERT_NO_FATAL_FAILURE(testEffect.setParam<int16_t>(key, value1));
+ } else {
+ ASSERT_NO_FATAL_FAILURE(testEffect.setParam<int32_t>(key, value2));
+ ASSERT_NO_FATAL_FAILURE(testEffect.setParam<int32_t>(key, value1));
+ }
+
+ pInput += totalNumSamples / 2;
+ pOutput += totalNumSamples / 2;
+ ASSERT_NO_FATAL_FAILURE(testEffect.process(pInput, pOutput));
+ ASSERT_NO_FATAL_FAILURE(testEffect.releaseEffect());
+ ASSERT_TRUE(areNearlySame(refOutput.data(), testOutput.data(), totalNumSamples))
+ << "Outputs do not match with default setParam calls";
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ EffectBundleTestAll, SingleEffectDefaultSetParamTest,
+ ::testing::Combine(::testing::Range(0, (int)EffectTestHelper::kNumChMasks),
+ ::testing::Range(0, (int)EffectTestHelper::kNumFrameCounts),
+ ::testing::Range(0, (int)kNumEffectUuids)));
+
int main(int argc, char** argv) {
::testing::InitGoogleTest(&argc, argv);
int status = RUN_ALL_TESTS();
diff --git a/media/libeffects/lvm/tests/EffectTestHelper.cpp b/media/libeffects/lvm/tests/EffectTestHelper.cpp
index 625c15a..ec727c7 100644
--- a/media/libeffects/lvm/tests/EffectTestHelper.cpp
+++ b/media/libeffects/lvm/tests/EffectTestHelper.cpp
@@ -50,23 +50,6 @@
ASSERT_EQ(reply, 0) << "cmd_enable reply non zero " << reply;
}
-void EffectTestHelper::setParam(uint32_t type, uint32_t value) {
- int reply = 0;
- uint32_t replySize = sizeof(reply);
- uint32_t paramData[2] = {type, value};
- auto effectParam = new effect_param_t[sizeof(effect_param_t) + sizeof(paramData)];
- memcpy(&effectParam->data[0], ¶mData[0], sizeof(paramData));
- effectParam->psize = sizeof(paramData[0]);
- effectParam->vsize = sizeof(paramData[1]);
- int status = (*mEffectHandle)
- ->command(mEffectHandle, EFFECT_CMD_SET_PARAM,
- sizeof(effect_param_t) + sizeof(paramData), effectParam,
- &replySize, &reply);
- delete[] effectParam;
- ASSERT_EQ(status, 0) << "set_param returned an error " << status;
- ASSERT_EQ(reply, 0) << "set_param reply non zero " << reply;
-}
-
void EffectTestHelper::process(float* input, float* output) {
audio_buffer_t inBuffer = {.frameCount = mFrameCount, .f32 = input};
audio_buffer_t outBuffer = {.frameCount = mFrameCount, .f32 = output};
diff --git a/media/libeffects/lvm/tests/EffectTestHelper.h b/media/libeffects/lvm/tests/EffectTestHelper.h
index 3854d46..bcee84e 100644
--- a/media/libeffects/lvm/tests/EffectTestHelper.h
+++ b/media/libeffects/lvm/tests/EffectTestHelper.h
@@ -50,6 +50,23 @@
return snr;
}
+template <typename T>
+static float areNearlySame(const T* ref, const T* tst, size_t count) {
+ T delta;
+ if constexpr (std::is_floating_point_v<T>) {
+ delta = std::numeric_limits<T>::epsilon();
+ } else {
+ delta = 1;
+ }
+ for (size_t i = 0; i < count; ++i) {
+ const double diff(tst[i] - ref[i]);
+ if (abs(diff) > delta) {
+ return false;
+ }
+ }
+ return true;
+}
+
class EffectTestHelper {
public:
EffectTestHelper(const effect_uuid_t* uuid, size_t inChMask, size_t outChMask,
@@ -65,7 +82,25 @@
void createEffect();
void releaseEffect();
void setConfig();
- void setParam(uint32_t type, uint32_t val);
+ template <typename VALUE_DTYPE>
+ void setParam(uint32_t type, VALUE_DTYPE const value) {
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+
+ uint8_t paramData[sizeof(effect_param_t) + sizeof(type) + sizeof(value)];
+ auto effectParam = (effect_param_t*)paramData;
+
+ memcpy(&effectParam->data[0], &type, sizeof(type));
+ memcpy(&effectParam->data[sizeof(type)], &value, sizeof(value));
+ effectParam->psize = sizeof(type);
+ effectParam->vsize = sizeof(value);
+ int status = (*mEffectHandle)
+ ->command(mEffectHandle, EFFECT_CMD_SET_PARAM,
+ sizeof(effect_param_t) + sizeof(type) + sizeof(value),
+ effectParam, &replySize, &reply);
+ ASSERT_EQ(status, 0) << "set_param returned an error " << status;
+ ASSERT_EQ(reply, 0) << "set_param reply non zero " << reply;
+ };
void process(float* input, float* output);
// Corresponds to SNR for 1 bit difference between two int16_t signals
diff --git a/media/libeffects/preprocessing/.clang-format b/media/libeffects/preprocessing/.clang-format
deleted file mode 120000
index f1b4f69..0000000
--- a/media/libeffects/preprocessing/.clang-format
+++ /dev/null
@@ -1 +0,0 @@
-../../../../../build/soong/scripts/system-clang-format
\ No newline at end of file
diff --git a/media/libeffects/preprocessing/tests/correlation.cpp b/media/libeffects/preprocessing/tests/correlation.cpp
index eb56fc3..0853673 100644
--- a/media/libeffects/preprocessing/tests/correlation.cpp
+++ b/media/libeffects/preprocessing/tests/correlation.cpp
@@ -36,7 +36,7 @@
const int16_t* sigY, int len,
int16_t enableCrossCorr) {
float maxCorrVal = 0.f, prevCorrVal = 0.f;
- int delay = 0, peakIndex = 0, flag = 0;
+ int peakIndex = 0, flag = 0;
int loopLim = (1 == enableCrossCorr) ? len : kMinLoopLimitValue;
std::vector<int> peakIndexVect(kNumPeaks, 0);
std::vector<float> peakValueVect(kNumPeaks, 0.f);
@@ -47,7 +47,6 @@
}
corrVal /= len - i;
if (corrVal > maxCorrVal) {
- delay = i;
maxCorrVal = corrVal;
}
// Correlation peaks are expected to be observed at equal intervals. The interval length is
diff --git a/media/libheadtracking/Android.bp b/media/libheadtracking/Android.bp
index 6141824..63b769e 100644
--- a/media/libheadtracking/Android.bp
+++ b/media/libheadtracking/Android.bp
@@ -1,3 +1,12 @@
+package {
+ // See: http://go/android-license-faq
+ // A large-scale-change added 'default_applicable_licenses' to import
+ // all of the 'license_kinds' from "frameworks_av_license"
+ // to get the below license kinds:
+ // SPDX-license-identifier-Apache-2.0
+ default_applicable_licenses: ["frameworks_av_license"],
+}
+
cc_library {
name: "libheadtracking",
host_supported: true,
@@ -22,6 +31,35 @@
],
}
+cc_library {
+ name: "libheadtracking-binding",
+ srcs: [
+ "SensorPoseProvider.cpp",
+ ],
+ shared_libs: [
+ "libheadtracking",
+ "liblog",
+ "libsensor",
+ "libutils",
+ ],
+ export_shared_lib_headers: [
+ "libheadtracking",
+ ],
+}
+
+cc_binary {
+ name: "SensorPoseProvider-example",
+ srcs: [
+ "SensorPoseProvider-example.cpp",
+ ],
+ shared_libs: [
+ "libheadtracking",
+ "libheadtracking-binding",
+ "libsensor",
+ "libutils",
+ ],
+}
+
cc_test_host {
name: "libheadtracking-test",
srcs: [
diff --git a/media/libheadtracking/HeadTrackingProcessor-test.cpp b/media/libheadtracking/HeadTrackingProcessor-test.cpp
index ae5567d..299192f 100644
--- a/media/libheadtracking/HeadTrackingProcessor-test.cpp
+++ b/media/libheadtracking/HeadTrackingProcessor-test.cpp
@@ -48,10 +48,15 @@
std::unique_ptr<HeadTrackingProcessor> processor =
createHeadTrackingProcessor(Options{}, HeadTrackingMode::SCREEN_RELATIVE);
+
+ // Establish a baseline for the drift compensators.
+ processor->setWorldToHeadPose(0, Pose3f(), Twist3f());
+ processor->setWorldToScreenPose(0, Pose3f());
+
+ processor->setDisplayOrientation(physicalToLogical);
processor->setWorldToHeadPose(0, worldToHead, Twist3f());
processor->setWorldToScreenPose(0, worldToScreen);
processor->setScreenToStagePose(screenToStage);
- processor->setDisplayOrientation(physicalToLogical);
processor->calculate(0);
ASSERT_EQ(processor->getActualMode(), HeadTrackingMode::SCREEN_RELATIVE);
EXPECT_EQ(processor->getHeadToStagePose(), worldToHead.inverse() * worldToScreen *
@@ -76,6 +81,11 @@
std::unique_ptr<HeadTrackingProcessor> processor = createHeadTrackingProcessor(
Options{.predictionDuration = 2.f}, HeadTrackingMode::WORLD_RELATIVE);
+
+ // Establish a baseline for the drift compensators.
+ processor->setWorldToHeadPose(0, Pose3f(), Twist3f());
+ processor->setWorldToScreenPose(0, Pose3f());
+
processor->setWorldToHeadPose(0, worldToHead, headTwist);
processor->setWorldToScreenPose(0, worldToScreen);
processor->calculate(0);
@@ -100,6 +110,10 @@
std::unique_ptr<HeadTrackingProcessor> processor = createHeadTrackingProcessor(
Options{.maxTranslationalVelocity = 1}, HeadTrackingMode::STATIC);
+ // Establish a baseline for the drift compensators.
+ processor->setWorldToHeadPose(0, Pose3f(), Twist3f());
+ processor->setWorldToScreenPose(0, Pose3f());
+
processor->calculate(0);
processor->setDesiredMode(HeadTrackingMode::WORLD_RELATIVE);
diff --git a/media/libheadtracking/HeadTrackingProcessor.cpp b/media/libheadtracking/HeadTrackingProcessor.cpp
index ee60fa5..47f7cf0 100644
--- a/media/libheadtracking/HeadTrackingProcessor.cpp
+++ b/media/libheadtracking/HeadTrackingProcessor.cpp
@@ -57,7 +57,14 @@
}
void setWorldToScreenPose(int64_t timestamp, const Pose3f& worldToScreen) override {
- mScreenPoseDriftCompensator.setInput(timestamp, worldToScreen);
+ if (mPhysicalToLogicalAngle != mPendingPhysicalToLogicalAngle) {
+ // We're introducing an artificial discontinuity. Enable the rate limiter.
+ mRateLimiter.enable();
+ mPhysicalToLogicalAngle = mPendingPhysicalToLogicalAngle;
+ }
+
+ mScreenPoseDriftCompensator.setInput(
+ timestamp, worldToScreen * Pose3f(rotateY(-mPhysicalToLogicalAngle)));
mWorldToScreenTimestamp = timestamp;
}
@@ -66,10 +73,7 @@
}
void setDisplayOrientation(float physicalToLogicalAngle) override {
- if (mPhysicalToLogicalAngle != physicalToLogicalAngle) {
- mRateLimiter.enable();
- }
- mPhysicalToLogicalAngle = physicalToLogicalAngle;
+ mPendingPhysicalToLogicalAngle = physicalToLogicalAngle;
}
void calculate(int64_t timestamp) override {
@@ -80,8 +84,7 @@
}
if (mWorldToScreenTimestamp.has_value()) {
- const Pose3f worldToLogicalScreen = mScreenPoseDriftCompensator.getOutput() *
- Pose3f(rotateY(-mPhysicalToLogicalAngle));
+ const Pose3f worldToLogicalScreen = mScreenPoseDriftCompensator.getOutput();
mScreenHeadFusion.setWorldToScreenPose(mWorldToScreenTimestamp.value(),
worldToLogicalScreen);
}
@@ -108,15 +111,30 @@
HeadTrackingMode getActualMode() const override { return mModeSelector.getActualMode(); }
- void recenter() override {
- mHeadPoseDriftCompensator.recenter();
- mScreenPoseDriftCompensator.recenter();
- mRateLimiter.enable();
+ void recenter(bool recenterHead, bool recenterScreen) override {
+ if (recenterHead) {
+ mHeadPoseDriftCompensator.recenter();
+ }
+ if (recenterScreen) {
+ mScreenPoseDriftCompensator.recenter();
+ }
+
+ // If a sensor being recentered is included in the current mode, apply rate limiting to
+ // avoid discontinuities.
+ HeadTrackingMode mode = mModeSelector.getActualMode();
+ if ((recenterHead && (mode == HeadTrackingMode::WORLD_RELATIVE ||
+ mode == HeadTrackingMode::SCREEN_RELATIVE)) ||
+ (recenterScreen && mode == HeadTrackingMode::SCREEN_RELATIVE)) {
+ mRateLimiter.enable();
+ }
}
private:
const Options mOptions;
float mPhysicalToLogicalAngle = 0;
+ // We store the physical to logical angle as "pending" until the next world-to-screen sample it
+ // applies to arrives.
+ float mPendingPhysicalToLogicalAngle = 0;
std::optional<int64_t> mWorldToHeadTimestamp;
std::optional<int64_t> mWorldToScreenTimestamp;
Pose3f mHeadToStagePose;
diff --git a/media/libheadtracking/Pose-test.cpp b/media/libheadtracking/Pose-test.cpp
index 3ff6a9b..a9e18ce 100644
--- a/media/libheadtracking/Pose-test.cpp
+++ b/media/libheadtracking/Pose-test.cpp
@@ -110,6 +110,29 @@
EXPECT_TRUE(std::get<1>(result));
}
+TEST(Pose, FloatVectorRoundTrip1) {
+ // Rotation vector magnitude must be less than Pi.
+ std::vector<float> vec = { 1, 2, 3, 0.4, 0.5, 0.6};
+ std::optional<Pose3f> pose = Pose3f::fromVector(vec);
+ ASSERT_TRUE(pose.has_value());
+ std::vector<float> reconstructed = pose->toVector();
+ EXPECT_EQ(vec, reconstructed);
+}
+
+TEST(Pose, FloatVectorRoundTrip2) {
+ Pose3f pose({1, 2, 3}, Quaternionf::UnitRandom());
+ std::vector<float> vec = pose.toVector();
+ std::optional<Pose3f> reconstructed = Pose3f::fromVector(vec);
+ ASSERT_TRUE(reconstructed.has_value());
+ EXPECT_EQ(pose, reconstructed.value());
+}
+
+TEST(Pose, FloatVectorInvalid) {
+ EXPECT_FALSE(Pose3f::fromVector({}).has_value());
+ EXPECT_FALSE(Pose3f::fromVector({1, 2, 3, 4, 5}).has_value());
+ EXPECT_FALSE(Pose3f::fromVector({1, 2, 3, 4, 5, 6, 7}).has_value());
+}
+
} // namespace
} // namespace media
} // namespace android
diff --git a/media/libheadtracking/Pose.cpp b/media/libheadtracking/Pose.cpp
index 9eeb2b1..47241ce 100644
--- a/media/libheadtracking/Pose.cpp
+++ b/media/libheadtracking/Pose.cpp
@@ -16,10 +16,25 @@
#include "media/Pose.h"
#include "media/Twist.h"
+#include "QuaternionUtil.h"
namespace android {
namespace media {
+using Eigen::Vector3f;
+
+std::optional<Pose3f> Pose3f::fromVector(const std::vector<float>& vec) {
+ if (vec.size() != 6) {
+ return std::nullopt;
+ }
+ return Pose3f({vec[0], vec[1], vec[2]}, rotationVectorToQuaternion({vec[3], vec[4], vec[5]}));
+}
+
+std::vector<float> Pose3f::toVector() const {
+ Eigen::Vector3f rot = quaternionToRotationVector(mRotation);
+ return {mTranslation[0], mTranslation[1], mTranslation[2], rot[0], rot[1], rot[2]};
+}
+
std::tuple<Pose3f, bool> moveWithRateLimit(const Pose3f& from, const Pose3f& to, float t,
float maxTranslationalVelocity,
float maxRotationalVelocity) {
diff --git a/media/libheadtracking/PoseDriftCompensator-test.cpp b/media/libheadtracking/PoseDriftCompensator-test.cpp
index 74f4bee..df0a05f 100644
--- a/media/libheadtracking/PoseDriftCompensator-test.cpp
+++ b/media/libheadtracking/PoseDriftCompensator-test.cpp
@@ -39,17 +39,22 @@
Pose3f pose2({4, 5, 6}, Quaternionf::UnitRandom());
PoseDriftCompensator comp(Options{});
+ // First pose sets the baseline.
comp.setInput(1000, pose1);
- EXPECT_EQ(comp.getOutput(), pose1);
+ EXPECT_EQ(comp.getOutput(), Pose3f());
comp.setInput(2000, pose2);
- EXPECT_EQ(comp.getOutput(), pose2);
+ EXPECT_EQ(comp.getOutput(), pose1.inverse() * pose2);
+ // Recentering resets the baseline.
comp.recenter();
EXPECT_EQ(comp.getOutput(), Pose3f());
comp.setInput(3000, pose1);
- EXPECT_EQ(comp.getOutput(), pose2.inverse() * pose1);
+ EXPECT_EQ(comp.getOutput(), Pose3f());
+
+ comp.setInput(4000, pose2);
+ EXPECT_EQ(comp.getOutput(), pose1.inverse() * pose2);
}
TEST(PoseDriftCompensator, NoDriftZeroTime) {
@@ -58,16 +63,19 @@
PoseDriftCompensator comp(Options{});
comp.setInput(1000, pose1);
- EXPECT_EQ(comp.getOutput(), pose1);
+ EXPECT_EQ(comp.getOutput(), Pose3f());
comp.setInput(1000, pose2);
- EXPECT_EQ(comp.getOutput(), pose2);
+ EXPECT_EQ(comp.getOutput(), pose1.inverse() * pose2);
comp.recenter();
EXPECT_EQ(comp.getOutput(), Pose3f());
comp.setInput(1000, pose1);
- EXPECT_EQ(comp.getOutput(), pose2.inverse() * pose1);
+ EXPECT_EQ(comp.getOutput(), Pose3f());
+
+ comp.setInput(1000, pose2);
+ EXPECT_EQ(comp.getOutput(), pose1.inverse() * pose2);
}
TEST(PoseDriftCompensator, Asymptotic) {
@@ -92,16 +100,19 @@
Options{.translationalDriftTimeConstant = 1e7, .rotationalDriftTimeConstant = 1e7});
comp.setInput(0, pose1);
- EXPECT_EQ(comp.getOutput(), pose1);
+ EXPECT_EQ(comp.getOutput(), Pose3f());
comp.setInput(1, pose2);
- EXPECT_EQ(comp.getOutput(), pose2);
+ EXPECT_EQ(comp.getOutput(), pose1.inverse() * pose2);
comp.recenter();
EXPECT_EQ(comp.getOutput(), Pose3f());
comp.setInput(2, pose1);
- EXPECT_EQ(comp.getOutput(), pose2.inverse() * pose1);
+ EXPECT_EQ(comp.getOutput(), Pose3f());
+
+ comp.setInput(3, pose2);
+ EXPECT_EQ(comp.getOutput(), pose1.inverse() * pose2);
}
TEST(PoseDriftCompensator, Drift) {
@@ -109,6 +120,9 @@
PoseDriftCompensator comp(
Options{.translationalDriftTimeConstant = 500, .rotationalDriftTimeConstant = 1000});
+ // Establish a baseline.
+ comp.setInput(1000, Pose3f());
+
// Initial pose is used as is.
comp.setInput(1000, pose1);
EXPECT_EQ(comp.getOutput(), pose1);
diff --git a/media/libheadtracking/PoseDriftCompensator.cpp b/media/libheadtracking/PoseDriftCompensator.cpp
index 9dfe172..0e90cad 100644
--- a/media/libheadtracking/PoseDriftCompensator.cpp
+++ b/media/libheadtracking/PoseDriftCompensator.cpp
@@ -29,10 +29,8 @@
PoseDriftCompensator::PoseDriftCompensator(const Options& options) : mOptions(options) {}
void PoseDriftCompensator::setInput(int64_t timestamp, const Pose3f& input) {
- if (!mTimestamp.has_value()) {
- // First input sample sets the output directly.
- mOutput = input;
- } else {
+ if (mTimestamp.has_value()) {
+ // Avoid computation upon first input (only sets the initial state).
Pose3f prevInputToInput = mPrevInput.inverse() * input;
mOutput = scale(mOutput, timestamp - mTimestamp.value()) * prevInputToInput;
}
@@ -41,6 +39,7 @@
}
void PoseDriftCompensator::recenter() {
+ mTimestamp.reset();
mOutput = Pose3f();
}
diff --git a/media/libheadtracking/PoseProcessingGraph.png b/media/libheadtracking/PoseProcessingGraph.png
index 8e6dfd2..0363068 100644
--- a/media/libheadtracking/PoseProcessingGraph.png
+++ b/media/libheadtracking/PoseProcessingGraph.png
Binary files differ
diff --git a/media/libheadtracking/SensorPoseProvider-example.cpp b/media/libheadtracking/SensorPoseProvider-example.cpp
new file mode 100644
index 0000000..a246e8b
--- /dev/null
+++ b/media/libheadtracking/SensorPoseProvider-example.cpp
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <unistd.h>
+#include <iostream>
+
+#include <android/sensor.h>
+#include <hardware/sensors.h>
+#include <utils/SystemClock.h>
+
+#include <media/SensorPoseProvider.h>
+#include <sensor/Sensor.h>
+#include <sensor/SensorManager.h>
+
+using android::elapsedRealtimeNano;
+using android::Sensor;
+using android::SensorManager;
+using android::String16;
+using android::media::Pose3f;
+using android::media::SensorPoseProvider;
+using android::media::Twist3f;
+
+using namespace std::chrono_literals;
+
+const char kPackageName[] = "SensorPoseProvider-example";
+
+class Listener : public SensorPoseProvider::Listener {
+ public:
+ void onPose(int64_t timestamp, int32_t handle, const Pose3f& pose,
+ const std::optional<Twist3f>& twist) override {
+ int64_t now = elapsedRealtimeNano();
+
+ std::cout << "onPose t=" << timestamp
+ << " lag=" << ((now - timestamp) / 1e6) << "[ms]"
+ << " sensor=" << handle
+ << " pose=" << pose
+ << " twist=";
+ if (twist.has_value()) {
+ std::cout << twist.value();
+ } else {
+ std::cout << "<none>";
+ }
+ std::cout << std::endl;
+ }
+};
+
+int main() {
+ SensorManager& sensorManager = SensorManager::getInstanceForPackage(String16(kPackageName));
+
+ const Sensor* headSensor = sensorManager.getDefaultSensor(SENSOR_TYPE_GAME_ROTATION_VECTOR);
+ const Sensor* screenSensor = sensorManager.getDefaultSensor(SENSOR_TYPE_ROTATION_VECTOR);
+
+ Listener listener;
+
+ std::unique_ptr<SensorPoseProvider> provider =
+ SensorPoseProvider::create(kPackageName, &listener);
+ int32_t headHandle = provider->startSensor(headSensor->getHandle(), 500ms);
+ sleep(2);
+ provider->startSensor(screenSensor->getHandle(), 500ms);
+ sleep(2);
+ provider->stopSensor(headHandle);
+ sleep(2);
+ return 0;
+}
diff --git a/media/libheadtracking/SensorPoseProvider.cpp b/media/libheadtracking/SensorPoseProvider.cpp
new file mode 100644
index 0000000..c4c031d
--- /dev/null
+++ b/media/libheadtracking/SensorPoseProvider.cpp
@@ -0,0 +1,250 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/SensorPoseProvider.h>
+
+#define LOG_TAG "SensorPoseProvider"
+
+#include <inttypes.h>
+
+#include <future>
+#include <map>
+#include <thread>
+
+#include <log/log_main.h>
+#include <sensor/Sensor.h>
+#include <sensor/SensorEventQueue.h>
+#include <sensor/SensorManager.h>
+#include <utils/Looper.h>
+
+#include "QuaternionUtil.h"
+
+namespace android {
+namespace media {
+namespace {
+
+// Identifier to use for our event queue on the loop.
+// The number 19 is arbitrary, only useful if using multiple objects on the same looper.
+constexpr int kIdent = 19;
+
+static inline Looper* ALooper_to_Looper(ALooper* alooper) {
+ return reinterpret_cast<Looper*>(alooper);
+}
+
+static inline ALooper* Looper_to_ALooper(Looper* looper) {
+ return reinterpret_cast<ALooper*>(looper);
+}
+
+/**
+ * RAII-wrapper around SensorEventQueue, which unregisters it on destruction.
+ */
+class EventQueueGuard {
+ public:
+ EventQueueGuard(const sp<SensorEventQueue>& queue, Looper* looper) : mQueue(queue) {
+ mQueue->looper = Looper_to_ALooper(looper);
+ mQueue->requestAdditionalInfo = false;
+ looper->addFd(mQueue->getFd(), kIdent, ALOOPER_EVENT_INPUT, nullptr, nullptr);
+ }
+
+ ~EventQueueGuard() {
+ if (mQueue) {
+ ALooper_to_Looper(mQueue->looper)->removeFd(mQueue->getFd());
+ }
+ }
+
+ EventQueueGuard(const EventQueueGuard&) = delete;
+ EventQueueGuard& operator=(const EventQueueGuard&) = delete;
+
+ [[nodiscard]] SensorEventQueue* get() const { return mQueue.get(); }
+
+ private:
+ sp<SensorEventQueue> mQueue;
+};
+
+/**
+ * RAII-wrapper around an enabled sensor, which disables it upon destruction.
+ */
+class SensorEnableGuard {
+ public:
+ SensorEnableGuard(const sp<SensorEventQueue>& queue, int32_t sensor)
+ : mQueue(queue), mSensor(sensor) {}
+
+ ~SensorEnableGuard() {
+ if (mSensor != SensorPoseProvider::INVALID_HANDLE) {
+ int ret = mQueue->disableSensor(mSensor);
+ if (ret) {
+ ALOGE("Failed to disable sensor: %s\n", strerror(ret));
+ }
+ }
+ }
+
+ SensorEnableGuard(const SensorEnableGuard&) = delete;
+ SensorEnableGuard& operator=(const SensorEnableGuard&) = delete;
+
+ // Enable moving.
+ SensorEnableGuard(SensorEnableGuard&& other) : mQueue(other.mQueue), mSensor(other.mSensor) {
+ other.mSensor = SensorPoseProvider::INVALID_HANDLE;
+ }
+
+ private:
+ sp<SensorEventQueue> const mQueue;
+ int32_t mSensor;
+};
+
+/**
+ * Streams the required events to a PoseListener, based on events originating from the Sensor stack.
+ */
+class SensorPoseProviderImpl : public SensorPoseProvider {
+ public:
+ static std::unique_ptr<SensorPoseProvider> create(const char* packageName, Listener* listener) {
+ std::unique_ptr<SensorPoseProviderImpl> result(
+ new SensorPoseProviderImpl(packageName, listener));
+ return result->waitInitFinished() ? std::move(result) : nullptr;
+ }
+
+ ~SensorPoseProviderImpl() override {
+ // Disable all active sensors.
+ mEnabledSensors.clear();
+ mLooper->wake();
+ mThread.join();
+ }
+
+ bool startSensor(int32_t sensor, std::chrono::microseconds samplingPeriod) override {
+ // Enable the sensor.
+ if (mQueue->enableSensor(sensor, samplingPeriod.count(), 0, 0)) {
+ ALOGE("Failed to enable sensor");
+ return false;
+ }
+
+ mEnabledSensors.emplace(sensor, SensorEnableGuard(mQueue.get(), sensor));
+ return true;
+ }
+
+ void stopSensor(int handle) override { mEnabledSensors.erase(handle); }
+
+ private:
+ sp<Looper> mLooper;
+ Listener* const mListener;
+
+ std::thread mThread;
+ std::map<int32_t, SensorEnableGuard> mEnabledSensors;
+ sp<SensorEventQueue> mQueue;
+
+ // We must do some of the initialization operations on the worker thread, because the API relies
+ // on the thread-local looper. In addition, as a matter of convenience, we store some of the
+ // state on the stack.
+ // For that reason, we use a two-step initialization approach, where the ctor mostly just starts
+ // the worker thread and that thread would notify, via the promise below whenever initialization
+ // is finished, and whether it was successful.
+ std::promise<bool> mInitPromise;
+
+ SensorPoseProviderImpl(const char* packageName, Listener* listener)
+ : mListener(listener),
+ mThread([this, p = std::string(packageName)] { threadFunc(p.c_str()); }) {}
+
+ void initFinished(bool success) { mInitPromise.set_value(success); }
+
+ bool waitInitFinished() { return mInitPromise.get_future().get(); }
+
+ void threadFunc(const char* packageName) {
+ // Obtain looper.
+ mLooper = Looper::prepare(ALOOPER_PREPARE_ALLOW_NON_CALLBACKS);
+
+ // Obtain sensor manager.
+ SensorManager& sensorManager = SensorManager::getInstanceForPackage(String16(packageName));
+
+ // Create event queue.
+ mQueue = sensorManager.createEventQueue();
+
+ if (mQueue == nullptr) {
+ ALOGE("Failed to create a sensor event queue");
+ initFinished(false);
+ return;
+ }
+
+ EventQueueGuard eventQueueGuard(mQueue, mLooper.get());
+
+ initFinished(true);
+
+ while (true) {
+ int ret = mLooper->pollOnce(-1 /* no timeout */, nullptr, nullptr, nullptr);
+
+ switch (ret) {
+ case ALOOPER_POLL_WAKE:
+ // Normal way to exit.
+ return;
+
+ case kIdent:
+ // Possible events on our queue.
+ break;
+
+ default:
+ ALOGE("Unexpected status out of Looper::pollOnce: %d", ret);
+ }
+
+ // Process an event.
+ ASensorEvent event;
+ ssize_t actual = mQueue->read(&event, 1);
+ if (actual > 0) {
+ mQueue->sendAck(&event, actual);
+ }
+ ssize_t size = mQueue->filterEvents(&event, actual);
+
+ if (size < 0 || size > 1) {
+ ALOGE("Unexpected return value from SensorEventQueue::filterEvents: %zd", size);
+ break;
+ }
+ if (size == 0) {
+ // No events.
+ continue;
+ }
+
+ handleEvent(event);
+ }
+ }
+
+ void handleEvent(const ASensorEvent& event) {
+ auto value = parseEvent(event);
+ mListener->onPose(event.timestamp, event.sensor, std::get<0>(value), std::get<1>(value));
+ }
+
+ static std::tuple<Pose3f, std::optional<Twist3f>> parseEvent(const ASensorEvent& event) {
+ // TODO(ytai): Add more types.
+ switch (event.type) {
+ case ASENSOR_TYPE_ROTATION_VECTOR:
+ case ASENSOR_TYPE_GAME_ROTATION_VECTOR: {
+ Eigen::Quaternionf quat(event.data[3], event.data[0], event.data[1], event.data[2]);
+ // Adapt to different frame convention.
+ quat *= rotateX(-M_PI_2);
+ return std::make_tuple(Pose3f(quat), std::optional<Twist3f>());
+ }
+
+ default:
+ ALOGE("Unsupported sensor type: %" PRId32, event.type);
+ return std::make_tuple(Pose3f(), std::optional<Twist3f>());
+ }
+ }
+};
+
+} // namespace
+
+std::unique_ptr<SensorPoseProvider> SensorPoseProvider::create(const char* packageName,
+ Listener* listener) {
+ return SensorPoseProviderImpl::create(packageName, listener);
+}
+
+} // namespace media
+} // namespace android
diff --git a/media/libheadtracking/include/media/HeadTrackingProcessor.h b/media/libheadtracking/include/media/HeadTrackingProcessor.h
index 23de540..9fea273 100644
--- a/media/libheadtracking/include/media/HeadTrackingProcessor.h
+++ b/media/libheadtracking/include/media/HeadTrackingProcessor.h
@@ -89,9 +89,9 @@
virtual HeadTrackingMode getActualMode() const = 0;
/**
- * This causes the current poses for both the head and screen to be considered "center".
+ * This causes the current poses for both the head and/or screen to be considered "center".
*/
- virtual void recenter() = 0;
+ virtual void recenter(bool recenterHead = true, bool recenterScreen = true) = 0;
};
/**
diff --git a/media/libheadtracking/include/media/Pose.h b/media/libheadtracking/include/media/Pose.h
index 06b33f3..e660bb9 100644
--- a/media/libheadtracking/include/media/Pose.h
+++ b/media/libheadtracking/include/media/Pose.h
@@ -15,6 +15,8 @@
*/
#pragma once
+#include <optional>
+#include <vector>
#include <Eigen/Geometry>
namespace android {
@@ -45,6 +47,22 @@
Pose3f(const Pose3f& other) { *this = other; }
+ /**
+ * Create instance from a vector-of-floats representation.
+ * The vector is expected to have exactly 6 elements, where the first three are a translation
+ * vector and the last three are a rotation vector.
+ *
+ * Returns nullopt if the input vector is illegal.
+ */
+ static std::optional<Pose3f> fromVector(const std::vector<float>& vec);
+
+ /**
+ * Convert instance to a vector-of-floats representation.
+ * The vector will have exactly 6 elements, where the first three are a translation vector and
+ * the last three are a rotation vector.
+ */
+ std::vector<float> toVector() const;
+
Pose3f& operator=(const Pose3f& other) {
mTranslation = other.mTranslation;
mRotation = other.mRotation;
diff --git a/media/libheadtracking/include/media/SensorPoseProvider.h b/media/libheadtracking/include/media/SensorPoseProvider.h
new file mode 100644
index 0000000..1a5deb0
--- /dev/null
+++ b/media/libheadtracking/include/media/SensorPoseProvider.h
@@ -0,0 +1,97 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#pragma once
+
+#include <chrono>
+#include <memory>
+#include <optional>
+
+#include <android/sensor.h>
+
+#include "Pose.h"
+#include "Twist.h"
+
+namespace android {
+namespace media {
+
+/**
+ * A utility providing streaming of pose data from motion sensors provided by the Sensor Framework.
+ *
+ * A live instance of this interface keeps around some resources required for accessing sensor
+ * readings (e.g. a thread and a queue). Those would be released when the instance is deleted.
+ *
+ * Once alive, individual sensors can be subscribed to using startSensor() and updates can be
+ * stopped via stopSensor(). Those two methods should not be called concurrently and correct usage
+ * is assumed.
+ */
+class SensorPoseProvider {
+ public:
+ static constexpr int32_t INVALID_HANDLE = ASENSOR_INVALID;
+
+ /**
+ * Interface for consuming pose-related sensor events.
+ *
+ * The listener will be provided with a stream of events, each including:
+ * - A handle of the sensor responsible for the event.
+ * - Timestamp.
+ * - Pose.
+ * - Optional twist (time-derivative of pose).
+ *
+ * Sensors having only orientation data will have the translation part of the pose set to
+ * identity.
+ *
+ * Events are delivered in a serialized manner (i.e. callbacks do not need to be reentrant).
+ * Callbacks should not block.
+ */
+ class Listener {
+ public:
+ virtual ~Listener() = default;
+
+ virtual void onPose(int64_t timestamp, int32_t handle, const Pose3f& pose,
+ const std::optional<Twist3f>& twist) = 0;
+ };
+
+ /**
+ * Creates a new SensorPoseProvider instance.
+ * Events will be delivered to the listener as long as the returned instance is kept alive.
+ * @param packageName Client's package name.
+ * @param listener The listener that will get the events.
+ * @return The new instance, or nullptr in case of failure.
+ */
+ static std::unique_ptr<SensorPoseProvider> create(const char* packageName, Listener* listener);
+
+ virtual ~SensorPoseProvider() = default;
+
+ /**
+ * Start receiving pose updates from a given sensor.
+ * Attempting to start a sensor that has already been started results in undefined behavior.
+ * @param sensor The sensor to subscribe to.
+ * @param samplingPeriod Sampling interval, in microseconds. Actual rate might be slightly
+ * different.
+ * @return true iff succeeded.
+ */
+ virtual bool startSensor(int32_t sensor, std::chrono::microseconds samplingPeriod) = 0;
+
+ /**
+ * Stop a sensor, previously started with startSensor(). It is not required to stop all sensors
+ * before deleting the SensorPoseProvider instance.
+ * @param handle The sensor handle, as provided to startSensor().
+ */
+ virtual void stopSensor(int32_t handle) = 0;
+};
+
+} // namespace media
+} // namespace android
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 9c1b563..4a2523f 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -347,6 +347,7 @@
shared_libs: [
"android.hidl.token@1.0-utils",
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"av-types-aidl-cpp",
"liblog",
diff --git a/media/libmedia/tests/codeclist/Android.bp b/media/libmedia/tests/codeclist/Android.bp
index 57af9a9..2ed3126 100644
--- a/media/libmedia/tests/codeclist/Android.bp
+++ b/media/libmedia/tests/codeclist/Android.bp
@@ -28,6 +28,22 @@
test_suites: ["device-tests", "mts"],
gtest: true,
+ // Support multilib variants (using different suffix per sub-architecture), which is needed on
+ // build targets with secondary architectures, as the MTS test suite packaging logic flattens
+ // all test artifacts into a single `testcases` directory.
+ compile_multilib: "both",
+ multilib: {
+ lib32: {
+ suffix: "32",
+ },
+ lib64: {
+ suffix: "64",
+ },
+ },
+
+ // used within mainline MTS, but only to R, not to Q.
+ min_sdk_version: "30",
+
srcs: [
"CodecListTest.cpp",
],
@@ -35,13 +51,13 @@
shared_libs: [
"libbinder",
"liblog",
- "libmedia_codeclist",
+ "libmedia_codeclist", // available >= R
"libstagefright",
"libstagefright_foundation",
"libstagefright_xmlparser",
"libutils",
],
- compile_multilib: "first",
+
cflags: [
"-Werror",
"-Wall",
diff --git a/media/libmedia/tests/codeclist/AndroidTest.xml b/media/libmedia/tests/codeclist/AndroidTest.xml
new file mode 100644
index 0000000..eeaab8e
--- /dev/null
+++ b/media/libmedia/tests/codeclist/AndroidTest.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for CodecList unit tests">
+ <option name="test-suite-tag" value="CodecListTest" />
+ <object type="module_controller" class="com.android.tradefed.testtype.suite.module.Sdk30ModuleController" />
+
+ <target_preparer class="com.android.compatibility.common.tradefed.targetprep.FilePusher">
+ <option name="cleanup" value="true" />
+ <option name="append-bitness" value="true" />
+ <option name="push" value="CodecListTest->/data/local/tmp/CodecListTest" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="CodecListTest" />
+ </test>
+
+
+</configuration>
diff --git a/media/libmedia/xsd/vts/OWNERS b/media/libmedia/xsd/vts/OWNERS
new file mode 100644
index 0000000..9af2eba
--- /dev/null
+++ b/media/libmedia/xsd/vts/OWNERS
@@ -0,0 +1,2 @@
+# Bug component: 151862
+sundongahn@google.com
diff --git a/media/libmediahelper/TypeConverter.cpp b/media/libmediahelper/TypeConverter.cpp
index d3a517f..97b5b95 100644
--- a/media/libmediahelper/TypeConverter.cpp
+++ b/media/libmediahelper/TypeConverter.cpp
@@ -50,6 +50,8 @@
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_MUTE_HAPTIC),
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_NO_SYSTEM_CAPTURE),
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_CAPTURE_PRIVATE),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_CONTENT_SPATIALIZED),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_NEVER_SPATIALIZE),
TERMINATOR
};
diff --git a/media/libmediametrics/IMediaMetricsService.cpp b/media/libmediametrics/IMediaMetricsService.cpp
deleted file mode 100644
index b5675e6..0000000
--- a/media/libmediametrics/IMediaMetricsService.cpp
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "MediaMetrics"
-
-#include <stdint.h>
-#include <inttypes.h>
-#include <sys/types.h>
-
-#include <binder/Parcel.h>
-#include <binder/IMemory.h>
-#include <binder/IPCThreadState.h>
-
-#include <utils/Errors.h> // for status_t
-#include <utils/List.h>
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include <media/MediaMetricsItem.h>
-#include <media/IMediaMetricsService.h>
-
-namespace android {
-
-// TODO: Currently ONE_WAY transactions, make both ONE_WAY and synchronous options.
-
-enum {
- SUBMIT_ITEM = IBinder::FIRST_CALL_TRANSACTION,
- SUBMIT_BUFFER,
-};
-
-class BpMediaMetricsService: public BpInterface<IMediaMetricsService>
-{
-public:
- explicit BpMediaMetricsService(const sp<IBinder>& impl)
- : BpInterface<IMediaMetricsService>(impl)
- {
- }
-
- status_t submit(mediametrics::Item *item) override
- {
- if (item == nullptr) {
- return BAD_VALUE;
- }
- ALOGV("%s: (ONEWAY) item=%s", __func__, item->toString().c_str());
-
- Parcel data;
- data.writeInterfaceToken(IMediaMetricsService::getInterfaceDescriptor());
-
- status_t status = item->writeToParcel(&data);
- if (status != NO_ERROR) { // assume failure logged in item
- return status;
- }
-
- status = remote()->transact(
- SUBMIT_ITEM, data, nullptr /* reply */, IBinder::FLAG_ONEWAY);
- ALOGW_IF(status != NO_ERROR, "%s: bad response from service for submit, status=%d",
- __func__, status);
- return status;
- }
-
- status_t submitBuffer(const char *buffer, size_t length) override
- {
- if (buffer == nullptr || length > INT32_MAX) {
- return BAD_VALUE;
- }
- ALOGV("%s: (ONEWAY) length:%zu", __func__, length);
-
- Parcel data;
- data.writeInterfaceToken(IMediaMetricsService::getInterfaceDescriptor());
-
- status_t status = data.writeInt32(length)
- ?: data.write((uint8_t*)buffer, length);
- if (status != NO_ERROR) {
- return status;
- }
-
- status = remote()->transact(
- SUBMIT_BUFFER, data, nullptr /* reply */, IBinder::FLAG_ONEWAY);
- ALOGW_IF(status != NO_ERROR, "%s: bad response from service for submit, status=%d",
- __func__, status);
- return status;
- }
-};
-
-IMPLEMENT_META_INTERFACE(MediaMetricsService, "android.media.IMediaMetricsService");
-
-// ----------------------------------------------------------------------
-
-status_t BnMediaMetricsService::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
- switch (code) {
- case SUBMIT_ITEM: {
- CHECK_INTERFACE(IMediaMetricsService, data, reply);
-
- mediametrics::Item * const item = mediametrics::Item::create();
- status_t status = item->readFromParcel(data);
- if (status != NO_ERROR) { // assume failure logged in item
- return status;
- }
- status = submitInternal(item, true /* release */);
- // assume failure logged by submitInternal
- return NO_ERROR;
- }
- case SUBMIT_BUFFER: {
- CHECK_INTERFACE(IMediaMetricsService, data, reply);
- int32_t length;
- status_t status = data.readInt32(&length);
- if (status != NO_ERROR || length <= 0) {
- return BAD_VALUE;
- }
- const void *ptr = data.readInplace(length);
- if (ptr == nullptr) {
- return BAD_VALUE;
- }
- status = submitBuffer(static_cast<const char *>(ptr), length);
- // assume failure logged by submitBuffer
- return NO_ERROR;
- }
-
- default:
- return BBinder::onTransact(code, data, reply, flags);
- }
-}
-
-// ----------------------------------------------------------------------------
-
-} // namespace android
diff --git a/media/libmediaplayerservice/Android.bp b/media/libmediaplayerservice/Android.bp
index f55678d..37f8103 100644
--- a/media/libmediaplayerservice/Android.bp
+++ b/media/libmediaplayerservice/Android.bp
@@ -65,6 +65,7 @@
"libstagefright_foundation",
"libstagefright_httplive",
"libutils",
+ "packagemanager_aidl-cpp",
],
header_libs: [
@@ -82,6 +83,7 @@
export_shared_lib_headers: [
"libmedia",
"framework-permission-aidl-cpp",
+ "packagemanager_aidl-cpp",
],
include_dirs: [
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index d278a01..f85887e 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -21,6 +21,7 @@
#define LOG_TAG "MediaPlayerService"
#include <utils/Log.h>
+#include <chrono>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/time.h>
@@ -2467,8 +2468,13 @@
void MediaPlayerService::AudioOutput::pause()
{
ALOGV("pause");
+ // We use pauseAndWait() instead of pause() to ensure tracks ramp to silence before
+ // any flush. We choose 40 ms timeout to allow 1 deep buffer mixer period
+ // to occur. Often waiting is 0 - 20 ms.
+ using namespace std::chrono_literals;
+ constexpr auto TIMEOUT_MS = 40ms;
Mutex::Autolock lock(mLock);
- if (mTrack != 0) mTrack->pause();
+ if (mTrack != 0) mTrack->pauseAndWait(TIMEOUT_MS);
}
void MediaPlayerService::AudioOutput::close()
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index a914006..4aa80be 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -126,8 +126,13 @@
}
if ((as == AUDIO_SOURCE_FM_TUNER
- && !(captureAudioOutputAllowed(mAttributionSource)
+ && !(captureAudioOutputAllowed(mAttributionSource)
|| captureTunerAudioInputAllowed(mAttributionSource)))
+ || (as == AUDIO_SOURCE_REMOTE_SUBMIX
+ && !(captureAudioOutputAllowed(mAttributionSource)
+ || modifyAudioRoutingAllowed(mAttributionSource)))
+ || (as == AUDIO_SOURCE_ECHO_REFERENCE
+ && !captureAudioOutputAllowed(mAttributionSource))
|| !recordingAllowed(mAttributionSource, (audio_source_t)as)) {
return PERMISSION_DENIED;
}
diff --git a/media/libmediaplayerservice/StagefrightMetadataRetriever.cpp b/media/libmediaplayerservice/StagefrightMetadataRetriever.cpp
index 2aabd53..a372b7f 100644
--- a/media/libmediaplayerservice/StagefrightMetadataRetriever.cpp
+++ b/media/libmediaplayerservice/StagefrightMetadataRetriever.cpp
@@ -225,10 +225,17 @@
"media.stagefright.thumbnail.prefer_hw_codecs", false);
uint32_t flags = preferhw ? 0 : MediaCodecList::kPreferSoftwareCodecs;
Vector<AString> matchingCodecs;
+ sp<AMessage> format = new AMessage;
+ status_t err = convertMetaDataToMessage(trackMeta, &format);
+ if (err != OK) {
+ format = NULL;
+ }
+
MediaCodecList::findMatchingCodecs(
mime,
false, /* encoder */
flags,
+ format,
&matchingCodecs);
for (size_t i = 0; i < matchingCodecs.size(); ++i) {
@@ -348,11 +355,18 @@
bool preferhw = property_get_bool(
"media.stagefright.thumbnail.prefer_hw_codecs", false);
uint32_t flags = preferhw ? 0 : MediaCodecList::kPreferSoftwareCodecs;
+ sp<AMessage> format = new AMessage;
+ status_t err = convertMetaDataToMessage(trackMeta, &format);
+ if (err != OK) {
+ format = NULL;
+ }
+
Vector<AString> matchingCodecs;
MediaCodecList::findMatchingCodecs(
mime,
false, /* encoder */
flags,
+ format,
&matchingCodecs);
for (size_t i = 0; i < matchingCodecs.size(); ++i) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 4a65f71..2828d44 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -1673,24 +1673,18 @@
mDrainAudioQueuePending = false;
- if (offloadingAudio()) {
- mAudioSink->pause();
- mAudioSink->flush();
- if (!mPaused) {
- mAudioSink->start();
- }
- } else {
- mAudioSink->pause();
- mAudioSink->flush();
+ mAudioSink->pause();
+ mAudioSink->flush();
+ if (!offloadingAudio()) {
// Call stop() to signal to the AudioSink to completely fill the
// internal buffer before resuming playback.
// FIXME: this is ignored after flush().
mAudioSink->stop();
- if (!mPaused) {
- mAudioSink->start();
- }
mNumFramesWritten = 0;
}
+ if (!mPaused) {
+ mAudioSink->start();
+ }
mNextAudioClockUpdateTimeUs = -1;
} else {
flushQueue(&mVideoQueue);
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 3ef89bb..a4fbbbc 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -5433,6 +5433,7 @@
notify->setInt32("channel-count", params.nChannels);
notify->setInt32("sample-rate", params.nSampleRate);
notify->setInt32("bitrate", params.nBitRate);
+ notify->setInt32("aac-profile", params.eAACProfile);
break;
}
@@ -9207,4 +9208,19 @@
return OK;
}
+status_t ACodec::querySupportedParameters(std::vector<std::string> *names) {
+ if (!names) {
+ return BAD_VALUE;
+ }
+ return OK;
+}
+
+status_t ACodec::subscribeToParameters([[maybe_unused]] const std::vector<std::string> &names) {
+ return OK;
+}
+
+status_t ACodec::unsubscribeFromParameters([[maybe_unused]] const std::vector<std::string> &names) {
+ return OK;
+}
+
} // namespace android
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index a052a70..1e9d4fe 100644
--- a/media/libstagefright/Android.bp
+++ b/media/libstagefright/Android.bp
@@ -109,6 +109,7 @@
srcs: [
"CodecBase.cpp",
+ "DataConverter.cpp",
"FrameRenderTracker.cpp",
"MediaCodecListWriter.cpp",
"SkipCutBuffer.cpp",
@@ -125,6 +126,7 @@
],
shared_libs: [
+ "libaudioutils",
"libgui",
"libhidlallocatorutils",
"liblog",
@@ -266,7 +268,6 @@
"CallbackMediaSource.cpp",
"CameraSource.cpp",
"CameraSourceTimeLapse.cpp",
- "DataConverter.cpp",
"FrameDecoder.cpp",
"HevcUtils.cpp",
"InterfaceUtils.cpp",
@@ -340,6 +341,7 @@
"android.hardware.media.omx@1.0",
"framework-permission-aidl-cpp",
"libaudioclient_aidl_conversion",
+ "packagemanager_aidl-cpp",
],
static_libs: [
@@ -371,6 +373,7 @@
"libmedia",
"android.hidl.allocator@1.0",
"framework-permission-aidl-cpp",
+ "packagemanager_aidl-cpp",
],
export_include_dirs: [
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index 94a0424..01cb9b3 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -793,8 +793,16 @@
if (overrideMeta == NULL) {
// check if we're dealing with a tiled heif
int32_t tileWidth, tileHeight, gridRows, gridCols;
+ int32_t widthColsProduct = 0;
+ int32_t heightRowsProduct = 0;
if (findGridInfo(trackMeta(), &tileWidth, &tileHeight, &gridRows, &gridCols)) {
- if (mWidth <= tileWidth * gridCols && mHeight <= tileHeight * gridRows) {
+ if (__builtin_mul_overflow(tileWidth, gridCols, &widthColsProduct) ||
+ __builtin_mul_overflow(tileHeight, gridRows, &heightRowsProduct)) {
+ ALOGE("Multiplication overflowed Grid size: %dx%d, Picture size: %dx%d",
+ gridCols, gridRows, tileWidth, tileHeight);
+ return nullptr;
+ }
+ if (mWidth <= widthColsProduct && mHeight <= heightRowsProduct) {
ALOGV("grid: %dx%d, tile size: %dx%d, picture size: %dx%d",
gridCols, gridRows, tileWidth, tileHeight, mWidth, mHeight);
@@ -880,9 +888,18 @@
}
int32_t width, height, stride;
- CHECK(outputFormat->findInt32("width", &width));
- CHECK(outputFormat->findInt32("height", &height));
- CHECK(outputFormat->findInt32("stride", &stride));
+ if (outputFormat->findInt32("width", &width) == false) {
+ ALOGE("MediaImageDecoder::onOutputReceived:width is missing in outputFormat");
+ return ERROR_MALFORMED;
+ }
+ if (outputFormat->findInt32("height", &height) == false) {
+ ALOGE("MediaImageDecoder::onOutputReceived:height is missing in outputFormat");
+ return ERROR_MALFORMED;
+ }
+ if (outputFormat->findInt32("stride", &stride) == false) {
+ ALOGE("MediaImageDecoder::onOutputReceived:stride is missing in outputFormat");
+ return ERROR_MALFORMED;
+ }
if (mFrame == NULL) {
sp<IMemory> frameMem = allocVideoFrame(
diff --git a/media/libstagefright/MediaAppender.cpp b/media/libstagefright/MediaAppender.cpp
index 5d80b30..21dcfa1 100644
--- a/media/libstagefright/MediaAppender.cpp
+++ b/media/libstagefright/MediaAppender.cpp
@@ -75,10 +75,21 @@
return status;
}
- if (strcmp("MPEG4Extractor", mExtractor->getName()) == 0) {
+ sp<AMessage> fileFormat;
+ status = mExtractor->getFileFormat(&fileFormat);
+ if (status != OK) {
+ ALOGE("extractor_getFileFormat failed, status :%d", status);
+ return status;
+ }
+
+ AString fileMime;
+ fileFormat->findString("mime", &fileMime);
+ // only compare the end of the file MIME type to allow for vendor customized mime type
+ if (fileMime.endsWith("mp4")){
mFormat = MediaMuxer::OUTPUT_FORMAT_MPEG_4;
} else {
- ALOGE("Unsupported format, extractor name:%s", mExtractor->getName());
+ ALOGE("Unsupported file format, extractor name:%s, fileformat %s",
+ mExtractor->getName(), fileMime.c_str());
return ERROR_UNSUPPORTED;
}
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index a366506..00c38d5 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -3185,8 +3185,11 @@
mediametrics_setInt32(mMetricsHandle, kCodecSecure, 0);
}
- if (mIsVideo) {
- // audio codec is currently ignored.
+ MediaCodecInfo::Attributes attr = mCodecInfo
+ ? mCodecInfo->getAttributes()
+ : MediaCodecInfo::Attributes(0);
+ if (!(attr & MediaCodecInfo::kFlagIsSoftwareOnly)) {
+ // software codec is currently ignored.
mResourceManagerProxy->addResource(
MediaResource::CodecResource(mFlags & kFlagIsSecure, mIsVideo));
}
@@ -3502,6 +3505,20 @@
case kWhatDrainThisBuffer:
{
+ if ((mFlags & kFlagUseBlockModel) == 0 && mTunneled) {
+ sp<RefBase> obj;
+ CHECK(msg->findObject("buffer", &obj));
+ sp<MediaCodecBuffer> buffer = static_cast<MediaCodecBuffer *>(obj.get());
+ if (mFlags & kFlagIsAsync) {
+ // In asynchronous mode, output format change is processed immediately.
+ handleOutputFormatChangeIfNeeded(buffer);
+ } else {
+ postActivityNotificationIfPossible();
+ }
+ mBufferChannel->discardBuffer(buffer);
+ break;
+ }
+
/* size_t index = */updateBuffers(kPortIndexOutput, msg);
if (mState == FLUSHING
diff --git a/media/libstagefright/MediaCodecList.cpp b/media/libstagefright/MediaCodecList.cpp
index 6243828..2ffe728 100644
--- a/media/libstagefright/MediaCodecList.cpp
+++ b/media/libstagefright/MediaCodecList.cpp
@@ -509,6 +509,29 @@
}
}
}
+
+ int32_t profile = -1;
+ if (format->findInt32("profile", &profile)) {
+ int32_t level = -1;
+ format->findInt32("level", &level);
+ Vector<MediaCodecInfo::ProfileLevel> profileLevels;
+ capabilities->getSupportedProfileLevels(&profileLevels);
+ auto it = profileLevels.begin();
+ for (; it != profileLevels.end(); ++it) {
+ if (profile != it->mProfile) {
+ continue;
+ }
+ if (level > -1 && level > it->mLevel) {
+ continue;
+ }
+ break;
+ }
+
+ if (it == profileLevels.end()) {
+ ALOGV("Codec does not support profile %d with level %d", profile, level);
+ return false;
+ }
+ }
}
// haven't found a reason to discard this one
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
index 0107c32..b07f8f7 100644
--- a/media/libstagefright/MediaCodecSource.cpp
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -943,10 +943,17 @@
sp<MediaCodecBuffer> outbuf;
status_t err = mEncoder->getOutputBuffer(index, &outbuf);
- if (err != OK || outbuf == NULL || outbuf->data() == NULL
- || outbuf->size() == 0) {
+ if (err != OK || outbuf == NULL || outbuf->data() == NULL) {
signalEOS();
break;
+ } else if (outbuf->size() == 0) {
+ // Zero length CSD buffers are not treated as an error
+ if (flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) {
+ mEncoder->releaseOutputBuffer(index);
+ } else {
+ signalEOS();
+ }
+ break;
}
MediaBufferBase *mbuf = new MediaBuffer(outbuf->size());
diff --git a/media/libstagefright/SkipCutBuffer.cpp b/media/libstagefright/SkipCutBuffer.cpp
index ee9016d..de91533 100644
--- a/media/libstagefright/SkipCutBuffer.cpp
+++ b/media/libstagefright/SkipCutBuffer.cpp
@@ -145,7 +145,19 @@
if (available < num) {
int32_t newcapacity = mCapacity + (num - available);
char * newbuffer = new char[newcapacity];
- memcpy(newbuffer, mCutBuffer, mCapacity);
+ if (mWriteHead < mReadHead) {
+ // data isn't continuous, need to memcpy twice
+ // to move previous data to new buffer.
+ size_t copyLeft = mCapacity - mReadHead;
+ memcpy(newbuffer, mCutBuffer + mReadHead, copyLeft);
+ memcpy(newbuffer + copyLeft, mCutBuffer, mWriteHead);
+ mReadHead = 0;
+ mWriteHead += copyLeft;
+ } else {
+ memcpy(newbuffer, mCutBuffer + mReadHead, mWriteHead - mReadHead);
+ mWriteHead -= mReadHead;
+ mReadHead = 0;
+ }
delete [] mCutBuffer;
mCapacity = newcapacity;
mCutBuffer = newbuffer;
diff --git a/media/libstagefright/colorconversion/Android.bp b/media/libstagefright/colorconversion/Android.bp
index 06cebd3..7ff9b10 100644
--- a/media/libstagefright/colorconversion/Android.bp
+++ b/media/libstagefright/colorconversion/Android.bp
@@ -25,10 +25,6 @@
"SoftwareRenderer.cpp",
],
- include_dirs: [
- "frameworks/native/include/media/openmax",
- ],
-
shared_libs: [
"libui",
"libnativewindow",
@@ -37,6 +33,7 @@
header_libs: [
"libstagefright_headers",
"libstagefright_foundation_headers",
+ "media_plugin_headers",
],
static_libs: ["libyuv_static"],
diff --git a/media/libstagefright/filters/Android.bp b/media/libstagefright/filters/Android.bp
index acc9e87..b46a271 100644
--- a/media/libstagefright/filters/Android.bp
+++ b/media/libstagefright/filters/Android.bp
@@ -22,10 +22,6 @@
"ZeroFilter.cpp",
],
- include_dirs: [
- "frameworks/native/include/media/openmax",
- ],
-
cflags: [
"-Wno-multichar",
"-Werror",
diff --git a/media/libstagefright/foundation/Android.bp b/media/libstagefright/foundation/Android.bp
index dd2c66f..5f86c22 100644
--- a/media/libstagefright/foundation/Android.bp
+++ b/media/libstagefright/foundation/Android.bp
@@ -33,18 +33,13 @@
},
host_supported: true,
double_loadable: true,
- include_dirs: [
- "frameworks/av/include",
- "frameworks/native/include",
- "frameworks/native/libs/arect/include",
- "frameworks/native/libs/nativebase/include",
- ],
local_include_dirs: [
"include/media/stagefright/foundation",
],
header_libs: [
+ "av-headers",
// this is only needed for the vendor variant that removes libbinder, but vendor
// target below does not allow adding header_libs.
"libbinder_headers",
diff --git a/media/libstagefright/foundation/MediaDefs.cpp b/media/libstagefright/foundation/MediaDefs.cpp
index 0ec5ad5..5c4ec17 100644
--- a/media/libstagefright/foundation/MediaDefs.cpp
+++ b/media/libstagefright/foundation/MediaDefs.cpp
@@ -60,6 +60,10 @@
const char *MEDIA_MIMETYPE_AUDIO_AC4 = "audio/ac4";
const char *MEDIA_MIMETYPE_AUDIO_MPEGH_MHA1 = "audio/mha1";
const char *MEDIA_MIMETYPE_AUDIO_MPEGH_MHM1 = "audio/mhm1";
+const char *MEDIA_MIMETYPE_AUDIO_MPEGH_BL_L3 = "audio/mhm1.03";
+const char *MEDIA_MIMETYPE_AUDIO_MPEGH_BL_L4 = "audio/mhm1.04";
+const char *MEDIA_MIMETYPE_AUDIO_MPEGH_LC_L3 = "audio/mhm1.0d";
+const char *MEDIA_MIMETYPE_AUDIO_MPEGH_LC_L4 = "audio/mhm1.0e";
const char *MEDIA_MIMETYPE_AUDIO_SCRAMBLED = "audio/scrambled";
const char *MEDIA_MIMETYPE_AUDIO_ALAC = "audio/alac";
const char *MEDIA_MIMETYPE_AUDIO_WMA = "audio/x-ms-wma";
@@ -75,7 +79,47 @@
const char *MEDIA_MIMETYPE_AUDIO_AMR_WB_PLUS = "audio/amr-wb+";
const char *MEDIA_MIMETYPE_AUDIO_APTX = "audio/aptx";
const char *MEDIA_MIMETYPE_AUDIO_DRA = "audio/vnd.dra";
-const char *MEDIA_MIMETYPE_AUDIO_AAC_FORMAT = "audio/aac";
+// Note: not in the IANA registry.
+const char *MEDIA_MIMETYPE_AUDIO_DOLBY_MAT = "audio/vnd.dolby.mat";
+// Note: not in the IANA registry.
+const char *MEDIA_MIMETYPE_AUDIO_DOLBY_MAT_1_0 = "audio/vnd.dolby.mat.1.0";
+// Note: not in the IANA registry.
+const char *MEDIA_MIMETYPE_AUDIO_DOLBY_MAT_2_0 = "audio/vnd.dolby.mat.2.0";
+// Note: not in the IANA registry.
+const char *MEDIA_MIMETYPE_AUDIO_DOLBY_MAT_2_1 = "audio/vnd.dolby.mat.2.1";
+const char *MEDIA_MIMETYPE_AUDIO_DOLBY_TRUEHD = "audio/vnd.dolby.mlp";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_MP4 = "audio/mp4a.40";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_MAIN = "audio/mp4a.40.01";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_LC = "audio/mp4a.40.02";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_SSR = "audio/mp4a.40.03";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_LTP = "audio/mp4a.40.04";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_HE_V1 = "audio/mp4a.40.05";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_SCALABLE = "audio/mp4a.40.06";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ERLC = "audio/mp4a.40.17";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_LD = "audio/mp4a.40.23";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_HE_V2 = "audio/mp4a.40.29";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ELD = "audio/mp4a.40.39";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_XHE = "audio/mp4a.40.42";
+// Note: not in the IANA registry.
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADIF = "audio/aac-adif";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_MAIN = "audio/aac-adts.01";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LC = "audio/aac-adts.02";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_SSR = "audio/aac-adts.03";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LTP = "audio/aac-adts.04";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_HE_V1 = "audio/aac-adts.05";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_SCALABLE = "audio/aac-adts.06";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_ERLC = "audio/aac-adts.17";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LD = "audio/aac-adts.23";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_HE_V2 = "audio/aac-adts.29";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_ELD = "audio/aac-adts.39";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_XHE = "audio/aac-adts.42";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_LATM_LC = "audio/mp4a-latm.02";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_LATM_HE_V1 = "audio/mp4a-latm.05";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_LATM_HE_V2 = "audio/mp4a-latm.29";
+// Note: not in the IANA registry.
+const char *MEDIA_MIMETYPE_AUDIO_IEC61937 = "audio/x-iec61937";
+// Note: not in the IANA registry.
+const char *MEDIA_MIMETYPE_AUDIO_IEC60958 = "audio/x-iec60958";
const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4";
const char *MEDIA_MIMETYPE_CONTAINER_WAV = "audio/x-wav";
diff --git a/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h b/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
index afa0c6d..fb8c299 100644
--- a/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
+++ b/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
@@ -62,6 +62,10 @@
extern const char *MEDIA_MIMETYPE_AUDIO_AC4;
extern const char *MEDIA_MIMETYPE_AUDIO_MPEGH_MHA1;
extern const char *MEDIA_MIMETYPE_AUDIO_MPEGH_MHM1;
+extern const char *MEDIA_MIMETYPE_AUDIO_MPEGH_BL_L3;
+extern const char *MEDIA_MIMETYPE_AUDIO_MPEGH_BL_L4;
+extern const char *MEDIA_MIMETYPE_AUDIO_MPEGH_LC_L3;
+extern const char *MEDIA_MIMETYPE_AUDIO_MPEGH_LC_L4;
extern const char *MEDIA_MIMETYPE_AUDIO_SCRAMBLED;
extern const char *MEDIA_MIMETYPE_AUDIO_ALAC;
extern const char *MEDIA_MIMETYPE_AUDIO_WMA;
@@ -77,7 +81,40 @@
extern const char *MEDIA_MIMETYPE_AUDIO_AMR_WB_PLUS;
extern const char *MEDIA_MIMETYPE_AUDIO_APTX;
extern const char *MEDIA_MIMETYPE_AUDIO_DRA;
-extern const char *MEDIA_MIMETYPE_AUDIO_AAC_FORMAT;
+extern const char *MEDIA_MIMETYPE_AUDIO_DOLBY_MAT;
+extern const char *MEDIA_MIMETYPE_AUDIO_DOLBY_MAT_1_0;
+extern const char *MEDIA_MIMETYPE_AUDIO_DOLBY_MAT_2_0;
+extern const char *MEDIA_MIMETYPE_AUDIO_DOLBY_MAT_2_1;
+extern const char *MEDIA_MIMETYPE_AUDIO_DOLBY_TRUEHD;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_MP4;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_MAIN;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_LC;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_SSR;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_LTP;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_HE_V1;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_SCALABLE;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ERLC;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_LD;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_HE_V2;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ELD;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_XHE;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADIF;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_MAIN;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LC;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_SSR;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LTP;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_HE_V1;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_SCALABLE;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_ERLC;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_LD;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_HE_V2;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_ELD;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS_XHE;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_LATM_LC;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_LATM_HE_V1;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_LATM_HE_V2;
+extern const char *MEDIA_MIMETYPE_AUDIO_IEC61937;
+extern const char *MEDIA_MIMETYPE_AUDIO_IEC60958;
extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4;
extern const char *MEDIA_MIMETYPE_CONTAINER_WAV;
diff --git a/media/libstagefright/foundation/tests/AVCUtils/Android.bp b/media/libstagefright/foundation/tests/AVCUtils/Android.bp
index 594da56..ee7db21 100644
--- a/media/libstagefright/foundation/tests/AVCUtils/Android.bp
+++ b/media/libstagefright/foundation/tests/AVCUtils/Android.bp
@@ -43,10 +43,6 @@
"libstagefright_foundation",
],
- include_dirs: [
- "frameworks/av/media/libstagefright/foundation",
- ],
-
cflags: [
"-Werror",
"-Wall",
diff --git a/media/libstagefright/foundation/tests/Android.bp b/media/libstagefright/foundation/tests/Android.bp
index e50742e..e72ce43 100644
--- a/media/libstagefright/foundation/tests/Android.bp
+++ b/media/libstagefright/foundation/tests/Android.bp
@@ -18,10 +18,6 @@
"-Wall",
],
- include_dirs: [
- "frameworks/av/include",
- ],
-
shared_libs: [
"liblog",
"libstagefright_foundation",
diff --git a/media/libstagefright/httplive/Android.bp b/media/libstagefright/httplive/Android.bp
index 0b0acbf..7acf735 100644
--- a/media/libstagefright/httplive/Android.bp
+++ b/media/libstagefright/httplive/Android.bp
@@ -29,7 +29,6 @@
],
include_dirs: [
- "frameworks/av/media/libstagefright",
"frameworks/native/include/media/openmax",
],
@@ -65,6 +64,8 @@
header_libs: [
"libbase_headers",
+ "libstagefright_headers",
+ "libstagefright_httplive_headers",
],
static_libs: [
@@ -74,3 +75,8 @@
],
}
+
+cc_library_headers {
+ name: "libstagefright_httplive_headers",
+ export_include_dirs: ["."],
+}
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 3bad015..0d7cadd 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -23,7 +23,7 @@
#include "M3UParser.h"
#include "PlaylistFetcher.h"
-#include "mpeg2ts/AnotherPacketSource.h"
+#include <AnotherPacketSource.h>
#include <cutils/properties.h>
#include <media/MediaHTTPService.h>
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 7a6d487..ceea41d 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -24,7 +24,7 @@
#include <utils/String8.h>
-#include "mpeg2ts/ATSParser.h"
+#include <ATSParser.h>
namespace android {
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index b23aa8a..907b326 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -24,9 +24,9 @@
#include "HTTPDownloader.h"
#include "LiveSession.h"
#include "M3UParser.h"
-#include "include/ID3.h"
-#include "mpeg2ts/AnotherPacketSource.h"
-#include "mpeg2ts/HlsSampleDecryptor.h"
+#include <ID3.h>
+#include <AnotherPacketSource.h>
+#include <HlsSampleDecryptor.h>
#include <datasource/DataURISource.h>
#include <media/stagefright/foundation/ABitReader.h>
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 5d3f9c1..2e28164 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -21,7 +21,7 @@
#include <media/stagefright/foundation/AHandler.h>
#include <openssl/aes.h>
-#include "mpeg2ts/ATSParser.h"
+#include <ATSParser.h>
#include "LiveSession.h"
namespace android {
diff --git a/media/libstagefright/httplive/fuzzer/Android.bp b/media/libstagefright/httplive/fuzzer/Android.bp
new file mode 100644
index 0000000..14097b0
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/Android.bp
@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package {
+ // See: http://go/android-license-faq
+ // A large-scale-change added 'default_applicable_licenses' to import
+ // all of the 'license_kinds' from "frameworks_av_media_libstagefright_httplive_license"
+ // to get the below license kinds:
+ // SPDX-license-identifier-Apache-2.0
+ default_applicable_licenses: [
+ "frameworks_av_media_libstagefright_httplive_license",
+ ],
+}
+
+cc_fuzz {
+ name: "httplive_fuzzer",
+ srcs: [
+ "httplive_fuzzer.cpp",
+ ],
+ static_libs: [
+ "libstagefright_httplive",
+ "libstagefright_id3",
+ "libstagefright_metadatautils",
+ "libstagefright_mpeg2support",
+ "liblog",
+ "libcutils",
+ "libdatasource",
+ "libmedia",
+ "libstagefright",
+ "libutils",
+ ],
+ header_libs: [
+ "libbase_headers",
+ "libstagefright_foundation_headers",
+ "libstagefright_headers",
+ "libstagefright_httplive_headers",
+ ],
+ shared_libs: [
+ "libcrypto",
+ "libstagefright_foundation",
+ "libhidlbase",
+ "libhidlmemory",
+ "android.hidl.allocator@1.0",
+ ],
+ corpus: ["corpus/*"],
+ dictionary: "httplive_fuzzer.dict",
+ fuzz_config: {
+ cc: [
+ "android-media-fuzzing-reports@google.com",
+ ],
+ componentid: 155276,
+ },
+}
diff --git a/media/libstagefright/httplive/fuzzer/README.md b/media/libstagefright/httplive/fuzzer/README.md
new file mode 100644
index 0000000..3a64ea4
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/README.md
@@ -0,0 +1,56 @@
+# Fuzzer for libstagefright_httplive
+
+## Plugin Design Considerations
+The fuzzer plugin for libstagefright_httplive is designed based on the understanding of the library and tries to achieve the following:
+
+##### Maximize code coverage
+The configuration parameters are not hardcoded, but instead selected based on
+incoming data.Also, several .m3u8 files are hand-crafted and added to the corpus directory to increase the code coverage. This ensures more code paths are reached by the fuzzer.
+
+libstagefright_httplive supports the following parameters:
+1. Final Result (parameter name: `finalResult`)
+2. Flags (parameter name: `flags`)
+3. Time Us (parameter name: `timeUs`)
+4. Track Index (parameter name: `trackIndex`)
+5. Index (parameter name: `index`)
+6. Select (parameter name: `select`)
+
+| Parameter| Valid Values| Configured Value|
+|------------- |-------------| ----- |
+| `finalResult` | `-34` to `-1` | Value obtained from FuzzedDataProvider|
+| `flags` | `0` to `1` | Value obtained from FuzzedDataProvider|
+| `timeUs` | `0` to `10000000` | Value obtained from FuzzedDataProvider|
+| `trackIndex` | `UINT32_MIN` to `UINT32_MAX` | Value obtained from FuzzedDataProvider|
+| `index` | `UINT32_MIN` to `UINT32_MAX` | Value obtained from FuzzedDataProvider|
+| `select` | `True` to `False` | Value obtained from FuzzedDataProvider|
+
+This also ensures that the plugin is always deterministic for any given input.
+
+##### Maximize utilization of input data
+The plugin feeds the entire input data to the httplive module.
+This ensures that the plugin tolerates any kind of input (empty, huge,
+malformed, etc) and doesnt `exit()` on any input and thereby increasing the
+chance of identifying vulnerabilities.
+
+## Build
+
+This describes steps to build httplive_fuzzer binary.
+
+### Android
+
+#### Steps to build
+Build the fuzzer
+```
+ $ mm -j$(nproc) httplive_fuzzer
+```
+#### Steps to run
+To run on device
+```
+ $ adb push $ANDROID_PRODUCT_OUT/data/fuzz/$(TARGET_ARCH)/lib /data/fuzz/$(TARGET_ARCH)/lib
+ $ adb push $ANDROID_PRODUCT_OUT/data/fuzz/$(TARGET_ARCH)/httplive_fuzzer /data/fuzz/$(TARGET_ARCH)/httplive_fuzzer
+ $ adb shell /data/fuzz/${TARGET_ARCH}/httplive_fuzzer/httplive_fuzzer /data/fuzz/${TARGET_ARCH}/httplive_fuzzer/corpus
+```
+
+## References:
+ * http://llvm.org/docs/LibFuzzer.html
+ * https://github.com/google/oss-fuzz
diff --git a/media/libstagefright/httplive/fuzzer/corpus/crypt.key b/media/libstagefright/httplive/fuzzer/corpus/crypt.key
new file mode 100644
index 0000000..f9d5d7f
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/crypt.key
@@ -0,0 +1,2 @@
+Û
+ÏüÐ5Ð_xïHÎ3
diff --git a/media/libstagefright/httplive/fuzzer/corpus/encrypted.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/encrypted.m3u8
new file mode 100644
index 0000000..32b0eac
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/encrypted.m3u8
@@ -0,0 +1,12 @@
+#EXTM3U
+#EXT-X-TARGETDURATION:10
+#EXT-X-ALLOW-CACHE:YES
+#EXT-X-PLAYLIST-TYPE:VOD
+#EXT-X-VERSION:3
+#EXT-X-MEDIA-SEQUENCE:1
+#EXT-X-KEY:METHOD=AES-128,URI="../../fuzz/arm64/httplive_fuzzer/corpus/crypt.key"
+#EXTINF:10.000,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXTINF:5.092,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXT-X-ENDLIST
diff --git a/media/libstagefright/httplive/fuzzer/corpus/hls.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/hls.m3u8
new file mode 100644
index 0000000..9338e04
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/hls.m3u8
@@ -0,0 +1,8 @@
+#EXTM3U
+#EXT-X-TARGETDURATION:10
+#EXT-X-MEDIA-SEQUENCE:0
+#EXTINF:10, no desc
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXTINF:10, no desc
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence2.ts
+#EXT-X-ENDLIST
diff --git a/media/libstagefright/httplive/fuzzer/corpus/index1.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/index1.m3u8
new file mode 100644
index 0000000..e1eff58
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/index1.m3u8
@@ -0,0 +1,14 @@
+#EXTM3U
+#EXT-X-VERSION:4
+#EXT-X-TARGETDURATION:5
+#EXT-X-KEY:METHOD=NONE
+#EXT-X-DISCONTINUITY-SEQUENCE:0
+#EXT-X-PLAYLIST-TYPE:VOD
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence0.ts
+#EXT-X-DISCONTINUITY
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence2.ts
+#EXT-X-ENDLIST
diff --git a/media/libstagefright/httplive/fuzzer/corpus/index2.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/index2.m3u8
new file mode 100644
index 0000000..37a0189
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/index2.m3u8
@@ -0,0 +1,6 @@
+#EXTM3U
+#EXT-X-INDEPENDENT-SEGMENTS
+#EXT-X-STREAM-INF:CLOSED-CAPTIONS=NONE,BANDWIDTH=165340,RESOLUTION=256x144,CODECS="mp4a.40.5,avc1.42c00b"
+https://non.existentsite.com/test-doesnt-dereference-these-paths/prog_index.m3u8
+#EXT-X-STREAM-INF:CLOSED-CAPTIONS=NONE,BANDWIDTH=344388,RESOLUTION=426x240,CODECS="mp4a.40.5,avc1.4d4015"
+https://non.existentsite.com/test-doesnt-dereference-these-paths/prog_index1.m3u8
diff --git a/media/libstagefright/httplive/fuzzer/corpus/index3.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/index3.m3u8
new file mode 100644
index 0000000..1b7f489
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/index3.m3u8
@@ -0,0 +1,13 @@
+#EXTM3U
+#EXT-X-VERSION:4
+#EXT-X-TARGETDURATION:5
+#EXT-X-KEY:METHOD=AES-128,URI="https://demo.unified-streaming.com/video/tears-of-steel/aes.key",IV=0X99b74007b6254e4bd1c6e03631cad15b
+#EXT-X-PLAYLIST-TYPE:VOD
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXT-X-DISCONTINUITY
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence2.ts
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence3.ts
+#EXT-X-ENDLIST
diff --git a/media/libstagefright/httplive/fuzzer/corpus/index4.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/index4.m3u8
new file mode 100644
index 0000000..89ba37c
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/index4.m3u8
@@ -0,0 +1,15 @@
+#EXTM3U
+#EXT-X-VERSION:4
+#EXT-X-TARGETDURATION:5
+#EXT-X-KEY:METHOD=SAMPLE-AES,URI="data:text/plain;charset=utf-8,a4cd9995a1aa91e1",IV=0X99b74007b6254e4bd1c6e03631cad15b
+#EXT-X-DISCONTINUITY-SEQUENCE:0
+#EXT-X-PLAYLIST-TYPE:VOD
+#EXT-X-DISCONTINUITY
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence0.ts
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXT-X-DISCONTINUITY
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence2.ts
+#EXT-X-ENDLIST
diff --git a/media/libstagefright/httplive/fuzzer/corpus/index5.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/index5.m3u8
new file mode 100644
index 0000000..2120de4
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/index5.m3u8
@@ -0,0 +1,14 @@
+#EXTM3U
+#EXT-X-TARGETDURATION:11
+#EXT-X-KEY:METHOD=NONE
+#EXT-X-MEDIA-SEQUENCE:0
+#EXT-X-VERSION:4
+#EXTINF:10.0,
+#EXT-X-BYTERANGE:10@0
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXTINF:10.0,
+#EXT-X-BYTERANGE:20@10
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXTINF:10.0,
+#EXT-X-BYTERANGE:80
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
diff --git a/media/libstagefright/httplive/fuzzer/corpus/index6.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/index6.m3u8
new file mode 100644
index 0000000..588368a
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/index6.m3u8
@@ -0,0 +1,12 @@
+#EXTM3U
+#EXT-X-VERSION:4
+#EXT-X-TARGETDURATION:5
+#EXT-X-KEY:METHOD=AES-128,URI="data:text/plain;charset=utf-8,a4cd9995a1aa91e1",IV=0x30303030303030303030303030303030
+#EXT-X-PLAYLIST-TYPE:VOD
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence2.ts
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence3.ts
+#EXT-X-ENDLIST
diff --git a/media/libstagefright/httplive/fuzzer/corpus/index7.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/index7.m3u8
new file mode 100644
index 0000000..b09948e
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/index7.m3u8
@@ -0,0 +1,46 @@
+#EXTM3U
+#EXT-X-VERSION:4
+## Created with Unified Streaming Platform (version=1.11.3-24438)
+#EXT-X-SESSION-KEY:METHOD=AES-128,URI="https://demo.unified-streaming.com/video/tears-of-steel/aes.key"
+
+# AUDIO groups
+#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio-aacl-64",LANGUAGE="en",NAME="English",DEFAULT=YES,AUTOSELECT=YES,CHANNELS="2"
+#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio-aacl-128",LANGUAGE="en",NAME="English",DEFAULT=YES,AUTOSELECT=YES,CHANNELS="2"
+
+# SUBTITLES groups
+#EXT-X-MEDIA:TYPE=SUBTITLES,GROUP-ID="textstream",LANGUAGE="en",NAME="English",DEFAULT=YES,AUTOSELECT=YES,URI="tears-of-steel-aes-textstream_eng=1000.m3u8"
+#EXT-X-MEDIA:TYPE=SUBTITLES,GROUP-ID="textstream",LANGUAGE="ru",NAME="Russian",AUTOSELECT=YES,URI="tears-of-steel-aes-textstream_rus=1000.m3u8"
+
+# variants
+#EXT-X-STREAM-INF:BANDWIDTH=494000,CODECS="mp4a.40.2,avc1.42C00D",RESOLUTION=224x100,FRAME-RATE=24,AUDIO="audio-aacl-64",SUBTITLES="textstream",CLOSED-CAPTIONS=NONE
+tears-of-steel-aes-audio_eng=64008-video_eng=401000.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=933000,CODECS="mp4a.40.2,avc1.42C016",RESOLUTION=448x200,FRAME-RATE=24,AUDIO="audio-aacl-128",SUBTITLES="textstream",CLOSED-CAPTIONS=NONE
+tears-of-steel-aes-audio_eng=128002-video_eng=751000.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=1198000,CODECS="mp4a.40.2,avc1.4D401F",RESOLUTION=784x350,FRAME-RATE=24,AUDIO="audio-aacl-128",SUBTITLES="textstream",CLOSED-CAPTIONS=NONE
+tears-of-steel-aes-audio_eng=128002-video_eng=1001000.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=1728000,CODECS="mp4a.40.2,avc1.640028",RESOLUTION=1680x750,FRAME-RATE=24,VIDEO-RANGE=SDR,AUDIO="audio-aacl-128",SUBTITLES="textstream",CLOSED-CAPTIONS=NONE
+tears-of-steel-aes-audio_eng=128002-video_eng=1501000.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=2469000,CODECS="mp4a.40.2,avc1.640028",RESOLUTION=1680x750,FRAME-RATE=24,VIDEO-RANGE=SDR,AUDIO="audio-aacl-128",SUBTITLES="textstream",CLOSED-CAPTIONS=NONE
+tears-of-steel-aes-audio_eng=128002-video_eng=2200000.m3u8
+
+# variants
+#EXT-X-STREAM-INF:BANDWIDTH=1025000,CODECS="mp4a.40.2,hvc1.1.6.L150.90",RESOLUTION=1680x750,FRAME-RATE=24,VIDEO-RANGE=SDR,AUDIO="audio-aacl-64",SUBTITLES="textstream",CLOSED-CAPTIONS=NONE
+tears-of-steel-aes-audio_eng=64008-video_eng_1=902000.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=1368000,CODECS="mp4a.40.2,hvc1.1.6.L150.90",RESOLUTION=2576x1150,FRAME-RATE=24,VIDEO-RANGE=SDR,AUDIO="audio-aacl-128",SUBTITLES="textstream",CLOSED-CAPTIONS=NONE
+tears-of-steel-aes-audio_eng=128002-video_eng_1=1161000.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=1815000,CODECS="mp4a.40.2,hvc1.1.6.L150.90",RESOLUTION=3360x1500,FRAME-RATE=24,VIDEO-RANGE=SDR,AUDIO="audio-aacl-128",SUBTITLES="textstream",CLOSED-CAPTIONS=NONE
+tears-of-steel-aes-audio_eng=128002-video_eng_1=1583000.m3u8
+
+# variants
+#EXT-X-STREAM-INF:BANDWIDTH=69000,CODECS="mp4a.40.2",AUDIO="audio-aacl-64",SUBTITLES="textstream"
+tears-of-steel-aes-audio_eng=64008.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=137000,CODECS="mp4a.40.2",AUDIO="audio-aacl-128",SUBTITLES="textstream"
+tears-of-steel-aes-audio_eng=128002.m3u8
+
+# keyframes
+#EXT-X-I-FRAME-STREAM-INF:BANDWIDTH=54000,CODECS="avc1.42C00D",RESOLUTION=224x100,URI="keyframes/tears-of-steel-aes-video_eng=401000.m3u8"
+#EXT-X-I-FRAME-STREAM-INF:BANDWIDTH=100000,CODECS="avc1.42C016",RESOLUTION=448x200,URI="keyframes/tears-of-steel-aes-video_eng=751000.m3u8"
+#EXT-X-I-FRAME-STREAM-INF:BANDWIDTH=133000,CODECS="avc1.4D401F",RESOLUTION=784x350,URI="keyframes/tears-of-steel-aes-video_eng=1001000.m3u8"
+#EXT-X-I-FRAME-STREAM-INF:BANDWIDTH=120000,CODECS="hvc1.1.6.L150.90",RESOLUTION=1680x750,VIDEO-RANGE=SDR,URI="keyframes/tears-of-steel-aes-video_eng_1=902000.m3u8"
+#EXT-X-I-FRAME-STREAM-INF:BANDWIDTH=154000,CODECS="hvc1.1.6.L150.90",RESOLUTION=2576x1150,VIDEO-RANGE=SDR,URI="keyframes/tears-of-steel-aes-video_eng_1=1161000.m3u8"
+#EXT-X-I-FRAME-STREAM-INF:BANDWIDTH=210000,CODECS="hvc1.1.6.L150.90",RESOLUTION=3360x1500,VIDEO-RANGE=SDR,URI="keyframes/tears-of-steel-aes-video_eng_1=1583000.m3u8"
diff --git a/media/libstagefright/httplive/fuzzer/corpus/index8.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/index8.m3u8
new file mode 100644
index 0000000..353d589
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/index8.m3u8
@@ -0,0 +1,13 @@
+#EXTM3U
+#EXT-X-VERSION:5
+
+#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",NAME="English stereo",LANGUAGE="en",AUTOSELECT=YES,URI="../../fuzz/arm64/httplive_fuzzer/index1.m3u8"
+
+#EXT-X-STREAM-INF:BANDWIDTH=628000,CODECS="avc1.42c00d,mp4a.40.2",RESOLUTION=320x180,AUDIO="audio"
+../../fuzz/arm64/httplive_fuzzer/index1.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=928000,CODECS="avc1.42c00d,mp4a.40.2",RESOLUTION=480x270,AUDIO="audio"
+../../fuzz/arm64/httplive_fuzzer/index2.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=1728000,CODECS="avc1.42c00d,mp4a.40.2",RESOLUTION=640x360,AUDIO="audio"
+../../fuzz/arm64/httplive_fuzzer/index3.m3u8
+#EXT-X-STREAM-INF:BANDWIDTH=2528000,CODECS="avc1.42c00d,mp4a.40.2",RESOLUTION=960x540,AUDIO="audio"
+../../fuzz/arm64/httplive_fuzzer/index1.m3u8
diff --git a/media/libstagefright/httplive/fuzzer/corpus/prog_index.m3u8 b/media/libstagefright/httplive/fuzzer/corpus/prog_index.m3u8
new file mode 100644
index 0000000..eb88422
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/corpus/prog_index.m3u8
@@ -0,0 +1,17 @@
+#EXTM3U
+#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="eng",NAME="English",AUTOSELECT=YES,DEFAULT=YES,URI="corpus/index1.m3u8"
+#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="fre",NAME="Français",AUTOSELECT=YES,DEFAULT=NO,URI="corpus/index1.m3u8"
+#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="sp",NAME="Espanol",AUTOSELECT=YES,DEFAULT=NO,URI="corpus/index1.m3u8"
+#EXT-X-VERSION:4
+#EXT-X-TARGETDURATION:5
+#EXT-X-KEY:METHOD=NONE
+#EXT-X-DISCONTINUITY-SEQUENCE:0
+#EXT-X-PLAYLIST-TYPE:VOD
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXT-X-DISCONTINUITY
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXTINF:5,
+https://non.existentsite.com/test-doesnt-dereference-these-paths/fileSequence1.ts
+#EXT-X-ENDLIST
diff --git a/media/libstagefright/httplive/fuzzer/httplive_fuzzer.cpp b/media/libstagefright/httplive/fuzzer/httplive_fuzzer.cpp
new file mode 100644
index 0000000..aa777b3
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/httplive_fuzzer.cpp
@@ -0,0 +1,298 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <fstream>
+#include <fuzzer/FuzzedDataProvider.h>
+#include <LiveDataSource.h>
+#include <LiveSession.h>
+#include <media/MediaHTTPConnection.h>
+#include <media/MediaHTTPService.h>
+#include <media/mediaplayer_common.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/ALooperRoster.h>
+#include <string>
+#include <utils/Log.h>
+
+using namespace std;
+using namespace android;
+
+constexpr char kFileNamePrefix[] = "/data/local/tmp/httplive-";
+constexpr char kFileNameSuffix[] = ".m3u8";
+constexpr char kFileUrlPrefix[] = "file://";
+constexpr int64_t kOffSet = 0;
+constexpr int32_t kReadyMarkMs = 5000;
+constexpr int32_t kPrepareMarkMs = 1500;
+constexpr int32_t kErrorNoMax = -1;
+constexpr int32_t kErrorNoMin = -34;
+constexpr int32_t kMaxTimeUs = 1000;
+constexpr int32_t kRandomStringLength = 64;
+constexpr int32_t kRangeMin = 0;
+constexpr int32_t kRangeMax = 1000;
+
+constexpr LiveSession::StreamType kValidStreamType[] = {
+ LiveSession::STREAMTYPE_AUDIO, LiveSession::STREAMTYPE_VIDEO,
+ LiveSession::STREAMTYPE_SUBTITLES, LiveSession::STREAMTYPE_METADATA};
+
+constexpr MediaSource::ReadOptions::SeekMode kValidSeekMode[] = {
+ MediaSource::ReadOptions::SeekMode::SEEK_PREVIOUS_SYNC,
+ MediaSource::ReadOptions::SeekMode::SEEK_NEXT_SYNC,
+ MediaSource::ReadOptions::SeekMode::SEEK_CLOSEST_SYNC,
+ MediaSource::ReadOptions::SeekMode::SEEK_CLOSEST,
+ MediaSource::ReadOptions::SeekMode::SEEK_FRAME_INDEX};
+
+constexpr media_track_type kValidMediaTrackType[] = {
+ MEDIA_TRACK_TYPE_UNKNOWN, MEDIA_TRACK_TYPE_VIDEO,
+ MEDIA_TRACK_TYPE_AUDIO, MEDIA_TRACK_TYPE_TIMEDTEXT,
+ MEDIA_TRACK_TYPE_SUBTITLE, MEDIA_TRACK_TYPE_METADATA};
+
+struct TestAHandler : public AHandler {
+public:
+ TestAHandler(std::function<void()> signalEosFunction)
+ : mSignalEosFunction(signalEosFunction) {}
+ virtual ~TestAHandler() {}
+
+protected:
+ void onMessageReceived(const sp<AMessage> &msg) override {
+ int32_t what = -1;
+ msg->findInt32("what", &what);
+ switch (what) {
+ case LiveSession::kWhatError:
+ case LiveSession::kWhatPrepared:
+ case LiveSession::kWhatPreparationFailed: {
+ mSignalEosFunction();
+ break;
+ }
+ }
+ return;
+ }
+
+private:
+ std::function<void()> mSignalEosFunction;
+};
+
+struct TestMediaHTTPConnection : public MediaHTTPConnection {
+public:
+ TestMediaHTTPConnection() {}
+ virtual ~TestMediaHTTPConnection() {}
+
+ virtual bool connect(const char * /*uri*/,
+ const KeyedVector<String8, String8> * /*headers*/) {
+ return true;
+ }
+
+ virtual void disconnect() { return; }
+
+ virtual ssize_t readAt(off64_t /*offset*/, void * /*data*/, size_t size) {
+ return size;
+ }
+
+ virtual off64_t getSize() { return 0; }
+ virtual status_t getMIMEType(String8 * /*mimeType*/) { return NO_ERROR; }
+ virtual status_t getUri(String8 * /*uri*/) { return NO_ERROR; }
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(TestMediaHTTPConnection);
+};
+
+struct TestMediaHTTPService : public MediaHTTPService {
+public:
+ TestMediaHTTPService() {}
+ ~TestMediaHTTPService(){};
+
+ virtual sp<MediaHTTPConnection> makeHTTPConnection() {
+ mediaHTTPConnection = sp<TestMediaHTTPConnection>::make();
+ return mediaHTTPConnection;
+ }
+
+private:
+ sp<TestMediaHTTPConnection> mediaHTTPConnection = nullptr;
+ DISALLOW_EVIL_CONSTRUCTORS(TestMediaHTTPService);
+};
+
+class HttpLiveFuzzer {
+public:
+ void process(const uint8_t *data, size_t size);
+ void deInitLiveSession();
+ ~HttpLiveFuzzer() { deInitLiveSession(); }
+
+private:
+ void invokeLiveDataSource();
+ void createM3U8File(const uint8_t *data, size_t size);
+ void initLiveDataSource();
+ void invokeLiveSession();
+ void initLiveSession();
+ void invokeDequeueAccessUnit();
+ void invokeConnectAsync();
+ void invokeSeekTo();
+ void invokeGetConfig();
+ void signalEos();
+ string generateFileName();
+ sp<LiveDataSource> mLiveDataSource = nullptr;
+ sp<LiveSession> mLiveSession = nullptr;
+ sp<ALooper> mLiveLooper = nullptr;
+ sp<TestMediaHTTPService> httpService = nullptr;
+ sp<TestAHandler> mHandler = nullptr;
+ FuzzedDataProvider *mFDP = nullptr;
+ bool mEosReached = false;
+ std::mutex mDownloadCompleteMutex;
+ std::condition_variable mConditionalVariable;
+};
+
+string HttpLiveFuzzer::generateFileName() {
+ return kFileNamePrefix + to_string(getpid()) + kFileNameSuffix;
+}
+
+void HttpLiveFuzzer::createM3U8File(const uint8_t *data, size_t size) {
+ ofstream m3u8File;
+ string currentFileName = generateFileName();
+ m3u8File.open(currentFileName, ios::out | ios::binary);
+ m3u8File.write((char *)data, size);
+ m3u8File.close();
+}
+
+void HttpLiveFuzzer::initLiveDataSource() {
+ mLiveDataSource = sp<LiveDataSource>::make();
+}
+
+void HttpLiveFuzzer::invokeLiveDataSource() {
+ initLiveDataSource();
+ size_t size = mFDP->ConsumeIntegralInRange<size_t>(kRangeMin, kRangeMax);
+ sp<ABuffer> buffer = new ABuffer(size);
+ mLiveDataSource->queueBuffer(buffer);
+ uint8_t *data = new uint8_t[size];
+ mLiveDataSource->readAtNonBlocking(kOffSet, data, size);
+ int32_t finalResult = mFDP->ConsumeIntegralInRange(kErrorNoMin, kErrorNoMax);
+ mLiveDataSource->queueEOS(finalResult);
+ mLiveDataSource->reset();
+ mLiveDataSource->countQueuedBuffers();
+ mLiveDataSource->initCheck();
+ delete[] data;
+}
+
+void HttpLiveFuzzer::initLiveSession() {
+ ALooperRoster looperRoster;
+ mHandler =
+ sp<TestAHandler>::make(std::bind(&HttpLiveFuzzer::signalEos, this));
+ mLiveLooper = sp<ALooper>::make();
+ mLiveLooper->setName("http live");
+ mLiveLooper->start();
+ sp<AMessage> notify = sp<AMessage>::make(0, mHandler);
+ httpService = new TestMediaHTTPService();
+ uint32_t flags = mFDP->ConsumeIntegral<uint32_t>();
+ mLiveSession = sp<LiveSession>::make(notify, flags, httpService);
+ mLiveLooper->registerHandler(mLiveSession);
+ looperRoster.registerHandler(mLiveLooper, mHandler);
+}
+
+void HttpLiveFuzzer::invokeDequeueAccessUnit() {
+ LiveSession::StreamType stream = mFDP->PickValueInArray(kValidStreamType);
+ sp<ABuffer> buffer;
+ mLiveSession->dequeueAccessUnit(stream, &buffer);
+}
+
+void HttpLiveFuzzer::invokeSeekTo() {
+ int64_t timeUs = mFDP->ConsumeIntegralInRange<int64_t>(0, kMaxTimeUs);
+ MediaSource::ReadOptions::SeekMode mode =
+ mFDP->PickValueInArray(kValidSeekMode);
+ mLiveSession->seekTo(timeUs, mode);
+}
+
+void HttpLiveFuzzer::invokeGetConfig() {
+ mLiveSession->getTrackCount();
+ size_t trackIndex = mFDP->ConsumeIntegral<size_t>();
+ mLiveSession->getTrackInfo(trackIndex);
+ media_track_type type = mFDP->PickValueInArray(kValidMediaTrackType);
+ mLiveSession->getSelectedTrack(type);
+ sp<MetaData> meta;
+ LiveSession::StreamType stream = mFDP->PickValueInArray(kValidStreamType);
+ mLiveSession->getStreamFormatMeta(stream, &meta);
+ mLiveSession->getKeyForStream(stream);
+ if (stream != LiveSession::STREAMTYPE_SUBTITLES) {
+ mLiveSession->getSourceTypeForStream(stream);
+ }
+}
+
+void HttpLiveFuzzer::invokeConnectAsync() {
+ string currentFileName = generateFileName();
+ string url = kFileUrlPrefix + currentFileName;
+ string str_1 = mFDP->ConsumeRandomLengthString(kRandomStringLength);
+ string str_2 = mFDP->ConsumeRandomLengthString(kRandomStringLength);
+
+ KeyedVector<String8, String8> headers;
+ headers.add(String8(str_1.c_str()), String8(str_2.c_str()));
+ mLiveSession->connectAsync(url.c_str(), &headers);
+}
+
+void HttpLiveFuzzer::invokeLiveSession() {
+ initLiveSession();
+ BufferingSettings bufferingSettings;
+ bufferingSettings.mInitialMarkMs = kPrepareMarkMs;
+ bufferingSettings.mResumePlaybackMarkMs = kReadyMarkMs;
+ mLiveSession->setBufferingSettings(bufferingSettings);
+ invokeConnectAsync();
+ std::unique_lock waitForDownloadComplete(mDownloadCompleteMutex);
+ mConditionalVariable.wait(waitForDownloadComplete,
+ [this] { return mEosReached; });
+ if (mLiveSession->isSeekable()) {
+ invokeSeekTo();
+ }
+ invokeDequeueAccessUnit();
+ size_t index = mFDP->ConsumeIntegral<size_t>();
+ bool select = mFDP->ConsumeBool();
+ mLiveSession->selectTrack(index, select);
+ mLiveSession->hasDynamicDuration();
+ int64_t firstTimeUs =
+ mFDP->ConsumeIntegralInRange<int64_t>(kRangeMin, kRangeMax);
+ int64_t timeUs = mFDP->ConsumeIntegralInRange<int64_t>(kRangeMin, kRangeMax);
+ int32_t discontinuitySeq = mFDP->ConsumeIntegral<int32_t>();
+ mLiveSession->calculateMediaTimeUs(firstTimeUs, timeUs, discontinuitySeq);
+ invokeGetConfig();
+}
+
+void HttpLiveFuzzer::process(const uint8_t *data, size_t size) {
+ mFDP = new FuzzedDataProvider(data, size);
+ createM3U8File(data, size);
+ invokeLiveDataSource();
+ invokeLiveSession();
+ delete mFDP;
+}
+
+void HttpLiveFuzzer::deInitLiveSession() {
+ if (mLiveSession != nullptr) {
+ mLiveSession->disconnect();
+ mLiveLooper->unregisterHandler(mLiveSession->id());
+ mLiveLooper->stop();
+ }
+ mLiveSession.clear();
+ mLiveLooper.clear();
+}
+
+void HttpLiveFuzzer::signalEos() {
+ mEosReached = true;
+ {
+ std::lock_guard<std::mutex> waitForDownloadComplete(mDownloadCompleteMutex);
+ }
+ mConditionalVariable.notify_one();
+ return;
+}
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
+ HttpLiveFuzzer httpliveFuzzer;
+ httpliveFuzzer.process(data, size);
+ return 0;
+}
diff --git a/media/libstagefright/httplive/fuzzer/httplive_fuzzer.dict b/media/libstagefright/httplive/fuzzer/httplive_fuzzer.dict
new file mode 100644
index 0000000..703cc7e
--- /dev/null
+++ b/media/libstagefright/httplive/fuzzer/httplive_fuzzer.dict
@@ -0,0 +1,15 @@
+#m3u8-Tags
+kw1="#EXTM3U"
+kw2="#EXT-X-VERSION:"
+kw3="#EXT-X-TARGETDURATION:"
+kw4="#EXT-X-PLAYLIST-TYPE:"
+kw5="#EXTINF:"
+kw6="#EXT-X-ENDLIST"
+kw7="#EXT-X-MEDIA-SEQUENCE:"
+kw8="#EXT-X-KEY:METHOD=NONE"
+kw9="#EXT-X-DISCONTINUITY:"
+kw10="#EXT-X-DISCONTINUITY-SEQUENCE:0"
+kw11="#EXT-X-STREAM-INF:BANDWIDTH="
+kw12="#EXT-X-STREAM-INF:CODECS="
+kw13="#EXT-X-BYTERANGE:"
+kw14="#EXT-X-MEDIA"
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index c84cc10..632b32c 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -71,6 +71,9 @@
virtual void initiateSetInputSurface(const sp<PersistentSurface> &surface);
virtual void initiateStart();
virtual void initiateShutdown(bool keepComponentAllocated = false);
+ virtual status_t querySupportedParameters(std::vector<std::string> *names) override;
+ virtual status_t subscribeToParameters(const std::vector<std::string> &names) override;
+ virtual status_t unsubscribeFromParameters(const std::vector<std::string> &names) override;
status_t queryCapabilities(
const char* owner, const char* name,
diff --git a/media/libstagefright/mpeg2ts/Android.bp b/media/libstagefright/mpeg2ts/Android.bp
index a970224..fbfa8cc 100644
--- a/media/libstagefright/mpeg2ts/Android.bp
+++ b/media/libstagefright/mpeg2ts/Android.bp
@@ -27,11 +27,6 @@
"ESQueue.cpp",
],
- include_dirs: [
- "frameworks/av/media/libstagefright",
- "frameworks/native/include/media/openmax",
- ],
-
cflags: [
"-Werror",
"-Wall",
@@ -76,7 +71,6 @@
},
}
-
cc_library_static {
name: "libstagefright_mpeg2support",
defaults: [
diff --git a/media/libstagefright/mpeg2ts/test/Android.bp b/media/libstagefright/mpeg2ts/test/Android.bp
index 464b039..34a8d3e 100644
--- a/media/libstagefright/mpeg2ts/test/Android.bp
+++ b/media/libstagefright/mpeg2ts/test/Android.bp
@@ -57,11 +57,6 @@
"libstagefright_mpeg2support",
],
- include_dirs: [
- "frameworks/av/media/extractors/",
- "frameworks/av/media/libstagefright/",
- ],
-
header_libs: [
"libmedia_headers",
"libaudioclient_headers",
diff --git a/media/libstagefright/mpeg2ts/test/Mpeg2tsUnitTest.cpp b/media/libstagefright/mpeg2ts/test/Mpeg2tsUnitTest.cpp
index 79c233b..7f25d78 100644
--- a/media/libstagefright/mpeg2ts/test/Mpeg2tsUnitTest.cpp
+++ b/media/libstagefright/mpeg2ts/test/Mpeg2tsUnitTest.cpp
@@ -27,8 +27,8 @@
#include <media/stagefright/MetaDataBase.h>
#include <media/stagefright/foundation/AUtils.h>
-#include "mpeg2ts/ATSParser.h"
-#include "mpeg2ts/AnotherPacketSource.h"
+#include <ATSParser.h>
+#include <AnotherPacketSource.h>
#include "Mpeg2tsUnitTestEnvironment.h"
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index ffccbb1..0bd342a 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -508,8 +508,6 @@
if (n != (ssize_t)buffer->size()) {
ALOGW("failed to send RTCP TMMBR (%s).",
n >= 0 ? "connection gone" : strerror(errno));
-
- it = mStreams.erase(it);
continue;
}
}
@@ -560,8 +558,6 @@
if (n != (ssize_t)buffer->size()) {
ALOGW("failed to send RTCP receiver report (%s).",
n >= 0 ? "connection gone" : strerror(errno));
-
- it = mStreams.erase(it);
continue;
}
@@ -621,7 +617,14 @@
} while (nbytes < 0 && errno == EINTR);
if (nbytes <= 0) {
- return -ECONNRESET;
+ ALOGW("failed to recv rtp packet. cause=%s", strerror(errno));
+ // ECONNREFUSED may happen in next recvfrom() calling if one of
+ // outgoing packet can not be delivered to remote by using sendto()
+ if (errno == ECONNREFUSED) {
+ return -ECONNREFUSED;
+ } else {
+ return -ECONNRESET;
+ }
}
buffer->setRange(0, nbytes);
@@ -665,6 +668,10 @@
pRemoteRTCPAddr, sizeSockSt);
} while (n < 0 && errno == EINTR);
+ if (n < 0) {
+ ALOGW("failed to send rtcp packet. cause=%s", strerror(errno));
+ }
+
return n;
}
diff --git a/media/libstagefright/rtsp/JitterCalculator.cpp b/media/libstagefright/rtsp/JitterCalculator.cpp
index 93b5a83..7e60be2 100644
--- a/media/libstagefright/rtsp/JitterCalculator.cpp
+++ b/media/libstagefright/rtsp/JitterCalculator.cpp
@@ -38,14 +38,13 @@
mInterArrivalJitterUs = inter;
}
-void JitterCalc::putBaseData(int64_t rtpTime, int64_t arrivalTimeUs) {
- // A RTP time wraps around after UINT32_MAX. We must consider this case.
- const int64_t UINT32_MSB = 0x80000000;
- int64_t overflowMask = (mFirstTimeStamp & UINT32_MSB & ~rtpTime) << 1;
- int64_t tempRtpTime = overflowMask | rtpTime;
+void JitterCalc::putBaseData(uint32_t rtpTime, int64_t arrivalTimeUs) {
+ // A RTP time wraps around after UINT32_MAX. Overflow can present.
+ uint32_t diff = 0;
+ __builtin_usub_overflow(rtpTime, mFirstTimeStamp, &diff);
// Base jitter implementation can be various
- int64_t scheduledTimeUs = (tempRtpTime - (int64_t)mFirstTimeStamp) * 1000000ll / mClockRate;
+ int64_t scheduledTimeUs = ((int32_t)diff) * 1000000ll / mClockRate;
int64_t elapsedTimeUs = arrivalTimeUs - mFirstArrivalTimeUs;
int64_t correctionTimeUs = elapsedTimeUs - scheduledTimeUs; // additional propagation delay;
mBaseJitterUs = (mBaseJitterUs * 15 + correctionTimeUs) / 16;
@@ -53,18 +52,13 @@
(long long)mBaseJitterUs, (long long)correctionTimeUs);
}
-void JitterCalc::putInterArrivalData(int64_t rtpTime, int64_t arrivalTimeUs) {
- const int64_t UINT32_MSB = 0x80000000;
- int64_t tempRtpTime = rtpTime;
- int64_t tempLastTimeStamp = mLastTimeStamp;
-
- // A RTP time wraps around after UINT32_MAX. We must consider this case.
- int64_t overflowMask = (mLastTimeStamp ^ rtpTime) & UINT32_MSB;
- tempRtpTime |= ((overflowMask & ~rtpTime) << 1);
- tempLastTimeStamp |= ((overflowMask & ~mLastTimeStamp) << 1);
+void JitterCalc::putInterArrivalData(uint32_t rtpTime, int64_t arrivalTimeUs) {
+ // A RTP time wraps around after UINT32_MAX. Overflow can present.
+ uint32_t diff = 0;
+ __builtin_usub_overflow(rtpTime, mLastTimeStamp, &diff);
// 6.4.1 of RFC3550 defines this interarrival jitter value.
- int64_t diffTimeStampUs = abs(tempRtpTime - tempLastTimeStamp) * 1000000ll / mClockRate;
+ int64_t diffTimeStampUs = abs((int32_t)diff) * 1000000ll / mClockRate;
int64_t diffArrivalUs = arrivalTimeUs - mLastArrivalTimeUs; // Can't be minus
ALOGV("diffTimeStampUs %lld \t\t diffArrivalUs %lld",
(long long)diffTimeStampUs, (long long)diffArrivalUs);
@@ -72,7 +66,7 @@
int64_t varianceUs = diffArrivalUs - diffTimeStampUs;
mInterArrivalJitterUs = (mInterArrivalJitterUs * 15 + abs(varianceUs)) / 16;
- mLastTimeStamp = (uint32_t)rtpTime;
+ mLastTimeStamp = rtpTime;
mLastArrivalTimeUs = arrivalTimeUs;
}
diff --git a/media/libstagefright/rtsp/JitterCalculator.h b/media/libstagefright/rtsp/JitterCalculator.h
index ff36f1f..4f3b761 100644
--- a/media/libstagefright/rtsp/JitterCalculator.h
+++ b/media/libstagefright/rtsp/JitterCalculator.h
@@ -40,8 +40,8 @@
JitterCalc(int32_t clockRate);
void init(uint32_t rtpTime, int64_t arrivalTimeUs, int32_t base, int32_t inter);
- void putInterArrivalData(int64_t rtpTime, int64_t arrivalTime);
- void putBaseData(int64_t rtpTime, int64_t arrivalTimeUs);
+ void putInterArrivalData(uint32_t rtpTime, int64_t arrivalTime);
+ void putBaseData(uint32_t rtpTime, int64_t arrivalTimeUs);
int32_t getBaseJitterMs();
int32_t getInterArrivalJitterMs();
};
diff --git a/media/libstagefright/tests/HEVC/Android.bp b/media/libstagefright/tests/HEVC/Android.bp
index 91bf385..7a0ba52 100644
--- a/media/libstagefright/tests/HEVC/Android.bp
+++ b/media/libstagefright/tests/HEVC/Android.bp
@@ -44,10 +44,6 @@
"libstagefright_foundation",
],
- include_dirs: [
- "frameworks/av/media/libstagefright",
- ],
-
cflags: [
"-Werror",
"-Wall",
diff --git a/media/libstagefright/tests/HEVC/HEVCUtilsUnitTest.cpp b/media/libstagefright/tests/HEVC/HEVCUtilsUnitTest.cpp
index 324a042..c43e1f8 100644
--- a/media/libstagefright/tests/HEVC/HEVCUtilsUnitTest.cpp
+++ b/media/libstagefright/tests/HEVC/HEVCUtilsUnitTest.cpp
@@ -21,7 +21,7 @@
#include <fstream>
#include <media/stagefright/foundation/ABitReader.h>
-#include "include/HevcUtils.h"
+#include <HevcUtils.h>
#include "HEVCUtilsTestEnvironment.h"
diff --git a/media/libstagefright/tests/extractorFactory/Android.bp b/media/libstagefright/tests/extractorFactory/Android.bp
index 13d5b89..a067284 100644
--- a/media/libstagefright/tests/extractorFactory/Android.bp
+++ b/media/libstagefright/tests/extractorFactory/Android.bp
@@ -51,10 +51,6 @@
"libstagefright_foundation",
],
- include_dirs: [
- "frameworks/av/media/libstagefright",
- ],
-
// TODO: (b/150181583)
compile_multilib: "first",
diff --git a/media/libstagefright/tests/writer/Android.bp b/media/libstagefright/tests/writer/Android.bp
index 38d5ecc..49fb569 100644
--- a/media/libstagefright/tests/writer/Android.bp
+++ b/media/libstagefright/tests/writer/Android.bp
@@ -52,10 +52,6 @@
"libogg",
],
- include_dirs: [
- "frameworks/av/media/libstagefright",
- ],
-
cflags: [
"-Werror",
"-Wall",
diff --git a/media/libstagefright/tests/writer/WriterTest.cpp b/media/libstagefright/tests/writer/WriterTest.cpp
index d170e7c..398c592 100644
--- a/media/libstagefright/tests/writer/WriterTest.cpp
+++ b/media/libstagefright/tests/writer/WriterTest.cpp
@@ -36,7 +36,7 @@
#include <media/stagefright/MPEG2TSWriter.h>
#include <media/stagefright/MPEG4Writer.h>
#include <media/stagefright/OggWriter.h>
-#include <webm/WebmWriter.h>
+#include <WebmWriter.h>
#include "WriterTestEnvironment.h"
#include "WriterUtility.h"
diff --git a/media/libstagefright/webm/Android.bp b/media/libstagefright/webm/Android.bp
index 32a22ba..4209aea 100644
--- a/media/libstagefright/webm/Android.bp
+++ b/media/libstagefright/webm/Android.bp
@@ -33,7 +33,7 @@
"WebmWriter.cpp",
],
- include_dirs: ["frameworks/av/include"],
+ export_include_dirs: ["."],
shared_libs: [
"libdatasource",
@@ -44,6 +44,7 @@
],
header_libs: [
+ "av-headers",
"libmedia_headers",
"media_ndk_headers",
],
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index 1ae2b44..0e2de4e 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -452,17 +452,19 @@
uint32_t flags) {
sp<AMessage> nativeFormat;
AMediaFormat_getFormat(format, &nativeFormat);
- ALOGV("configure with format: %s", nativeFormat->debugString(0).c_str());
+ // create our shallow copy, so we aren't victim to any later changes.
+ sp<AMessage> dupNativeFormat = nativeFormat->dup();
+ ALOGV("configure with format: %s", dupNativeFormat->debugString(0).c_str());
sp<Surface> surface = NULL;
if (window != NULL) {
surface = (Surface*) window;
}
- status_t err = mData->mCodec->configure(nativeFormat, surface,
+ status_t err = mData->mCodec->configure(dupNativeFormat, surface,
crypto ? crypto->mCrypto : NULL, flags);
if (err != OK) {
ALOGE("configure: err(%d), failed with format: %s",
- err, nativeFormat->debugString(0).c_str());
+ err, dupNativeFormat->debugString(0).c_str());
}
return translate_error(err);
}
diff --git a/media/ndk/NdkMediaFormat.cpp b/media/ndk/NdkMediaFormat.cpp
index b035e5a..69ab242 100644
--- a/media/ndk/NdkMediaFormat.cpp
+++ b/media/ndk/NdkMediaFormat.cpp
@@ -200,8 +200,11 @@
AString tmp;
if (mData->mFormat->findString(name, &tmp)) {
String8 ret(tmp.c_str());
- mData->mStringCache.add(String8(name), ret);
- *out = ret.string();
+ ssize_t i = mData->mStringCache.add(String8(name), ret);
+ if (i < 0) {
+ return false;
+ }
+ *out = mData->mStringCache.valueAt(i).string();
return true;
}
return false;
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index 73c4e3b..6e29ac8 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -51,6 +51,7 @@
"libpermission",
"android.hardware.graphics.bufferqueue@1.0",
"android.hidl.token@1.0-utils",
+ "packagemanager_aidl-cpp",
],
export_static_lib_headers: [
"libbatterystats_aidl",
@@ -71,6 +72,7 @@
export_shared_lib_headers: [
"libpermission",
+ "packagemanager_aidl-cpp",
],
include_dirs: [
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index b91f302..f401fff 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -41,6 +41,7 @@
"FastThreadState.cpp",
"NBAIO_Tee.cpp",
"PatchPanel.cpp",
+ "PropertyUtils.cpp",
"SpdifStreamOut.cpp",
"StateQueue.cpp",
"Threads.cpp",
@@ -54,6 +55,7 @@
],
shared_libs: [
+ "android.media.audio.common.types-V1-cpp",
"audioflinger-aidl-cpp",
"audioclient-types-aidl-cpp",
"av-types-aidl-cpp",
@@ -81,6 +83,7 @@
"libmedia_helper",
"libshmemcompat",
"libvibrator",
+ "packagemanager_aidl-cpp",
],
static_libs: [
@@ -90,6 +93,7 @@
],
header_libs: [
+ "libaaudio_headers",
"libaudioclient_headers",
"libaudiohal_headers",
"libmedia_headers",
@@ -97,6 +101,7 @@
export_shared_lib_headers: [
"libpermission",
+ "packagemanager_aidl-cpp",
],
cflags: [
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 8a32b72..b359331 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -57,6 +57,7 @@
#include "AudioFlinger.h"
#include "NBAIO_Tee.h"
+#include "PropertyUtils.h"
#include <media/AudioResamplerPublic.h>
@@ -64,6 +65,7 @@
#include <system/audio_effects/effect_ns.h>
#include <system/audio_effects/effect_aec.h>
#include <system/audio_effects/effect_hapticgenerator.h>
+#include <system/audio_effects/effect_spatializer.h>
#include <audio_utils/primitives.h>
@@ -102,7 +104,11 @@
namespace android {
+#define MAX_AAUDIO_PROPERTY_DEVICE_HAL_VERSION 7.0
+
using media::IEffectClient;
+using media::audio::common::AudioMMapPolicyInfo;
+using media::audio::common::AudioMMapPolicyType;
using android::content::AttributionSourceState;
static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
@@ -300,6 +306,11 @@
mDevicesFactoryHalCallback = new DevicesFactoryHalCallbackImpl;
mDevicesFactoryHal->setCallbackOnce(mDevicesFactoryHalCallback);
+
+ if (mDevicesFactoryHal->getHalVersion() <= MAX_AAUDIO_PROPERTY_DEVICE_HAL_VERSION) {
+ mAAudioBurstsPerBuffer = getAAudioMixerBurstCountFromSystemProperty();
+ mAAudioHwBurstMinMicros = getAAudioHardwareBurstMinUsecFromSystemProperty();
+ }
}
status_t AudioFlinger::setAudioHalPids(const std::vector<pid_t>& pids) {
@@ -335,6 +346,44 @@
return NO_ERROR;
}
+status_t AudioFlinger::getMmapPolicyInfos(
+ AudioMMapPolicyType policyType, std::vector<AudioMMapPolicyInfo> *policyInfos) {
+ Mutex::Autolock _l(mLock);
+ if (const auto it = mPolicyInfos.find(policyType); it != mPolicyInfos.end()) {
+ *policyInfos = it->second;
+ return NO_ERROR;
+ }
+ if (mDevicesFactoryHal->getHalVersion() > MAX_AAUDIO_PROPERTY_DEVICE_HAL_VERSION) {
+ AutoMutex lock(mHardwareLock);
+ for (size_t i = 0; i < mAudioHwDevs.size(); ++i) {
+ AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
+ std::vector<AudioMMapPolicyInfo> infos;
+ status_t status = dev->getMmapPolicyInfos(policyType, &infos);
+ if (status != NO_ERROR) {
+ ALOGE("Failed to query mmap policy info of %d, error %d",
+ mAudioHwDevs.keyAt(i), status);
+ continue;
+ }
+ policyInfos->insert(policyInfos->end(), infos.begin(), infos.end());
+ }
+ mPolicyInfos[policyType] = *policyInfos;
+ } else {
+ getMmapPolicyInfosFromSystemProperty(policyType, policyInfos);
+ mPolicyInfos[policyType] = *policyInfos;
+ }
+ return NO_ERROR;
+}
+
+int32_t AudioFlinger::getAAudioMixerBurstCount() {
+ Mutex::Autolock _l(mLock);
+ return mAAudioBurstsPerBuffer;
+}
+
+int32_t AudioFlinger::getAAudioHardwareBurstMinUsec() {
+ Mutex::Autolock _l(mLock);
+ return mAAudioHwBurstMinMicros;
+}
+
// getDefaultVibratorInfo_l must be called with AudioFlinger lock held.
std::optional<media::AudioVibratorInfo> AudioFlinger::getDefaultVibratorInfo_l() {
if (mAudioVibratorInfos.empty()) {
@@ -566,10 +615,12 @@
String8 result;
result.append("Clients:\n");
+ result.append(" pid heap_size\n");
for (size_t i = 0; i < mClients.size(); ++i) {
sp<Client> client = mClients.valueAt(i).promote();
if (client != 0) {
- result.appendFormat(" pid: %d\n", client->pid());
+ result.appendFormat("%6d %12zu\n", client->pid(),
+ client->heap()->getMemoryHeap()->getSize());
}
}
@@ -2287,6 +2338,17 @@
mHardwareStatus = AUDIO_HW_IDLE;
}
+ if (mDevicesFactoryHal->getHalVersion() > MAX_AAUDIO_PROPERTY_DEVICE_HAL_VERSION) {
+ if (int32_t mixerBursts = dev->getAAudioMixerBurstCount();
+ mixerBursts > mAAudioBurstsPerBuffer) {
+ mAAudioBurstsPerBuffer = mixerBursts;
+ }
+ if (int32_t hwBurstMinMicros = dev->getAAudioHardwareBurstMinUsec();
+ hwBurstMinMicros < mAAudioHwBurstMinMicros || mAAudioHwBurstMinMicros == 0) {
+ mAAudioHwBurstMinMicros = hwBurstMinMicros;
+ }
+ }
+
mAudioHwDevs.add(handle, audioDevice);
ALOGI("loadHwModule() Loaded %s audio interface, handle %d", name, handle);
@@ -2857,7 +2919,7 @@
&config,
device.mType,
device.address().c_str(),
- VALUE_OR_RETURN_STATUS(aidl2legacy_AudioSourceType_audio_source_t(request.source)),
+ VALUE_OR_RETURN_STATUS(aidl2legacy_AudioSource_audio_source_t(request.source)),
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_input_flags_t_mask(request.flags)),
AUDIO_DEVICE_NONE,
String8{});
@@ -3757,6 +3819,15 @@
goto Exit;
}
+ // Only audio policy service can create a spatializer effect
+ if ((memcmp(&descOut.type, FX_IID_SPATIALIZER, sizeof(effect_uuid_t)) == 0) &&
+ (callingUid != AID_AUDIOSERVER || currentPid != getpid())) {
+ ALOGW("%s: attempt to create a spatializer effect from uid/pid %d/%d",
+ __func__, callingUid, currentPid);
+ lStatus = PERMISSION_DENIED;
+ goto Exit;
+ }
+
if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
// if the output returned by getOutputForEffect() is removed before we lock the
// mutex below, the call to checkPlaybackThread_l(io) below will detect it
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index d6bf0ae..8c546cc 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -282,6 +282,14 @@
virtual status_t updateSecondaryOutputs(
const TrackSecondaryOutputsMap& trackSecondaryOutputs);
+ virtual status_t getMmapPolicyInfos(
+ media::audio::common::AudioMMapPolicyType policyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos);
+
+ virtual int32_t getAAudioMixerBurstCount();
+
+ virtual int32_t getAAudioHardwareBurstMinUsec();
+
status_t onTransactWrapper(TransactionCode code, const Parcel& data, uint32_t flags,
const std::function<status_t()>& delegate) override;
@@ -1004,6 +1012,11 @@
// Keep in sync with java definition in media/java/android/media/AudioRecord.java
static constexpr int32_t kMaxSharedAudioHistoryMs = 5000;
+
+ std::map<media::audio::common::AudioMMapPolicyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo>> mPolicyInfos;
+ int32_t mAAudioBurstsPerBuffer = 0;
+ int32_t mAAudioHwBurstMinMicros = 0;
};
#undef INCLUDING_FROM_AUDIOFLINGER_H
diff --git a/services/audioflinger/AudioHwDevice.cpp b/services/audioflinger/AudioHwDevice.cpp
index 16b25f6..dee6161 100644
--- a/services/audioflinger/AudioHwDevice.cpp
+++ b/services/audioflinger/AudioHwDevice.cpp
@@ -29,6 +29,9 @@
namespace android {
+using media::audio::common::AudioMMapPolicyInfo;
+using media::audio::common::AudioMMapPolicyType;
+
// ----------------------------------------------------------------------------
status_t AudioHwDevice::openOutputStream(
@@ -102,5 +105,18 @@
return mHwDevice->getAudioPort(port);
}
+status_t AudioHwDevice::getMmapPolicyInfos(
+ AudioMMapPolicyType policyType, std::vector<AudioMMapPolicyInfo> *policyInfos) const {
+ return mHwDevice->getMmapPolicyInfos(policyType, policyInfos);
+}
+
+int32_t AudioHwDevice::getAAudioMixerBurstCount() const {
+ return mHwDevice->getAAudioMixerBurstCount();
+}
+
+int32_t AudioHwDevice::getAAudioHardwareBurstMinUsec() const {
+ return mHwDevice->getAAudioHardwareBurstMinUsec();
+}
+
}; // namespace android
diff --git a/services/audioflinger/AudioHwDevice.h b/services/audioflinger/AudioHwDevice.h
index fc2c693..8c5d239 100644
--- a/services/audioflinger/AudioHwDevice.h
+++ b/services/audioflinger/AudioHwDevice.h
@@ -22,6 +22,8 @@
#include <stdlib.h>
#include <sys/types.h>
+#include <android/media/audio/common/AudioMMapPolicyInfo.h>
+#include <android/media/audio/common/AudioMMapPolicyType.h>
#include <media/audiohal/DeviceHalInterface.h>
#include <utils/Errors.h>
#include <system/audio.h>
@@ -85,6 +87,14 @@
status_t getAudioPort(struct audio_port_v7 *port) const;
+ status_t getMmapPolicyInfos(
+ media::audio::common::AudioMMapPolicyType policyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos) const;
+
+ int32_t getAAudioMixerBurstCount() const;
+
+ int32_t getAAudioHardwareBurstMinUsec() const;
+
private:
const audio_module_handle_t mHandle;
const char * const mModuleName;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index e9e98ca..b069462 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -195,6 +195,10 @@
audio_output_flags_t getOutputFlags() const { return mFlags; }
float getSpeed() const { return mSpeed; }
+
+ bool canBeSpatialized() const { return (mAttr.flags
+ & (AUDIO_FLAG_CONTENT_SPATIALIZED | AUDIO_FLAG_NEVER_SPATIALIZE)) == 0; }
+
protected:
// for numerous
friend class PlaybackThread;
diff --git a/services/audioflinger/PropertyUtils.cpp b/services/audioflinger/PropertyUtils.cpp
new file mode 100644
index 0000000..65e2533
--- /dev/null
+++ b/services/audioflinger/PropertyUtils.cpp
@@ -0,0 +1,104 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <aaudio/AAudio.h>
+#include <aaudio/AAudioTesting.h>
+#include <android/media/audio/common/AudioMMapPolicy.h>
+#include <cutils/properties.h>
+
+#include "PropertyUtils.h"
+
+namespace android {
+
+using media::audio::common::AudioMMapPolicy;
+using media::audio::common::AudioMMapPolicyType;
+using media::audio::common::AudioMMapPolicyInfo;
+
+std::string getMmapPolicyProperty(AudioMMapPolicyType policyType) {
+ switch (policyType) {
+ case AudioMMapPolicyType::DEFAULT:
+ return "aaudio.mmap_policy";
+ case AudioMMapPolicyType::EXCLUSIVE:
+ return "aaudio.mmap_exclusive_policy";
+ default:
+ return "";
+ }
+}
+
+int getDefaultPolicyFromType(AudioMMapPolicyType policyType) {
+ switch (policyType) {
+ case AudioMMapPolicyType::EXCLUSIVE:
+ return AAUDIO_UNSPECIFIED;
+ case AudioMMapPolicyType::DEFAULT:
+ default:
+ return AAUDIO_POLICY_NEVER;
+ }
+}
+
+AudioMMapPolicy legacy2aidl_aaudio_policy_t_AudioMMapPolicy(aaudio_policy_t legacy) {
+ switch (legacy) {
+ case AAUDIO_POLICY_NEVER:
+ return AudioMMapPolicy::NEVER;
+ case AAUDIO_POLICY_AUTO:
+ return AudioMMapPolicy::AUTO;
+ case AAUDIO_POLICY_ALWAYS:
+ return AudioMMapPolicy::ALWAYS;
+ case AAUDIO_UNSPECIFIED:
+ return AudioMMapPolicy::UNSPECIFIED;
+ default:
+ ALOGE("%s unknown aaudio policy: %d", __func__, legacy);
+ return AudioMMapPolicy::UNSPECIFIED;
+ }
+}
+
+status_t getMmapPolicyInfosFromSystemProperty(
+ AudioMMapPolicyType policyType, std::vector<AudioMMapPolicyInfo> *policyInfos) {
+ AudioMMapPolicyInfo policyInfo;
+ const std::string propertyStr = getMmapPolicyProperty(policyType);
+ if (propertyStr.empty()) {
+ return BAD_VALUE;
+ }
+ policyInfo.mmapPolicy = legacy2aidl_aaudio_policy_t_AudioMMapPolicy(
+ property_get_int32(propertyStr.c_str(), getDefaultPolicyFromType(policyType)));
+ policyInfos->push_back(policyInfo);
+ return NO_ERROR;
+}
+
+int32_t getAAudioMixerBurstCountFromSystemProperty() {
+ static const int32_t sDefaultBursts = 2; // arbitrary, use 2 for double buffered
+ static const int32_t sMaxBursts = 1024; // arbitrary
+ static const char* sPropMixerBursts = "aaudio.mixer_bursts";
+ int32_t prop = property_get_int32(sPropMixerBursts, sDefaultBursts);
+ if (prop <= 0 || prop > sMaxBursts) {
+ ALOGE("%s: invalid value %d, use default %d", __func__, prop, sDefaultBursts);
+ prop = sDefaultBursts;
+ }
+ return prop;
+}
+
+int32_t getAAudioHardwareBurstMinUsecFromSystemProperty() {
+ static const int32_t sDefaultMicros = 1000; // arbitrary
+ static const int32_t sMaxMicros = 1000 * 1000; // arbitrary
+ static const char* sPropHwBurstMinUsec = "aaudio.hw_burst_min_usec";
+ int32_t prop = property_get_int32(sPropHwBurstMinUsec, sDefaultMicros);
+ if (prop <= 0 || prop > sMaxMicros) {
+ ALOGE("%s invalid value %d, use default %d", __func__, prop, sDefaultMicros);
+ prop = sDefaultMicros;
+ }
+ return prop;
+}
+
+} // namespace android
diff --git a/services/audioflinger/PropertyUtils.h b/services/audioflinger/PropertyUtils.h
new file mode 100644
index 0000000..fbf651a
--- /dev/null
+++ b/services/audioflinger/PropertyUtils.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <android/media/audio/common/AudioMMapPolicyType.h>
+#include <android/media/audio/common/AudioMMapPolicyInfo.h>
+
+namespace android {
+
+status_t getMmapPolicyInfosFromSystemProperty(
+ media::audio::common::AudioMMapPolicyType policyType,
+ std::vector<media::audio::common::AudioMMapPolicyInfo> *policyInfos);
+
+int32_t getAAudioMixerBurstCountFromSystemProperty();
+
+int32_t getAAudioHardwareBurstMinUsecFromSystemProperty();
+
+} // namespace android
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 9242f89..a14c1dd 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1343,6 +1343,13 @@
return BAD_VALUE;
}
+ if (memcmp(&desc->type, FX_IID_SPATIALIZER, sizeof(effect_uuid_t)) == 0
+ && mType != SPATIALIZER) {
+ ALOGW("%s: attempt to create a spatializer effect on a thread of type %d",
+ __func__, mType);
+ return BAD_VALUE;
+ }
+
switch (mType) {
case MIXER: {
#ifndef MULTICHANNEL_EFFECT_CHAIN
@@ -2052,6 +2059,7 @@
free(mSinkBuffer);
free(mMixerBuffer);
free(mEffectBuffer);
+ free(mEffectToSinkBuffer);
}
// Thread virtuals
@@ -3004,11 +3012,18 @@
// Originally this was int16_t[] array, need to remove legacy implications.
free(mSinkBuffer);
mSinkBuffer = NULL;
+ free(mEffectToSinkBuffer);
+ mEffectToSinkBuffer = nullptr;
+
// For sink buffer size, we use the frame size from the downstream sink to avoid problems
// with non PCM formats for compressed music, e.g. AAC, and Offload threads.
const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
(void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
+ if (mType == SPATIALIZER) {
+ (void)posix_memalign(&mEffectToSinkBuffer, 32, sinkBufferSize);
+ }
+
// We resize the mMixerBuffer according to the requirements of the sink buffer which
// drives the output.
free(mMixerBuffer);
@@ -3839,11 +3854,11 @@
//
// mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
// TODO use mSleepTimeUs == 0 as an additional condition.
+ uint32_t mixerChannelCount = mEffectBufferValid ?
+ audio_channel_count_from_out_mask(mMixerChannelMask) : mChannelCount;
if (mMixerBufferValid) {
void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
- uint32_t channelCount = mEffectBufferValid ?
- audio_channel_count_from_out_mask(mMixerChannelMask) : mChannelCount;
// mono blend occurs for mixer threads only (not direct or offloaded)
// and is handled here if we're going directly to the sink.
@@ -3861,7 +3876,7 @@
}
memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
- mNormalFrameCount * (channelCount + mHapticChannelCount));
+ mNormalFrameCount * (mixerChannelCount + mHapticChannelCount));
// If we're going directly to the sink and there are haptic channels,
// we should adjust channels as the sample data is partially interleaved
@@ -3894,8 +3909,11 @@
&& activeHapticSessionId == effectChains[i]->sessionId()) {
// Haptic data is active in this case, copy it directly from
// in buffer to out buffer.
+ uint32_t channelCount =
+ effectChains[i]->sessionId() == AUDIO_SESSION_OUTPUT_STAGE ?
+ mixerChannelCount : mChannelCount;
const size_t audioBufferSize = mNormalFrameCount
- * audio_bytes_per_frame(mChannelCount, EFFECT_BUFFER_FORMAT);
+ * audio_bytes_per_frame(channelCount, EFFECT_BUFFER_FORMAT);
memcpy_by_audio_format(
(uint8_t*)effectChains[i]->outBuffer() + audioBufferSize,
EFFECT_BUFFER_FORMAT,
@@ -3935,8 +3953,23 @@
mBalance.process((float *)mEffectBuffer, mNormalFrameCount);
}
- memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
- mNormalFrameCount * (mChannelCount + mHapticChannelCount));
+ if (mType == SPATIALIZER) {
+ memcpy_by_audio_format(mEffectToSinkBuffer, mFormat, mEffectBuffer,
+ mEffectBufferFormat,
+ mNormalFrameCount * (mChannelCount + mHapticChannelCount));
+ accumulate_by_audio_format(mSinkBuffer, mEffectToSinkBuffer, mFormat,
+ mNormalFrameCount * mChannelCount);
+ const size_t audioBufferSize = mNormalFrameCount
+ * audio_bytes_per_frame(mChannelCount, mFormat);
+ memcpy_by_audio_format(
+ (uint8_t*)mSinkBuffer + audioBufferSize,
+ mFormat,
+ (uint8_t*)mEffectToSinkBuffer + audioBufferSize,
+ mFormat, mNormalFrameCount * mHapticChannelCount);
+ } else {
+ memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
+ mNormalFrameCount * (mChannelCount + mHapticChannelCount));
+ }
// The sample data is partially interleaved when haptic channels exist,
// we need to adjust channels here.
if (mHapticChannelCount > 0) {
@@ -4598,6 +4631,7 @@
initFastMixer = mFrameCount < mNormalFrameCount;
break;
}
+ ALOG_ASSERT(initFastMixer && mType == SPATIALIZER);
ALOGW_IF(initFastMixer == false && mFrameCount < mNormalFrameCount,
"FastMixer is preferred for this sink as frameCount %zu is less than threshold %zu",
mFrameCount, mNormalFrameCount);
@@ -4956,6 +4990,9 @@
// before effects processing or output.
if (mMixerBufferValid) {
memset(mMixerBuffer, 0, mMixerBufferSize);
+ if (mType == SPATIALIZER) {
+ memset(mSinkBuffer, 0, mSinkBufferSize);
+ }
} else {
memset(mSinkBuffer, 0, mSinkBufferSize);
}
@@ -5184,7 +5221,7 @@
break;
case TrackBase::IDLE:
default:
- LOG_ALWAYS_FATAL("unexpected track state %d", track->mState);
+ LOG_ALWAYS_FATAL("unexpected track state %d", (int)track->mState);
}
if (isActive) {
@@ -5244,7 +5281,7 @@
// TODO Remove the ALOGW when this theory is confirmed.
ALOGW("fast track %d should have been active; "
"mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d",
- j, track->mState, state->mTrackMask, recentUnderruns,
+ j, (int)track->mState, state->mTrackMask, recentUnderruns,
track->sharedBuffer() != 0);
// Since the FastMixer state already has the track inactive, do nothing here.
}
@@ -5448,11 +5485,21 @@
trackId,
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
- mAudioMixer->setParameter(
- trackId,
- AudioMixer::TRACK,
- AudioMixer::MIXER_CHANNEL_MASK,
- (void *)(uintptr_t)(mMixerChannelMask | mHapticChannelMask));
+
+ if (mType == SPATIALIZER && !track->canBeSpatialized()) {
+ mAudioMixer->setParameter(
+ trackId,
+ AudioMixer::TRACK,
+ AudioMixer::MIXER_CHANNEL_MASK,
+ (void *)(uintptr_t)(mChannelMask | mHapticChannelMask));
+ } else {
+ mAudioMixer->setParameter(
+ trackId,
+ AudioMixer::TRACK,
+ AudioMixer::MIXER_CHANNEL_MASK,
+ (void *)(uintptr_t)(mMixerChannelMask | mHapticChannelMask));
+ }
+
// limit track sample rate to 2 x output sample rate, which changes at re-configuration
uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
uint32_t reqSampleRate = proxy->getSampleRate();
@@ -5489,16 +5536,27 @@
if (mMixerBufferEnabled
&& (track->mainBuffer() == mSinkBuffer
|| track->mainBuffer() == mMixerBuffer)) {
- mAudioMixer->setParameter(
- trackId,
- AudioMixer::TRACK,
- AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
- mAudioMixer->setParameter(
- trackId,
- AudioMixer::TRACK,
- AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
- // TODO: override track->mainBuffer()?
- mMixerBufferValid = true;
+ if (mType == SPATIALIZER && !track->canBeSpatialized()) {
+ mAudioMixer->setParameter(
+ trackId,
+ AudioMixer::TRACK,
+ AudioMixer::MIXER_FORMAT, (void *)mFormat);
+ mAudioMixer->setParameter(
+ trackId,
+ AudioMixer::TRACK,
+ AudioMixer::MAIN_BUFFER, (void *)mSinkBuffer);
+ } else {
+ mAudioMixer->setParameter(
+ trackId,
+ AudioMixer::TRACK,
+ AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
+ mAudioMixer->setParameter(
+ trackId,
+ AudioMixer::TRACK,
+ AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
+ // TODO: override track->mainBuffer()?
+ mMixerBufferValid = true;
+ }
} else {
mAudioMixer->setParameter(
trackId,
@@ -5688,8 +5746,10 @@
// sink or mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to the sink or mix buffer
// and track effects will accumulate into it
- if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
- (mixedTracks == 0 && fastTracks > 0))) {
+ // always clear sink buffer for spatializer output as the output of the spatializer
+ // effect will be accumulated into it
+ if ((mBytesRemaining == 0) && (((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
+ (mixedTracks == 0 && fastTracks > 0)) || (mType == SPATIALIZER))) {
// FIXME as a performance optimization, should remember previous zero status
if (mMixerBufferValid) {
memset(mMixerBuffer, 0, mMixerBufferSize);
@@ -8205,7 +8265,7 @@
ALOGV("active record track PAUSING -> ACTIVE");
recordTrack->mState = TrackBase::ACTIVE;
} else {
- ALOGV("active record track state %d", recordTrack->mState);
+ ALOGV("active record track state %d", (int)recordTrack->mState);
}
return status;
}
@@ -8231,7 +8291,7 @@
}
if (recordTrack->mState != TrackBase::STARTING_1) {
ALOGW("%s(%d): unsynchronized mState:%d change",
- __func__, recordTrack->id(), recordTrack->mState);
+ __func__, recordTrack->id(), (int)recordTrack->mState);
// Someone else has changed state, let them take over,
// leave mState in the new state.
recordTrack->clearSyncStartEvent();
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 300171e..5755c42 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1123,6 +1123,11 @@
// for any processing (including output processing).
bool mEffectBufferValid;
+ // Frame size aligned buffer used to convert mEffectBuffer samples to mSinkBuffer format prior
+ // to accumulate into mSinkBuffer on SPATIALIZER threads
+ void* mEffectToSinkBuffer = nullptr;
+
+
// suspend count, > 0 means suspended. While suspended, the thread continues to pull from
// tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle
// concurrent use of both of them, so Audio Policy Service suspends one of the threads to
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 92f129c..5311fe2 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -23,7 +23,7 @@
class TrackBase : public ExtendedAudioBufferProvider, public RefBase {
public:
- enum track_state {
+ enum track_state : int32_t {
IDLE,
FLUSHED, // for PlaybackTracks only
STOPPED,
@@ -271,6 +271,7 @@
void releaseCblk() {
if (mCblk != nullptr) {
+ mState.clear();
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
if (mClient == 0) {
free(mCblk);
@@ -355,7 +356,7 @@
// except for OutputTrack when it is in local memory
size_t mBufferSize; // size of mBuffer in bytes
// we don't really need a lock for these
- track_state mState;
+ MirroredVariable<track_state> mState;
const audio_attributes_t mAttr;
const uint32_t mSampleRate; // initial sample rate only; for tracks which
// support dynamic rates, the current value is in control block
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index d2a30b1..e0c5fa5 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -234,7 +234,11 @@
#ifdef TEE_SINK
mTee.set(sampleRate, mChannelCount, format, NBAIO_Tee::TEE_FLAG_TRACK);
#endif
-
+ // mState is mirrored for the client to read.
+ mState.setMirror(&mCblk->mState);
+ // ensure our state matches up until we consolidate the enumeration.
+ static_assert(CBLK_STATE_IDLE == IDLE);
+ static_assert(CBLK_STATE_PAUSING == PAUSING);
}
}
@@ -933,7 +937,7 @@
buffer->raw = buf.mRaw;
if (buf.mFrameCount == 0 && !isStopping() && !isStopped() && !isPaused() && !isOffloaded()) {
ALOGV("%s(%d): underrun, framesReady(%zu) < framesDesired(%zd), state: %d",
- __func__, mId, buf.mFrameCount, desiredFrames, mState);
+ __func__, mId, buf.mFrameCount, desiredFrames, (int)mState);
mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
} else {
mAudioTrackServerProxy->tallyUnderrunFrames(0);
@@ -1590,7 +1594,7 @@
(mState == STOPPED)))) {
ALOGW("%s(%d): in invalid state %d on session %d %s mode, framesReady %zu",
__func__, mId,
- mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
+ (int)mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
event->cancel();
return INVALID_OPERATION;
}
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 8613b2e..add0684 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -32,30 +32,42 @@
// ----------------------------------------------------------------------------
-// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces
-// between the platform specific audio policy manager and Android generic audio policy manager.
-// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class.
+// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication
+// interfaces between the platform specific audio policy manager and Android generic audio policy
+// manager.
+// The platform specific audio policy manager must implement methods of the AudioPolicyInterface
+// class.
// This implementation makes use of the AudioPolicyClientInterface to control the activity and
// configuration of audio input and output streams.
//
// The platform specific audio policy manager is in charge of the audio routing and volume control
// policies for a given platform.
// The main roles of this module are:
-// - keep track of current system state (removable device connections, phone state, user requests...).
-// System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface.
+// - keep track of current system state (removable device connections, phone state,
+// user requests...).
+// System state changes and user actions are notified to audio policy manager with methods of the
+// AudioPolicyInterface.
// - process getOutput() queries received when AudioTrack objects are created: Those queries
-// return a handler on an output that has been selected, configured and opened by the audio policy manager and that
-// must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method.
-// When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide
-// to close or reconfigure the output depending on other streams using this output and current system state.
-// - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs.
-// - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value
-// applicable to each output as a function of platform specific settings and current output route (destination device). It
-// also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries).
+// return a handler on an output that has been selected, configured and opened by the audio
+// policy manager and that must be used by the AudioTrack when registering to the AudioFlinger
+// with the createTrack() method.
+// When the AudioTrack object is released, a putOutput() query is received and the audio policy
+// manager can decide to close or reconfigure the output depending on other streams using this
+// output and current system state.
+// - similarly process getInput() and putInput() queries received from AudioRecord objects and
+// configure audio inputs.
+// - process volume control requests: the stream volume is converted from an index value
+// (received from UI) to a float value applicable to each output as a function of platform
+// specificsettings and current output route (destination device). It also make sure that streams
+// are not muted if not allowed (e.g. camera shutter sound in some countries).
//
-// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so)
-// and is linked with libaudioflinger.so
-
+// The platform specific audio policy manager is provided as a shared library by platform vendors
+// (as for libaudio.so) and is linked with libaudioflinger.so
+//
+// NOTE: by convention, the implementation of the AudioPolicyInterface in AudioPolicyManager does
+// not have to perform any nullptr check on input arguments: The caller of this API is
+// AudioPolicyService running in the same process and in charge of validating arguments received
+// from incoming binder calls before calling AudioPolicyManager.
// Audio Policy Manager Interface
class AudioPolicyInterface
@@ -100,7 +112,7 @@
audio_format_t encodedFormat) = 0;
// retrieve a device connection status
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
- const char *device_address) = 0;
+ const char *device_address) = 0;
// indicate a change in device configuration
virtual status_t handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
@@ -134,9 +146,11 @@
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs,
output_type_t *outputType) = 0;
- // indicates to the audio policy manager that the output starts being used by corresponding stream.
+ // indicates to the audio policy manager that the output starts being used by corresponding
+ // stream.
virtual status_t startOutput(audio_port_handle_t portId) = 0;
- // indicates to the audio policy manager that the output stops being used by corresponding stream.
+ // indicates to the audio policy manager that the output stops being used by corresponding
+ // stream.
virtual status_t stopOutput(audio_port_handle_t portId) = 0;
// releases the output, return true if the output descriptor is reopened.
virtual bool releaseOutput(audio_port_handle_t portId) = 0;
@@ -398,10 +412,13 @@
// Audio output Control functions
//
- // opens an audio output with the requested parameters. The parameter values can indicate to use the default values
- // in case the audio policy manager has no specific requirements for the output being opened.
- // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream.
- // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
+ // opens an audio output with the requested parameters. The parameter values can indicate to
+ // use the default values in case the audio policy manager has no specific requirements for the
+ // output being opened.
+ // When the function returns, the parameter values reflect the actual values used by the audio
+ // hardware output stream.
+ // The audio policy manager can check if the proposed parameters are suitable or not and act
+ // accordingly.
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *halConfig,
@@ -409,13 +426,15 @@
const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags) = 0;
- // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
- // a special mixer thread in the AudioFlinger.
- virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0;
+ // creates a special output that is duplicated to the two outputs passed as arguments.
+ // The duplication is performed by a special mixer thread in the AudioFlinger.
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2) = 0;
// closes the output stream
virtual status_t closeOutput(audio_io_handle_t output) = 0;
- // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in
- // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded.
+ // suspends the output. When an output is suspended, the corresponding audio hardware output
+ // stream is placed in standby and the AudioTracks attached to the mixer thread are still
+ // processed but the output mix is discarded.
virtual status_t suspendOutput(audio_io_handle_t output) = 0;
// restores a suspended output.
virtual status_t restoreOutput(audio_io_handle_t output) = 0;
@@ -438,16 +457,21 @@
// misc control functions
//
- // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
+ // set a stream volume for a particular output. For the same user setting, a given stream type
+ // can have different volumes
// for each output (destination device) it is attached to.
- virtual status_t setStreamVolume(audio_stream_type_t stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0;
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float volume,
+ audio_io_handle_t output, int delayMs = 0) = 0;
// invalidate a stream type, causing a reroute to an unspecified new output
virtual status_t invalidateStream(audio_stream_type_t stream) = 0;
- // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
- virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0) = 0;
- // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
+ // function enabling to send proprietary informations directly from audio policy manager to
+ // audio hardware interface.
+ virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs,
+ int delayMs = 0) = 0;
+ // function enabling to receive proprietary informations directly from audio hardware interface
+ // to audio policy manager.
virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0;
// set down link audio volume.
@@ -510,7 +534,8 @@
// These are the signatures of createAudioPolicyManager/destroyAudioPolicyManager
// methods respectively, expected by AudioPolicyService, needs to be exposed by
// libaudiopolicymanagercustom.
- using CreateAudioPolicyManagerInstance = AudioPolicyInterface* (*)(AudioPolicyClientInterface*);
+ using CreateAudioPolicyManagerInstance =
+ AudioPolicyInterface* (*)(AudioPolicyClientInterface*);
using DestroyAudioPolicyManagerInstance = void (*)(AudioPolicyInterface*);
} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 7c7f02d..123011a 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -307,6 +307,8 @@
DeviceVector mDevices; /**< current devices this output is routed to */
wp<AudioPolicyMix> mPolicyMix; // non NULL when used by a dynamic policy
+ virtual uint32_t getRecommendedMuteDurationMs() const { return 0; }
+
protected:
const sp<PolicyAudioPort> mPolicyAudioPort;
AudioPolicyClientInterface * const mClientInterface;
@@ -415,6 +417,8 @@
*/
DeviceVector filterSupportedDevices(const DeviceVector &devices) const;
+ uint32_t getRecommendedMuteDurationMs() const override;
+
const sp<IOProfile> mProfile; // I/O profile this output derives from
audio_io_handle_t mIoHandle; // output handle
uint32_t mLatency; //
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index a74cefa..81828ed 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -207,6 +207,8 @@
// Number of streams currently active for this profile. This is not the number of active clients
// (AudioTrack or AudioRecord) but the number of active HAL streams.
uint32_t curActiveCount;
+ // Mute duration while changing device on this output profile.
+ uint32_t recommendedMuteDurationMs = 0;
private:
DeviceVector mSupportedDevices; // supported devices: this input/output can be routed from/to
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index f3d2326..34b5c1a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -670,6 +670,15 @@
return NO_ERROR;
}
+uint32_t SwAudioOutputDescriptor::getRecommendedMuteDurationMs() const
+{
+ if (isDuplicated()) {
+ return std::max(mOutput1->getRecommendedMuteDurationMs(),
+ mOutput2->getRecommendedMuteDurationMs());
+ }
+ return mProfile->recommendedMuteDurationMs;
+}
+
// HwAudioOutputDescriptor implementation
HwAudioOutputDescriptor::HwAudioOutputDescriptor(const sp<SourceClientDescriptor>& source,
AudioPolicyClientInterface *clientInterface)
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 1722032..c9c8ede 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -58,15 +58,6 @@
mDeclaredAddress(DeviceDescriptorBase::address())
{
mCurrentEncodedFormat = AUDIO_FORMAT_DEFAULT;
- /* If framework runs against a pre 5.0 Audio HAL, encoded formats are absent from the config.
- * FIXME: APM should know the version of the HAL and don't add the formats for V5.0.
- * For now, the workaround to remove AC3 and IEC61937 support on HDMI is to declare
- * something like 'encodedFormats="AUDIO_FORMAT_PCM_16_BIT"' on the HDMI devicePort.
- */
- if (mDeviceTypeAddr.mType == AUDIO_DEVICE_OUT_HDMI && mEncodedFormats.empty()) {
- mEncodedFormats.push_back(AUDIO_FORMAT_AC3);
- mEncodedFormats.push_back(AUDIO_FORMAT_IEC61937);
- }
}
void DeviceDescriptor::attach(const sp<HwModule>& module)
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index 09b614d..624ad95 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -137,6 +137,7 @@
maxOpenCount, curOpenCount);
dst->appendFormat(" - maxActiveCount: %u - curActiveCount: %u\n",
maxActiveCount, curActiveCount);
+ dst->appendFormat(" - recommendedMuteDurationMs: %u ms\n", recommendedMuteDurationMs);
}
void IOProfile::log()
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index 84ed656..4dfef73 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -123,6 +123,7 @@
static constexpr const char *flags = "flags";
static constexpr const char *maxOpenCount = "maxOpenCount";
static constexpr const char *maxActiveCount = "maxActiveCount";
+ static constexpr const char *recommendedMuteDurationMs = "recommendedMuteDurationMs";
};
// Children: GainTraits
@@ -496,6 +497,13 @@
if (!maxActiveCount.empty()) {
convertTo(maxActiveCount, mixPort->maxActiveCount);
}
+
+ std::string recommendedmuteDurationMsLiteral =
+ getXmlAttribute(child, Attributes::recommendedMuteDurationMs);
+ if (!recommendedmuteDurationMsLiteral.empty()) {
+ convertTo(recommendedmuteDurationMsLiteral, mixPort->recommendedMuteDurationMs);
+ }
+
// Deserialize children
AudioGainTraits::Collection gains;
status = deserializeCollection<AudioGainTraits>(child, &gains, NULL);
diff --git a/services/audiopolicy/engine/common/src/EngineDefaultConfig.h b/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
index d39eff6..665c2dd 100644
--- a/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
+++ b/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
@@ -69,12 +69,6 @@
{
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION, AUDIO_SOURCE_DEFAULT,
AUDIO_FLAG_NONE, ""},
- {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
- AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
- {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
- AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
- {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
- AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_EVENT,
AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}
}
diff --git a/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_product_strategies.xml b/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_product_strategies.xml
index bc32416..0ddf66d 100644
--- a/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_product_strategies.xml
+++ b/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_product_strategies.xml
@@ -57,9 +57,6 @@
<ProductStrategy name="STRATEGY_SONIFICATION_RESPECTFUL">
<AttributesGroup streamType="AUDIO_STREAM_NOTIFICATION" volumeGroup="notification">
<Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED"/> </Attributes>
<Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_EVENT"/> </Attributes>
</AttributesGroup>
</ProductStrategy>
diff --git a/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_product_strategies.xml b/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_product_strategies.xml
index bc32416..0ddf66d 100644
--- a/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_product_strategies.xml
+++ b/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_product_strategies.xml
@@ -57,9 +57,6 @@
<ProductStrategy name="STRATEGY_SONIFICATION_RESPECTFUL">
<AttributesGroup streamType="AUDIO_STREAM_NOTIFICATION" volumeGroup="notification">
<Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED"/> </Attributes>
<Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_EVENT"/> </Attributes>
</AttributesGroup>
</ProductStrategy>
diff --git a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
index 7000cd9..8584702 100644
--- a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
+++ b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
@@ -163,7 +163,9 @@
AUDIO_FLAG_BYPASS_MUTE, AUDIO_FLAG_LOW_LATENCY,
AUDIO_FLAG_DEEP_BUFFER, AUDIO_FLAG_NO_MEDIA_PROJECTION,
AUDIO_FLAG_MUTE_HAPTIC, AUDIO_FLAG_NO_SYSTEM_CAPTURE,
- AUDIO_FLAG_CAPTURE_PRIVATE};
+ AUDIO_FLAG_CAPTURE_PRIVATE, AUDIO_FLAG_CONTENT_SPATIALIZED,
+ AUDIO_FLAG_NEVER_SPATIALIZE,
+ };
std::vector<audio_policy_dev_state_t> kAudioPolicyDeviceStates = {
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 6f87bf0..a46da41 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -248,9 +248,7 @@
// been opened by checkOutputsForDevice() to query dynamic parameters
if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)
|| (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
- (desc->mDirectOpenCount == 0))
- || (((desc->mFlags & AUDIO_OUTPUT_FLAG_SPATIALIZER) != 0) &&
- (desc != mSpatializerOutput))) {
+ (desc->mDirectOpenCount == 0))) {
clearAudioSourcesForOutput(output);
closeOutput(output);
}
@@ -928,8 +926,7 @@
}
sp<IOProfile> AudioPolicyManager::getSpatializerOutputProfile(
- const audio_config_t *config __unused, const AudioDeviceTypeAddrVector &devices,
- bool forOpening) const
+ const audio_config_t *config __unused, const AudioDeviceTypeAddrVector &devices) const
{
for (const auto& hwModule : mHwModules) {
for (const auto& curProfile : hwModule->getOutputProfiles()) {
@@ -947,9 +944,6 @@
continue;
}
}
- if (forOpening && !curProfile->canOpenNewIo()) {
- continue;
- }
ALOGV("%s found profile %s", __func__, curProfile->getName().c_str());
return curProfile;
}
@@ -4844,6 +4838,21 @@
return source;
}
+/* static */
+bool AudioPolicyManager::isChannelMaskSpatialized(audio_channel_mask_t channels) {
+ switch (channels) {
+ case AUDIO_CHANNEL_OUT_5POINT1:
+ case AUDIO_CHANNEL_OUT_5POINT1POINT2:
+ case AUDIO_CHANNEL_OUT_5POINT1POINT4:
+ case AUDIO_CHANNEL_OUT_7POINT1:
+ case AUDIO_CHANNEL_OUT_7POINT1POINT2:
+ case AUDIO_CHANNEL_OUT_7POINT1POINT4:
+ return true;
+ default:
+ return false;
+ }
+}
+
bool AudioPolicyManager::canBeSpatialized(const audio_attributes_t *attr,
const audio_config_t *config,
const AudioDeviceTypeAddrVector &devices) const
@@ -4852,9 +4861,13 @@
// the AUDIO_ATTRIBUTES_INITIALIZER value.
// If attributes are specified, current policy is to only allow spatialization for media
// and game usages.
- if (attr != nullptr && *attr != AUDIO_ATTRIBUTES_INITIALIZER &&
- attr->usage != AUDIO_USAGE_MEDIA && attr->usage != AUDIO_USAGE_GAME) {
- return false;
+ if (attr != nullptr && *attr != AUDIO_ATTRIBUTES_INITIALIZER) {
+ if (attr->usage != AUDIO_USAGE_MEDIA && attr->usage != AUDIO_USAGE_GAME) {
+ return false;
+ }
+ if ((attr->flags & (AUDIO_FLAG_CONTENT_SPATIALIZED | AUDIO_FLAG_NEVER_SPATIALIZE)) != 0) {
+ return false;
+ }
}
// The caller can have the devices criteria ignored by passing and empty vector, and
@@ -4862,7 +4875,7 @@
// Otherwise an output profile supporting a spatializer effect that can be routed
// to the specified devices must exist.
sp<IOProfile> profile =
- getSpatializerOutputProfile(config, devices, false /*forOpening*/);
+ getSpatializerOutputProfile(config, devices);
if (profile == nullptr) {
return false;
}
@@ -4870,37 +4883,36 @@
// The caller can have the audio config criteria ignored by either passing a null ptr or
// the AUDIO_CONFIG_INITIALIZER value.
// If an audio config is specified, current policy is to only allow spatialization for
- // 5.1, 7.1and 7.1.4 audio.
+ // some positional channel masks.
// If the spatializer output is already opened, only channel masks included in the
// spatializer output mixer channel mask are allowed.
+
if (config != nullptr && *config != AUDIO_CONFIG_INITIALIZER) {
- if (config->channel_mask != AUDIO_CHANNEL_OUT_5POINT1
- && config->channel_mask != AUDIO_CHANNEL_OUT_7POINT1
- && config->channel_mask != AUDIO_CHANNEL_OUT_7POINT1POINT4) {
+ if (!isChannelMaskSpatialized(config->channel_mask)) {
return false;
}
- if (mSpatializerOutput != nullptr) {
+ if (mSpatializerOutput != nullptr && mSpatializerOutput->mProfile == profile) {
if ((config->channel_mask & mSpatializerOutput->mMixerChannelMask)
!= config->channel_mask) {
return false;
}
}
}
-
return true;
}
void AudioPolicyManager::checkVirtualizerClientRoutes() {
std::set<audio_stream_type_t> streamsToInvalidate;
for (size_t i = 0; i < mOutputs.size(); i++) {
- const sp<SwAudioOutputDescriptor>& outputDescriptor = mOutputs[i];
- for (const sp<TrackClientDescriptor>& client : outputDescriptor->getClientIterable()) {
+ const sp<SwAudioOutputDescriptor>& desc = mOutputs[i];
+ for (const sp<TrackClientDescriptor>& client : desc->getClientIterable()) {
audio_attributes_t attr = client->attributes();
DeviceVector devices = mEngine->getOutputDevicesForAttributes(attr, nullptr, false);
AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector();
audio_config_base_t clientConfig = client->config();
audio_config_t config = audio_config_initializer(&clientConfig);
- if (canBeSpatialized(&attr, &config, devicesTypeAddress)) {
+ if (desc != mSpatializerOutput
+ && canBeSpatialized(&attr, &config, devicesTypeAddress)) {
streamsToInvalidate.insert(client->stream());
}
}
@@ -4916,10 +4928,6 @@
audio_io_handle_t *output) {
*output = AUDIO_IO_HANDLE_NONE;
- if (mSpatializerOutput != nullptr) {
- return INVALID_OPERATION;
- }
-
DeviceVector devices = mEngine->getOutputDevicesForAttributes(*attr, nullptr, false);
AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector();
audio_config_t *configPtr = nullptr;
@@ -4929,35 +4937,87 @@
configPtr = &config;
}
if (!canBeSpatialized(attr, configPtr, devicesTypeAddress)) {
+ ALOGW("%s provided attributes or mixer config cannot be spatialized", __func__);
return BAD_VALUE;
}
sp<IOProfile> profile =
- getSpatializerOutputProfile(configPtr, devicesTypeAddress, true /*forOpening*/);
+ getSpatializerOutputProfile(configPtr, devicesTypeAddress);
if (profile == nullptr) {
+ ALOGW("%s no suitable output profile for provided attributes or mixer config", __func__);
return BAD_VALUE;
}
- mSpatializerOutput = new SwAudioOutputDescriptor(profile, mpClientInterface);
- status_t status = mSpatializerOutput->open(nullptr, mixerConfig, devices,
+ if (mSpatializerOutput != nullptr && mSpatializerOutput->mProfile == profile
+ && configPtr != nullptr
+ && configPtr->channel_mask == mSpatializerOutput->mMixerChannelMask) {
+ *output = mSpatializerOutput->mIoHandle;
+ ALOGV("%s returns current spatializer output %d", __func__, *output);
+ return NO_ERROR;
+ }
+ mSpatializerOutput.clear();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && desc->mProfile == profile) {
+ mSpatializerOutput = desc;
+ break;
+ }
+ }
+ if (mSpatializerOutput == nullptr) {
+ ALOGW("%s no opened spatializer output for profile %s",
+ __func__, profile->getName().c_str());
+ return BAD_VALUE;
+ }
+
+ if (configPtr != nullptr
+ && configPtr->channel_mask != mSpatializerOutput->mMixerChannelMask) {
+ audio_config_base_t savedMixerConfig = {
+ .sample_rate = mSpatializerOutput->getSamplingRate(),
+ .format = mSpatializerOutput->getFormat(),
+ .channel_mask = mSpatializerOutput->mMixerChannelMask,
+ };
+ DeviceVector savedDevices = mSpatializerOutput->devices();
+
+ closeOutput(mSpatializerOutput->mIoHandle);
+ mSpatializerOutput.clear();
+
+ const sp<SwAudioOutputDescriptor> desc =
+ new SwAudioOutputDescriptor(profile, mpClientInterface);
+ status_t status = desc->open(nullptr, mixerConfig, devices,
mEngine->getStreamTypeForAttributes(*attr),
AUDIO_OUTPUT_FLAG_SPATIALIZER, output);
- if (status != NO_ERROR) {
- ALOGV("%s failed opening output: status %d, output %d", __func__, status, *output);
- if (*output != AUDIO_IO_HANDLE_NONE) {
- mSpatializerOutput->close();
+ if (status != NO_ERROR) {
+ ALOGW("%s failed opening output: status %d, output %d", __func__, status, *output);
+ if (*output != AUDIO_IO_HANDLE_NONE) {
+ desc->close();
+ }
+ // re open the spatializer output with previous channel mask
+ status_t newStatus = desc->open(nullptr, &savedMixerConfig, savedDevices,
+ mEngine->getStreamTypeForAttributes(*attr),
+ AUDIO_OUTPUT_FLAG_SPATIALIZER, output);
+ if (newStatus != NO_ERROR) {
+ if (*output != AUDIO_IO_HANDLE_NONE) {
+ desc->close();
+ }
+ ALOGE("%s failed to re-open mSpatializerOutput, status %d", __func__, newStatus);
+ } else {
+ mSpatializerOutput = desc;
+ addOutput(*output, desc);
+ }
+ mPreviousOutputs = mOutputs;
+ mpClientInterface->onAudioPortListUpdate();
+ *output = AUDIO_IO_HANDLE_NONE;
+ return status;
}
- mSpatializerOutput.clear();
- *output = AUDIO_IO_HANDLE_NONE;
- return status;
+ mSpatializerOutput = desc;
+ addOutput(*output, desc);
+ mPreviousOutputs = mOutputs;
+ mpClientInterface->onAudioPortListUpdate();
}
checkVirtualizerClientRoutes();
- addOutput(*output, mSpatializerOutput);
- mPreviousOutputs = mOutputs;
- mpClientInterface->onAudioPortListUpdate();
-
+ *output = mSpatializerOutput->mIoHandle;
ALOGV("%s returns new spatializer output %d", __func__, *output);
return NO_ERROR;
}
@@ -4969,8 +5029,11 @@
if (mSpatializerOutput->mIoHandle != output) {
return BAD_VALUE;
}
- closeOutput(output);
+
mSpatializerOutput.clear();
+
+ checkVirtualizerClientRoutes();
+
return NO_ERROR;
}
@@ -5187,8 +5250,7 @@
outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = outputDesc;
}
- if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0
- || (outProfile->getFlags() & AUDIO_OUTPUT_FLAG_SPATIALIZER) != 0 ) {
+ if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
outputDesc->close();
} else {
addOutput(output, outputDesc);
@@ -6294,11 +6356,18 @@
// different per device volumes
if (outputDesc->isActive() && (devices != prevDevices)) {
uint32_t tempMuteWaitMs = outputDesc->latency() * 2;
- // temporary mute duration is conservatively set to 4 times the reported latency
- uint32_t tempMuteDurationMs = outputDesc->latency() * 4;
+
if (muteWaitMs < tempMuteWaitMs) {
muteWaitMs = tempMuteWaitMs;
}
+
+ // If recommended duration is defined, replace temporary mute duration to avoid
+ // truncated notifications at beginning, which depends on duration of changing path in HAL.
+ // Otherwise, temporary mute duration is conservatively set to 4 times the reported latency.
+ uint32_t tempRecommendedMuteDuration = outputDesc->getRecommendedMuteDurationMs();
+ uint32_t tempMuteDurationMs = tempRecommendedMuteDuration > 0 ?
+ tempRecommendedMuteDuration : outputDesc->latency() * 4;
+
for (const auto &activeVs : outputDesc->getActiveVolumeSources()) {
// make sure that we do not start the temporary mute period too early in case of
// delayed device change
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 2b3f3b4..0eda10e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -962,8 +962,9 @@
audio_io_handle_t *output);
sp<IOProfile> getSpatializerOutputProfile(const audio_config_t *config,
- const AudioDeviceTypeAddrVector &devices,
- bool forOpening) const;
+ const AudioDeviceTypeAddrVector &devices) const;
+
+ static bool isChannelMaskSpatialized(audio_channel_mask_t channels);
void checkVirtualizerClientRoutes();
diff --git a/services/audiopolicy/service/Android.bp b/services/audiopolicy/service/Android.bp
index 5ffddc2..cdad9a6 100644
--- a/services/audiopolicy/service/Android.bp
+++ b/services/audiopolicy/service/Android.bp
@@ -17,6 +17,7 @@
"AudioPolicyService.cpp",
"CaptureStateNotifier.cpp",
"Spatializer.cpp",
+ "SpatializerPoseController.cpp",
],
include_dirs: [
@@ -36,20 +37,27 @@
"libcutils",
"libeffectsconfig",
"libhardware_legacy",
+ "libheadtracking",
+ "libheadtracking-binding",
"liblog",
"libmedia_helper",
"libmediametrics",
"libmediautils",
"libpermission",
+ "libsensor",
"libsensorprivacy",
"libshmemcompat",
"libutils",
+ "libstagefright_foundation",
+ "android.media.audio.common.types-V1-cpp",
"audioclient-types-aidl-cpp",
"audioflinger-aidl-cpp",
"audiopolicy-aidl-cpp",
"audiopolicy-types-aidl-cpp",
"capture_state_listener-aidl-cpp",
"framework-permission-aidl-cpp",
+ "packagemanager_aidl-cpp",
+ "spatializer-aidl-cpp",
],
static_libs: [
@@ -74,6 +82,8 @@
export_shared_lib_headers: [
"libactivitymanager_aidl",
+ "libheadtracking",
+ "libheadtracking-binding",
"libsensorprivacy",
"framework-permission-aidl-cpp",
],
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index c8db45b..aaf6fba 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -139,7 +139,7 @@
request.config = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_t_AudioConfig(*config, true /*isInput*/));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(deviceTypeAddr));
- request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSourceType(source));
+ request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSource(source));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
media::OpenInputResponse response;
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 95c1bab..29baa22 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -32,6 +32,9 @@
if (!_tmp.ok()) return aidl_utils::binderStatusFromStatusT(_tmp.error()); \
std::move(_tmp.value()); })
+#define RETURN_BINDER_STATUS_IF_ERROR(x) \
+ if (status_t _tmp = (x); _tmp != OK) return aidl_utils::binderStatusFromStatusT(_tmp);
+
#define RETURN_IF_BINDER_ERROR(x) \
{ \
binder::Status _tmp = (x); \
@@ -44,6 +47,19 @@
using binder::Status;
using aidl_utils::binderStatusFromStatusT;
using content::AttributionSourceState;
+using media::audio::common::AudioConfig;
+using media::audio::common::AudioConfigBase;
+using media::audio::common::AudioDevice;
+using media::audio::common::AudioDeviceAddress;
+using media::audio::common::AudioDeviceDescription;
+using media::audio::common::AudioFormatDescription;
+using media::audio::common::AudioMode;
+using media::audio::common::AudioOffloadInfo;
+using media::audio::common::AudioSource;
+using media::audio::common::AudioStreamType;
+using media::audio::common::AudioUsage;
+using media::audio::common::AudioUuid;
+using media::audio::common::Int;
const std::vector<audio_usage_t>& SYSTEM_USAGES = {
AUDIO_USAGE_CALL_ASSISTANT,
@@ -96,12 +112,14 @@
}
Status AudioPolicyService::setDeviceConnectionState(
- const media::AudioDevice& deviceAidl,
+ const AudioDevice& deviceAidl,
media::AudioPolicyDeviceState stateAidl,
const std::string& deviceNameAidl,
- const media::AudioFormatDescription& encodedFormatAidl) {
- audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl.type));
+ const AudioFormatDescription& encodedFormatAidl) {
+ audio_devices_t device;
+ std::string address;
+ RETURN_BINDER_STATUS_IF_ERROR(
+ aidl2legacy_AudioDevice_audio_device(deviceAidl, &device, &address));
audio_policy_dev_state_t state = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioPolicyDeviceState_audio_policy_dev_state_t(stateAidl));
audio_format_t encodedFormat = VALUE_OR_RETURN_BINDER_STATUS(
@@ -121,17 +139,20 @@
ALOGV("setDeviceConnectionState()");
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
- return binderStatusFromStatusT(
- mAudioPolicyManager->setDeviceConnectionState(device, state,
- deviceAidl.address.c_str(),
- deviceNameAidl.c_str(),
- encodedFormat));
+ status_t status = mAudioPolicyManager->setDeviceConnectionState(
+ device, state, address.c_str(), deviceNameAidl.c_str(), encodedFormat);
+ if (status == NO_ERROR) {
+ onCheckSpatializer_l();
+ }
+ return binderStatusFromStatusT(status);
}
-Status AudioPolicyService::getDeviceConnectionState(const media::AudioDevice& deviceAidl,
+Status AudioPolicyService::getDeviceConnectionState(const AudioDevice& deviceAidl,
media::AudioPolicyDeviceState* _aidl_return) {
- audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl.type));
+ audio_devices_t device;
+ std::string address;
+ RETURN_BINDER_STATUS_IF_ERROR(
+ aidl2legacy_AudioDevice_audio_device(deviceAidl, &device, &address));
if (mAudioPolicyManager == NULL) {
*_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
legacy2aidl_audio_policy_dev_state_t_AudioPolicyDeviceState(
@@ -141,17 +162,19 @@
AutoCallerClear acc;
*_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
legacy2aidl_audio_policy_dev_state_t_AudioPolicyDeviceState(
- mAudioPolicyManager->getDeviceConnectionState(device,
- deviceAidl.address.c_str())));
+ mAudioPolicyManager->getDeviceConnectionState(
+ device, address.c_str())));
return Status::ok();
}
Status AudioPolicyService::handleDeviceConfigChange(
- const media::AudioDevice& deviceAidl,
+ const AudioDevice& deviceAidl,
const std::string& deviceNameAidl,
- const media::AudioFormatDescription& encodedFormatAidl) {
- audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl.type));
+ const AudioFormatDescription& encodedFormatAidl) {
+ audio_devices_t device;
+ std::string address;
+ RETURN_BINDER_STATUS_IF_ERROR(
+ aidl2legacy_AudioDevice_audio_device(deviceAidl, &device, &address));
audio_format_t encodedFormat = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioFormatDescription_audio_format_t(encodedFormatAidl));
@@ -165,12 +188,16 @@
ALOGV("handleDeviceConfigChange()");
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
- return binderStatusFromStatusT(
- mAudioPolicyManager->handleDeviceConfigChange(device, deviceAidl.address.c_str(),
- deviceNameAidl.c_str(), encodedFormat));
+ status_t status = mAudioPolicyManager->handleDeviceConfigChange(
+ device, address.c_str(), deviceNameAidl.c_str(), encodedFormat);
+
+ if (status == NO_ERROR) {
+ onCheckSpatializer_l();
+ }
+ return binderStatusFromStatusT(status);
}
-Status AudioPolicyService::setPhoneState(media::AudioMode stateAidl, int32_t uidAidl)
+Status AudioPolicyService::setPhoneState(AudioMode stateAidl, int32_t uidAidl)
{
audio_mode_t state = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioMode_audio_mode_t(stateAidl));
@@ -202,7 +229,7 @@
return Status::ok();
}
-Status AudioPolicyService::getPhoneState(media::AudioMode* _aidl_return) {
+Status AudioPolicyService::getPhoneState(AudioMode* _aidl_return) {
Mutex::Autolock _l(mLock);
*_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(legacy2aidl_audio_mode_t_AudioMode(mPhoneState));
return Status::ok();
@@ -234,6 +261,7 @@
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
mAudioPolicyManager->setForceUse(usage, config);
+ onCheckSpatializer_l();
return Status::ok();
}
@@ -257,7 +285,7 @@
return Status::ok();
}
-Status AudioPolicyService::getOutput(media::AudioStreamType streamAidl, int32_t* _aidl_return)
+Status AudioPolicyService::getOutput(AudioStreamType streamAidl, int32_t* _aidl_return)
{
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
@@ -281,7 +309,7 @@
Status AudioPolicyService::getOutputForAttr(const media::AudioAttributesInternal& attrAidl,
int32_t sessionAidl,
const AttributionSourceState& attributionSource,
- const media::AudioConfig& configAidl,
+ const AudioConfig& configAidl,
int32_t flagsAidl,
int32_t selectedDeviceIdAidl,
media::GetOutputForAttrResponse* _aidl_return)
@@ -510,7 +538,7 @@
int32_t riidAidl,
int32_t sessionAidl,
const AttributionSourceState& attributionSource,
- const media::AudioConfigBase& configAidl,
+ const AudioConfigBase& configAidl,
int32_t flagsAidl,
int32_t selectedDeviceIdAidl,
media::GetInputForAttrResponse* _aidl_return) {
@@ -581,28 +609,35 @@
adjAttributionSource)));
// check calling permissions.
- // Capturing from FM_TUNER source is controlled by captureTunerAudioInputAllowed() and
- // captureAudioOutputAllowed() (deprecated) as this does not affect users privacy
- // as does capturing from an actual microphone.
- if (!(recordingAllowed(adjAttributionSource, attr.source)
- || attr.source == AUDIO_SOURCE_FM_TUNER)) {
+ // Capturing from the following sources does not require permission RECORD_AUDIO
+ // as the captured audio does not come from a microphone:
+ // - FM_TUNER source is controlled by captureTunerAudioInputAllowed() or
+ // captureAudioOutputAllowed() (deprecated).
+ // - REMOTE_SUBMIX source is controlled by captureAudioOutputAllowed() if the input
+ // type is API_INPUT_MIX_EXT_POLICY_REROUTE and by AudioService if a media projection
+ // is used and input type is API_INPUT_MIX_PUBLIC_CAPTURE_PLAYBACK
+ // - ECHO_REFERENCE source is controlled by captureAudioOutputAllowed()
+ if (!(recordingAllowed(adjAttributionSource, inputSource)
+ || inputSource == AUDIO_SOURCE_FM_TUNER
+ || inputSource == AUDIO_SOURCE_REMOTE_SUBMIX
+ || inputSource == AUDIO_SOURCE_ECHO_REFERENCE)) {
ALOGE("%s permission denied: recording not allowed for %s",
__func__, adjAttributionSource.toString().c_str());
return binderStatusFromStatusT(PERMISSION_DENIED);
}
bool canCaptureOutput = captureAudioOutputAllowed(adjAttributionSource);
- if ((inputSource == AUDIO_SOURCE_VOICE_UPLINK ||
- inputSource == AUDIO_SOURCE_VOICE_DOWNLINK ||
- inputSource == AUDIO_SOURCE_VOICE_CALL ||
- inputSource == AUDIO_SOURCE_ECHO_REFERENCE)
- && !canCaptureOutput) {
+ if ((inputSource == AUDIO_SOURCE_VOICE_UPLINK
+ || inputSource == AUDIO_SOURCE_VOICE_DOWNLINK
+ || inputSource == AUDIO_SOURCE_VOICE_CALL
+ || inputSource == AUDIO_SOURCE_ECHO_REFERENCE)
+ && !canCaptureOutput) {
return binderStatusFromStatusT(PERMISSION_DENIED);
}
if (inputSource == AUDIO_SOURCE_FM_TUNER
- && !captureTunerAudioInputAllowed(adjAttributionSource)
- && !canCaptureOutput) {
+ && !canCaptureOutput
+ && !captureTunerAudioInputAllowed(adjAttributionSource)) {
return binderStatusFromStatusT(PERMISSION_DENIED);
}
@@ -730,8 +765,10 @@
// check calling permissions
if (!(startRecording(client->attributionSource, String16(msg.str().c_str()),
- client->attributes.source)
- || client->attributes.source == AUDIO_SOURCE_FM_TUNER)) {
+ client->attributes.source)
+ || client->attributes.source == AUDIO_SOURCE_FM_TUNER
+ || client->attributes.source == AUDIO_SOURCE_REMOTE_SUBMIX
+ || client->attributes.source == AUDIO_SOURCE_ECHO_REFERENCE)) {
ALOGE("%s permission denied: recording not allowed for attribution source %s",
__func__, client->attributionSource.toString().c_str());
return binderStatusFromStatusT(PERMISSION_DENIED);
@@ -901,7 +938,7 @@
return Status::ok();
}
-Status AudioPolicyService::initStreamVolume(media::AudioStreamType streamAidl,
+Status AudioPolicyService::initStreamVolume(AudioStreamType streamAidl,
int32_t indexMinAidl,
int32_t indexMaxAidl) {
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
@@ -924,8 +961,8 @@
return binderStatusFromStatusT(NO_ERROR);
}
-Status AudioPolicyService::setStreamVolumeIndex(media::AudioStreamType streamAidl,
- const media::AudioDeviceDescription& deviceAidl,
+Status AudioPolicyService::setStreamVolumeIndex(AudioStreamType streamAidl,
+ const AudioDeviceDescription& deviceAidl,
int32_t indexAidl) {
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
@@ -949,8 +986,8 @@
device));
}
-Status AudioPolicyService::getStreamVolumeIndex(media::AudioStreamType streamAidl,
- const media::AudioDeviceDescription& deviceAidl,
+Status AudioPolicyService::getStreamVolumeIndex(AudioStreamType streamAidl,
+ const AudioDeviceDescription& deviceAidl,
int32_t* _aidl_return) {
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
@@ -974,7 +1011,7 @@
Status AudioPolicyService::setVolumeIndexForAttributes(
const media::AudioAttributesInternal& attrAidl,
- const media::AudioDeviceDescription& deviceAidl, int32_t indexAidl) {
+ const AudioDeviceDescription& deviceAidl, int32_t indexAidl) {
audio_attributes_t attributes = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioAttributesInternal_audio_attributes_t(attrAidl));
int index = VALUE_OR_RETURN_BINDER_STATUS(convertIntegral<int>(indexAidl));
@@ -997,7 +1034,7 @@
Status AudioPolicyService::getVolumeIndexForAttributes(
const media::AudioAttributesInternal& attrAidl,
- const media::AudioDeviceDescription& deviceAidl, int32_t* _aidl_return) {
+ const AudioDeviceDescription& deviceAidl, int32_t* _aidl_return) {
audio_attributes_t attributes = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioAttributesInternal_audio_attributes_t(attrAidl));
audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
@@ -1055,7 +1092,7 @@
return Status::ok();
}
-Status AudioPolicyService::getStrategyForStream(media::AudioStreamType streamAidl,
+Status AudioPolicyService::getStrategyForStream(AudioStreamType streamAidl,
int32_t* _aidl_return) {
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
@@ -1080,13 +1117,13 @@
//audio policy: use audio_device_t appropriately
Status AudioPolicyService::getDevicesForStream(
- media::AudioStreamType streamAidl,
- std::vector<media::AudioDeviceDescription>* _aidl_return) {
+ AudioStreamType streamAidl,
+ std::vector<AudioDeviceDescription>* _aidl_return) {
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) {
- *_aidl_return = std::vector<media::AudioDeviceDescription>{};
+ *_aidl_return = std::vector<AudioDeviceDescription>{};
return Status::ok();
}
if (mAudioPolicyManager == NULL) {
@@ -1095,14 +1132,14 @@
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
*_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
- convertContainer<std::vector<media::AudioDeviceDescription>>(
+ convertContainer<std::vector<AudioDeviceDescription>>(
mAudioPolicyManager->getDevicesForStream(stream),
legacy2aidl_audio_devices_t_AudioDeviceDescription));
return Status::ok();
}
Status AudioPolicyService::getDevicesForAttributes(const media::AudioAttributesEx& attrAidl,
- std::vector<media::AudioDevice>* _aidl_return)
+ std::vector<AudioDevice>* _aidl_return)
{
AudioAttributes aa = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioAttributesEx_AudioAttributes(attrAidl));
@@ -1116,8 +1153,8 @@
RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
mAudioPolicyManager->getDevicesForAttributes(aa.getAttributes(), &devices)));
*_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
return Status::ok();
}
@@ -1203,7 +1240,7 @@
return binderStatusFromStatusT(mAudioPolicyManager->moveEffectsToIo(ids, io));
}
-Status AudioPolicyService::isStreamActive(media::AudioStreamType streamAidl, int32_t inPastMsAidl,
+Status AudioPolicyService::isStreamActive(AudioStreamType streamAidl, int32_t inPastMsAidl,
bool* _aidl_return) {
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
@@ -1222,7 +1259,7 @@
return Status::ok();
}
-Status AudioPolicyService::isStreamActiveRemotely(media::AudioStreamType streamAidl,
+Status AudioPolicyService::isStreamActiveRemotely(AudioStreamType streamAidl,
int32_t inPastMsAidl,
bool* _aidl_return) {
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
@@ -1242,9 +1279,9 @@
return Status::ok();
}
-Status AudioPolicyService::isSourceActive(media::AudioSourceType sourceAidl, bool* _aidl_return) {
+Status AudioPolicyService::isSourceActive(AudioSource sourceAidl, bool* _aidl_return) {
audio_source_t source = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioSourceType_audio_source_t(sourceAidl));
+ aidl2legacy_AudioSource_audio_source_t(sourceAidl));
if (mAudioPolicyManager == NULL) {
return binderStatusFromStatusT(NO_INIT);
}
@@ -1272,7 +1309,7 @@
Status AudioPolicyService::queryDefaultPreProcessing(
int32_t audioSessionAidl,
- media::Int* countAidl,
+ Int* countAidl,
std::vector<media::EffectDescriptor>* _aidl_return) {
audio_session_t audioSession = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_int32_t_audio_session_t(audioSessionAidl));
@@ -1296,11 +1333,11 @@
return Status::ok();
}
-Status AudioPolicyService::addSourceDefaultEffect(const media::AudioUuid& typeAidl,
+Status AudioPolicyService::addSourceDefaultEffect(const AudioUuid& typeAidl,
const std::string& opPackageNameAidl,
- const media::AudioUuid& uuidAidl,
+ const AudioUuid& uuidAidl,
int32_t priority,
- media::AudioSourceType sourceAidl,
+ AudioSource sourceAidl,
int32_t* _aidl_return) {
effect_uuid_t type = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioUuid_audio_uuid_t(typeAidl));
@@ -1309,7 +1346,7 @@
effect_uuid_t uuid = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioUuid_audio_uuid_t(uuidAidl));
audio_source_t source = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioSourceType_audio_source_t(sourceAidl));
+ aidl2legacy_AudioSource_audio_source_t(sourceAidl));
audio_unique_id_t id;
sp<AudioPolicyEffects>audioPolicyEffects;
@@ -1323,10 +1360,10 @@
return Status::ok();
}
-Status AudioPolicyService::addStreamDefaultEffect(const media::AudioUuid& typeAidl,
+Status AudioPolicyService::addStreamDefaultEffect(const AudioUuid& typeAidl,
const std::string& opPackageNameAidl,
- const media::AudioUuid& uuidAidl,
- int32_t priority, media::AudioUsage usageAidl,
+ const AudioUuid& uuidAidl,
+ int32_t priority, AudioUsage usageAidl,
int32_t* _aidl_return) {
effect_uuid_t type = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioUuid_audio_uuid_t(typeAidl));
@@ -1374,7 +1411,7 @@
}
Status AudioPolicyService::setSupportedSystemUsages(
- const std::vector<media::AudioUsage>& systemUsagesAidl) {
+ const std::vector<AudioUsage>& systemUsagesAidl) {
size_t size = systemUsagesAidl.size();
if (size > MAX_ITEMS_PER_LIST) {
size = MAX_ITEMS_PER_LIST;
@@ -1413,7 +1450,7 @@
mAudioPolicyManager->setAllowedCapturePolicy(uid, capturePolicy));
}
-Status AudioPolicyService::getOffloadSupport(const media::AudioOffloadInfo& infoAidl,
+Status AudioPolicyService::getOffloadSupport(const AudioOffloadInfo& infoAidl,
media::AudioOffloadMode* _aidl_return) {
audio_offload_info_t info = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioOffloadInfo_audio_offload_info_t(infoAidl));
@@ -1429,7 +1466,7 @@
}
Status AudioPolicyService::isDirectOutputSupported(
- const media::AudioConfigBase& configAidl,
+ const AudioConfigBase& configAidl,
const media::AudioAttributesInternal& attributesAidl,
bool* _aidl_return) {
audio_config_base_t config = VALUE_OR_RETURN_BINDER_STATUS(
@@ -1453,7 +1490,7 @@
Status AudioPolicyService::listAudioPorts(media::AudioPortRole roleAidl,
- media::AudioPortType typeAidl, media::Int* count,
+ media::AudioPortType typeAidl, Int* count,
std::vector<media::AudioPort>* portsAidl,
int32_t* _aidl_return) {
audio_port_role_t role = VALUE_OR_RETURN_BINDER_STATUS(
@@ -1541,7 +1578,7 @@
IPCThreadState::self()->getCallingUid()));
}
-Status AudioPolicyService::listAudioPatches(media::Int* count,
+Status AudioPolicyService::listAudioPatches(Int* count,
std::vector<media::AudioPatch>* patchesAidl,
int32_t* _aidl_return) {
unsigned int num_patches = VALUE_OR_RETURN_BINDER_STATUS(
@@ -1678,7 +1715,7 @@
Status AudioPolicyService::setUidDeviceAffinities(
int32_t uidAidl,
- const std::vector<media::AudioDevice>& devicesAidl) {
+ const std::vector<AudioDevice>& devicesAidl) {
uid_t uid = VALUE_OR_RETURN_BINDER_STATUS(aidl2legacy_int32_t_uid_t(uidAidl));
AudioDeviceTypeAddrVector devices = VALUE_OR_RETURN_BINDER_STATUS(
convertContainer<AudioDeviceTypeAddrVector>(devicesAidl,
@@ -1711,7 +1748,7 @@
Status AudioPolicyService::setUserIdDeviceAffinities(
int32_t userIdAidl,
- const std::vector<media::AudioDevice>& devicesAidl) {
+ const std::vector<AudioDevice>& devicesAidl) {
int userId = VALUE_OR_RETURN_BINDER_STATUS(convertReinterpret<int>(userIdAidl));
AudioDeviceTypeAddrVector devices = VALUE_OR_RETURN_BINDER_STATUS(
convertContainer<AudioDeviceTypeAddrVector>(devicesAidl,
@@ -1809,8 +1846,8 @@
Status AudioPolicyService::getStreamVolumeDB(
- media::AudioStreamType streamAidl, int32_t indexAidl,
- const media::AudioDeviceDescription& deviceAidl, float* _aidl_return) {
+ AudioStreamType streamAidl, int32_t indexAidl,
+ const AudioDeviceDescription& deviceAidl, float* _aidl_return) {
audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
int index = VALUE_OR_RETURN_BINDER_STATUS(convertIntegral<int>(indexAidl));
@@ -1826,8 +1863,8 @@
return Status::ok();
}
-Status AudioPolicyService::getSurroundFormats(media::Int* count,
- std::vector<media::AudioFormatDescription>* formats,
+Status AudioPolicyService::getSurroundFormats(Int* count,
+ std::vector<AudioFormatDescription>* formats,
std::vector<bool>* formatsEnabled) {
unsigned int numSurroundFormats = VALUE_OR_RETURN_BINDER_STATUS(
convertIntegral<unsigned int>(count->value));
@@ -1860,7 +1897,7 @@
}
Status AudioPolicyService::getReportedSurroundFormats(
- media::Int* count, std::vector<media::AudioFormatDescription>* formats) {
+ Int* count, std::vector<AudioFormatDescription>* formats) {
unsigned int numSurroundFormats = VALUE_OR_RETURN_BINDER_STATUS(
convertIntegral<unsigned int>(count->value));
if (numSurroundFormats > MAX_ITEMS_PER_LIST) {
@@ -1887,7 +1924,7 @@
}
Status AudioPolicyService::getHwOffloadEncodingFormatsSupportedForA2DP(
- std::vector<media::AudioFormatDescription>* _aidl_return) {
+ std::vector<AudioFormatDescription>* _aidl_return) {
std::vector<audio_format_t> formats;
if (mAudioPolicyManager == NULL) {
@@ -1898,14 +1935,14 @@
RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
mAudioPolicyManager->getHwOffloadEncodingFormatsSupportedForA2DP(&formats)));
*_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
- convertContainer<std::vector<media::AudioFormatDescription>>(
+ convertContainer<std::vector<AudioFormatDescription>>(
formats,
legacy2aidl_audio_format_t_AudioFormatDescription));
return Status::ok();
}
Status AudioPolicyService::setSurroundFormatEnabled(
- const media::AudioFormatDescription& audioFormatAidl, bool enabled) {
+ const AudioFormatDescription& audioFormatAidl, bool enabled) {
audio_format_t audioFormat = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioFormatDescription_audio_format_t(audioFormatAidl));
if (mAudioPolicyManager == NULL) {
@@ -2058,7 +2095,7 @@
Status AudioPolicyService::setDevicesRoleForStrategy(
int32_t strategyAidl,
media::DeviceRole roleAidl,
- const std::vector<media::AudioDevice>& devicesAidl) {
+ const std::vector<AudioDevice>& devicesAidl) {
product_strategy_t strategy = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_int32_t_product_strategy_t(strategyAidl));
device_role_t role = VALUE_OR_RETURN_BINDER_STATUS(
@@ -2071,8 +2108,11 @@
return binderStatusFromStatusT(NO_INIT);
}
Mutex::Autolock _l(mLock);
- return binderStatusFromStatusT(
- mAudioPolicyManager->setDevicesRoleForStrategy(strategy, role, devices));
+ status_t status = mAudioPolicyManager->setDevicesRoleForStrategy(strategy, role, devices);
+ if (status == NO_ERROR) {
+ onCheckSpatializer_l();
+ }
+ return binderStatusFromStatusT(status);
}
Status AudioPolicyService::removeDevicesRoleForStrategy(int32_t strategyAidl,
@@ -2085,14 +2125,17 @@
return binderStatusFromStatusT(NO_INIT);
}
Mutex::Autolock _l(mLock);
- return binderStatusFromStatusT(
- mAudioPolicyManager->removeDevicesRoleForStrategy(strategy, role));
+ status_t status = mAudioPolicyManager->removeDevicesRoleForStrategy(strategy, role);
+ if (status == NO_ERROR) {
+ onCheckSpatializer_l();
+ }
+ return binderStatusFromStatusT(status);
}
Status AudioPolicyService::getDevicesForRoleAndStrategy(
int32_t strategyAidl,
media::DeviceRole roleAidl,
- std::vector<media::AudioDevice>* _aidl_return) {
+ std::vector<AudioDevice>* _aidl_return) {
product_strategy_t strategy = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_int32_t_product_strategy_t(strategyAidl));
device_role_t role = VALUE_OR_RETURN_BINDER_STATUS(
@@ -2106,8 +2149,8 @@
RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
mAudioPolicyManager->getDevicesForRoleAndStrategy(strategy, role, devices)));
*_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
return Status::ok();
}
@@ -2118,11 +2161,11 @@
}
Status AudioPolicyService::setDevicesRoleForCapturePreset(
- media::AudioSourceType audioSourceAidl,
+ AudioSource audioSourceAidl,
media::DeviceRole roleAidl,
- const std::vector<media::AudioDevice>& devicesAidl) {
+ const std::vector<AudioDevice>& devicesAidl) {
audio_source_t audioSource = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioSourceType_audio_source_t(audioSourceAidl));
+ aidl2legacy_AudioSource_audio_source_t(audioSourceAidl));
device_role_t role = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_DeviceRole_device_role_t(roleAidl));
AudioDeviceTypeAddrVector devices = VALUE_OR_RETURN_BINDER_STATUS(
@@ -2138,11 +2181,11 @@
}
Status AudioPolicyService::addDevicesRoleForCapturePreset(
- media::AudioSourceType audioSourceAidl,
+ AudioSource audioSourceAidl,
media::DeviceRole roleAidl,
- const std::vector<media::AudioDevice>& devicesAidl) {
+ const std::vector<AudioDevice>& devicesAidl) {
audio_source_t audioSource = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioSourceType_audio_source_t(audioSourceAidl));
+ aidl2legacy_AudioSource_audio_source_t(audioSourceAidl));
device_role_t role = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_DeviceRole_device_role_t(roleAidl));
AudioDeviceTypeAddrVector devices = VALUE_OR_RETURN_BINDER_STATUS(
@@ -2158,11 +2201,11 @@
}
Status AudioPolicyService::removeDevicesRoleForCapturePreset(
- media::AudioSourceType audioSourceAidl,
+ AudioSource audioSourceAidl,
media::DeviceRole roleAidl,
- const std::vector<media::AudioDevice>& devicesAidl) {
+ const std::vector<AudioDevice>& devicesAidl) {
audio_source_t audioSource = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioSourceType_audio_source_t(audioSourceAidl));
+ aidl2legacy_AudioSource_audio_source_t(audioSourceAidl));
device_role_t role = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_DeviceRole_device_role_t(roleAidl));
AudioDeviceTypeAddrVector devices = VALUE_OR_RETURN_BINDER_STATUS(
@@ -2177,10 +2220,10 @@
mAudioPolicyManager->removeDevicesRoleForCapturePreset(audioSource, role, devices));
}
-Status AudioPolicyService::clearDevicesRoleForCapturePreset(media::AudioSourceType audioSourceAidl,
+Status AudioPolicyService::clearDevicesRoleForCapturePreset(AudioSource audioSourceAidl,
media::DeviceRole roleAidl) {
audio_source_t audioSource = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioSourceType_audio_source_t(audioSourceAidl));
+ aidl2legacy_AudioSource_audio_source_t(audioSourceAidl));
device_role_t role = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_DeviceRole_device_role_t(roleAidl));
@@ -2193,11 +2236,11 @@
}
Status AudioPolicyService::getDevicesForRoleAndCapturePreset(
- media::AudioSourceType audioSourceAidl,
+ AudioSource audioSourceAidl,
media::DeviceRole roleAidl,
- std::vector<media::AudioDevice>* _aidl_return) {
+ std::vector<AudioDevice>* _aidl_return) {
audio_source_t audioSource = VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_AudioSourceType_audio_source_t(audioSourceAidl));
+ aidl2legacy_AudioSource_audio_source_t(audioSourceAidl));
device_role_t role = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_DeviceRole_device_role_t(roleAidl));
AudioDeviceTypeAddrVector devices;
@@ -2209,8 +2252,8 @@
RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
mAudioPolicyManager->getDevicesForRoleAndCapturePreset(audioSource, role, devices)));
*_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
- convertContainer<std::vector<media::AudioDevice>>(devices,
- legacy2aidl_AudioDeviceTypeAddress));
+ convertContainer<std::vector<AudioDevice>>(devices,
+ legacy2aidl_AudioDeviceTypeAddress));
return Status::ok();
}
@@ -2218,16 +2261,21 @@
const sp<media::INativeSpatializerCallback>& callback,
media::GetSpatializerResponse* _aidl_return) {
_aidl_return->spatializer = nullptr;
- LOG_ALWAYS_FATAL_IF(callback == nullptr);
- RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(mSpatializer->registerCallback(callback)));
- _aidl_return->spatializer = mSpatializer;
+ if (callback == nullptr) {
+ return binderStatusFromStatusT(BAD_VALUE);
+ }
+ if (mSpatializer != nullptr) {
+ RETURN_IF_BINDER_ERROR(
+ binderStatusFromStatusT(mSpatializer->registerCallback(callback)));
+ _aidl_return->spatializer = mSpatializer;
+ }
return Status::ok();
}
Status AudioPolicyService::canBeSpatialized(
const std::optional<media::AudioAttributesInternal>& attrAidl,
- const std::optional<media::AudioConfig>& configAidl,
- const std::vector<media::AudioDevice>& devicesAidl,
+ const std::optional<AudioConfig>& configAidl,
+ const std::vector<AudioDevice>& devicesAidl,
bool* _aidl_return) {
if (mAudioPolicyManager == nullptr) {
return binderStatusFromStatusT(NO_INIT);
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 86994ed..0b4a059 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -142,11 +142,14 @@
sensorPrivacyPolicy->registerSelf();
// Create spatializer if supported
- const audio_attributes_t attr = attributes_initializer(AUDIO_USAGE_MEDIA);
- AudioDeviceTypeAddrVector devices;
- bool hasSpatializer = mAudioPolicyManager->canBeSpatialized(&attr, nullptr, devices);
- if (hasSpatializer) {
- mSpatializer = Spatializer::create(this);
+ if (mAudioPolicyManager != nullptr) {
+ Mutex::Autolock _l(mLock);
+ const audio_attributes_t attr = attributes_initializer(AUDIO_USAGE_MEDIA);
+ AudioDeviceTypeAddrVector devices;
+ bool hasSpatializer = mAudioPolicyManager->canBeSpatialized(&attr, nullptr, devices);
+ if (hasSpatializer) {
+ mSpatializer = Spatializer::create(this);
+ }
}
AudioSystem::audioPolicyReady();
}
@@ -366,41 +369,52 @@
void AudioPolicyService::onCheckSpatializer()
{
Mutex::Autolock _l(mLock);
- mOutputCommandThread->checkSpatializerCommand();
+ onCheckSpatializer_l();
+}
+
+void AudioPolicyService::onCheckSpatializer_l()
+{
+ if (mSpatializer != nullptr) {
+ mOutputCommandThread->checkSpatializerCommand();
+ }
}
void AudioPolicyService::doOnCheckSpatializer()
{
- sp<Spatializer> spatializer;
- {
- Mutex::Autolock _l(mLock);
- spatializer = mSpatializer;
+ Mutex::Autolock _l(mLock);
- if (spatializer != nullptr) {
- audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- if (spatializer->getLevel() != media::SpatializationLevel::NONE
- && spatializer->getOutput() == AUDIO_IO_HANDLE_NONE) {
- const audio_attributes_t attr = attributes_initializer(AUDIO_USAGE_MEDIA);
- audio_config_base_t config = spatializer->getAudioInConfig();
- status_t status =
- mAudioPolicyManager->getSpatializerOutput(&config, &attr, &output);
- if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
- return;
- }
- mLock.unlock();
- status = spatializer->attachOutput(output);
+ if (mSpatializer != nullptr) {
+ // Note: mSpatializer != nullptr => mAudioPolicyManager != nullptr
+ if (mSpatializer->getLevel() != media::SpatializationLevel::NONE) {
+ audio_io_handle_t currentOutput = mSpatializer->getOutput();
+ audio_io_handle_t newOutput;
+ const audio_attributes_t attr = attributes_initializer(AUDIO_USAGE_MEDIA);
+ audio_config_base_t config = mSpatializer->getAudioInConfig();
+ status_t status =
+ mAudioPolicyManager->getSpatializerOutput(&config, &attr, &newOutput);
+
+ if (status == NO_ERROR && currentOutput == newOutput) {
+ return;
+ }
+ mLock.unlock();
+ // It is OK to call detachOutput() is none is already attached.
+ mSpatializer->detachOutput();
+ if (status != NO_ERROR || newOutput == AUDIO_IO_HANDLE_NONE) {
mLock.lock();
- if (status != NO_ERROR) {
- mAudioPolicyManager->releaseSpatializerOutput(output);
- }
- } else if (spatializer->getLevel() == media::SpatializationLevel::NONE
- && spatializer->getOutput() != AUDIO_IO_HANDLE_NONE) {
- mLock.unlock();
- output = spatializer->detachOutput();
- mLock.lock();
- if (output != AUDIO_IO_HANDLE_NONE) {
- mAudioPolicyManager->releaseSpatializerOutput(output);
- }
+ return;
+ }
+ status = mSpatializer->attachOutput(newOutput);
+ mLock.lock();
+ if (status != NO_ERROR) {
+ mAudioPolicyManager->releaseSpatializerOutput(newOutput);
+ }
+ } else if (mSpatializer->getLevel() == media::SpatializationLevel::NONE
+ && mSpatializer->getOutput() != AUDIO_IO_HANDLE_NONE) {
+ mLock.unlock();
+ audio_io_handle_t output = mSpatializer->detachOutput();
+ mLock.lock();
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ mAudioPolicyManager->releaseSpatializerOutput(output);
}
}
}
@@ -495,14 +509,14 @@
int32_t eventAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(event));
media::RecordClientInfo clientInfoAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_record_client_info_t_RecordClientInfo(*clientInfo));
- media::AudioConfigBase clientConfigAidl = VALUE_OR_RETURN_STATUS(
+ AudioConfigBase clientConfigAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_base_t_AudioConfigBase(
*clientConfig, true /*isInput*/));
std::vector<media::EffectDescriptor> clientEffectsAidl = VALUE_OR_RETURN_STATUS(
convertContainer<std::vector<media::EffectDescriptor>>(
clientEffects,
legacy2aidl_effect_descriptor_t_EffectDescriptor));
- media::AudioConfigBase deviceConfigAidl = VALUE_OR_RETURN_STATUS(
+ AudioConfigBase deviceConfigAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_config_base_t_AudioConfigBase(
*deviceConfig, true /*isInput*/));
std::vector<media::EffectDescriptor> effectsAidl = VALUE_OR_RETURN_STATUS(
@@ -511,8 +525,8 @@
legacy2aidl_effect_descriptor_t_EffectDescriptor));
int32_t patchHandleAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_patch_handle_t_int32_t(patchHandle));
- media::AudioSourceType sourceAidl = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_source_t_AudioSourceType(source));
+ media::audio::common::AudioSource sourceAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_source_t_AudioSource(source));
return aidl_utils::statusTFromBinderStatus(
mAudioPolicyServiceClient->onRecordingConfigurationUpdate(eventAidl,
clientInfoAidl,
@@ -894,6 +908,7 @@
switch (source) {
case AUDIO_SOURCE_FM_TUNER:
case AUDIO_SOURCE_ECHO_REFERENCE:
+ case AUDIO_SOURCE_REMOTE_SUBMIX:
return false;
default:
break;
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 97d4c00..03241a2 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -49,6 +49,17 @@
namespace android {
using content::AttributionSourceState;
+using media::audio::common::AudioConfig;
+using media::audio::common::AudioConfigBase;
+using media::audio::common::AudioDevice;
+using media::audio::common::AudioDeviceDescription;
+using media::audio::common::AudioFormatDescription;
+using media::audio::common::AudioMode;
+using media::audio::common::AudioSource;
+using media::audio::common::AudioStreamType;
+using media::audio::common::AudioUsage;
+using media::audio::common::AudioUuid;
+using media::audio::common::Int;
// ----------------------------------------------------------------------------
@@ -71,25 +82,25 @@
//
binder::Status onNewAudioModulesAvailable() override;
binder::Status setDeviceConnectionState(
- const media::AudioDevice& device,
+ const AudioDevice& device,
media::AudioPolicyDeviceState state,
const std::string& deviceName,
- const media::AudioFormatDescription& encodedFormat) override;
- binder::Status getDeviceConnectionState(const media::AudioDevice& device,
+ const AudioFormatDescription& encodedFormat) override;
+ binder::Status getDeviceConnectionState(const AudioDevice& device,
media::AudioPolicyDeviceState* _aidl_return) override;
binder::Status handleDeviceConfigChange(
- const media::AudioDevice& device,
+ const AudioDevice& device,
const std::string& deviceName,
- const media::AudioFormatDescription& encodedFormat) override;
- binder::Status setPhoneState(media::AudioMode state, int32_t uid) override;
+ const AudioFormatDescription& encodedFormat) override;
+ binder::Status setPhoneState(AudioMode state, int32_t uid) override;
binder::Status setForceUse(media::AudioPolicyForceUse usage,
media::AudioPolicyForcedConfig config) override;
binder::Status getForceUse(media::AudioPolicyForceUse usage,
media::AudioPolicyForcedConfig* _aidl_return) override;
- binder::Status getOutput(media::AudioStreamType stream, int32_t* _aidl_return) override;
+ binder::Status getOutput(AudioStreamType stream, int32_t* _aidl_return) override;
binder::Status getOutputForAttr(const media::AudioAttributesInternal& attr, int32_t session,
const AttributionSourceState &attributionSource,
- const media::AudioConfig& config,
+ const AudioConfig& config,
int32_t flags, int32_t selectedDeviceId,
media::GetOutputForAttrResponse* _aidl_return) override;
binder::Status startOutput(int32_t portId) override;
@@ -98,37 +109,37 @@
binder::Status getInputForAttr(const media::AudioAttributesInternal& attr, int32_t input,
int32_t riid, int32_t session,
const AttributionSourceState &attributionSource,
- const media::AudioConfigBase& config, int32_t flags,
+ const AudioConfigBase& config, int32_t flags,
int32_t selectedDeviceId,
media::GetInputForAttrResponse* _aidl_return) override;
binder::Status startInput(int32_t portId) override;
binder::Status stopInput(int32_t portId) override;
binder::Status releaseInput(int32_t portId) override;
- binder::Status initStreamVolume(media::AudioStreamType stream, int32_t indexMin,
+ binder::Status initStreamVolume(AudioStreamType stream, int32_t indexMin,
int32_t indexMax) override;
- binder::Status setStreamVolumeIndex(media::AudioStreamType stream,
- const media::AudioDeviceDescription& device,
+ binder::Status setStreamVolumeIndex(AudioStreamType stream,
+ const AudioDeviceDescription& device,
int32_t index) override;
- binder::Status getStreamVolumeIndex(media::AudioStreamType stream,
- const media::AudioDeviceDescription& device,
+ binder::Status getStreamVolumeIndex(AudioStreamType stream,
+ const AudioDeviceDescription& device,
int32_t* _aidl_return) override;
binder::Status setVolumeIndexForAttributes(const media::AudioAttributesInternal& attr,
- const media::AudioDeviceDescription& device,
+ const AudioDeviceDescription& device,
int32_t index) override;
binder::Status getVolumeIndexForAttributes(const media::AudioAttributesInternal& attr,
- const media::AudioDeviceDescription& device,
+ const AudioDeviceDescription& device,
int32_t* _aidl_return) override;
binder::Status getMaxVolumeIndexForAttributes(const media::AudioAttributesInternal& attr,
int32_t* _aidl_return) override;
binder::Status getMinVolumeIndexForAttributes(const media::AudioAttributesInternal& attr,
int32_t* _aidl_return) override;
- binder::Status getStrategyForStream(media::AudioStreamType stream,
+ binder::Status getStrategyForStream(AudioStreamType stream,
int32_t* _aidl_return) override;
binder::Status getDevicesForStream(
- media::AudioStreamType stream,
- std::vector<media::AudioDeviceDescription>* _aidl_return) override;
+ AudioStreamType stream,
+ std::vector<AudioDeviceDescription>* _aidl_return) override;
binder::Status getDevicesForAttributes(const media::AudioAttributesEx& attr,
- std::vector<media::AudioDevice>* _aidl_return) override;
+ std::vector<AudioDevice>* _aidl_return) override;
binder::Status getOutputForEffect(const media::EffectDescriptor& desc,
int32_t* _aidl_return) override;
binder::Status registerEffect(const media::EffectDescriptor& desc, int32_t io, int32_t strategy,
@@ -136,42 +147,42 @@
binder::Status unregisterEffect(int32_t id) override;
binder::Status setEffectEnabled(int32_t id, bool enabled) override;
binder::Status moveEffectsToIo(const std::vector<int32_t>& ids, int32_t io) override;
- binder::Status isStreamActive(media::AudioStreamType stream, int32_t inPastMs,
+ binder::Status isStreamActive(AudioStreamType stream, int32_t inPastMs,
bool* _aidl_return) override;
- binder::Status isStreamActiveRemotely(media::AudioStreamType stream, int32_t inPastMs,
+ binder::Status isStreamActiveRemotely(AudioStreamType stream, int32_t inPastMs,
bool* _aidl_return) override;
- binder::Status isSourceActive(media::AudioSourceType source, bool* _aidl_return) override;
+ binder::Status isSourceActive(AudioSource source, bool* _aidl_return) override;
binder::Status queryDefaultPreProcessing(
- int32_t audioSession, media::Int* count,
+ int32_t audioSession, Int* count,
std::vector<media::EffectDescriptor>* _aidl_return) override;
- binder::Status addSourceDefaultEffect(const media::AudioUuid& type,
+ binder::Status addSourceDefaultEffect(const AudioUuid& type,
const std::string& opPackageName,
- const media::AudioUuid& uuid, int32_t priority,
- media::AudioSourceType source,
+ const AudioUuid& uuid, int32_t priority,
+ AudioSource source,
int32_t* _aidl_return) override;
- binder::Status addStreamDefaultEffect(const media::AudioUuid& type,
+ binder::Status addStreamDefaultEffect(const AudioUuid& type,
const std::string& opPackageName,
- const media::AudioUuid& uuid, int32_t priority,
- media::AudioUsage usage, int32_t* _aidl_return) override;
+ const AudioUuid& uuid, int32_t priority,
+ AudioUsage usage, int32_t* _aidl_return) override;
binder::Status removeSourceDefaultEffect(int32_t id) override;
binder::Status removeStreamDefaultEffect(int32_t id) override;
binder::Status setSupportedSystemUsages(
- const std::vector<media::AudioUsage>& systemUsages) override;
+ const std::vector<AudioUsage>& systemUsages) override;
binder::Status setAllowedCapturePolicy(int32_t uid, int32_t capturePolicy) override;
- binder::Status getOffloadSupport(const media::AudioOffloadInfo& info,
+ binder::Status getOffloadSupport(const media::audio::common::AudioOffloadInfo& info,
media::AudioOffloadMode* _aidl_return) override;
- binder::Status isDirectOutputSupported(const media::AudioConfigBase& config,
+ binder::Status isDirectOutputSupported(const AudioConfigBase& config,
const media::AudioAttributesInternal& attributes,
bool* _aidl_return) override;
binder::Status listAudioPorts(media::AudioPortRole role, media::AudioPortType type,
- media::Int* count, std::vector<media::AudioPort>* ports,
+ Int* count, std::vector<media::AudioPort>* ports,
int32_t* _aidl_return) override;
binder::Status getAudioPort(const media::AudioPort& port,
media::AudioPort* _aidl_return) override;
binder::Status createAudioPatch(const media::AudioPatch& patch, int32_t handle,
int32_t* _aidl_return) override;
binder::Status releaseAudioPatch(int32_t handle) override;
- binder::Status listAudioPatches(media::Int* count, std::vector<media::AudioPatch>* patches,
+ binder::Status listAudioPatches(Int* count, std::vector<media::AudioPatch>* patches,
int32_t* _aidl_return) override;
binder::Status setAudioPortConfig(const media::AudioPortConfig& config) override;
binder::Status registerClient(const sp<media::IAudioPolicyServiceClient>& client) override;
@@ -179,15 +190,15 @@
binder::Status setAudioVolumeGroupCallbacksEnabled(bool enabled) override;
binder::Status acquireSoundTriggerSession(media::SoundTriggerSession* _aidl_return) override;
binder::Status releaseSoundTriggerSession(int32_t session) override;
- binder::Status getPhoneState(media::AudioMode* _aidl_return) override;
+ binder::Status getPhoneState(AudioMode* _aidl_return) override;
binder::Status registerPolicyMixes(const std::vector<media::AudioMix>& mixes,
bool registration) override;
binder::Status setUidDeviceAffinities(int32_t uid,
- const std::vector<media::AudioDevice>& devices) override;
+ const std::vector<AudioDevice>& devices) override;
binder::Status removeUidDeviceAffinities(int32_t uid) override;
binder::Status setUserIdDeviceAffinities(
int32_t userId,
- const std::vector<media::AudioDevice>& devices) override;
+ const std::vector<AudioDevice>& devices) override;
binder::Status removeUserIdDeviceAffinities(int32_t userId) override;
binder::Status startAudioSource(const media::AudioPortConfig& source,
const media::AudioAttributesInternal& attributes,
@@ -195,17 +206,17 @@
binder::Status stopAudioSource(int32_t portId) override;
binder::Status setMasterMono(bool mono) override;
binder::Status getMasterMono(bool* _aidl_return) override;
- binder::Status getStreamVolumeDB(media::AudioStreamType stream, int32_t index,
- const media::AudioDeviceDescription& device,
+ binder::Status getStreamVolumeDB(AudioStreamType stream, int32_t index,
+ const AudioDeviceDescription& device,
float* _aidl_return) override;
- binder::Status getSurroundFormats(media::Int* count,
- std::vector<media::AudioFormatDescription>* formats,
+ binder::Status getSurroundFormats(Int* count,
+ std::vector<AudioFormatDescription>* formats,
std::vector<bool>* formatsEnabled) override;
binder::Status getReportedSurroundFormats(
- media::Int* count, std::vector<media::AudioFormatDescription>* formats) override;
+ Int* count, std::vector<AudioFormatDescription>* formats) override;
binder::Status getHwOffloadEncodingFormatsSupportedForA2DP(
- std::vector<media::AudioFormatDescription>* _aidl_return) override;
- binder::Status setSurroundFormatEnabled(const media::AudioFormatDescription& audioFormat,
+ std::vector<AudioFormatDescription>* _aidl_return) override;
+ binder::Status setSurroundFormatEnabled(const AudioFormatDescription& audioFormat,
bool enabled) override;
binder::Status setAssistantUid(int32_t uid) override;
binder::Status setHotwordDetectionServiceUid(int32_t uid) override;
@@ -226,29 +237,29 @@
binder::Status isCallScreenModeSupported(bool* _aidl_return) override;
binder::Status setDevicesRoleForStrategy(
int32_t strategy, media::DeviceRole role,
- const std::vector<media::AudioDevice>& devices) override;
+ const std::vector<AudioDevice>& devices) override;
binder::Status removeDevicesRoleForStrategy(int32_t strategy, media::DeviceRole role) override;
binder::Status getDevicesForRoleAndStrategy(
int32_t strategy, media::DeviceRole role,
- std::vector<media::AudioDevice>* _aidl_return) override;
+ std::vector<AudioDevice>* _aidl_return) override;
binder::Status setDevicesRoleForCapturePreset(
- media::AudioSourceType audioSource,
+ AudioSource audioSource,
media::DeviceRole role,
- const std::vector<media::AudioDevice>& devices) override;
+ const std::vector<AudioDevice>& devices) override;
binder::Status addDevicesRoleForCapturePreset(
- media::AudioSourceType audioSource,
+ AudioSource audioSource,
media::DeviceRole role,
- const std::vector<media::AudioDevice>& devices) override;
+ const std::vector<AudioDevice>& devices) override;
binder::Status removeDevicesRoleForCapturePreset(
- media::AudioSourceType audioSource,
+ AudioSource audioSource,
media::DeviceRole role,
- const std::vector<media::AudioDevice>& devices) override;
- binder::Status clearDevicesRoleForCapturePreset(media::AudioSourceType audioSource,
+ const std::vector<AudioDevice>& devices) override;
+ binder::Status clearDevicesRoleForCapturePreset(AudioSource audioSource,
media::DeviceRole role) override;
binder::Status getDevicesForRoleAndCapturePreset(
- media::AudioSourceType audioSource,
+ AudioSource audioSource,
media::DeviceRole role,
- std::vector<media::AudioDevice>* _aidl_return) override;
+ std::vector<AudioDevice>* _aidl_return) override;
binder::Status registerSoundTriggerCaptureStateListener(
const sp<media::ICaptureStateListener>& listener, bool* _aidl_return) override;
@@ -256,8 +267,8 @@
media::GetSpatializerResponse* _aidl_return) override;
binder::Status canBeSpatialized(
const std::optional<media::AudioAttributesInternal>& attr,
- const std::optional<media::AudioConfig>& config,
- const std::vector<media::AudioDevice>& devices,
+ const std::optional<AudioConfig>& config,
+ const std::vector<AudioDevice>& devices,
bool* _aidl_return) override;
status_t onTransact(uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) override;
@@ -333,6 +344,7 @@
* by audio policy manager and attach/detach the spatializer effect accordingly.
*/
void onCheckSpatializer() override;
+ void onCheckSpatializer_l();
void doOnCheckSpatializer();
void setEffectSuspended(int effectId,
diff --git a/services/audiopolicy/service/Spatializer.cpp b/services/audiopolicy/service/Spatializer.cpp
index 345665c..502a8d0 100644
--- a/services/audiopolicy/service/Spatializer.cpp
+++ b/services/audiopolicy/service/Spatializer.cpp
@@ -27,8 +27,11 @@
#include <android/content/AttributionSourceState.h>
#include <audio_utils/fixedfft.h>
#include <cutils/bitops.h>
-#include <media/ShmemCompat.h>
+#include <hardware/sensors.h>
#include <media/audiohal/EffectsFactoryHalInterface.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/ShmemCompat.h>
#include <mediautils/ServiceUtilities.h>
#include <utils/Thread.h>
@@ -41,21 +44,105 @@
using android::content::AttributionSourceState;
using binder::Status;
using media::HeadTrackingMode;
+using media::Pose3f;
using media::SpatializationLevel;
+using media::SpatializationMode;
+using media::SpatializerHeadTrackingMode;
+using media::SensorPoseProvider;
+
+using namespace std::chrono_literals;
#define VALUE_OR_RETURN_BINDER_STATUS(x) \
({ auto _tmp = (x); \
if (!_tmp.ok()) return aidl_utils::binderStatusFromStatusT(_tmp.error()); \
std::move(_tmp.value()); })
-#define RETURN_IF_BINDER_ERROR(x) \
- { \
- binder::Status _tmp = (x); \
- if (!_tmp.isOk()) return _tmp; \
- }
-
// ---------------------------------------------------------------------------
+class Spatializer::EngineCallbackHandler : public AHandler {
+public:
+ EngineCallbackHandler(wp<Spatializer> spatializer)
+ : mSpatializer(spatializer) {
+ }
+
+ enum {
+ // Device state callbacks
+ kWhatOnFramesProcessed, // AudioEffect::EVENT_FRAMES_PROCESSED
+ kWhatOnHeadToStagePose, // SpatializerPoseController::Listener::onHeadToStagePose
+ kWhatOnActualModeChange, // SpatializerPoseController::Listener::onActualModeChange
+ };
+ static constexpr const char *kNumFramesKey = "numFrames";
+ static constexpr const char *kModeKey = "mode";
+ static constexpr const char *kTranslation0Key = "translation0";
+ static constexpr const char *kTranslation1Key = "translation1";
+ static constexpr const char *kTranslation2Key = "translation2";
+ static constexpr const char *kRotation0Key = "rotation0";
+ static constexpr const char *kRotation1Key = "rotation1";
+ static constexpr const char *kRotation2Key = "rotation2";
+
+ void onMessageReceived(const sp<AMessage> &msg) override {
+ switch (msg->what()) {
+ case kWhatOnFramesProcessed: {
+ sp<Spatializer> spatializer = mSpatializer.promote();
+ if (spatializer == nullptr) {
+ ALOGW("%s: Cannot promote spatializer", __func__);
+ return;
+ }
+ int numFrames;
+ if (!msg->findInt32(kNumFramesKey, &numFrames)) {
+ ALOGE("%s: Cannot find num frames!", __func__);
+ return;
+ }
+ if (numFrames > 0) {
+ spatializer->calculateHeadPose();
+ }
+ } break;
+ case kWhatOnHeadToStagePose: {
+ sp<Spatializer> spatializer = mSpatializer.promote();
+ if (spatializer == nullptr) {
+ ALOGW("%s: Cannot promote spatializer", __func__);
+ return;
+ }
+ std::vector<float> headToStage(sHeadPoseKeys.size());
+ for (size_t i = 0 ; i < sHeadPoseKeys.size(); i++) {
+ if (!msg->findFloat(sHeadPoseKeys[i], &headToStage[i])) {
+ ALOGE("%s: Cannot find kTranslation0Key!", __func__);
+ return;
+ }
+ }
+ spatializer->onHeadToStagePoseMsg(headToStage);
+ } break;
+ case kWhatOnActualModeChange: {
+ sp<Spatializer> spatializer = mSpatializer.promote();
+ if (spatializer == nullptr) {
+ ALOGW("%s: Cannot promote spatializer", __func__);
+ return;
+ }
+ int mode;
+ if (!msg->findInt32(EngineCallbackHandler::kModeKey, &mode)) {
+ ALOGE("%s: Cannot find actualMode!", __func__);
+ return;
+ }
+ spatializer->onActualModeChangeMsg(static_cast<HeadTrackingMode>(mode));
+ } break;
+ default:
+ LOG_ALWAYS_FATAL("Invalid callback message %d", msg->what());
+ }
+ }
+private:
+ wp<Spatializer> mSpatializer;
+};
+
+const std::vector<const char *> Spatializer::sHeadPoseKeys = {
+ Spatializer::EngineCallbackHandler::kTranslation0Key,
+ Spatializer::EngineCallbackHandler::kTranslation1Key,
+ Spatializer::EngineCallbackHandler::kTranslation2Key,
+ Spatializer::EngineCallbackHandler::kRotation0Key,
+ Spatializer::EngineCallbackHandler::kRotation1Key,
+ Spatializer::EngineCallbackHandler::kRotation2Key,
+};
+
+// ---------------------------------------------------------------------------
sp<Spatializer> Spatializer::create(SpatializerPolicyCallback *callback) {
sp<Spatializer> spatializer;
@@ -84,28 +171,87 @@
if (status == NO_ERROR && effect != nullptr) {
spatializer = new Spatializer(descriptors[0], callback);
- // TODO: Read supported config from engine
- audio_config_base_t config = AUDIO_CONFIG_BASE_INITIALIZER;
- config.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
- spatializer->setAudioInConfig(config);
+ if (spatializer->loadEngineConfiguration(effect) != NO_ERROR) {
+ spatializer.clear();
+ }
}
return spatializer;
}
-Spatializer::Spatializer(effect_descriptor_t engineDescriptor,
- SpatializerPolicyCallback *callback)
- : mEngineDescriptor(engineDescriptor), mPolicyCallback(callback) {
+Spatializer::Spatializer(effect_descriptor_t engineDescriptor, SpatializerPolicyCallback* callback)
+ : mEngineDescriptor(engineDescriptor),
+ mPolicyCallback(callback) {
ALOGV("%s", __func__);
}
+void Spatializer::onFirstRef() {
+ mLooper = new ALooper;
+ mLooper->setName("Spatializer-looper");
+ mLooper->start(
+ /*runOnCallingThread*/false,
+ /*canCallJava*/ false,
+ PRIORITY_AUDIO);
+
+ mHandler = new EngineCallbackHandler(this);
+ mLooper->registerHandler(mHandler);
+}
+
Spatializer::~Spatializer() {
ALOGV("%s", __func__);
+ if (mLooper != nullptr) {
+ mLooper->stop();
+ mLooper->unregisterHandler(mHandler->id());
+ }
+ mLooper.clear();
+ mHandler.clear();
+}
+
+status_t Spatializer::loadEngineConfiguration(sp<EffectHalInterface> effect) {
+ ALOGV("%s", __func__);
+
+ std::vector<bool> supportsHeadTracking;
+ status_t status = getHalParameter<false>(effect, SPATIALIZER_PARAM_HEADTRACKING_SUPPORTED,
+ &supportsHeadTracking);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ mSupportsHeadTracking = supportsHeadTracking[0];
+
+ status = getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_LEVELS, &mLevels);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ status = getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_SPATIALIZATION_MODES,
+ &mSpatializationModes);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ status = getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_CHANNEL_MASKS,
+ &mChannelMasks);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ return NO_ERROR;
+}
+
+/** Gets the channel mask, sampling rate and format set for the spatializer input. */
+audio_config_base_t Spatializer::getAudioInConfig() const {
+ std::lock_guard lock(mLock);
+ audio_config_base_t config = AUDIO_CONFIG_BASE_INITIALIZER;
+ // For now use highest supported channel count
+ uint32_t maxCount = 0;
+ for ( auto mask : mChannelMasks) {
+ if (audio_channel_count_from_out_mask(mask) > maxCount) {
+ config.channel_mask = mask;
+ }
+ }
+ return config;
}
status_t Spatializer::registerCallback(
const sp<media::INativeSpatializerCallback>& callback) {
- Mutex::Autolock _l(mLock);
+ std::lock_guard lock(mLock);
if (callback == nullptr) {
return BAD_VALUE;
}
@@ -122,7 +268,7 @@
// IBinder::DeathRecipient
void Spatializer::binderDied(__unused const wp<IBinder> &who) {
{
- Mutex::Autolock _l(mLock);
+ std::lock_guard lock(mLock);
mLevel = SpatializationLevel::NONE;
mSpatializerCallback.clear();
}
@@ -136,25 +282,28 @@
if (levels == nullptr) {
return binderStatusFromStatusT(BAD_VALUE);
}
- //TODO: get this from engine
levels->push_back(SpatializationLevel::NONE);
- levels->push_back(SpatializationLevel::SPATIALIZER_MULTICHANNEL);
+ levels->insert(levels->end(), mLevels.begin(), mLevels.end());
return Status::ok();
}
-Status Spatializer::setLevel(media::SpatializationLevel level) {
+Status Spatializer::setLevel(SpatializationLevel level) {
ALOGV("%s level %d", __func__, (int)level);
if (level != SpatializationLevel::NONE
- && level != SpatializationLevel::SPATIALIZER_MULTICHANNEL) {
+ && std::find(mLevels.begin(), mLevels.end(), level) == mLevels.end()) {
return binderStatusFromStatusT(BAD_VALUE);
}
sp<media::INativeSpatializerCallback> callback;
bool levelChanged = false;
{
- Mutex::Autolock _l(mLock);
+ std::lock_guard lock(mLock);
levelChanged = mLevel != level;
mLevel = level;
callback = mSpatializerCallback;
+
+ if (levelChanged && mEngine != nullptr) {
+ setEffectParameter_l(SPATIALIZER_PARAM_LEVEL, std::vector<SpatializationLevel>{level});
+ }
}
if (levelChanged) {
@@ -166,61 +315,109 @@
return Status::ok();
}
-Status Spatializer::getLevel(media::SpatializationLevel *level) {
+Status Spatializer::getLevel(SpatializationLevel *level) {
if (level == nullptr) {
return binderStatusFromStatusT(BAD_VALUE);
}
- Mutex::Autolock _l(mLock);
+ std::lock_guard lock(mLock);
*level = mLevel;
ALOGV("%s level %d", __func__, (int)*level);
return Status::ok();
}
+Status Spatializer::isHeadTrackingSupported(bool *supports) {
+ ALOGV("%s mSupportsHeadTracking %d", __func__, mSupportsHeadTracking);
+ if (supports == nullptr) {
+ return binderStatusFromStatusT(BAD_VALUE);
+ }
+ std::lock_guard lock(mLock);
+ *supports = mSupportsHeadTracking;
+ return Status::ok();
+}
+
Status Spatializer::getSupportedHeadTrackingModes(
- std::vector<media::HeadTrackingMode>* modes) {
+ std::vector<SpatializerHeadTrackingMode>* modes) {
+ std::lock_guard lock(mLock);
ALOGV("%s", __func__);
if (modes == nullptr) {
return binderStatusFromStatusT(BAD_VALUE);
}
- //TODO: get this from:
- // - The engine capabilities
- // - If a head tracking sensor is registered and linked to a connected audio device
- // - if we have indications on the screen orientation
- modes->push_back(HeadTrackingMode::RELATIVE_WORLD);
- return Status::ok();
-}
-Status Spatializer::setDesiredHeadTrackingMode(media::HeadTrackingMode mode) {
- ALOGV("%s level %d", __func__, (int)mode);
- if (mode != HeadTrackingMode::DISABLED
- && mode != HeadTrackingMode::RELATIVE_WORLD) {
- return binderStatusFromStatusT(BAD_VALUE);
- }
- {
- Mutex::Autolock _l(mLock);
- mHeadTrackingMode = mode;
+ modes->push_back(SpatializerHeadTrackingMode::DISABLED);
+ if (mSupportsHeadTracking) {
+ if (mHeadSensor != SpatializerPoseController::INVALID_SENSOR) {
+ modes->push_back(SpatializerHeadTrackingMode::RELATIVE_WORLD);
+ if (mScreenSensor != SpatializerPoseController::INVALID_SENSOR) {
+ modes->push_back(SpatializerHeadTrackingMode::RELATIVE_SCREEN);
+ }
+ }
}
return Status::ok();
}
-Status Spatializer::getActualHeadTrackingMode(media::HeadTrackingMode *mode) {
+Status Spatializer::setDesiredHeadTrackingMode(SpatializerHeadTrackingMode mode) {
+ ALOGV("%s mode %d", __func__, (int)mode);
+
+ if (!mSupportsHeadTracking) {
+ return binderStatusFromStatusT(INVALID_OPERATION);
+ }
+ std::lock_guard lock(mLock);
+ switch (mode) {
+ case SpatializerHeadTrackingMode::OTHER:
+ return binderStatusFromStatusT(BAD_VALUE);
+ case SpatializerHeadTrackingMode::DISABLED:
+ mDesiredHeadTrackingMode = HeadTrackingMode::STATIC;
+ break;
+ case SpatializerHeadTrackingMode::RELATIVE_WORLD:
+ mDesiredHeadTrackingMode = HeadTrackingMode::WORLD_RELATIVE;
+ break;
+ case SpatializerHeadTrackingMode::RELATIVE_SCREEN:
+ mDesiredHeadTrackingMode = HeadTrackingMode::SCREEN_RELATIVE;
+ break;
+ }
+
+ if (mPoseController != nullptr) {
+ mPoseController->setDesiredMode(mDesiredHeadTrackingMode);
+ }
+
+ return Status::ok();
+}
+
+Status Spatializer::getActualHeadTrackingMode(SpatializerHeadTrackingMode *mode) {
if (mode == nullptr) {
return binderStatusFromStatusT(BAD_VALUE);
}
- Mutex::Autolock _l(mLock);
- *mode = mHeadTrackingMode;
+ std::lock_guard lock(mLock);
+ *mode = mActualHeadTrackingMode;
ALOGV("%s mode %d", __func__, (int)*mode);
return Status::ok();
}
-Status Spatializer::recenterHeadtracker() {
+Status Spatializer::recenterHeadTracker() {
+ if (!mSupportsHeadTracking) {
+ return binderStatusFromStatusT(INVALID_OPERATION);
+ }
+ std::lock_guard lock(mLock);
+ if (mPoseController != nullptr) {
+ mPoseController->recenter();
+ }
return Status::ok();
}
Status Spatializer::setGlobalTransform(const std::vector<float>& screenToStage) {
- Mutex::Autolock _l(mLock);
- mScreenToStageTransform = screenToStage;
ALOGV("%s", __func__);
+ if (!mSupportsHeadTracking) {
+ return binderStatusFromStatusT(INVALID_OPERATION);
+ }
+ std::optional<Pose3f> maybePose = Pose3f::fromVector(screenToStage);
+ if (!maybePose.has_value()) {
+ ALOGW("Invalid screenToStage vector.");
+ return binderStatusFromStatusT(BAD_VALUE);
+ }
+ std::lock_guard lock(mLock);
+ if (mPoseController != nullptr) {
+ mPoseController->setScreenToStagePose(maybePose.value());
+ }
return Status::ok();
}
@@ -228,7 +425,7 @@
ALOGV("%s", __func__);
bool levelChanged = false;
{
- Mutex::Autolock _l(mLock);
+ std::lock_guard lock(mLock);
if (mSpatializerCallback == nullptr) {
return binderStatusFromStatusT(INVALID_OPERATION);
}
@@ -247,132 +444,283 @@
return Status::ok();
}
+Status Spatializer::setHeadSensor(int sensorHandle) {
+ ALOGV("%s sensorHandle %d", __func__, sensorHandle);
+ if (!mSupportsHeadTracking) {
+ return binderStatusFromStatusT(INVALID_OPERATION);
+ }
+ std::lock_guard lock(mLock);
+ mHeadSensor = sensorHandle;
+ if (mPoseController != nullptr) {
+ mPoseController->setHeadSensor(mHeadSensor);
+ }
+ return Status::ok();
+}
+
+Status Spatializer::setScreenSensor(int sensorHandle) {
+ ALOGV("%s sensorHandle %d", __func__, sensorHandle);
+ if (!mSupportsHeadTracking) {
+ return binderStatusFromStatusT(INVALID_OPERATION);
+ }
+ std::lock_guard lock(mLock);
+ mScreenSensor = sensorHandle;
+ if (mPoseController != nullptr) {
+ mPoseController->setScreenSensor(mScreenSensor);
+ }
+ return Status::ok();
+}
+
+Status Spatializer::setDisplayOrientation(float physicalToLogicalAngle) {
+ ALOGV("%s physicalToLogicalAngle %f", __func__, physicalToLogicalAngle);
+ if (!mSupportsHeadTracking) {
+ return binderStatusFromStatusT(INVALID_OPERATION);
+ }
+ std::lock_guard lock(mLock);
+ mDisplayOrientation = physicalToLogicalAngle;
+ if (mPoseController != nullptr) {
+ mPoseController->setDisplayOrientation(mDisplayOrientation);
+ }
+ if (mEngine != nullptr) {
+ setEffectParameter_l(
+ SPATIALIZER_PARAM_DISPLAY_ORIENTATION, std::vector<float>{physicalToLogicalAngle});
+ }
+ return Status::ok();
+}
+
+Status Spatializer::setHingeAngle(float hingeAngle) {
+ std::lock_guard lock(mLock);
+ ALOGV("%s hingeAngle %f", __func__, hingeAngle);
+ if (mEngine != nullptr) {
+ setEffectParameter_l(SPATIALIZER_PARAM_HINGE_ANGLE, std::vector<float>{hingeAngle});
+ }
+ return Status::ok();
+}
+
+Status Spatializer::getSupportedModes(std::vector<SpatializationMode> *modes) {
+ ALOGV("%s", __func__);
+ if (modes == nullptr) {
+ return binderStatusFromStatusT(BAD_VALUE);
+ }
+ *modes = mSpatializationModes;
+ return Status::ok();
+}
+
+Status Spatializer::registerHeadTrackingCallback(
+ const sp<media::ISpatializerHeadTrackingCallback>& callback) {
+ ALOGV("%s callback %p", __func__, callback.get());
+ std::lock_guard lock(mLock);
+ if (!mSupportsHeadTracking) {
+ return binderStatusFromStatusT(INVALID_OPERATION);
+ }
+ mHeadTrackingCallback = callback;
+ return Status::ok();
+}
+
+Status Spatializer::setParameter(int key, const std::vector<unsigned char>& value) {
+ ALOGV("%s key %d", __func__, key);
+ std::lock_guard lock(mLock);
+ status_t status = INVALID_OPERATION;
+ if (mEngine != nullptr) {
+ status = setEffectParameter_l(key, value);
+ }
+ return binderStatusFromStatusT(status);
+}
+
+Status Spatializer::getParameter(int key, std::vector<unsigned char> *value) {
+ ALOGV("%s key %d value size %d", __func__, key,
+ (value != nullptr ? (int)value->size() : -1));
+ if (value == nullptr) {
+ return binderStatusFromStatusT(BAD_VALUE);
+ }
+ std::lock_guard lock(mLock);
+ status_t status = INVALID_OPERATION;
+ if (mEngine != nullptr) {
+ ALOGV("%s key %d mEngine %p", __func__, key, mEngine.get());
+ status = getEffectParameter_l(key, value);
+ }
+ return binderStatusFromStatusT(status);
+}
+
+Status Spatializer::getOutput(int *output) {
+ ALOGV("%s", __func__);
+ if (output == nullptr) {
+ binderStatusFromStatusT(BAD_VALUE);
+ }
+ std::lock_guard lock(mLock);
+ *output = VALUE_OR_RETURN_BINDER_STATUS(legacy2aidl_audio_io_handle_t_int32_t(mOutput));
+ ALOGV("%s got output %d", __func__, *output);
+ return Status::ok();
+}
+
+// SpatializerPoseController::Listener
+void Spatializer::onHeadToStagePose(const Pose3f& headToStage) {
+ ALOGV("%s", __func__);
+ LOG_ALWAYS_FATAL_IF(!mSupportsHeadTracking,
+ "onHeadToStagePose() called with no head tracking support!");
+
+ auto vec = headToStage.toVector();
+ LOG_ALWAYS_FATAL_IF(vec.size() != sHeadPoseKeys.size(),
+ "%s invalid head to stage vector size %zu", __func__, vec.size());
+
+ sp<AMessage> msg =
+ new AMessage(EngineCallbackHandler::kWhatOnHeadToStagePose, mHandler);
+ for (size_t i = 0 ; i < sHeadPoseKeys.size(); i++) {
+ msg->setFloat(sHeadPoseKeys[i], vec[i]);
+ }
+ msg->post();
+}
+
+void Spatializer::onHeadToStagePoseMsg(const std::vector<float>& headToStage) {
+ ALOGV("%s", __func__);
+ sp<media::ISpatializerHeadTrackingCallback> callback;
+ {
+ std::lock_guard lock(mLock);
+ callback = mHeadTrackingCallback;
+ if (mEngine != nullptr) {
+ setEffectParameter_l(SPATIALIZER_PARAM_HEAD_TO_STAGE, headToStage);
+ }
+ }
+
+ if (callback != nullptr) {
+ callback->onHeadToSoundStagePoseUpdated(headToStage);
+ }
+}
+
+void Spatializer::onActualModeChange(HeadTrackingMode mode) {
+ ALOGV("%s(%d)", __func__, (int)mode);
+ sp<AMessage> msg =
+ new AMessage(EngineCallbackHandler::kWhatOnActualModeChange, mHandler);
+ msg->setInt32(EngineCallbackHandler::kModeKey, static_cast<int>(mode));
+ msg->post();
+}
+
+void Spatializer::onActualModeChangeMsg(HeadTrackingMode mode) {
+ ALOGV("%s(%d)", __func__, (int) mode);
+ sp<media::ISpatializerHeadTrackingCallback> callback;
+ SpatializerHeadTrackingMode spatializerMode;
+ {
+ std::lock_guard lock(mLock);
+ if (!mSupportsHeadTracking) {
+ spatializerMode = SpatializerHeadTrackingMode::DISABLED;
+ } else {
+ switch (mode) {
+ case HeadTrackingMode::STATIC:
+ spatializerMode = SpatializerHeadTrackingMode::DISABLED;
+ break;
+ case HeadTrackingMode::WORLD_RELATIVE:
+ spatializerMode = SpatializerHeadTrackingMode::RELATIVE_WORLD;
+ break;
+ case HeadTrackingMode::SCREEN_RELATIVE:
+ spatializerMode = SpatializerHeadTrackingMode::RELATIVE_SCREEN;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("Unknown mode: %d", mode);
+ }
+ }
+ mActualHeadTrackingMode = spatializerMode;
+ callback = mHeadTrackingCallback;
+ }
+ if (callback != nullptr) {
+ callback->onHeadTrackingModeChanged(spatializerMode);
+ }
+}
+
status_t Spatializer::attachOutput(audio_io_handle_t output) {
- Mutex::Autolock _l(mLock);
- ALOGV("%s output %d mOutput %d", __func__, (int)output, (int)mOutput);
- if (mOutput != AUDIO_IO_HANDLE_NONE) {
- LOG_ALWAYS_FATAL_IF(mEngine != nullptr, "%s output set without FX engine", __func__);
- // remove FX instance
- mEngine->setEnabled(false);
- mEngine.clear();
+ std::shared_ptr<SpatializerPoseController> poseController;
+ {
+ std::lock_guard lock(mLock);
+ ALOGV("%s output %d mOutput %d", __func__, (int)output, (int)mOutput);
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ LOG_ALWAYS_FATAL_IF(mEngine == nullptr, "%s output set without FX engine", __func__);
+ // remove FX instance
+ mEngine->setEnabled(false);
+ mEngine.clear();
+ }
+ // create FX instance on output
+ AttributionSourceState attributionSource = AttributionSourceState();
+ mEngine = new AudioEffect(attributionSource);
+ mEngine->set(nullptr, &mEngineDescriptor.uuid, 0, Spatializer::engineCallback /* cbf */,
+ this /* user */, AUDIO_SESSION_OUTPUT_STAGE, output, {} /* device */,
+ false /* probe */, true /* notifyFramesProcessed */);
+ status_t status = mEngine->initCheck();
+ ALOGV("%s mEngine create status %d", __func__, (int)status);
+ if (status != NO_ERROR) {
+ return status;
+ }
+
+ setEffectParameter_l(SPATIALIZER_PARAM_LEVEL,
+ std::vector<SpatializationLevel>{mLevel});
+ setEffectParameter_l(SPATIALIZER_PARAM_HEADTRACKING_MODE,
+ std::vector<SpatializerHeadTrackingMode>{mActualHeadTrackingMode});
+
+ mEngine->setEnabled(true);
+ mOutput = output;
+
+ if (mSupportsHeadTracking) {
+ mPoseController = std::make_shared<SpatializerPoseController>(
+ static_cast<SpatializerPoseController::Listener*>(this), 10ms, 50ms);
+ LOG_ALWAYS_FATAL_IF(mPoseController == nullptr,
+ "%s could not allocate pose controller", __func__);
+
+ mPoseController->setDesiredMode(mDesiredHeadTrackingMode);
+ mPoseController->setHeadSensor(mHeadSensor);
+ mPoseController->setScreenSensor(mScreenSensor);
+ mPoseController->setDisplayOrientation(mDisplayOrientation);
+ poseController = mPoseController;
+ }
}
- // create FX instance on output
- AttributionSourceState attributionSource = AttributionSourceState();
- mEngine = new AudioEffect(attributionSource);
- mEngine->set(nullptr, &mEngineDescriptor.uuid, 0,
- Spatializer::engineCallback /* cbf */, this /* user */,
- AUDIO_SESSION_OUTPUT_STAGE, output,
- {} /* device */, false /* probe */, true /* notifyFramesProcessed */);
- status_t status = mEngine->initCheck();
- ALOGV("%s mEngine create status %d", __func__, (int)status);
- if (status != NO_ERROR) {
- return status;
+ if (poseController != nullptr) {
+ poseController->waitUntilCalculated();
}
- mEngine->setEnabled(true);
- mOutput = output;
return NO_ERROR;
}
audio_io_handle_t Spatializer::detachOutput() {
- Mutex::Autolock _l(mLock);
+ std::lock_guard lock(mLock);
ALOGV("%s mOutput %d", __func__, (int)mOutput);
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
if (mOutput == AUDIO_IO_HANDLE_NONE) {
- return AUDIO_IO_HANDLE_NONE;
+ return output;
}
// remove FX instance
mEngine->setEnabled(false);
mEngine.clear();
- audio_io_handle_t output = mOutput;
+ output = mOutput;
mOutput = AUDIO_IO_HANDLE_NONE;
+ mPoseController.reset();
return output;
}
-void Spatializer::engineCallback(int32_t event, void *user, void *info) {
+void Spatializer::calculateHeadPose() {
+ ALOGV("%s", __func__);
+ std::lock_guard lock(mLock);
+ if (mPoseController != nullptr) {
+ mPoseController->calculateAsync();
+ }
+}
+void Spatializer::engineCallback(int32_t event, void *user, void *info) {
if (user == nullptr) {
return;
}
- const Spatializer * const me = reinterpret_cast<Spatializer *>(user);
+ Spatializer* const me = reinterpret_cast<Spatializer *>(user);
switch (event) {
case AudioEffect::EVENT_FRAMES_PROCESSED: {
- int frames = info == nullptr ? 0 : *(int *)info;
+ int frames = info == nullptr ? 0 : *(int*)info;
ALOGD("%s frames processed %d for me %p", __func__, frames, me);
- } break;
+ me->postFramesProcessedMsg(frames);
+ } break;
default:
ALOGD("%s event %d", __func__, event);
break;
}
}
-// ---------------------------------------------------------------------------
-
-Spatializer::EffectClient::EffectClient(const sp<media::IEffectClient>& effectClient,
- Spatializer& parent)
- : BnEffect(),
- mEffectClient(effectClient), mParent(parent) {
-}
-
-Spatializer::EffectClient::~EffectClient() {
-}
-
-// IEffect
-
-#define RETURN(code) \
- *_aidl_return = (code); \
- return Status::ok();
-
-// Write a POD value into a vector of bytes (clears the previous buffer
-// content).
-template<typename T>
-void writeToBuffer(const T& value, std::vector<uint8_t>* buffer) {
- buffer->clear();
- appendToBuffer(value, buffer);
-}
-
-Status Spatializer::EffectClient::enable(int32_t* _aidl_return) {
- RETURN(OK);
-}
-
-Status Spatializer::EffectClient::disable(int32_t* _aidl_return) {
- RETURN(OK);
-}
-
-Status Spatializer::EffectClient::command(int32_t cmdCode,
- const std::vector<uint8_t>& cmdData __unused,
- int32_t maxResponseSize __unused,
- std::vector<uint8_t>* response __unused,
- int32_t* _aidl_return) {
-
- // reject commands reserved for internal use by audio framework if coming from outside
- // of audioserver
- switch(cmdCode) {
- case EFFECT_CMD_ENABLE:
- case EFFECT_CMD_DISABLE:
- case EFFECT_CMD_SET_PARAM_DEFERRED:
- case EFFECT_CMD_SET_PARAM_COMMIT:
- RETURN(BAD_VALUE);
- case EFFECT_CMD_SET_PARAM:
- case EFFECT_CMD_GET_PARAM:
- break;
- default:
- if (cmdCode >= EFFECT_CMD_FIRST_PROPRIETARY) {
- break;
- }
- android_errorWriteLog(0x534e4554, "62019992");
- RETURN(BAD_VALUE);
- }
- (void)mParent;
- RETURN(OK);
-}
-
-Status Spatializer::EffectClient::disconnect() {
- mDisconnected = true;
- return Status::ok();
-}
-
-Status Spatializer::EffectClient::getCblk(media::SharedFileRegion* _aidl_return) {
- LOG_ALWAYS_FATAL_IF(!convertIMemoryToSharedFileRegion(mCblkMemory, _aidl_return));
- return Status::ok();
+void Spatializer::postFramesProcessedMsg(int frames) {
+ sp<AMessage> msg =
+ new AMessage(EngineCallbackHandler::kWhatOnFramesProcessed, mHandler);
+ msg->setInt32(EngineCallbackHandler::kNumFramesKey, frames);
+ msg->post();
}
} // namespace android
diff --git a/services/audiopolicy/service/Spatializer.h b/services/audiopolicy/service/Spatializer.h
index f0ab605..4d77b78 100644
--- a/services/audiopolicy/service/Spatializer.h
+++ b/services/audiopolicy/service/Spatializer.h
@@ -19,12 +19,15 @@
#include <android/media/BnEffect.h>
#include <android/media/BnSpatializer.h>
-#include <android/media/HeadTrackingMode.h>
#include <android/media/SpatializationLevel.h>
-
+#include <android/media/SpatializationMode.h>
+#include <android/media/SpatializerHeadTrackingMode.h>
+#include <media/audiohal/EffectHalInterface.h>
+#include <media/stagefright/foundation/ALooper.h>
#include <media/AudioEffect.h>
#include <system/audio_effects/effect_spatializer.h>
+#include "SpatializerPoseController.h"
namespace android {
@@ -80,24 +83,41 @@
* made to audio policy manager to release and close the spatializer output stream and the
* spatializer mixer thread is destroyed.
*/
-class Spatializer : public media::BnSpatializer, public IBinder::DeathRecipient {
-public:
-
+class Spatializer : public media::BnSpatializer,
+ public IBinder::DeathRecipient,
+ private SpatializerPoseController::Listener {
+ public:
static sp<Spatializer> create(SpatializerPolicyCallback *callback);
~Spatializer() override;
+ /** RefBase */
+ void onFirstRef();
+
/** ISpatializer, see ISpatializer.aidl */
binder::Status release() override;
binder::Status getSupportedLevels(std::vector<media::SpatializationLevel>* levels) override;
binder::Status setLevel(media::SpatializationLevel level) override;
binder::Status getLevel(media::SpatializationLevel *level) override;
+ binder::Status isHeadTrackingSupported(bool *supports);
binder::Status getSupportedHeadTrackingModes(
- std::vector<media::HeadTrackingMode>* modes) override;
- binder::Status setDesiredHeadTrackingMode(media::HeadTrackingMode mode) override;
- binder::Status getActualHeadTrackingMode(media::HeadTrackingMode *mode) override;
- binder::Status recenterHeadtracker() override;
+ std::vector<media::SpatializerHeadTrackingMode>* modes) override;
+ binder::Status setDesiredHeadTrackingMode(
+ media::SpatializerHeadTrackingMode mode) override;
+ binder::Status getActualHeadTrackingMode(
+ media::SpatializerHeadTrackingMode* mode) override;
+ binder::Status recenterHeadTracker() override;
binder::Status setGlobalTransform(const std::vector<float>& screenToStage) override;
+ binder::Status setHeadSensor(int sensorHandle) override;
+ binder::Status setScreenSensor(int sensorHandle) override;
+ binder::Status setDisplayOrientation(float physicalToLogicalAngle) override;
+ binder::Status setHingeAngle(float hingeAngle) override;
+ binder::Status getSupportedModes(std::vector<media::SpatializationMode>* modes) override;
+ binder::Status registerHeadTrackingCallback(
+ const sp<media::ISpatializerHeadTrackingCallback>& callback) override;
+ binder::Status setParameter(int key, const std::vector<unsigned char>& value) override;
+ binder::Status getParameter(int key, std::vector<unsigned char> *value) override;
+ binder::Status getOutput(int *output);
/** IBinder::DeathRecipient. Listen to the death of the INativeSpatializerCallback. */
virtual void binderDied(const wp<IBinder>& who);
@@ -107,8 +127,10 @@
*/
status_t registerCallback(const sp<media::INativeSpatializerCallback>& callback);
+ status_t loadEngineConfiguration(sp<EffectHalInterface> effect);
+
/** Level getter for use by local classes. */
- media::SpatializationLevel getLevel() const { Mutex::Autolock _l(mLock); return mLevel; }
+ media::SpatializationLevel getLevel() const { std::lock_guard lock(mLock); return mLevel; }
/** Called by audio policy service when the special output mixer dedicated to spatialization
* is opened and the spatializer engine must be created.
@@ -119,84 +141,194 @@
*/
audio_io_handle_t detachOutput();
/** Returns the output stream the spatializer is attached to. */
- audio_io_handle_t getOutput() const { Mutex::Autolock _l(mLock); return mOutput; }
-
- /** Sets the channel mask, sampling rate and format for the spatializer input. */
- void setAudioInConfig(const audio_config_base_t& config) {
- Mutex::Autolock _l(mLock);
- mAudioInConfig = config;
- }
+ audio_io_handle_t getOutput() const { std::lock_guard lock(mLock); return mOutput; }
/** Gets the channel mask, sampling rate and format set for the spatializer input. */
- audio_config_base_t getAudioInConfig() const {
- Mutex::Autolock _l(mLock);
- return mAudioInConfig;
- }
+ audio_config_base_t getAudioInConfig() const;
- /** An implementation of an IEffect interface that can be used to pass advanced parameters to
- * the spatializer engine. All APis are noop (i.e. the interface cannot be used to control
- * the effect) except for passing parameters via the command() API. */
- class EffectClient: public android::media::BnEffect {
- public:
-
- EffectClient(const sp<media::IEffectClient>& effectClient,
- Spatializer& parent);
- virtual ~EffectClient();
-
- // IEffect
- android::binder::Status enable(int32_t* _aidl_return) override;
- android::binder::Status disable(int32_t* _aidl_return) override;
- android::binder::Status command(int32_t cmdCode,
- const std::vector<uint8_t>& cmdData,
- int32_t maxResponseSize,
- std::vector<uint8_t>* response,
- int32_t* _aidl_return) override;
- android::binder::Status disconnect() override;
- android::binder::Status getCblk(media::SharedFileRegion* _aidl_return) override;
-
- private:
- const sp<media::IEffectClient> mEffectClient;
- sp<IMemory> mCblkMemory;
- const Spatializer& mParent;
- bool mDisconnected = false;
- };
+ void calculateHeadPose();
private:
-
Spatializer(effect_descriptor_t engineDescriptor,
SpatializerPolicyCallback *callback);
-
static void engineCallback(int32_t event, void* user, void *info);
+ // From VirtualizerStageController::Listener
+ void onHeadToStagePose(const media::Pose3f& headToStage) override;
+ void onActualModeChange(media::HeadTrackingMode mode) override;
+
+ void onHeadToStagePoseMsg(const std::vector<float>& headToStage);
+ void onActualModeChangeMsg(media::HeadTrackingMode mode);
+
+ static constexpr int kMaxEffectParamValues = 10;
+ /**
+ * Get a parameter from spatializer engine by calling the effect HAL command method directly.
+ * To be used when the engine instance mEngine is not yet created in the effect framework.
+ * When MULTI_VALUES is false, the expected reply is only one value of type T.
+ * When MULTI_VALUES is true, the expected reply is made of a number (of type T) indicating
+ * how many values are returned, followed by this number for values of type T.
+ */
+ template<bool MULTI_VALUES, typename T>
+ status_t getHalParameter(sp<EffectHalInterface> effect, uint32_t type,
+ std::vector<T> *values) {
+ static_assert(sizeof(T) <= sizeof(uint32_t), "The size of T must less than 32 bits");
+
+ uint32_t cmd[sizeof(effect_param_t) / sizeof(uint32_t) + 1];
+ uint32_t reply[sizeof(effect_param_t) / sizeof(uint32_t) + 2 + kMaxEffectParamValues];
+
+ effect_param_t *p = (effect_param_t *)cmd;
+ p->psize = sizeof(uint32_t);
+ if (MULTI_VALUES) {
+ p->vsize = (kMaxEffectParamValues + 1) * sizeof(T);
+ } else {
+ p->vsize = sizeof(T);
+ }
+ *(uint32_t *)p->data = type;
+ uint32_t replySize = sizeof(effect_param_t) + p->psize + p->vsize;
+
+ status_t status = effect->command(EFFECT_CMD_GET_PARAM,
+ sizeof(effect_param_t) + sizeof(uint32_t), cmd,
+ &replySize, reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ if (p->status != NO_ERROR) {
+ return p->status;
+ }
+ if (replySize <
+ sizeof(effect_param_t) + sizeof(uint32_t) + (MULTI_VALUES ? 2 : 1) * sizeof(T)) {
+ return BAD_VALUE;
+ }
+
+ T *params = (T *)((uint8_t *)reply + sizeof(effect_param_t) + sizeof(uint32_t));
+ int numParams = 1;
+ if (MULTI_VALUES) {
+ numParams = (int)*params++;
+ }
+ if (numParams > kMaxEffectParamValues) {
+ return BAD_VALUE;
+ }
+ (*values).clear();
+ std::copy(¶ms[0], ¶ms[numParams], back_inserter(*values));
+ return NO_ERROR;
+ }
+
+ /**
+ * Set a parameter to spatializer engine by calling setParameter on mEngine AudioEffect object.
+ * It is possible to pass more than one value of type T according to the parameter type
+ * according to values vector size.
+ */
+ template<typename T>
+ status_t setEffectParameter_l(uint32_t type, const std::vector<T>& values) REQUIRES(mLock) {
+ static_assert(sizeof(T) <= sizeof(uint32_t), "The size of T must less than 32 bits");
+
+ uint32_t cmd[sizeof(effect_param_t) / sizeof(uint32_t) + 1 + values.size()];
+ effect_param_t *p = (effect_param_t *)cmd;
+ p->psize = sizeof(uint32_t);
+ p->vsize = sizeof(T) * values.size();
+ *(uint32_t *)p->data = type;
+ memcpy((uint32_t *)p->data + 1, values.data(), sizeof(T) * values.size());
+
+ status_t status = mEngine->setParameter(p);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ if (p->status != NO_ERROR) {
+ return p->status;
+ }
+ return NO_ERROR;
+ }
+
+ /**
+ * Get a parameter from spatializer engine by calling getParameter on AudioEffect object.
+ * It is possible to read more than one value of type T according to the parameter type
+ * by specifying values vector size.
+ */
+ template<typename T>
+ status_t getEffectParameter_l(uint32_t type, std::vector<T> *values) REQUIRES(mLock) {
+ static_assert(sizeof(T) <= sizeof(uint32_t), "The size of T must less than 32 bits");
+
+ uint32_t cmd[sizeof(effect_param_t) / sizeof(uint32_t) + 1 + values->size()];
+ effect_param_t *p = (effect_param_t *)cmd;
+ p->psize = sizeof(uint32_t);
+ p->vsize = sizeof(T) * values->size();
+ *(uint32_t *)p->data = type;
+
+ status_t status = mEngine->getParameter(p);
+
+ if (status != NO_ERROR) {
+ return status;
+ }
+ if (p->status != NO_ERROR) {
+ return p->status;
+ }
+
+ int numValues = std::min(p->vsize / sizeof(T), values->size());
+ (*values).clear();
+ T *retValues = (T *)((uint8_t *)p->data + sizeof(uint32_t));
+ std::copy(&retValues[0], &retValues[numValues], back_inserter(*values));
+
+ return NO_ERROR;
+ }
+
+ void postFramesProcessedMsg(int frames);
+
/** Effect engine descriptor */
const effect_descriptor_t mEngineDescriptor;
/** Callback interface to parent audio policy service */
SpatializerPolicyCallback* mPolicyCallback;
/** Mutex protecting internal state */
- mutable Mutex mLock;
+ mutable std::mutex mLock;
/** Client AudioEffect for the engine */
sp<AudioEffect> mEngine GUARDED_BY(mLock);
/** Output stream the spatializer mixer thread is attached to */
audio_io_handle_t mOutput GUARDED_BY(mLock) = AUDIO_IO_HANDLE_NONE;
- /** Virtualizer engine input configuration */
- audio_config_base_t mAudioInConfig GUARDED_BY(mLock) = AUDIO_CONFIG_BASE_INITIALIZER;
/** Callback interface to the client (AudioService) controlling this`Spatializer */
sp<media::INativeSpatializerCallback> mSpatializerCallback GUARDED_BY(mLock);
+ /** Callback interface for head tracking */
+ sp<media::ISpatializerHeadTrackingCallback> mHeadTrackingCallback GUARDED_BY(mLock);
+
/** Requested spatialization level */
media::SpatializationLevel mLevel GUARDED_BY(mLock) = media::SpatializationLevel::NONE;
- /** Requested head tracking mode */
- media::HeadTrackingMode mHeadTrackingMode GUARDED_BY(mLock)
- = media::HeadTrackingMode::DISABLED;
- /** Configured screen to stage transform */
- std::vector<float> mScreenToStageTransform GUARDED_BY(mLock);
- /** Extended IEffect interface is one has been created */
- sp<EffectClient> mEffectClient GUARDED_BY(mLock);
+ /** Control logic for head-tracking, etc. */
+ std::shared_ptr<SpatializerPoseController> mPoseController GUARDED_BY(mLock);
+
+ /** Last requested head tracking mode */
+ media::HeadTrackingMode mDesiredHeadTrackingMode GUARDED_BY(mLock)
+ = media::HeadTrackingMode::STATIC;
+
+ /** Last-reported actual head-tracking mode. */
+ media::SpatializerHeadTrackingMode mActualHeadTrackingMode GUARDED_BY(mLock)
+ = media::SpatializerHeadTrackingMode::DISABLED;
+
+ /** Selected Head pose sensor */
+ int32_t mHeadSensor GUARDED_BY(mLock) = SpatializerPoseController::INVALID_SENSOR;
+
+ /** Selected Screen pose sensor */
+ int32_t mScreenSensor GUARDED_BY(mLock) = SpatializerPoseController::INVALID_SENSOR;
+
+ /** Last display orientation received */
+ static constexpr float kDisplayOrientationInvalid = 1000;
+ float mDisplayOrientation GUARDED_BY(mLock) = kDisplayOrientationInvalid;
+
+ std::vector<media::SpatializationLevel> mLevels;
+ std::vector<media::SpatializationMode> mSpatializationModes;
+ std::vector<audio_channel_mask_t> mChannelMasks;
+ bool mSupportsHeadTracking;
+
+ // Looper thread for mEngine callbacks
+ class EngineCallbackHandler;
+
+ sp<ALooper> mLooper;
+ sp<EngineCallbackHandler> mHandler;
+
+ static const std::vector<const char *> sHeadPoseKeys;
};
diff --git a/services/audiopolicy/service/SpatializerPoseController.cpp b/services/audiopolicy/service/SpatializerPoseController.cpp
new file mode 100644
index 0000000..eb23298
--- /dev/null
+++ b/services/audiopolicy/service/SpatializerPoseController.cpp
@@ -0,0 +1,237 @@
+/*
+ * Copyright 2021, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include "SpatializerPoseController.h"
+
+#define LOG_TAG "SpatializerPoseController"
+//#define LOG_NDEBUG 0
+#include <sensor/Sensor.h>
+#include <utils/Log.h>
+#include <utils/SystemClock.h>
+
+namespace android {
+
+using media::createHeadTrackingProcessor;
+using media::HeadTrackingMode;
+using media::HeadTrackingProcessor;
+using media::Pose3f;
+using media::SensorPoseProvider;
+using media::Twist3f;
+
+using namespace std::chrono_literals;
+
+namespace {
+
+// This is how fast, in m/s, we allow position to shift during rate-limiting.
+constexpr auto kMaxTranslationalVelocity = 2;
+
+// This is how fast, in rad/s, we allow rotation angle to shift during rate-limiting.
+constexpr auto kMaxRotationalVelocity = 4 * M_PI;
+
+// This should be set to the typical time scale that the translation sensors used drift in. This
+// means, loosely, for how long we can trust the reading to be "accurate enough". This would
+// determine the time constants used for high-pass filtering those readings. If the value is set
+// too high, we may experience drift. If it is set too low, we may experience poses tending toward
+// identity too fast.
+constexpr auto kTranslationalDriftTimeConstant = 20s;
+
+// This should be set to the typical time scale that the rotation sensors used drift in. This
+// means, loosely, for how long we can trust the reading to be "accurate enough". This would
+// determine the time constants used for high-pass filtering those readings. If the value is set
+// too high, we may experience drift. If it is set too low, we may experience poses tending toward
+// identity too fast.
+constexpr auto kRotationalDriftTimeConstant = 20s;
+
+// This is how far into the future we predict the head pose, using linear extrapolation based on
+// twist (velocity). It should be set to a value that matches the characteristic durations of moving
+// one's head. The higher we set this, the more latency we are able to reduce, but setting this too
+// high will result in high prediction errors whenever the head accelerates (changes velocity).
+constexpr auto kPredictionDuration = 10ms;
+
+// After losing this many consecutive samples from either sensor, we would treat the measurement as
+// stale;
+constexpr auto kMaxLostSamples = 4;
+
+// Time units for system clock ticks. This is what the Sensor Framework timestamps represent and
+// what we use for pose filtering.
+using Ticks = std::chrono::nanoseconds;
+
+// How many ticks in a second.
+constexpr auto kTicksPerSecond = Ticks::period::den;
+
+} // namespace
+
+SpatializerPoseController::SpatializerPoseController(Listener* listener,
+ std::chrono::microseconds sensorPeriod,
+ std::chrono::microseconds maxUpdatePeriod)
+ : mListener(listener),
+ mSensorPeriod(sensorPeriod),
+ mProcessor(createHeadTrackingProcessor(HeadTrackingProcessor::Options{
+ .maxTranslationalVelocity = kMaxTranslationalVelocity / kTicksPerSecond,
+ .maxRotationalVelocity = kMaxRotationalVelocity / kTicksPerSecond,
+ .translationalDriftTimeConstant = Ticks(kTranslationalDriftTimeConstant).count(),
+ .rotationalDriftTimeConstant = Ticks(kRotationalDriftTimeConstant).count(),
+ .freshnessTimeout = Ticks(sensorPeriod * kMaxLostSamples).count(),
+ .predictionDuration = Ticks(kPredictionDuration).count(),
+ })),
+ mPoseProvider(SensorPoseProvider::create("headtracker", this)),
+ mThread([this, maxUpdatePeriod] {
+ while (true) {
+ Pose3f headToStage;
+ std::optional<HeadTrackingMode> modeIfChanged;
+ {
+ std::unique_lock lock(mMutex);
+ mCondVar.wait_for(lock, maxUpdatePeriod,
+ [this] { return mShouldExit || mShouldCalculate; });
+ if (mShouldExit) {
+ ALOGV("Exiting thread");
+ return;
+ }
+
+ // Calculate.
+ std::tie(headToStage, modeIfChanged) = calculate_l();
+ }
+
+ // Invoke the callbacks outside the lock.
+ mListener->onHeadToStagePose(headToStage);
+ if (modeIfChanged) {
+ mListener->onActualModeChange(modeIfChanged.value());
+ }
+
+ {
+ std::lock_guard lock(mMutex);
+ if (!mCalculated) {
+ mCalculated = true;
+ mCondVar.notify_all();
+ }
+ mShouldCalculate = false;
+ }
+ }
+ }) {}
+
+SpatializerPoseController::~SpatializerPoseController() {
+ {
+ std::unique_lock lock(mMutex);
+ mShouldExit = true;
+ mCondVar.notify_all();
+ }
+ mThread.join();
+}
+
+void SpatializerPoseController::setHeadSensor(int32_t sensor) {
+ std::lock_guard lock(mMutex);
+ // Stop current sensor, if valid and different from the other sensor.
+ if (mHeadSensor != INVALID_SENSOR && mHeadSensor != mScreenSensor) {
+ mPoseProvider->stopSensor(mHeadSensor);
+ }
+
+ if (sensor != INVALID_SENSOR) {
+ if (sensor != mScreenSensor) {
+ // Start new sensor.
+ mHeadSensor =
+ mPoseProvider->startSensor(sensor, mSensorPeriod) ? sensor : INVALID_SENSOR;
+ } else {
+ // Sensor is already enabled.
+ mHeadSensor = mScreenSensor;
+ }
+ } else {
+ mHeadSensor = INVALID_SENSOR;
+ }
+
+ mProcessor->recenter(true, false);
+}
+
+void SpatializerPoseController::setScreenSensor(int32_t sensor) {
+ std::lock_guard lock(mMutex);
+ // Stop current sensor, if valid and different from the other sensor.
+ if (mScreenSensor != INVALID_SENSOR && mScreenSensor != mHeadSensor) {
+ mPoseProvider->stopSensor(mScreenSensor);
+ }
+
+ if (sensor != INVALID_SENSOR) {
+ if (sensor != mHeadSensor) {
+ // Start new sensor.
+ mScreenSensor =
+ mPoseProvider->startSensor(sensor, mSensorPeriod) ? sensor : INVALID_SENSOR;
+ } else {
+ // Sensor is already enabled.
+ mScreenSensor = mHeadSensor;
+ }
+ } else {
+ mScreenSensor = INVALID_SENSOR;
+ }
+
+ mProcessor->recenter(false, true);
+}
+
+void SpatializerPoseController::setDesiredMode(HeadTrackingMode mode) {
+ std::lock_guard lock(mMutex);
+ mProcessor->setDesiredMode(mode);
+}
+
+void SpatializerPoseController::setScreenToStagePose(const Pose3f& screenToStage) {
+ std::lock_guard lock(mMutex);
+ mProcessor->setScreenToStagePose(screenToStage);
+}
+
+void SpatializerPoseController::setDisplayOrientation(float physicalToLogicalAngle) {
+ std::lock_guard lock(mMutex);
+ mProcessor->setDisplayOrientation(physicalToLogicalAngle);
+}
+
+void SpatializerPoseController::calculateAsync() {
+ std::lock_guard lock(mMutex);
+ mShouldCalculate = true;
+ mCondVar.notify_all();
+}
+
+void SpatializerPoseController::waitUntilCalculated() {
+ std::unique_lock lock(mMutex);
+ mCondVar.wait(lock, [this] { return mCalculated; });
+}
+
+std::tuple<media::Pose3f, std::optional<media::HeadTrackingMode>>
+SpatializerPoseController::calculate_l() {
+ Pose3f headToStage;
+ HeadTrackingMode mode;
+ std::optional<media::HeadTrackingMode> modeIfChanged;
+
+ mProcessor->calculate(elapsedRealtimeNano());
+ headToStage = mProcessor->getHeadToStagePose();
+ mode = mProcessor->getActualMode();
+ if (!mActualMode.has_value() || mActualMode.value() != mode) {
+ mActualMode = mode;
+ modeIfChanged = mode;
+ }
+ return std::make_tuple(headToStage, modeIfChanged);
+}
+
+void SpatializerPoseController::recenter() {
+ std::lock_guard lock(mMutex);
+ mProcessor->recenter();
+}
+
+void SpatializerPoseController::onPose(int64_t timestamp, int32_t sensor, const Pose3f& pose,
+ const std::optional<Twist3f>& twist) {
+ std::lock_guard lock(mMutex);
+ if (sensor == mHeadSensor) {
+ mProcessor->setWorldToHeadPose(timestamp, pose, twist.value_or(Twist3f()));
+ }
+ if (sensor == mScreenSensor) {
+ mProcessor->setWorldToScreenPose(timestamp, pose);
+ }
+}
+
+} // namespace android
diff --git a/services/audiopolicy/service/SpatializerPoseController.h b/services/audiopolicy/service/SpatializerPoseController.h
new file mode 100644
index 0000000..c579622
--- /dev/null
+++ b/services/audiopolicy/service/SpatializerPoseController.h
@@ -0,0 +1,142 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#pragma once
+
+#include <chrono>
+#include <condition_variable>
+#include <limits>
+#include <memory>
+#include <mutex>
+#include <thread>
+
+#include <media/HeadTrackingProcessor.h>
+#include <media/SensorPoseProvider.h>
+
+namespace android {
+
+/**
+ * This class encapsulates the logic for pose processing, intended for driving a spatializer effect.
+ * This includes integration with the Sensor sub-system for retrieving sensor data, doing all the
+ * necessary processing, etc.
+ *
+ * Calculations happen on a dedicated thread and published to the client via the Listener interface.
+ * A calculation may be triggered in one of two ways:
+ * - By calling calculateAsync() - calculation will be kicked off in the background.
+ * - By setting a timeout in the ctor, a calculation will be triggered after the timeout elapsed
+ * from the last calculateAsync() call.
+ *
+ * This class is thread-safe.
+ */
+class SpatializerPoseController : private media::SensorPoseProvider::Listener {
+ public:
+ static constexpr int32_t INVALID_SENSOR = media::SensorPoseProvider::INVALID_HANDLE;
+
+ /**
+ * Listener interface for getting pose and mode updates.
+ * Methods will always be invoked from a designated thread.
+ */
+ class Listener {
+ public:
+ virtual ~Listener() = default;
+
+ virtual void onHeadToStagePose(const media::Pose3f&) = 0;
+ virtual void onActualModeChange(media::HeadTrackingMode) = 0;
+ };
+
+ /**
+ * Ctor.
+ * sensorPeriod determines how often to receive updates from the sensors (input rate).
+ * maxUpdatePeriod determines how often to produce an output when calculateAsync() isn't
+ * invoked.
+ */
+ SpatializerPoseController(Listener* listener, std::chrono::microseconds sensorPeriod,
+ std::chrono::microseconds maxUpdatePeriod);
+
+ /** Dtor. */
+ ~SpatializerPoseController();
+
+ /**
+ * Set the sensor that is to be used for head-tracking.
+ * INVALID_SENSOR can be used to disable head-tracking.
+ */
+ void setHeadSensor(int32_t sensor);
+
+ /**
+ * Set the sensor that is to be used for screen-tracking.
+ * INVALID_SENSOR can be used to disable screen-tracking.
+ */
+ void setScreenSensor(int32_t sensor);
+
+ /** Sets the desired head-tracking mode. */
+ void setDesiredMode(media::HeadTrackingMode mode);
+
+ /**
+ * Set the screen-to-stage pose, used in all modes.
+ */
+ void setScreenToStagePose(const media::Pose3f& screenToStage);
+
+ /**
+ * Sets the display orientation.
+ * Orientation is expressed in the angle of rotation from the physical "up" side of the screen
+ * to the logical "up" side of the content displayed the screen. Counterclockwise angles, as
+ * viewed while facing the screen are positive.
+ */
+ void setDisplayOrientation(float physicalToLogicalAngle);
+
+ /**
+ * This causes the current poses for both the head and screen to be considered "center".
+ */
+ void recenter();
+
+ /**
+ * This call triggers the recalculation of the output and the invocation of the relevant
+ * callbacks. This call is async and the callbacks will be triggered shortly after.
+ */
+ void calculateAsync();
+
+ /**
+ * Blocks until calculation and invocation of the respective callbacks has happened at least
+ * once. Do not call from within callbacks.
+ */
+ void waitUntilCalculated();
+
+ private:
+ mutable std::mutex mMutex;
+ Listener* const mListener;
+ const std::chrono::microseconds mSensorPeriod;
+ // Order matters for the following two members to ensure correct destruction.
+ std::unique_ptr<media::HeadTrackingProcessor> mProcessor;
+ std::unique_ptr<media::SensorPoseProvider> mPoseProvider;
+ int32_t mHeadSensor = media::SensorPoseProvider::INVALID_HANDLE;
+ int32_t mScreenSensor = media::SensorPoseProvider::INVALID_HANDLE;
+ std::optional<media::HeadTrackingMode> mActualMode;
+ std::thread mThread;
+ std::condition_variable mCondVar;
+ bool mShouldCalculate = true;
+ bool mShouldExit = false;
+ bool mCalculated = false;
+
+ void onPose(int64_t timestamp, int32_t sensor, const media::Pose3f& pose,
+ const std::optional<media::Twist3f>& twist) override;
+
+ /**
+ * Calculates the new outputs and updates internal state. Must be called with the lock held.
+ * Returns values that should be passed to the respective callbacks.
+ */
+ std::tuple<media::Pose3f, std::optional<media::HeadTrackingMode>> calculate_l();
+};
+
+} // namespace android
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClientForHdmi.h b/services/audiopolicy/tests/AudioPolicyManagerTestClientForHdmi.h
index a5ad9b1..7343b9b 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClientForHdmi.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClientForHdmi.h
@@ -28,19 +28,26 @@
class AudioPolicyManagerTestClientForHdmi : public AudioPolicyManagerTestClient {
public:
String8 getParameters(audio_io_handle_t /* ioHandle */, const String8& /* keys*/ ) override {
+ AudioParameter mAudioParameters;
+ std::string formats;
+ for (const auto& f : mSupportedFormats) {
+ if (!formats.empty()) formats += AUDIO_PARAMETER_VALUE_LIST_SEPARATOR;
+ formats += audio_format_to_string(f);
+ }
+ mAudioParameters.add(
+ String8(AudioParameter::keyStreamSupportedFormats),
+ String8(formats.c_str()));
+ mAudioParameters.addInt(String8(AudioParameter::keyStreamSupportedSamplingRates), 48000);
+ mAudioParameters.add(String8(AudioParameter::keyStreamSupportedChannels), String8(""));
return mAudioParameters.toString();
}
void addSupportedFormat(audio_format_t format) override {
- mAudioParameters.add(
- String8(AudioParameter::keyStreamSupportedFormats),
- String8(audio_format_to_string(format)));
- mAudioParameters.addInt(String8(AudioParameter::keyStreamSupportedSamplingRates), 48000);
- mAudioParameters.add(String8(AudioParameter::keyStreamSupportedChannels), String8(""));
+ mSupportedFormats.insert(format);
}
private:
- AudioParameter mAudioParameters;
+ std::set<audio_format_t> mSupportedFormats;
};
-} // namespace android
\ No newline at end of file
+} // namespace android
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index a16ab7d..9c1adc6 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -151,7 +151,7 @@
void AudioPolicyManagerTest::SetUp() {
mClient.reset(getClient());
mManager.reset(new AudioPolicyTestManager(mClient.get()));
- SetUpManagerConfig(); // Subclasses may want to customize the config.
+ ASSERT_NO_FATAL_FAILURE(SetUpManagerConfig()); // Subclasses may want to customize the config.
ASSERT_EQ(NO_ERROR, mManager->initialize());
ASSERT_EQ(NO_ERROR, mManager->initCheck());
}
@@ -401,7 +401,7 @@
void AudioPolicyManagerTestMsd::SetUpManagerConfig() {
// TODO: Consider using Serializer to load part of the config from a string.
- AudioPolicyManagerTest::SetUpManagerConfig();
+ ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTest::SetUpManagerConfig());
AudioPolicyConfig& config = mManager->getConfig();
mMsdOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_BUS);
sp<AudioProfile> pcmOutputProfile = new AudioProfile(
@@ -660,6 +660,7 @@
void AudioPolicyManagerTestWithConfigurationFile::SetUpManagerConfig() {
status_t status = deserializeAudioPolicyFile(getConfigFile().c_str(), &mManager->getConfig());
ASSERT_EQ(NO_ERROR, status);
+ mManager->getConfig().setSource(getConfigFile());
}
TEST_F(AudioPolicyManagerTestWithConfigurationFile, InitSuccess) {
@@ -803,7 +804,8 @@
}
class AudioPolicyManagerTestForHdmi
- : public AudioPolicyManagerTestWithConfigurationFile {
+ : public AudioPolicyManagerTestWithConfigurationFile,
+ public testing::WithParamInterface<audio_format_t> {
protected:
void SetUp() override;
std::string getConfigFile() override { return sTvConfig; }
@@ -824,7 +826,8 @@
"test_settop_box_surround_configuration.xml";
void AudioPolicyManagerTestForHdmi::SetUp() {
- AudioPolicyManagerTest::SetUp();
+ ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTest::SetUp());
+ mClient->addSupportedFormat(AUDIO_FORMAT_AC3);
mClient->addSupportedFormat(AUDIO_FORMAT_E_AC3);
mManager->setDeviceConnectionState(
AUDIO_DEVICE_OUT_HDMI, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
@@ -914,76 +917,90 @@
return formats;
}
-TEST_F(AudioPolicyManagerTestForHdmi, GetSurroundFormatsReturnsSupportedFormats) {
+TEST_P(AudioPolicyManagerTestForHdmi, GetSurroundFormatsReturnsSupportedFormats) {
mManager->setForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND, AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS);
auto surroundFormats = getSurroundFormatsHelper();
- ASSERT_EQ(1, surroundFormats.count(AUDIO_FORMAT_E_AC3));
+ ASSERT_EQ(1, surroundFormats.count(GetParam()));
}
-TEST_F(AudioPolicyManagerTestForHdmi,
+TEST_P(AudioPolicyManagerTestForHdmi,
GetSurroundFormatsReturnsManipulatedFormats) {
mManager->setForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND, AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL);
status_t ret =
- mManager->setSurroundFormatEnabled(AUDIO_FORMAT_E_AC3, false /*enabled*/);
+ mManager->setSurroundFormatEnabled(GetParam(), false /*enabled*/);
ASSERT_EQ(NO_ERROR, ret);
auto surroundFormats = getSurroundFormatsHelper();
- ASSERT_EQ(1, surroundFormats.count(AUDIO_FORMAT_E_AC3));
- ASSERT_FALSE(surroundFormats[AUDIO_FORMAT_E_AC3]);
+ ASSERT_EQ(1, surroundFormats.count(GetParam()));
+ ASSERT_FALSE(surroundFormats[GetParam()]);
- ret = mManager->setSurroundFormatEnabled(AUDIO_FORMAT_E_AC3, true /*enabled*/);
+ ret = mManager->setSurroundFormatEnabled(GetParam(), true /*enabled*/);
ASSERT_EQ(NO_ERROR, ret);
surroundFormats = getSurroundFormatsHelper();
- ASSERT_EQ(1, surroundFormats.count(AUDIO_FORMAT_E_AC3));
- ASSERT_TRUE(surroundFormats[AUDIO_FORMAT_E_AC3]);
+ ASSERT_EQ(1, surroundFormats.count(GetParam()));
+ ASSERT_TRUE(surroundFormats[GetParam()]);
- ret = mManager->setSurroundFormatEnabled(AUDIO_FORMAT_E_AC3, false /*enabled*/);
+ ret = mManager->setSurroundFormatEnabled(GetParam(), false /*enabled*/);
ASSERT_EQ(NO_ERROR, ret);
surroundFormats = getSurroundFormatsHelper();
- ASSERT_EQ(1, surroundFormats.count(AUDIO_FORMAT_E_AC3));
- ASSERT_FALSE(surroundFormats[AUDIO_FORMAT_E_AC3]);
+ ASSERT_EQ(1, surroundFormats.count(GetParam()));
+ ASSERT_FALSE(surroundFormats[GetParam()]);
}
-TEST_F(AudioPolicyManagerTestForHdmi,
+TEST_P(AudioPolicyManagerTestForHdmi,
ListAudioPortsReturnManipulatedHdmiFormats) {
mManager->setForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND, AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL);
- ASSERT_EQ(NO_ERROR, mManager->setSurroundFormatEnabled(AUDIO_FORMAT_E_AC3, false /*enabled*/));
+ ASSERT_EQ(NO_ERROR, mManager->setSurroundFormatEnabled(GetParam(), false /*enabled*/));
auto formats = getFormatsFromPorts();
- ASSERT_EQ(0, formats.count(AUDIO_FORMAT_E_AC3));
+ ASSERT_EQ(0, formats.count(GetParam()));
- ASSERT_EQ(NO_ERROR, mManager->setSurroundFormatEnabled(AUDIO_FORMAT_E_AC3, true /*enabled*/));
+ ASSERT_EQ(NO_ERROR, mManager->setSurroundFormatEnabled(GetParam(), true /*enabled*/));
formats = getFormatsFromPorts();
- ASSERT_EQ(1, formats.count(AUDIO_FORMAT_E_AC3));
+ ASSERT_EQ(1, formats.count(GetParam()));
}
-TEST_F(AudioPolicyManagerTestForHdmi,
+TEST_P(AudioPolicyManagerTestForHdmi,
GetReportedSurroundFormatsReturnsHdmiReportedFormats) {
mManager->setForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND, AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS);
auto surroundFormats = getReportedSurroundFormatsHelper();
- ASSERT_EQ(1, std::count(surroundFormats.begin(), surroundFormats.end(), AUDIO_FORMAT_E_AC3));
+ ASSERT_EQ(1, std::count(surroundFormats.begin(), surroundFormats.end(), GetParam()));
}
-TEST_F(AudioPolicyManagerTestForHdmi,
+TEST_P(AudioPolicyManagerTestForHdmi,
GetReportedSurroundFormatsReturnsNonManipulatedHdmiReportedFormats) {
mManager->setForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND, AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL);
- status_t ret = mManager->setSurroundFormatEnabled(AUDIO_FORMAT_E_AC3, false /*enabled*/);
+ status_t ret = mManager->setSurroundFormatEnabled(GetParam(), false /*enabled*/);
ASSERT_EQ(NO_ERROR, ret);
auto surroundFormats = getReportedSurroundFormatsHelper();
- ASSERT_EQ(1, std::count(surroundFormats.begin(), surroundFormats.end(), AUDIO_FORMAT_E_AC3));
+ ASSERT_EQ(1, std::count(surroundFormats.begin(), surroundFormats.end(), GetParam()));
- ret = mManager->setSurroundFormatEnabled(AUDIO_FORMAT_E_AC3, true /*enabled*/);
+ ret = mManager->setSurroundFormatEnabled(GetParam(), true /*enabled*/);
ASSERT_EQ(NO_ERROR, ret);
surroundFormats = getReportedSurroundFormatsHelper();
- ASSERT_EQ(1, std::count(surroundFormats.begin(), surroundFormats.end(), AUDIO_FORMAT_E_AC3));
+ ASSERT_EQ(1, std::count(surroundFormats.begin(), surroundFormats.end(), GetParam()));
}
+TEST_P(AudioPolicyManagerTestForHdmi, GetSurroundFormatsIgnoresSupportedFormats) {
+ mManager->setForceUse(
+ AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND, AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER);
+ auto surroundFormats = getSurroundFormatsHelper();
+ ASSERT_EQ(1, surroundFormats.count(GetParam()));
+ ASSERT_FALSE(surroundFormats[GetParam()]);
+}
+
+INSTANTIATE_TEST_SUITE_P(SurroundFormatSupport, AudioPolicyManagerTestForHdmi,
+ testing::Values(AUDIO_FORMAT_AC3, AUDIO_FORMAT_E_AC3),
+ [](const ::testing::TestParamInfo<AudioPolicyManagerTestForHdmi::ParamType>& info) {
+ return audio_format_to_string(info.param);
+ });
+
class AudioPolicyManagerTestDPNoRemoteSubmixModule : public AudioPolicyManagerTestDynamicPolicy {
protected:
std::string getConfigFile() override { return sPrimaryOnlyConfig; }
@@ -1035,7 +1052,7 @@
};
void AudioPolicyManagerTestDPPlaybackReRouting::SetUp() {
- AudioPolicyManagerTestDynamicPolicy::SetUp();
+ ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTestDynamicPolicy::SetUp());
mTracker.reset(new RecordingActivityTracker());
@@ -1221,7 +1238,7 @@
};
void AudioPolicyManagerTestDPMixRecordInjection::SetUp() {
- AudioPolicyManagerTestDynamicPolicy::SetUp();
+ ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTestDynamicPolicy::SetUp());
mTracker.reset(new RecordingActivityTracker());
@@ -1375,7 +1392,8 @@
if (type == AUDIO_DEVICE_OUT_HDMI) {
// Set device connection state failed due to no device descriptor found
// For HDMI case, it is easier to simulate device descriptor not found error
- // by using a undeclared encoded format.
+ // by using an encoded format which isn't listed in the 'encodedFormats'
+ // attribute for this devicePort.
ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.c_str(), name.c_str(), AUDIO_FORMAT_MAT_2_1));
@@ -1519,7 +1537,7 @@
};
void AudioPolicyManagerDynamicHwModulesTest::SetUpManagerConfig() {
- AudioPolicyManagerTestWithConfigurationFile::SetUpManagerConfig();
+ ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTestWithConfigurationFile::SetUpManagerConfig());
// Only allow successful opening of "primary" hw module during APM initialization.
mClient->swapAllowedModuleNames({"primary"});
}
diff --git a/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
index 87f0ab9..41ed70c 100644
--- a/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
+++ b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
@@ -50,7 +50,8 @@
</devicePort>
<devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
</devicePort>
- <devicePort tagName="Hdmi" type="AUDIO_DEVICE_OUT_HDMI" role="sink">
+ <devicePort tagName="Hdmi" type="AUDIO_DEVICE_OUT_HDMI" role="sink"
+ encodedFormats="AUDIO_FORMAT_AC3">
</devicePort>
<devicePort tagName="Hdmi-In Mic" type="AUDIO_DEVICE_IN_HDMI" role="source">
</devicePort>
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 19a111d..54f6520 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -1336,7 +1336,7 @@
auto clientSp = current->getValue();
if (clientSp.get() != nullptr) { // should never be needed
if (!clientSp->canCastToApiClient(effectiveApiLevel)) {
- ALOGW("CameraService connect called from same client, but with a different"
+ ALOGW("CameraService connect called with a different"
" API level, evicting prior client...");
} else if (clientSp->getRemote() == remoteCallback) {
ALOGI("CameraService::connect X (PID %d) (second call from same"
@@ -1836,8 +1836,9 @@
// Set rotate-and-crop override behavior
if (mOverrideRotateAndCropMode != ANDROID_SCALER_ROTATE_AND_CROP_AUTO) {
client->setRotateAndCropOverride(mOverrideRotateAndCropMode);
- } else if (CameraServiceProxyWrapper::isRotateAndCropOverrideNeeded(clientPackageName,
- orientation, facing)) {
+ } else if ((effectiveApiLevel == API_2) &&
+ CameraServiceProxyWrapper::isRotateAndCropOverrideNeeded(clientPackageName,
+ orientation, facing) ) {
client->setRotateAndCropOverride(ANDROID_SCALER_ROTATE_AND_CROP_90);
}
@@ -2077,6 +2078,11 @@
id.string());
errorCode = ERROR_ILLEGAL_ARGUMENT;
break;
+ case -EBUSY:
+ msg = String8::format("Camera \"%s\" is in use",
+ id.string());
+ errorCode = ERROR_CAMERA_IN_USE;
+ break;
default:
msg = String8::format(
"Setting torch mode of camera \"%s\" to %d failed: %s (%d)",
@@ -2201,7 +2207,6 @@
newDeviceState |= vendorBits;
ALOGV("%s: New device state 0x%" PRIx64, __FUNCTION__, newDeviceState);
- Mutex::Autolock l(mServiceLock);
mCameraProviderManager->notifyDeviceStateChange(newDeviceState);
return Status::ok();
@@ -2235,7 +2240,7 @@
for (auto& current : clients) {
if (current != nullptr) {
const auto basicClient = current->getValue();
- if (basicClient.get() != nullptr) {
+ if (basicClient.get() != nullptr && basicClient->canCastToApiClient(API_2)) {
if (CameraServiceProxyWrapper::isRotateAndCropOverrideNeeded(
basicClient->getPackageName(), basicClient->getCameraOrientation(),
basicClient->getCameraFacing())) {
@@ -2396,7 +2401,8 @@
Mutex::Autolock lock(mCameraStatesLock);
for (auto& i : mCameraStates) {
cameraStatuses->emplace_back(i.first,
- mapToInterface(i.second->getStatus()), i.second->getUnavailablePhysicalIds());
+ mapToInterface(i.second->getStatus()), i.second->getUnavailablePhysicalIds(),
+ openCloseCallbackAllowed ? i.second->getClientPackage() : String8::empty());
}
}
// Remove the camera statuses that should be hidden from the client, we do
@@ -3766,6 +3772,16 @@
return count > 0;
}
+void CameraService::CameraState::setClientPackage(const String8& clientPackage) {
+ Mutex::Autolock lock(mStatusLock);
+ mClientPackage = clientPackage;
+}
+
+String8 CameraService::CameraState::getClientPackage() const {
+ Mutex::Autolock lock(mStatusLock);
+ return mClientPackage;
+}
+
// ----------------------------------------------------------------------------
// ClientEventListener
// ----------------------------------------------------------------------------
@@ -4329,6 +4345,18 @@
void CameraService::updateOpenCloseStatus(const String8& cameraId, bool open,
const String16& clientPackageName) {
+ auto state = getCameraState(cameraId);
+ if (state == nullptr) {
+ ALOGW("%s: Could not update the status for %s, no such device exists", __FUNCTION__,
+ cameraId.string());
+ return;
+ }
+ if (open) {
+ state->setClientPackage(String8(clientPackageName));
+ } else {
+ state->setClientPackage(String8::empty());
+ }
+
Mutex::Autolock lock(mStatusListenerLock);
for (const auto& it : mListenerList) {
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 224287e..7a69123 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -635,6 +635,12 @@
bool removeUnavailablePhysicalId(const String8& physicalId);
/**
+ * Set and get client package name.
+ */
+ void setClientPackage(const String8& clientPackage);
+ String8 getClientPackage() const;
+
+ /**
* Return the unavailable physical ids for this device.
*
* This method acquires mStatusLock.
@@ -646,6 +652,7 @@
const int mCost;
std::set<String8> mConflicting;
std::set<String8> mUnavailablePhysicalIds;
+ String8 mClientPackage;
mutable Mutex mStatusLock;
CameraParameters mShimParams;
const SystemCameraKind mSystemCameraKind;
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 80508e4..a406e62 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -864,7 +864,6 @@
if (fabs(maxDigitalZoom.data.f[0] - 1.f) > 0.00001f) {
params.set(CameraParameters::KEY_ZOOM, zoom);
- params.set(CameraParameters::KEY_MAX_ZOOM, NUM_ZOOM_STEPS - 1);
{
String8 zoomRatios;
@@ -872,18 +871,34 @@
float zoomIncrement = (maxDigitalZoom.data.f[0] - zoom) /
(NUM_ZOOM_STEPS-1);
bool addComma = false;
- for (size_t i=0; i < NUM_ZOOM_STEPS; i++) {
+ int previousZoom = -1;
+ size_t zoomSteps = 0;
+ for (size_t i = 0; i < NUM_ZOOM_STEPS; i++) {
+ int currentZoom = static_cast<int>(zoom * 100);
+ if (previousZoom == currentZoom) {
+ zoom += zoomIncrement;
+ continue;
+ }
if (addComma) zoomRatios += ",";
addComma = true;
- zoomRatios += String8::format("%d", static_cast<int>(zoom * 100));
+ zoomRatios += String8::format("%d", currentZoom);
zoom += zoomIncrement;
+ previousZoom = currentZoom;
+ zoomSteps++;
}
- params.set(CameraParameters::KEY_ZOOM_RATIOS, zoomRatios);
+
+ if (zoomSteps > 0) {
+ params.set(CameraParameters::KEY_ZOOM_RATIOS, zoomRatios);
+ params.set(CameraParameters::KEY_ZOOM_SUPPORTED,
+ CameraParameters::TRUE);
+ params.set(CameraParameters::KEY_MAX_ZOOM, zoomSteps - 1);
+ zoomAvailable = true;
+ } else {
+ params.set(CameraParameters::KEY_ZOOM_SUPPORTED,
+ CameraParameters::FALSE);
+ }
}
- params.set(CameraParameters::KEY_ZOOM_SUPPORTED,
- CameraParameters::TRUE);
- zoomAvailable = true;
} else {
params.set(CameraParameters::KEY_ZOOM_SUPPORTED,
CameraParameters::FALSE);
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index e2f8d011..1b2ceda 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -205,7 +205,7 @@
static const int MAX_INITIAL_PREVIEW_WIDTH = 1920;
static const int MAX_INITIAL_PREVIEW_HEIGHT = 1080;
// Aspect ratio tolerance
- static const CONSTEXPR float ASPECT_RATIO_TOLERANCE = 0.001;
+ static const CONSTEXPR float ASPECT_RATIO_TOLERANCE = 0.01;
// Threshold for slow jpeg mode
static const int64_t kSlowJpegModeThreshold = 33400000LL; // 33.4 ms
// Margin for checking FPS
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.cpp b/services/camera/libcameraservice/common/CameraProviderManager.cpp
index 4f2b878..9abb972 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.cpp
+++ b/services/camera/libcameraservice/common/CameraProviderManager.cpp
@@ -359,7 +359,13 @@
for (auto& provider : mProviders) {
ALOGV("%s: Notifying %s for new state 0x%" PRIx64,
__FUNCTION__, provider->mProviderName.c_str(), newState);
+ // b/199240726 Camera providers can for example try to add/remove
+ // camera devices as part of the state change notification. Holding
+ // 'mInterfaceMutex' while calling 'notifyDeviceStateChange' can
+ // result in a recursive deadlock.
+ mInterfaceMutex.unlock();
status_t singleRes = provider->notifyDeviceStateChange(mDeviceState);
+ mInterfaceMutex.lock();
if (singleRes != OK) {
ALOGE("%s: Unable to notify provider %s about device state change",
__FUNCTION__,
@@ -367,6 +373,7 @@
res = singleRes;
// continue to do the rest of the providers instead of returning now
}
+ provider->notifyDeviceInfoStateChangeLocked(mDeviceState);
}
return res;
}
@@ -1185,10 +1192,12 @@
}
bool CameraProviderManager::isHiddenPhysicalCamera(const std::string& cameraId) const {
+ std::lock_guard<std::mutex> lock(mInterfaceMutex);
return isHiddenPhysicalCameraInternal(cameraId).first;
}
status_t CameraProviderManager::filterSmallJpegSizes(const std::string& cameraId) {
+ std::lock_guard<std::mutex> lock(mInterfaceMutex);
for (auto& provider : mProviders) {
for (auto& deviceInfo : provider->mDevices) {
if (deviceInfo->mId == cameraId) {
@@ -2031,6 +2040,14 @@
return OK;
}
+void CameraProviderManager::ProviderInfo::notifyDeviceInfoStateChangeLocked(
+ hardware::hidl_bitfield<provider::V2_5::DeviceState> newDeviceState) {
+ std::lock_guard<std::mutex> lock(mLock);
+ for (auto it = mDevices.begin(); it != mDevices.end(); it++) {
+ (*it)->notifyDeviceStateChange(newDeviceState);
+ }
+}
+
status_t CameraProviderManager::ProviderInfo::notifyDeviceStateChange(
hardware::hidl_bitfield<provider::V2_5::DeviceState> newDeviceState) {
mDeviceState = newDeviceState;
@@ -2285,6 +2302,18 @@
return;
}
+ if (mCameraCharacteristics.exists(ANDROID_INFO_DEVICE_STATE_ORIENTATIONS)) {
+ const auto &stateMap = mCameraCharacteristics.find(ANDROID_INFO_DEVICE_STATE_ORIENTATIONS);
+ if ((stateMap.count > 0) && ((stateMap.count % 2) == 0)) {
+ for (size_t i = 0; i < stateMap.count; i += 2) {
+ mDeviceStateOrientationMap.emplace(stateMap.data.i64[i], stateMap.data.i64[i+1]);
+ }
+ } else {
+ ALOGW("%s: Invalid ANDROID_INFO_DEVICE_STATE_ORIENTATIONS map size: %zu", __FUNCTION__,
+ stateMap.count);
+ }
+ }
+
mSystemCameraKind = getSystemCameraKind();
status_t res = fixupMonochromeTags();
@@ -2413,6 +2442,16 @@
CameraProviderManager::ProviderInfo::DeviceInfo3::~DeviceInfo3() {}
+void CameraProviderManager::ProviderInfo::DeviceInfo3::notifyDeviceStateChange(
+ hardware::hidl_bitfield<hardware::camera::provider::V2_5::DeviceState> newState) {
+
+ if (!mDeviceStateOrientationMap.empty() &&
+ (mDeviceStateOrientationMap.find(newState) != mDeviceStateOrientationMap.end())) {
+ mCameraCharacteristics.update(ANDROID_SENSOR_ORIENTATION,
+ &mDeviceStateOrientationMap[newState], 1);
+ }
+}
+
status_t CameraProviderManager::ProviderInfo::DeviceInfo3::setTorchMode(bool enabled) {
return setTorchModeForDevice<InterfaceT>(enabled);
}
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.h b/services/camera/libcameraservice/common/CameraProviderManager.h
index 1bdbb44..e3763a1 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.h
+++ b/services/camera/libcameraservice/common/CameraProviderManager.h
@@ -40,7 +40,6 @@
#include <camera/VendorTagDescriptor.h>
namespace android {
-
/**
* The vendor tag descriptor class that takes HIDL vendor tag information as
* input. Not part of VendorTagDescriptor class because that class is used
@@ -284,12 +283,6 @@
sp<hardware::camera::device::V3_2::ICameraDeviceSession> *session);
/**
- * Save the ICameraProvider while it is being used by a camera or torch client
- */
- void saveRef(DeviceMode usageType, const std::string &cameraId,
- sp<hardware::camera::provider::V2_4::ICameraProvider> provider);
-
- /**
* Notify that the camera or torch is no longer being used by a camera client
*/
void removeRef(DeviceMode usageType, const std::string &cameraId);
@@ -434,6 +427,10 @@
/**
* Notify provider about top-level device physical state changes
+ *
+ * Note that 'mInterfaceMutex' should not be held when calling this method.
+ * It is possible for camera providers to add/remove devices and try to
+ * acquire it.
*/
status_t notifyDeviceStateChange(
hardware::hidl_bitfield<hardware::camera::provider::V2_5::DeviceState>
@@ -442,6 +439,15 @@
std::vector<std::unordered_set<std::string>> getConcurrentCameraIdCombinations();
/**
+ * Notify 'DeviceInfo' instanced about top-level device physical state changes
+ *
+ * Note that 'mInterfaceMutex' should be held when calling this method.
+ */
+ void notifyDeviceInfoStateChangeLocked(
+ hardware::hidl_bitfield<hardware::camera::provider::V2_5::DeviceState>
+ newDeviceState);
+
+ /**
* Query the camera provider for concurrent stream configuration support
*/
status_t isConcurrentSessionConfigurationSupported(
@@ -493,6 +499,9 @@
return INVALID_OPERATION;
}
virtual status_t filterSmallJpegSizes() = 0;
+ virtual void notifyDeviceStateChange(
+ hardware::hidl_bitfield<hardware::camera::provider::V2_5::DeviceState>
+ /*newState*/) {}
template<class InterfaceT>
sp<InterfaceT> startDeviceInterface();
@@ -553,6 +562,9 @@
bool *status /*out*/)
override;
virtual status_t filterSmallJpegSizes() override;
+ virtual void notifyDeviceStateChange(
+ hardware::hidl_bitfield<hardware::camera::provider::V2_5::DeviceState>
+ newState) override;
DeviceInfo3(const std::string& name, const metadata_vendor_id_t tagId,
const std::string &id, uint16_t minorVersion,
@@ -562,6 +574,8 @@
virtual ~DeviceInfo3();
private:
CameraMetadata mCameraCharacteristics;
+ // Map device states to sensor orientations
+ std::unordered_map<int64_t, int32_t> mDeviceStateOrientationMap;
// A copy of mCameraCharacteristics without performance class
// override
std::unique_ptr<CameraMetadata> mCameraCharNoPCOverride;
@@ -662,6 +676,12 @@
sp<hardware::camera::provider::V2_6::ICameraProvider> &interface2_6);
};
+ /**
+ * Save the ICameraProvider while it is being used by a camera or torch client
+ */
+ void saveRef(DeviceMode usageType, const std::string &cameraId,
+ sp<hardware::camera::provider::V2_4::ICameraProvider> provider);
+
// Utility to find a DeviceInfo by ID; pointer is only valid while mInterfaceMutex is held
// and the calling code doesn't mutate the list of providers or their lists of devices.
// Finds the first device of the given ID that falls within the requested version range
diff --git a/services/camera/libcameraservice/device3/BufferUtils.h b/services/camera/libcameraservice/device3/BufferUtils.h
index 1e1cd60..03112ec 100644
--- a/services/camera/libcameraservice/device3/BufferUtils.h
+++ b/services/camera/libcameraservice/device3/BufferUtils.h
@@ -104,7 +104,7 @@
// Return the removed buffer ID if input cache is found.
// Otherwise return BUFFER_ID_NO_BUFFER
- uint64_t removeOneBufferCache(int streamId, const native_handle_t* handle);
+ uint64_t removeOneBufferCache(int streamId, const native_handle_t* handle) override;
// Clear all caches for input stream, but do not remove the stream
// Removed buffers' ID are returned
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 7c3c6b7..67f99bf 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -2677,6 +2677,7 @@
}
mGroupIdPhysicalCameraMap.clear();
+ bool composerSurfacePresent = false;
for (size_t i = 0; i < mOutputStreams.size(); i++) {
// Don't configure bidi streams twice, nor add them twice to the list
@@ -2716,6 +2717,10 @@
const String8& physicalCameraId = mOutputStreams[i]->getPhysicalCameraId();
mGroupIdPhysicalCameraMap[streamGroupId].insert(physicalCameraId);
}
+
+ if (outputStream->usage & GraphicBuffer::USAGE_HW_COMPOSER) {
+ composerSurfacePresent = true;
+ }
}
config.streams = streams.editArray();
@@ -2783,6 +2788,8 @@
}
}
+ mRequestThread->setComposerSurface(composerSurfacePresent);
+
// Request thread needs to know to avoid using repeat-last-settings protocol
// across configure_streams() calls
if (notifyRequestThread) {
@@ -3084,10 +3091,11 @@
void Camera3Device::monitorMetadata(TagMonitor::eventSource source,
int64_t frameNumber, nsecs_t timestamp, const CameraMetadata& metadata,
const std::unordered_map<std::string, CameraMetadata>& physicalMetadata,
- const std::set<int32_t> &outputStreamIds, int32_t inputStreamId) {
+ const camera_stream_buffer_t *outputBuffers, uint32_t numOutputBuffers,
+ int32_t inputStreamId) {
mTagMonitor.monitorMetadata(source, frameNumber, timestamp, metadata,
- physicalMetadata, outputStreamIds, inputStreamId);
+ physicalMetadata, outputBuffers, numOutputBuffers, inputStreamId);
}
/**
@@ -4144,6 +4152,11 @@
return mBufferRecords.getBufferId(buf, streamId);
}
+uint64_t Camera3Device::HalInterface::removeOneBufferCache(int streamId,
+ const native_handle_t* handle) {
+ return mBufferRecords.removeOneBufferCache(streamId, handle);
+}
+
void Camera3Device::HalInterface::onBufferFreed(
int streamId, const native_handle_t* handle) {
uint32_t bufferId = mBufferRecords.removeOneBufferCache(streamId, handle);
@@ -4186,6 +4199,7 @@
mCurrentAfTriggerId(0),
mCurrentPreCaptureTriggerId(0),
mRotateAndCropOverride(ANDROID_SCALER_ROTATE_AND_CROP_NONE),
+ mComposerOutput(false),
mCameraMute(ANDROID_SENSOR_TEST_PATTERN_MODE_OFF),
mCameraMuteChanged(false),
mRepeatingLastFrameNumber(
@@ -4590,12 +4604,6 @@
sp<Camera3Device> parent = mParent.promote();
if (parent != NULL) {
- std::set<int32_t> outputStreamIds;
- for (size_t i = 0; i < halRequest.num_output_buffers; i++) {
- const camera_stream_buffer_t *src = halRequest.output_buffers + i;
- int32_t streamId = Camera3Stream::cast(src->stream)->getId();
- outputStreamIds.emplace(streamId);
- }
int32_t inputStreamId = -1;
if (halRequest.input_buffer != nullptr) {
inputStreamId = Camera3Stream::cast(halRequest.input_buffer->stream)->getId();
@@ -4603,7 +4611,8 @@
parent->monitorMetadata(TagMonitor::REQUEST,
halRequest.frame_number,
- 0, mLatestRequest, mLatestPhysicalRequest, outputStreamIds, inputStreamId);
+ 0, mLatestRequest, mLatestPhysicalRequest, halRequest.output_buffers,
+ halRequest.num_output_buffers, inputStreamId);
}
}
@@ -4801,6 +4810,26 @@
return submitRequestSuccess;
}
+status_t Camera3Device::removeFwkOnlyRegionKeys(CameraMetadata *request) {
+ static const std::array<uint32_t, 4> kFwkOnlyRegionKeys = {ANDROID_CONTROL_AF_REGIONS_SET,
+ ANDROID_CONTROL_AE_REGIONS_SET, ANDROID_CONTROL_AWB_REGIONS_SET,
+ ANDROID_SCALER_CROP_REGION_SET};
+ if (request == nullptr) {
+ ALOGE("%s request metadata nullptr", __FUNCTION__);
+ return BAD_VALUE;
+ }
+ status_t res = OK;
+ for (const auto &key : kFwkOnlyRegionKeys) {
+ if (request->exists(key)) {
+ res = request->erase(key);
+ if (res != OK) {
+ return res;
+ }
+ }
+ }
+ return OK;
+}
+
status_t Camera3Device::RequestThread::prepareHalRequests() {
ATRACE_CALL();
@@ -4827,7 +4856,11 @@
bool triggersMixedIn = (triggerCount > 0 || mPrevTriggers > 0);
mPrevTriggers = triggerCount;
- bool rotateAndCropChanged = overrideAutoRotateAndCrop(captureRequest);
+ // Do not override rotate&crop for stream configurations that include
+ // SurfaceViews(HW_COMPOSER) output. The display rotation there will be
+ // compensated by NATIVE_WINDOW_TRANSFORM_INVERSE_DISPLAY
+ bool rotateAndCropChanged = mComposerOutput ? false :
+ overrideAutoRotateAndCrop(captureRequest);
bool testPatternChanged = overrideTestPattern(captureRequest);
// If the request is the same as last, or we had triggers now or last time or
@@ -4860,6 +4893,12 @@
it != captureRequest->mSettingsList.end(); it++) {
if (parent->mUHRCropAndMeteringRegionMappers.find(it->cameraId) ==
parent->mUHRCropAndMeteringRegionMappers.end()) {
+ if (removeFwkOnlyRegionKeys(&(it->metadata)) != OK) {
+ SET_ERR("RequestThread: Unable to remove fwk-only keys from request"
+ "%d: %s (%d)", halRequest->frame_number, strerror(-res),
+ res);
+ return INVALID_OPERATION;
+ }
continue;
}
@@ -4874,6 +4913,12 @@
return INVALID_OPERATION;
}
captureRequest->mUHRCropAndMeteringRegionsUpdated = true;
+ if (removeFwkOnlyRegionKeys(&(it->metadata)) != OK) {
+ SET_ERR("RequestThread: Unable to remove fwk-only keys from request"
+ "%d: %s (%d)", halRequest->frame_number, strerror(-res),
+ res);
+ return INVALID_OPERATION;
+ }
}
}
@@ -5321,6 +5366,13 @@
return OK;
}
+status_t Camera3Device::RequestThread::setComposerSurface(bool composerSurfacePresent) {
+ ATRACE_CALL();
+ Mutex::Autolock l(mTriggerMutex);
+ mComposerOutput = composerSurfacePresent;
+ return OK;
+}
+
status_t Camera3Device::RequestThread::setCameraMute(int32_t muteMode) {
ATRACE_CALL();
Mutex::Autolock l(mTriggerMutex);
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index c0dc6f8..4ffdc5f 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -297,6 +297,7 @@
private:
status_t disconnectImpl();
+ static status_t removeFwkOnlyRegionKeys(CameraMetadata *request);
// internal typedefs
using RequestMetadataQueue = hardware::MessageQueue<uint8_t, hardware::kSynchronizedReadWrite>;
@@ -412,6 +413,8 @@
std::pair<bool, uint64_t> getBufferId(
const buffer_handle_t& buf, int streamId) override;
+ uint64_t removeOneBufferCache(int streamId, const native_handle_t* handle) override;
+
status_t popInflightBuffer(int32_t frameNumber, int32_t streamId,
/*out*/ buffer_handle_t **buffer) override;
@@ -916,6 +919,7 @@
status_t setRotateAndCropAutoBehavior(
camera_metadata_enum_android_scaler_rotate_and_crop_t rotateAndCropValue);
+ status_t setComposerSurface(bool composerSurfacePresent);
status_t setCameraMute(int32_t muteMode);
@@ -1068,6 +1072,7 @@
uint32_t mCurrentAfTriggerId;
uint32_t mCurrentPreCaptureTriggerId;
camera_metadata_enum_android_scaler_rotate_and_crop_t mRotateAndCropOverride;
+ bool mComposerOutput;
int32_t mCameraMute; // 0 = no mute, otherwise the TEST_PATTERN_MODE to use
bool mCameraMuteChanged;
@@ -1250,7 +1255,8 @@
void monitorMetadata(TagMonitor::eventSource source, int64_t frameNumber,
nsecs_t timestamp, const CameraMetadata& metadata,
const std::unordered_map<std::string, CameraMetadata>& physicalMetadata,
- const std::set<int32_t> &outputStreamIds, int32_t inputStreamId);
+ const camera_stream_buffer_t *outputBuffers, uint32_t numOutputBuffers,
+ int32_t inputStreamId);
metadata_vendor_id_t mVendorTagId;
diff --git a/services/camera/libcameraservice/device3/Camera3OutputInterface.h b/services/camera/libcameraservice/device3/Camera3OutputInterface.h
index 8817833..40eef1d 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputInterface.h
+++ b/services/camera/libcameraservice/device3/Camera3OutputInterface.h
@@ -50,6 +50,10 @@
// return pair of (newlySeenBuffer?, bufferId)
virtual std::pair<bool, uint64_t> getBufferId(const buffer_handle_t& buf, int streamId) = 0;
+ // Return the removed buffer ID if input cache is found.
+ // Otherwise return BUFFER_ID_NO_BUFFER
+ virtual uint64_t removeOneBufferCache(int streamId, const native_handle_t* handle) = 0;
+
// Find a buffer_handle_t based on frame number and stream ID
virtual status_t popInflightBuffer(int32_t frameNumber, int32_t streamId,
/*out*/ buffer_handle_t **buffer) = 0;
diff --git a/services/camera/libcameraservice/device3/Camera3OutputUtils.cpp b/services/camera/libcameraservice/device3/Camera3OutputUtils.cpp
index ef51ada..5a97f4b 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputUtils.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputUtils.cpp
@@ -368,7 +368,7 @@
}
states.tagMonitor.monitorMetadata(TagMonitor::RESULT,
frameNumber, sensorTimestamp, captureResult.mMetadata,
- monitoredPhysicalMetadata, std::set<int32_t>());
+ monitoredPhysicalMetadata);
insertResultLocked(states, &captureResult, frameNumber);
}
@@ -1305,6 +1305,7 @@
hardware::hidl_vec<StreamBuffer> tmpRetBuffers(numBuffersRequested);
bool currentReqSucceeds = true;
std::vector<camera_stream_buffer_t> streamBuffers(numBuffersRequested);
+ std::vector<buffer_handle_t> newBuffers;
size_t numAllocatedBuffers = 0;
size_t numPushedInflightBuffers = 0;
for (size_t b = 0; b < numBuffersRequested; b++) {
@@ -1344,6 +1345,9 @@
hBuf.buffer = (isNewBuffer) ? *buffer : nullptr;
hBuf.status = BufferStatus::OK;
hBuf.releaseFence = nullptr;
+ if (isNewBuffer) {
+ newBuffers.push_back(*buffer);
+ }
native_handle_t *acquireFence = nullptr;
if (sb.acquire_fence != -1) {
@@ -1386,6 +1390,9 @@
returnOutputBuffers(states.useHalBufManager, /*listener*/nullptr,
streamBuffers.data(), numAllocatedBuffers, 0, /*requested*/false,
/*requestTimeNs*/0, states.sessionStatsBuilder);
+ for (auto buf : newBuffers) {
+ states.bufferRecordsIntf.removeOneBufferCache(streamId, buf);
+ }
}
}
diff --git a/services/camera/libcameraservice/device3/Camera3OutputUtils.h b/services/camera/libcameraservice/device3/Camera3OutputUtils.h
index 142889a..3f54ac5 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputUtils.h
+++ b/services/camera/libcameraservice/device3/Camera3OutputUtils.h
@@ -42,66 +42,6 @@
namespace camera3 {
- typedef struct camera_stream_configuration {
- uint32_t num_streams;
- camera_stream_t **streams;
- uint32_t operation_mode;
- bool input_is_multi_resolution;
- } camera_stream_configuration_t;
-
- typedef struct camera_capture_request {
- uint32_t frame_number;
- const camera_metadata_t *settings;
- camera_stream_buffer_t *input_buffer;
- uint32_t num_output_buffers;
- const camera_stream_buffer_t *output_buffers;
- uint32_t num_physcam_settings;
- const char **physcam_id;
- const camera_metadata_t **physcam_settings;
- int32_t input_width;
- int32_t input_height;
- } camera_capture_request_t;
-
- typedef struct camera_capture_result {
- uint32_t frame_number;
- const camera_metadata_t *result;
- uint32_t num_output_buffers;
- const camera_stream_buffer_t *output_buffers;
- const camera_stream_buffer_t *input_buffer;
- uint32_t partial_result;
- uint32_t num_physcam_metadata;
- const char **physcam_ids;
- const camera_metadata_t **physcam_metadata;
- } camera_capture_result_t;
-
- typedef struct camera_shutter_msg {
- uint32_t frame_number;
- uint64_t timestamp;
- } camera_shutter_msg_t;
-
- typedef struct camera_error_msg {
- uint32_t frame_number;
- camera_stream_t *error_stream;
- int error_code;
- } camera_error_msg_t;
-
- typedef enum camera_error_msg_code {
- CAMERA_MSG_ERROR_DEVICE = 1,
- CAMERA_MSG_ERROR_REQUEST = 2,
- CAMERA_MSG_ERROR_RESULT = 3,
- CAMERA_MSG_ERROR_BUFFER = 4,
- CAMERA_MSG_NUM_ERRORS
- } camera_error_msg_code_t;
-
- typedef struct camera_notify_msg {
- int type;
-
- union {
- camera_error_msg_t error;
- camera_shutter_msg_t shutter;
- } message;
- } camera_notify_msg_t;
-
/**
* Helper methods shared between Camera3Device/Camera3OfflineSession for HAL callbacks
*/
diff --git a/services/camera/libcameraservice/device3/InFlightRequest.h b/services/camera/libcameraservice/device3/InFlightRequest.h
index 523a2c7..85e809c 100644
--- a/services/camera/libcameraservice/device3/InFlightRequest.h
+++ b/services/camera/libcameraservice/device3/InFlightRequest.h
@@ -30,6 +30,66 @@
namespace camera3 {
+typedef struct camera_stream_configuration {
+ uint32_t num_streams;
+ camera_stream_t **streams;
+ uint32_t operation_mode;
+ bool input_is_multi_resolution;
+} camera_stream_configuration_t;
+
+typedef struct camera_capture_request {
+ uint32_t frame_number;
+ const camera_metadata_t *settings;
+ camera_stream_buffer_t *input_buffer;
+ uint32_t num_output_buffers;
+ const camera_stream_buffer_t *output_buffers;
+ uint32_t num_physcam_settings;
+ const char **physcam_id;
+ const camera_metadata_t **physcam_settings;
+ int32_t input_width;
+ int32_t input_height;
+} camera_capture_request_t;
+
+typedef struct camera_capture_result {
+ uint32_t frame_number;
+ const camera_metadata_t *result;
+ uint32_t num_output_buffers;
+ const camera_stream_buffer_t *output_buffers;
+ const camera_stream_buffer_t *input_buffer;
+ uint32_t partial_result;
+ uint32_t num_physcam_metadata;
+ const char **physcam_ids;
+ const camera_metadata_t **physcam_metadata;
+} camera_capture_result_t;
+
+typedef struct camera_shutter_msg {
+ uint32_t frame_number;
+ uint64_t timestamp;
+} camera_shutter_msg_t;
+
+typedef struct camera_error_msg {
+ uint32_t frame_number;
+ camera_stream_t *error_stream;
+ int error_code;
+} camera_error_msg_t;
+
+typedef enum camera_error_msg_code {
+ CAMERA_MSG_ERROR_DEVICE = 1,
+ CAMERA_MSG_ERROR_REQUEST = 2,
+ CAMERA_MSG_ERROR_RESULT = 3,
+ CAMERA_MSG_ERROR_BUFFER = 4,
+ CAMERA_MSG_NUM_ERRORS
+} camera_error_msg_code_t;
+
+typedef struct camera_notify_msg {
+ int type;
+
+ union {
+ camera_error_msg_t error;
+ camera_shutter_msg_t shutter;
+ } message;
+} camera_notify_msg_t;
+
typedef enum {
// Cache the buffers with STATUS_ERROR within InFlightRequest
ERROR_BUF_CACHE,
diff --git a/services/camera/libcameraservice/hidl/HidlCameraService.cpp b/services/camera/libcameraservice/hidl/HidlCameraService.cpp
index 7d1b3cf..a812587 100644
--- a/services/camera/libcameraservice/hidl/HidlCameraService.cpp
+++ b/services/camera/libcameraservice/hidl/HidlCameraService.cpp
@@ -279,6 +279,9 @@
size_t numSections = sectionNames->size();
std::vector<std::vector<HVendorTag>> tagsBySection(numSections);
int tagCount = desc->getTagCount();
+ if (tagCount <= 0) {
+ continue;
+ }
std::vector<uint32_t> tags(tagCount);
desc->getTagArray(tags.data());
for (int i = 0; i < tagCount; i++) {
diff --git a/services/camera/libcameraservice/tests/ExifUtilsTest.cpp b/services/camera/libcameraservice/tests/ExifUtilsTest.cpp
new file mode 100644
index 0000000..3de4bf2
--- /dev/null
+++ b/services/camera/libcameraservice/tests/ExifUtilsTest.cpp
@@ -0,0 +1,55 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ExifUtilsTest"
+
+#include <camera/CameraMetadata.h>
+#include "../utils/ExifUtils.h"
+#include <gtest/gtest.h>
+
+using android::camera3::ExifUtils;
+using android::camera3::ExifOrientation;
+using android::CameraMetadata;
+
+uint32_t kImageWidth = 1920;
+uint32_t kImageHeight = 1440;
+ExifOrientation kExifOrientation = ExifOrientation::ORIENTATION_0_DEGREES;
+
+// Test that setFromMetadata works correctly, without errors.
+TEST(ExifUtilsTest, SetFromMetadataTest) {
+ std::unique_ptr<ExifUtils> utils(ExifUtils::create());
+ uint8_t invalidSensorPixelMode = 2;
+ uint8_t validSensorPixelMode = ANDROID_SENSOR_PIXEL_MODE_DEFAULT;
+ CameraMetadata metadata;
+ // Empty staticInfo
+ CameraMetadata staticInfo;
+ ASSERT_TRUE(utils->initializeEmpty());
+ ASSERT_TRUE(
+ metadata.update(ANDROID_SENSOR_PIXEL_MODE, &invalidSensorPixelMode, 1) == android::OK);
+ ASSERT_FALSE(utils->setFromMetadata(metadata, staticInfo, kImageWidth, kImageHeight));
+ ASSERT_TRUE(
+ metadata.update(ANDROID_SENSOR_PIXEL_MODE, &validSensorPixelMode, 1) == android::OK);
+ ASSERT_TRUE(utils->setFromMetadata(metadata, staticInfo, kImageWidth, kImageHeight));
+ ASSERT_TRUE(utils->setImageWidth(kImageWidth));
+ ASSERT_TRUE(utils->setImageHeight(kImageHeight));
+ ASSERT_TRUE(utils->setOrientationValue(kExifOrientation));
+ ASSERT_TRUE(utils->generateApp1());
+ const uint8_t* exifBuffer = utils->getApp1Buffer();
+ ASSERT_NE(exifBuffer, nullptr);
+ size_t exifBufferSize = utils->getApp1Length();
+ ASSERT_TRUE(exifBufferSize != 0);
+}
diff --git a/services/camera/libcameraservice/utils/ExifUtils.cpp b/services/camera/libcameraservice/utils/ExifUtils.cpp
index 485705c..21f02db 100644
--- a/services/camera/libcameraservice/utils/ExifUtils.cpp
+++ b/services/camera/libcameraservice/utils/ExifUtils.cpp
@@ -920,7 +920,7 @@
camera_metadata_ro_entry sensorPixelModeEntry = metadata.find(ANDROID_SENSOR_PIXEL_MODE);
if (sensorPixelModeEntry.count != 0) {
sensorPixelMode = sensorPixelModeEntry.data.u8[0];
- if (sensorPixelMode != ANDROID_SENSOR_PIXEL_MODE_DEFAULT ||
+ if (sensorPixelMode != ANDROID_SENSOR_PIXEL_MODE_DEFAULT &&
sensorPixelMode != ANDROID_SENSOR_PIXEL_MODE_MAXIMUM_RESOLUTION) {
ALOGE("%s: Request sensor pixel mode is not one of the valid values %d",
__FUNCTION__, sensorPixelMode);
diff --git a/services/camera/libcameraservice/utils/TagMonitor.cpp b/services/camera/libcameraservice/utils/TagMonitor.cpp
index 53a92e9..4488098 100644
--- a/services/camera/libcameraservice/utils/TagMonitor.cpp
+++ b/services/camera/libcameraservice/utils/TagMonitor.cpp
@@ -24,6 +24,7 @@
#include <utils/Log.h>
#include <camera/VendorTagDescriptor.h>
#include <camera_metadata_hidden.h>
+#include <device3/Camera3Stream.h>
namespace android {
@@ -119,7 +120,8 @@
void TagMonitor::monitorMetadata(eventSource source, int64_t frameNumber, nsecs_t timestamp,
const CameraMetadata& metadata,
const std::unordered_map<std::string, CameraMetadata>& physicalMetadata,
- const std::set<int32_t> &outputStreamIds, int32_t inputStreamId) {
+ const camera3::camera_stream_buffer_t *outputBuffers, uint32_t numOutputBuffers,
+ int32_t inputStreamId) {
if (!mMonitoringEnabled) return;
std::lock_guard<std::mutex> lock(mMonitorMutex);
@@ -127,7 +129,12 @@
if (timestamp == 0) {
timestamp = systemTime(SYSTEM_TIME_BOOTTIME);
}
-
+ std::unordered_set<int32_t> outputStreamIds;
+ for (size_t i = 0; i < numOutputBuffers; i++) {
+ const camera3::camera_stream_buffer_t *src = outputBuffers + i;
+ int32_t streamId = camera3::Camera3Stream::cast(src->stream)->getId();
+ outputStreamIds.emplace(streamId);
+ }
std::string emptyId;
for (auto tag : mMonitoredTagList) {
monitorSingleMetadata(source, frameNumber, timestamp, emptyId, tag, metadata,
@@ -142,7 +149,7 @@
void TagMonitor::monitorSingleMetadata(eventSource source, int64_t frameNumber, nsecs_t timestamp,
const std::string& cameraId, uint32_t tag, const CameraMetadata& metadata,
- const std::set<int32_t> &outputStreamIds, int32_t inputStreamId) {
+ const std::unordered_set<int32_t> &outputStreamIds, int32_t inputStreamId) {
CameraMetadata &lastValues = (source == REQUEST) ?
(cameraId.empty() ? mLastMonitoredRequestValues :
@@ -186,7 +193,8 @@
// Also monitor when the stream ids change, this helps visually see what
// monitored metadata values are for capture requests with different
// stream ids.
- if (inputStreamId != mLastInputStreamId || outputStreamIds != mLastStreamIds) {
+ if (source == REQUEST &&
+ (inputStreamId != mLastInputStreamId || outputStreamIds != mLastStreamIds)) {
mLastInputStreamId = inputStreamId;
mLastStreamIds = outputStreamIds;
isDifferent = true;
@@ -261,7 +269,7 @@
#define CAMERA_METADATA_ENUM_STRING_MAX_SIZE 29
void TagMonitor::printData(int fd, const uint8_t *data_ptr, uint32_t tag,
- int type, int count, int indentation, const std::set<int32_t> &outputStreamIds,
+ int type, int count, int indentation, const std::unordered_set<int32_t> &outputStreamIds,
int32_t inputStreamId) {
static int values_per_line[NUM_TYPES] = {
[TYPE_BYTE] = 16,
@@ -353,7 +361,8 @@
template<typename T>
TagMonitor::MonitorEvent::MonitorEvent(eventSource src, uint32_t frameNumber, nsecs_t timestamp,
- const T &value, const std::string& cameraId, const std::set<int32_t> &outputStreamIds,
+ const T &value, const std::string& cameraId,
+ const std::unordered_set<int32_t> &outputStreamIds,
int32_t inputStreamId) :
source(src),
frameNumber(frameNumber),
diff --git a/services/camera/libcameraservice/utils/TagMonitor.h b/services/camera/libcameraservice/utils/TagMonitor.h
index dfc424e..f6df4b7 100644
--- a/services/camera/libcameraservice/utils/TagMonitor.h
+++ b/services/camera/libcameraservice/utils/TagMonitor.h
@@ -20,7 +20,6 @@
#include <vector>
#include <atomic>
#include <mutex>
-#include <set>
#include <unordered_map>
#include <utils/RefBase.h>
@@ -31,6 +30,7 @@
#include <system/camera_metadata.h>
#include <system/camera_vendor_tags.h>
#include <camera/CameraMetadata.h>
+#include <device3/InFlightRequest.h>
namespace android {
@@ -68,7 +68,8 @@
void monitorMetadata(eventSource source, int64_t frameNumber,
nsecs_t timestamp, const CameraMetadata& metadata,
const std::unordered_map<std::string, CameraMetadata>& physicalMetadata,
- const std::set<int32_t> &outputStreamIds, int32_t inputStreamId = -1);
+ const camera3::camera_stream_buffer_t *outputBuffers = nullptr,
+ uint32_t numOutputBuffers = 0, int32_t inputStreamId = -1);
// Dump current event log to the provided fd
void dumpMonitoredMetadata(int fd);
@@ -76,12 +77,12 @@
private:
static void printData(int fd, const uint8_t *data_ptr, uint32_t tag,
- int type, int count, int indentation, const std::set<int32_t> &outputStreamIds,
- int32_t inputStreamId);
+ int type, int count, int indentation,
+ const std::unordered_set<int32_t> &outputStreamIds, int32_t inputStreamId);
void monitorSingleMetadata(TagMonitor::eventSource source, int64_t frameNumber,
nsecs_t timestamp, const std::string& cameraId, uint32_t tag,
- const CameraMetadata& metadata, const std::set<int32_t> &outputStreamIds,
+ const CameraMetadata& metadata, const std::unordered_set<int32_t> &outputStreamIds,
int32_t inputStreamId);
std::atomic<bool> mMonitoringEnabled;
@@ -98,7 +99,7 @@
std::unordered_map<std::string, CameraMetadata> mLastMonitoredPhysicalResultKeys;
int32_t mLastInputStreamId = -1;
- std::set<int32_t> mLastStreamIds;
+ std::unordered_set<int32_t> mLastStreamIds;
/**
* A monitoring event
@@ -109,7 +110,7 @@
template<typename T>
MonitorEvent(eventSource src, uint32_t frameNumber, nsecs_t timestamp,
const T &newValue, const std::string& cameraId,
- const std::set<int32_t> &outputStreamIds, int32_t inputStreamId);
+ const std::unordered_set<int32_t> &outputStreamIds, int32_t inputStreamId);
~MonitorEvent();
eventSource source;
@@ -119,7 +120,7 @@
uint8_t type;
std::vector<uint8_t> newData;
std::string cameraId;
- std::set<int32_t> outputStreamIds;
+ std::unordered_set<int32_t> outputStreamIds;
int32_t inputStreamId = 1;
};
diff --git a/services/mediacodec/seccomp_policy/mediaswcodec-arm.policy b/services/mediacodec/seccomp_policy/mediaswcodec-arm.policy
index 9058f10..41efce0 100644
--- a/services/mediacodec/seccomp_policy/mediaswcodec-arm.policy
+++ b/services/mediacodec/seccomp_policy/mediaswcodec-arm.policy
@@ -84,5 +84,6 @@
getgid32: 1
getegid32: 1
getgroups32: 1
+sysinfo: 1
@include /apex/com.android.media.swcodec/etc/seccomp_policy/code_coverage.arm.policy
diff --git a/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy b/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy
index 4c51a9c..4317ccc 100644
--- a/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy
+++ b/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy
@@ -78,5 +78,16 @@
getgid: 1
getegid: 1
getgroups: 1
+sysinfo: 1
+
+# Android profiler (heapprofd, traced_perf) additions, where not already
+# covered by the rest of the file, or by builtin minijail allow-listing of
+# logging-related syscalls.
+# TODO(b/197184220): this is a targeted addition for a specific investigation,
+# and addresses just the arm64 framework av service policies. In the future, we
+# should make this more general (e.g. a central file that can be @included in
+# other policy files).
+setsockopt: 1
+sendmsg: 1
@include /apex/com.android.media.swcodec/etc/seccomp_policy/code_coverage.arm64.policy
diff --git a/services/mediaextractor/seccomp_policy/mediaextractor-arm64.policy b/services/mediaextractor/seccomp_policy/mediaextractor-arm64.policy
index 9bbd53b..e54c918 100644
--- a/services/mediaextractor/seccomp_policy/mediaextractor-arm64.policy
+++ b/services/mediaextractor/seccomp_policy/mediaextractor-arm64.policy
@@ -46,5 +46,16 @@
# Required by Sanitizers
sched_yield: 1
+# Android profiler (heapprofd, traced_perf) additions, where not already
+# covered by the rest of the file, or by builtin minijail allow-listing of
+# logging-related syscalls.
+# TODO(b/197184220): this is a targeted addition for a specific investigation,
+# and addresses just the arm64 framework av service policies. In the future, we
+# should make this more general (e.g. a central file that can be @included in
+# other policy files).
+setsockopt: 1
+sendmsg: 1
+set_tid_address: 1
+
@include /apex/com.android.media/etc/seccomp_policy/crash_dump.arm64.policy
@include /apex/com.android.media/etc/seccomp_policy/code_coverage.arm64.policy
diff --git a/services/mediametrics/Android.bp b/services/mediametrics/Android.bp
index 0351d2d..04fc430 100644
--- a/services/mediametrics/Android.bp
+++ b/services/mediametrics/Android.bp
@@ -172,11 +172,13 @@
"libstatspull",
"libstatssocket",
"libutils",
+ "packagemanager_aidl-cpp",
],
export_shared_lib_headers: [
"libstatspull",
"libstatssocket",
+ "packagemanager_aidl-cpp",
],
static_libs: [
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index 340076e..390cd5c 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -193,7 +193,9 @@
if (direction == AAUDIO_DIRECTION_OUTPUT) {
flags = static_cast<audio_flags_mask_t>(AUDIO_FLAG_LOW_LATENCY
| AAudioConvert_allowCapturePolicyToAudioFlagsMask(
- params->getAllowedCapturePolicy()));
+ params->getAllowedCapturePolicy(),
+ params->getSpatializationBehavior(),
+ params->isContentSpatialized()));
} else {
flags = static_cast<audio_flags_mask_t>(AUDIO_FLAG_LOW_LATENCY
| AAudioConvert_privacySensitiveToAudioFlagsMask(params->isPrivacySensitive()));
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.cpp b/services/oboeservice/AAudioServiceEndpointPlay.cpp
index 4e46033..f590fc8 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.cpp
+++ b/services/oboeservice/AAudioServiceEndpointPlay.cpp
@@ -21,6 +21,7 @@
#include <assert.h>
#include <map>
#include <mutex>
+#include <media/AudioSystem.h>
#include <utils/Singleton.h>
#include "AAudioEndpointManager.h"
@@ -51,7 +52,7 @@
mMixer.allocate(getStreamInternal()->getSamplesPerFrame(),
getStreamInternal()->getFramesPerBurst());
- int32_t burstsPerBuffer = AAudioProperty_getMixerBursts();
+ int32_t burstsPerBuffer = AudioSystem::getAAudioMixerBurstCount();
if (burstsPerBuffer == 0) {
mLatencyTuningEnabled = true;
burstsPerBuffer = BURSTS_PER_BUFFER_DEFAULT;
diff --git a/services/tuner/Android.bp b/services/tuner/Android.bp
index b2b1c3b..ec62d4e 100644
--- a/services/tuner/Android.bp
+++ b/services/tuner/Android.bp
@@ -52,6 +52,7 @@
"liblog",
"libmedia",
"libutils",
+ "packagemanager_aidl-cpp",
"tv_tuner_aidl_interface-ndk",
"tv_tuner_resource_manager_aidl_interface-ndk",
],
diff --git a/services/tuner/hidl/TunerHidlFilter.cpp b/services/tuner/hidl/TunerHidlFilter.cpp
index 2e9e4a6..de82855 100644
--- a/services/tuner/hidl/TunerHidlFilter.cpp
+++ b/services/tuner/hidl/TunerHidlFilter.cpp
@@ -283,8 +283,6 @@
if (res == HidlResult::SUCCESS) {
*out_avMemory = dupToAidl(avMemory);
*_aidl_return = static_cast<int64_t>(avMemSize);
- } else {
- _aidl_return = nullptr;
}
});
