Merge "Move 'session' field to AudioPortExtSys AIDL"
diff --git a/PREUPLOAD.cfg b/PREUPLOAD.cfg
index 716b550..1f7083b 100644
--- a/PREUPLOAD.cfg
+++ b/PREUPLOAD.cfg
@@ -11,3 +11,4 @@
clang_format = --commit ${PREUPLOAD_COMMIT} --style file --extensions c,h,cc,cpp
media/libmediatranscoding/
services/mediatranscoding/
+ media/libaudioclient/tests/
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 15203d6..69d73ad 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -146,39 +146,6 @@
audio_channel_mask_t channelMask,
const AttributionSourceState& client,
size_t frameCount,
- legacy_callback_t callback,
- void* user,
- uint32_t notificationFrames,
- audio_session_t sessionId,
- transfer_type transferType,
- audio_input_flags_t flags,
- const audio_attributes_t* pAttributes,
- audio_port_handle_t selectedDeviceId,
- audio_microphone_direction_t selectedMicDirection,
- float microphoneFieldDimension)
- : mActive(false),
- mStatus(NO_INIT),
- mClientAttributionSource(client),
- mSessionId(AUDIO_SESSION_ALLOCATE),
- mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT),
- mProxy(NULL)
-{
- uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mClientAttributionSource.uid));
- pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
- (void)set(inputSource, sampleRate, format, channelMask, frameCount, callback, user,
- notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
- uid, pid, pAttributes, selectedDeviceId, selectedMicDirection,
- microphoneFieldDimension);
-}
-
-AudioRecord::AudioRecord(
- audio_source_t inputSource,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- const AttributionSourceState& client,
- size_t frameCount,
const wp<IAudioRecordCallback>& callback,
uint32_t notificationFrames,
audio_session_t sessionId,
@@ -255,37 +222,6 @@
mDeviceCallback.clear();
}
}
-namespace {
-class LegacyCallbackWrapper : public AudioRecord::IAudioRecordCallback {
- const AudioRecord::legacy_callback_t mCallback;
- void* const mData;
-
- public:
- LegacyCallbackWrapper(AudioRecord::legacy_callback_t callback, void* user)
- : mCallback(callback), mData(user) {}
-
- size_t onMoreData(const AudioRecord::Buffer& buffer) override {
- AudioRecord::Buffer copy = buffer;
- mCallback(AudioRecord::EVENT_MORE_DATA, mData, ©);
- return copy.size();
- }
-
- void onOverrun() override { mCallback(AudioRecord::EVENT_OVERRUN, mData, nullptr); }
-
- void onMarker(uint32_t markerPosition) override {
- mCallback(AudioRecord::EVENT_MARKER, mData, &markerPosition);
- }
-
- void onNewPos(uint32_t newPos) override {
- mCallback(AudioRecord::EVENT_NEW_POS, mData, &newPos);
- }
-
- void onNewIAudioRecord() override {
- mCallback(AudioRecord::EVENT_NEW_IAUDIORECORD, mData, nullptr);
- }
-};
-} // namespace
-
status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
@@ -479,37 +415,6 @@
return status;
}
-status_t AudioRecord::set(
- audio_source_t inputSource,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- legacy_callback_t callback,
- void* user,
- uint32_t notificationFrames,
- bool threadCanCallJava,
- audio_session_t sessionId,
- transfer_type transferType,
- audio_input_flags_t flags,
- uid_t uid,
- pid_t pid,
- const audio_attributes_t* pAttributes,
- audio_port_handle_t selectedDeviceId,
- audio_microphone_direction_t selectedMicDirection,
- float microphoneFieldDimension,
- int32_t maxSharedAudioHistoryMs)
-{
- if (callback != nullptr) {
- mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
- } else if (user) {
- LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
- }
- return set(inputSource, sampleRate, format, channelMask, frameCount, mLegacyCallbackWrapper,
- notificationFrames, threadCanCallJava, sessionId, transferType, flags, uid, pid,
- pAttributes, selectedDeviceId, selectedMicDirection, microphoneFieldDimension,
- maxSharedAudioHistoryMs);
-}
// -------------------------------------------------------------------------
status_t AudioRecord::start(AudioSystem::sync_event_t event, audio_session_t triggerSession)
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 6deef8f..96fc544 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -325,45 +325,6 @@
}
};
}
-
-AudioTrack::AudioTrack(
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- audio_output_flags_t flags,
- legacy_callback_t callback,
- void* user,
- int32_t notificationFrames,
- audio_session_t sessionId,
- transfer_type transferType,
- const audio_offload_info_t *offloadInfo,
- const AttributionSourceState& attributionSource,
- const audio_attributes_t* pAttributes,
- bool doNotReconnect,
- float maxRequiredSpeed,
- audio_port_handle_t selectedDeviceId)
- : mStatus(NO_INIT),
- mState(STATE_STOPPED),
- mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT),
- mPausedPosition(0),
- mAudioTrackCallback(new AudioTrackCallback())
-{
- mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
- if (callback != nullptr) {
- mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
- } else if (user) {
- LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
- }
- mSetParams = std::unique_ptr<SetParams>{new SetParams{
- streamType, sampleRate, format, channelMask, frameCount, flags, mLegacyCallbackWrapper,
- notificationFrames, 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId,
- transferType, offloadInfo, attributionSource, pAttributes, doNotReconnect,
- maxRequiredSpeed, selectedDeviceId}};
-}
-
AudioTrack::AudioTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
@@ -397,44 +358,6 @@
doNotReconnect, maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
}
-AudioTrack::AudioTrack(
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- const sp<IMemory>& sharedBuffer,
- audio_output_flags_t flags,
- legacy_callback_t callback,
- void* user,
- int32_t notificationFrames,
- audio_session_t sessionId,
- transfer_type transferType,
- const audio_offload_info_t *offloadInfo,
- const AttributionSourceState& attributionSource,
- const audio_attributes_t* pAttributes,
- bool doNotReconnect,
- float maxRequiredSpeed)
- : mStatus(NO_INIT),
- mState(STATE_STOPPED),
- mPreviousPriority(ANDROID_PRIORITY_NORMAL),
- mPreviousSchedulingGroup(SP_DEFAULT),
- mPausedPosition(0),
- mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
- mAudioTrackCallback(new AudioTrackCallback())
-{
- mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
- if (callback) {
- mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
- } else if (user) {
- LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
- }
- mSetParams = std::unique_ptr<SetParams>{new SetParams{
- streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
- mLegacyCallbackWrapper, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
- sessionId, transferType, offloadInfo, attributionSource, pAttributes, doNotReconnect,
- maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
-}
-
void AudioTrack::onFirstRef() {
if (mSetParams) {
set(*mSetParams);
@@ -496,38 +419,6 @@
mDeviceCallback.clear();
}
}
-
-status_t AudioTrack::set(
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- audio_output_flags_t flags,
- legacy_callback_t callback,
- void * user,
- int32_t notificationFrames,
- const sp<IMemory>& sharedBuffer,
- bool threadCanCallJava,
- audio_session_t sessionId,
- transfer_type transferType,
- const audio_offload_info_t *offloadInfo,
- const AttributionSourceState& attributionSource,
- const audio_attributes_t* pAttributes,
- bool doNotReconnect,
- float maxRequiredSpeed,
- audio_port_handle_t selectedDeviceId)
-{
- if (callback) {
- mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
- } else if (user) {
- LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
- }
- return set(streamType, sampleRate,format, channelMask, frameCount, flags,
- mLegacyCallbackWrapper, notificationFrames, sharedBuffer, threadCanCallJava,
- sessionId, transferType, offloadInfo, attributionSource, pAttributes,
- doNotReconnect, maxRequiredSpeed, selectedDeviceId);
-}
status_t AudioTrack::set(
audio_stream_type_t streamType,
uint32_t sampleRate,
diff --git a/media/libaudioclient/TEST_MAPPING b/media/libaudioclient/TEST_MAPPING
index 3751f80..d36cf10 100644
--- a/media/libaudioclient/TEST_MAPPING
+++ b/media/libaudioclient/TEST_MAPPING
@@ -4,6 +4,9 @@
"name": "audio_aidl_conversion_tests"
},
{
+ "name": "audio_aidl_status_tests"
+ },
+ {
"name": "CtsNativeMediaAAudioTestCases",
"options" : [
{
@@ -11,5 +14,22 @@
}
]
}
+ ],
+ "postsubmit": [
+ {
+ "name": "audieorecord_tests"
+ },
+ {
+ "name": "audioeffect_tests"
+ },
+ {
+ "name": "audiorouting_tests"
+ },
+ {
+ "name": "audioclient_serialization_tests"
+ },
+ {
+ "name": "trackplayerbase_tests"
+ }
]
}
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index cb05dd9..5a1ff65 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -46,27 +46,6 @@
{
public:
- /* Events used by AudioRecord callback function (legacy_callback_t).
- * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
- */
- enum event_type {
- EVENT_MORE_DATA = 0, // Request to read available data from buffer.
- // If this event is delivered but the callback handler
- // does not want to read the available data, the handler must
- // explicitly ignore the event by setting frameCount to zero.
- EVENT_OVERRUN = 1, // Buffer overrun occurred.
- EVENT_MARKER = 2, // Record head is at the specified marker position
- // (See setMarkerPosition()).
- EVENT_NEW_POS = 3, // Record head is at a new position
- // (See setPositionUpdatePeriod()).
- EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and
- // voluntary invalidation by mediaserver, or mediaserver crash.
- };
-
- /* Client should declare a Buffer and pass address to obtainBuffer()
- * and releaseBuffer(). See also legacy_callback_t for EVENT_MORE_DATA.
- */
-
class Buffer
{
friend AudioRecord;
@@ -122,7 +101,6 @@
* - EVENT_NEW_IAUDIORECORD: unused.
*/
- typedef void (*legacy_callback_t)(int event, void* user, void *info);
class IAudioRecordCallback : public virtual RefBase {
friend AudioRecord;
@@ -226,24 +204,6 @@
float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT);
- AudioRecord(audio_source_t inputSource,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- const android::content::AttributionSourceState& client,
- size_t frameCount,
- legacy_callback_t callback,
- void* user,
- uint32_t notificationFrames = 0,
- audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
- transfer_type transferType = TRANSFER_DEFAULT,
- audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
- const audio_attributes_t* pAttributes = nullptr,
- audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
- audio_microphone_direction_t
- selectedMicDirection = MIC_DIRECTION_UNSPECIFIED,
- float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT);
-
/* Terminates the AudioRecord and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioRecord.
*/
@@ -286,27 +246,6 @@
float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT,
int32_t maxSharedAudioHistoryMs = 0);
- status_t set(audio_source_t inputSource,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- legacy_callback_t callback,
- void* user,
- uint32_t notificationFrames = 0,
- bool threadCanCallJava = false,
- audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
- transfer_type transferType = TRANSFER_DEFAULT,
- audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
- uid_t uid = AUDIO_UID_INVALID,
- pid_t pid = -1,
- const audio_attributes_t* pAttributes = nullptr,
- audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
- audio_microphone_direction_t
- selectedMicDirection = MIC_DIRECTION_UNSPECIFIED,
- float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT,
- int32_t maxSharedAudioHistoryMs = 0);
-
/* Result of constructing the AudioRecord. This must be checked for successful initialization
* before using any AudioRecord API (except for set()), because using
* an uninitialized AudioRecord produces undefined results.
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 9f540e6..b6ee483 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -148,7 +148,6 @@
* - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp.
*/
- typedef void (*legacy_callback_t)(int event, void* user, void* info);
class IAudioTrackCallback : public virtual RefBase {
friend AudioTrack;
protected:
@@ -343,26 +342,6 @@
float maxRequiredSpeed = 1.0f,
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
-
- AudioTrack( audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- audio_output_flags_t flags,
- legacy_callback_t cbf,
- void* user = nullptr,
- int32_t notificationFrames = 0,
- audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
- transfer_type transferType = TRANSFER_DEFAULT,
- const audio_offload_info_t *offloadInfo = nullptr,
- const AttributionSourceState& attributionSource =
- AttributionSourceState(),
- const audio_attributes_t* pAttributes = nullptr,
- bool doNotReconnect = false,
- float maxRequiredSpeed = 1.0f,
- audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
-
/* Creates an audio track and registers it with AudioFlinger.
* With this constructor, the track is configured for static buffer mode.
* Data to be rendered is passed in a shared memory buffer
@@ -391,25 +370,6 @@
bool doNotReconnect = false,
float maxRequiredSpeed = 1.0f);
-
- AudioTrack( audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- const sp<IMemory>& sharedBuffer,
- audio_output_flags_t flags,
- legacy_callback_t cbf,
- void* user = nullptr,
- int32_t notificationFrames = 0,
- audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
- transfer_type transferType = TRANSFER_DEFAULT,
- const audio_offload_info_t *offloadInfo = nullptr,
- const AttributionSourceState& attributionSource =
- AttributionSourceState(),
- const audio_attributes_t* pAttributes = nullptr,
- bool doNotReconnect = false,
- float maxRequiredSpeed = 1.0f);
-
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioTrack.
*/
@@ -490,28 +450,8 @@
}
void onFirstRef() override;
public:
- status_t set(audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t frameCount,
- audio_output_flags_t flags,
- legacy_callback_t callback,
- void * user = nullptr,
- int32_t notificationFrames = 0,
- const sp<IMemory>& sharedBuffer = 0,
- bool threadCanCallJava = false,
- audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
- transfer_type transferType = TRANSFER_DEFAULT,
- const audio_offload_info_t *offloadInfo = nullptr,
- const AttributionSourceState& attributionSource =
- AttributionSourceState(),
- const audio_attributes_t* pAttributes = nullptr,
- bool doNotReconnect = false,
- float maxRequiredSpeed = 1.0f,
- audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
-
- // FIXME(b/169889714): Vendor code depends on the old method signature at link time
+ typedef void (*legacy_callback_t)(int event, void* user, void* info);
+ // FIXME(b/169889714): Vendor code depends on the old method signature at link time
status_t set(audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 891293e..6535b5b 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -93,3 +93,107 @@
],
data: ["record_test_input_*.txt"],
}
+
+cc_defaults {
+ name: "libaudioclient_gtests_defaults",
+ cflags: [
+ "-Wall",
+ "-Werror",
+ ],
+ shared_libs: [
+ "capture_state_listener-aidl-cpp",
+ "framework-permission-aidl-cpp",
+ "libbase",
+ "libbinder",
+ "libcgrouprc",
+ "libcutils",
+ "libdl",
+ "liblog",
+ "libmedia",
+ "libmediametrics",
+ "libmediautils",
+ "libmedia_helper",
+ "libnblog",
+ "libprocessgroup",
+ "libshmemcompat",
+ "libstagefright_foundation",
+ "libutils",
+ "libvibrator",
+ "mediametricsservice-aidl-cpp",
+ "packagemanager_aidl-cpp",
+ "shared-file-region-aidl-cpp",
+ ],
+ static_libs: [
+ "android.hardware.audio.common@7.0-enums",
+ "android.media.audio.common.types-V1-cpp",
+ "audioclient-types-aidl-cpp",
+ "audioflinger-aidl-cpp",
+ "audiopolicy-aidl-cpp",
+ "audiopolicy-types-aidl-cpp",
+ "av-types-aidl-cpp",
+ "effect-aidl-cpp",
+ "libaudioclient",
+ "libaudioclient_aidl_conversion",
+ "libaudiofoundation",
+ "libaudiomanager",
+ "libaudiopolicy",
+ "libaudioutils",
+ ],
+ data: ["bbb*.raw"],
+ test_config_template: "audio_test_template.xml",
+ test_suites: ["device-tests"],
+}
+
+cc_test {
+ name: "audiorecord_tests",
+ defaults: ["libaudioclient_gtests_defaults"],
+ srcs: [
+ "audiorecord_tests.cpp",
+ "audio_test_utils.cpp",
+ ],
+}
+
+cc_test {
+ name: "audiotrack_tests",
+ defaults: ["libaudioclient_gtests_defaults"],
+ srcs: [
+ "audiotrack_tests.cpp",
+ "audio_test_utils.cpp",
+ ],
+}
+
+cc_test {
+ name: "audioeffect_tests",
+ defaults: ["libaudioclient_gtests_defaults"],
+ srcs: [
+ "audioeffect_tests.cpp",
+ "audio_test_utils.cpp",
+ ],
+}
+
+cc_test {
+ name: "audiorouting_tests",
+ defaults: ["libaudioclient_gtests_defaults"],
+ srcs: [
+ "audiorouting_tests.cpp",
+ "audio_test_utils.cpp",
+ ],
+ shared_libs: [
+ "libxml2",
+ ],
+}
+
+cc_test {
+ name: "audioclient_serialization_tests",
+ defaults: ["libaudioclient_gtests_defaults"],
+ srcs: [
+ "audioclient_serialization_tests.cpp",
+ "audio_test_utils.cpp",
+ ],
+}
+
+cc_test {
+ name: "trackplayerbase_tests",
+ defaults: ["libaudioclient_gtests_defaults"],
+ srcs: ["trackplayerbase_tests.cpp"],
+}
diff --git a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
index 997f62a..9e663bc 100644
--- a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
+++ b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
@@ -16,22 +16,29 @@
#include <gtest/gtest.h>
-#include <media/AudioCommonTypes.h>
#include <media/AidlConversion.h>
+#include <media/AudioCommonTypes.h>
using namespace android;
using namespace android::aidl_utils;
+using android::media::AudioDirectMode;
using media::audio::common::AudioChannelLayout;
using media::audio::common::AudioDeviceDescription;
using media::audio::common::AudioDeviceType;
+using media::audio::common::AudioEncapsulationMetadataType;
+using media::audio::common::AudioEncapsulationType;
using media::audio::common::AudioFormatDescription;
using media::audio::common::AudioFormatType;
+using media::audio::common::AudioGainMode;
+using media::audio::common::AudioStandard;
+using media::audio::common::ExtraAudioDescriptor;
using media::audio::common::PcmType;
namespace {
-template<typename T> size_t hash(const T& t) {
+template <typename T>
+size_t hash(const T& t) {
return std::hash<T>{}(t);
}
@@ -52,10 +59,8 @@
return AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
// Use channels that exist both for input and output,
// but doesn't form a known layout mask.
- AudioChannelLayout::CHANNEL_FRONT_LEFT |
- AudioChannelLayout::CHANNEL_FRONT_RIGHT |
- AudioChannelLayout::CHANNEL_TOP_SIDE_LEFT |
- AudioChannelLayout::CHANNEL_TOP_SIDE_RIGHT);
+ AudioChannelLayout::CHANNEL_FRONT_LEFT | AudioChannelLayout::CHANNEL_FRONT_RIGHT |
+ AudioChannelLayout::CHANNEL_TOP_SIDE_LEFT | AudioChannelLayout::CHANNEL_TOP_SIDE_RIGHT);
}
AudioChannelLayout make_ACL_ChannelIndex2() {
@@ -74,7 +79,7 @@
}
AudioDeviceDescription make_AudioDeviceDescription(AudioDeviceType type,
- const std::string& connection = "") {
+ const std::string& connection = "") {
AudioDeviceDescription result;
result.type = type;
result.connection = connection;
@@ -95,12 +100,12 @@
AudioDeviceDescription make_ADD_WiredHeadset() {
return make_AudioDeviceDescription(AudioDeviceType::OUT_HEADSET,
- AudioDeviceDescription::CONNECTION_ANALOG());
+ AudioDeviceDescription::CONNECTION_ANALOG());
}
AudioDeviceDescription make_ADD_BtScoHeadset() {
return make_AudioDeviceDescription(AudioDeviceType::OUT_HEADSET,
- AudioDeviceDescription::CONNECTION_BT_SCO());
+ AudioDeviceDescription::CONNECTION_BT_SCO());
}
AudioFormatDescription make_AudioFormatDescription(AudioFormatType type) {
@@ -121,8 +126,7 @@
return result;
}
-AudioFormatDescription make_AudioFormatDescription(PcmType transport,
- const std::string& encoding) {
+AudioFormatDescription make_AudioFormatDescription(PcmType transport, const std::string& encoding) {
auto result = make_AudioFormatDescription(encoding);
result.pcm = transport;
return result;
@@ -154,6 +158,22 @@
return afd;
}
+android::media::TrackSecondaryOutputInfo make_TrackSecondaryOutputInfo() {
+ android::media::TrackSecondaryOutputInfo result;
+ result.portId = 1;
+ result.secondaryOutputIds = {0, 5, 7};
+ return result;
+}
+
+ExtraAudioDescriptor make_ExtraAudioDescriptor(AudioStandard audioStandard,
+ AudioEncapsulationType audioEncapsulationType) {
+ ExtraAudioDescriptor result;
+ result.standard = audioStandard;
+ result.audioDescriptor = {0xb4, 0xaf, 0x98, 0x1a};
+ result.encapsulationType = audioEncapsulationType;
+ return result;
+}
+
} // namespace
// Verify that two independently constructed ADDs/AFDs have the same hash.
@@ -163,7 +183,8 @@
// is identical to the same format description constructed by the framework.
class HashIdentityTest : public ::testing::Test {
public:
- template<typename T> void verifyHashIdentity(const std::vector<std::function<T()>>& valueGens) {
+ template <typename T>
+ void verifyHashIdentity(const std::vector<std::function<T()>>& valueGens) {
for (size_t i = 0; i < valueGens.size(); ++i) {
for (size_t j = 0; j < valueGens.size(); ++j) {
if (i == j) {
@@ -177,27 +198,25 @@
};
TEST_F(HashIdentityTest, AudioChannelLayoutHashIdentity) {
- verifyHashIdentity<AudioChannelLayout>({
- make_ACL_None, make_ACL_Invalid, make_ACL_Stereo,
- make_ACL_LayoutArbitrary, make_ACL_ChannelIndex2,
- make_ACL_ChannelIndexArbitrary, make_ACL_VoiceCall});
+ verifyHashIdentity<AudioChannelLayout>({make_ACL_None, make_ACL_Invalid, make_ACL_Stereo,
+ make_ACL_LayoutArbitrary, make_ACL_ChannelIndex2,
+ make_ACL_ChannelIndexArbitrary, make_ACL_VoiceCall});
}
TEST_F(HashIdentityTest, AudioDeviceDescriptionHashIdentity) {
- verifyHashIdentity<AudioDeviceDescription>({
- make_ADD_None, make_ADD_DefaultIn, make_ADD_DefaultOut, make_ADD_WiredHeadset,
- make_ADD_BtScoHeadset});
+ verifyHashIdentity<AudioDeviceDescription>({make_ADD_None, make_ADD_DefaultIn,
+ make_ADD_DefaultOut, make_ADD_WiredHeadset,
+ make_ADD_BtScoHeadset});
}
TEST_F(HashIdentityTest, AudioFormatDescriptionHashIdentity) {
- verifyHashIdentity<AudioFormatDescription>({
- make_AFD_Default, make_AFD_Invalid, make_AFD_Pcm16Bit, make_AFD_Bitstream,
- make_AFD_Encap, make_AFD_Encap_with_Enc});
+ 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<AudioChannelLayout, bool /*isInput*/>;
-class AudioChannelLayoutRoundTripTest :
- public testing::TestWithParam<ChannelLayoutParam> {};
+class AudioChannelLayoutRoundTripTest : public testing::TestWithParam<ChannelLayoutParam> {};
TEST_P(AudioChannelLayoutRoundTripTest, Aidl2Legacy2Aidl) {
const auto initial = std::get<0>(GetParam());
const bool isInput = std::get<1>(GetParam());
@@ -207,21 +226,82 @@
ASSERT_TRUE(convBack.ok());
EXPECT_EQ(initial, convBack.value());
}
-INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutRoundTrip,
- AudioChannelLayoutRoundTripTest,
+
+INSTANTIATE_TEST_SUITE_P(
+ AudioChannelLayoutRoundTrip, AudioChannelLayoutRoundTripTest,
testing::Combine(
testing::Values(AudioChannelLayout{}, make_ACL_Invalid(), make_ACL_Stereo(),
- make_ACL_LayoutArbitrary(), make_ACL_ChannelIndex2(),
- make_ACL_ChannelIndexArbitrary()),
+ make_ACL_LayoutArbitrary(), make_ACL_ChannelIndex2(),
+ make_ACL_ChannelIndexArbitrary(),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_FRONT_LEFT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_FRONT_RIGHT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_BACK_CENTER),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_BACK_LEFT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_BACK_RIGHT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_FRONT_CENTER),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_LOW_FREQUENCY),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_SIDE_LEFT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_SIDE_RIGHT)),
testing::Values(false, true)));
-INSTANTIATE_TEST_SUITE_P(AudioChannelVoiceRoundTrip,
- AudioChannelLayoutRoundTripTest,
- // In legacy constants the voice call is only defined for input.
- testing::Combine(testing::Values(make_ACL_VoiceCall()), testing::Values(true)));
+INSTANTIATE_TEST_SUITE_P(AudioChannelVoiceRoundTrip, AudioChannelLayoutRoundTripTest,
+ // In legacy constants the voice call is only defined for input.
+ testing::Combine(testing::Values(make_ACL_VoiceCall()),
+ testing::Values(true)));
+
+INSTANTIATE_TEST_SUITE_P(
+ OutAudioChannelLayoutLayoutRoundTrip, AudioChannelLayoutRoundTripTest,
+ testing::Combine(
+ testing::Values(AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_FRONT_LEFT_OF_CENTER),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_FRONT_RIGHT_OF_CENTER),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_SIDE_LEFT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_SIDE_RIGHT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_CENTER),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_FRONT_LEFT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_FRONT_CENTER),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_FRONT_RIGHT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_BACK_LEFT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_BACK_CENTER),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_TOP_BACK_RIGHT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_BOTTOM_FRONT_LEFT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_BOTTOM_FRONT_CENTER),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_BOTTOM_FRONT_RIGHT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_LOW_FREQUENCY_2),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_FRONT_WIDE_LEFT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_FRONT_WIDE_RIGHT),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_HAPTIC_A),
+ AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
+ AudioChannelLayout::CHANNEL_HAPTIC_B)),
+ testing::Values(false)));
using ChannelLayoutEdgeCaseParam = std::tuple<int /*legacy*/, bool /*isInput*/, bool /*isValid*/>;
-class AudioChannelLayoutEdgeCaseTest :
- public testing::TestWithParam<ChannelLayoutEdgeCaseParam> {};
+class AudioChannelLayoutEdgeCaseTest : public testing::TestWithParam<ChannelLayoutEdgeCaseParam> {};
TEST_P(AudioChannelLayoutEdgeCaseTest, Legacy2Aidl) {
const audio_channel_mask_t legacy = static_cast<audio_channel_mask_t>(std::get<0>(GetParam()));
const bool isInput = std::get<1>(GetParam());
@@ -229,8 +309,8 @@
auto conv = legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy, isInput);
EXPECT_EQ(isValid, conv.ok());
}
-INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutEdgeCase,
- AudioChannelLayoutEdgeCaseTest,
+INSTANTIATE_TEST_SUITE_P(
+ AudioChannelLayoutEdgeCase, AudioChannelLayoutEdgeCaseTest,
testing::Values(
// Valid legacy input masks.
std::make_tuple(AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO, true, true),
@@ -240,25 +320,26 @@
std::make_tuple(
// This has the same numerical representation as Mask 'A' below
AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
- AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT, false, true),
+ AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT,
+ false, true),
std::make_tuple(
// This has the same numerical representation as Mask 'B' below
AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
- AUDIO_CHANNEL_OUT_TOP_BACK_LEFT, false, true),
+ AUDIO_CHANNEL_OUT_TOP_BACK_LEFT,
+ false, true),
// Invalid legacy input masks.
std::make_tuple(AUDIO_CHANNEL_IN_6, true, false),
- std::make_tuple(
- AUDIO_CHANNEL_IN_6 | AUDIO_CHANNEL_IN_FRONT_PROCESSED, true, false),
- std::make_tuple(
- AUDIO_CHANNEL_IN_PRESSURE | AUDIO_CHANNEL_IN_X_AXIS |
- AUDIO_CHANNEL_IN_Y_AXIS | AUDIO_CHANNEL_IN_Z_AXIS, true, false),
+ std::make_tuple(AUDIO_CHANNEL_IN_6 | AUDIO_CHANNEL_IN_FRONT_PROCESSED, true, false),
+ std::make_tuple(AUDIO_CHANNEL_IN_PRESSURE | AUDIO_CHANNEL_IN_X_AXIS |
+ AUDIO_CHANNEL_IN_Y_AXIS | AUDIO_CHANNEL_IN_Z_AXIS,
+ true, false),
std::make_tuple( // Mask 'A'
AUDIO_CHANNEL_IN_STEREO | AUDIO_CHANNEL_IN_VOICE_UPLINK, true, false),
std::make_tuple( // Mask 'B'
AUDIO_CHANNEL_IN_STEREO | AUDIO_CHANNEL_IN_VOICE_DNLINK, true, false)));
-class AudioDeviceDescriptionRoundTripTest :
- public testing::TestWithParam<AudioDeviceDescription> {};
+class AudioDeviceDescriptionRoundTripTest : public testing::TestWithParam<AudioDeviceDescription> {
+};
TEST_P(AudioDeviceDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
const auto initial = GetParam();
auto conv = aidl2legacy_AudioDeviceDescription_audio_devices_t(initial);
@@ -267,13 +348,13 @@
ASSERT_TRUE(convBack.ok());
EXPECT_EQ(initial, convBack.value());
}
-INSTANTIATE_TEST_SUITE_P(AudioDeviceDescriptionRoundTrip,
- AudioDeviceDescriptionRoundTripTest,
- testing::Values(AudioDeviceDescription{}, make_ADD_DefaultIn(),
- make_ADD_DefaultOut(), make_ADD_WiredHeadset(), make_ADD_BtScoHeadset()));
+INSTANTIATE_TEST_SUITE_P(AudioDeviceDescriptionRoundTrip, AudioDeviceDescriptionRoundTripTest,
+ testing::Values(AudioDeviceDescription{}, make_ADD_DefaultIn(),
+ make_ADD_DefaultOut(), make_ADD_WiredHeadset(),
+ make_ADD_BtScoHeadset()));
-class AudioFormatDescriptionRoundTripTest :
- public testing::TestWithParam<AudioFormatDescription> {};
+class AudioFormatDescriptionRoundTripTest : public testing::TestWithParam<AudioFormatDescription> {
+};
TEST_P(AudioFormatDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
const auto initial = GetParam();
auto conv = aidl2legacy_AudioFormatDescription_audio_format_t(initial);
@@ -282,6 +363,140 @@
ASSERT_TRUE(convBack.ok());
EXPECT_EQ(initial, convBack.value());
}
-INSTANTIATE_TEST_SUITE_P(AudioFormatDescriptionRoundTrip,
- AudioFormatDescriptionRoundTripTest,
- testing::Values(make_AFD_Invalid(), AudioFormatDescription{}, make_AFD_Pcm16Bit()));
+INSTANTIATE_TEST_SUITE_P(AudioFormatDescriptionRoundTrip, AudioFormatDescriptionRoundTripTest,
+ testing::Values(make_AFD_Invalid(), AudioFormatDescription{},
+ make_AFD_Pcm16Bit()));
+
+class AudioDirectModeRoundTripTest : public testing::TestWithParam<AudioDirectMode> {};
+TEST_P(AudioDirectModeRoundTripTest, Aidl2Legacy2Aidl) {
+ const auto initial = GetParam();
+ auto conv = aidl2legacy_AudioDirectMode_audio_direct_mode_t(initial);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = legacy2aidl_audio_direct_mode_t_AudioDirectMode(conv.value());
+ ASSERT_TRUE(convBack.ok());
+ EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioDirectMode, AudioDirectModeRoundTripTest,
+ testing::Values(AudioDirectMode::NONE, AudioDirectMode::OFFLOAD,
+ AudioDirectMode::OFFLOAD_GAPLESS,
+ AudioDirectMode::BITSTREAM));
+
+class AudioStandardRoundTripTest : public testing::TestWithParam<AudioStandard> {};
+TEST_P(AudioStandardRoundTripTest, Aidl2Legacy2Aidl) {
+ const auto initial = GetParam();
+ auto conv = aidl2legacy_AudioStandard_audio_standard_t(initial);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = legacy2aidl_audio_standard_t_AudioStandard(conv.value());
+ ASSERT_TRUE(convBack.ok());
+ EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioStandard, AudioStandardRoundTripTest,
+ testing::Values(AudioStandard::NONE, AudioStandard::EDID));
+
+class AudioEncapsulationMetadataTypeRoundTripTest
+ : public testing::TestWithParam<AudioEncapsulationMetadataType> {};
+TEST_P(AudioEncapsulationMetadataTypeRoundTripTest, Aidl2Legacy2Aidl) {
+ const auto initial = GetParam();
+ auto conv =
+ aidl2legacy_AudioEncapsulationMetadataType_audio_encapsulation_metadata_type_t(initial);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = legacy2aidl_audio_encapsulation_metadata_type_t_AudioEncapsulationMetadataType(
+ conv.value());
+ ASSERT_TRUE(convBack.ok());
+ EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioEncapsulationMetadataType,
+ AudioEncapsulationMetadataTypeRoundTripTest,
+ testing::Values(AudioEncapsulationMetadataType::NONE,
+ AudioEncapsulationMetadataType::FRAMEWORK_TUNER,
+ AudioEncapsulationMetadataType::DVB_AD_DESCRIPTOR));
+
+class AudioGainModeRoundTripTest : public testing::TestWithParam<AudioGainMode> {};
+TEST_P(AudioGainModeRoundTripTest, Aidl2Legacy2Aidl) {
+ const auto initial = GetParam();
+ auto conv = aidl2legacy_AudioGainMode_audio_gain_mode_t(initial);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = legacy2aidl_audio_gain_mode_t_AudioGainMode(conv.value());
+ ASSERT_TRUE(convBack.ok());
+ EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioGainMode, AudioGainModeRoundTripTest,
+ testing::Values(AudioGainMode::JOINT, AudioGainMode::CHANNELS,
+ AudioGainMode::RAMP));
+
+TEST(AudioTrackSecondaryOutputInfoRoundTripTest, Aidl2Legacy2Aidl) {
+ const auto initial = make_TrackSecondaryOutputInfo();
+ auto conv = aidl2legacy_TrackSecondaryOutputInfo_TrackSecondaryOutputInfoPair(initial);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = legacy2aidl_TrackSecondaryOutputInfoPair_TrackSecondaryOutputInfo(conv.value());
+ ASSERT_TRUE(convBack.ok());
+ EXPECT_EQ(initial, convBack.value());
+}
+
+using ExtraAudioDescriptorParam = std::tuple<AudioStandard, AudioEncapsulationType>;
+class ExtraAudioDescriptorRoundTripTest : public testing::TestWithParam<ExtraAudioDescriptorParam> {
+};
+TEST_P(ExtraAudioDescriptorRoundTripTest, Aidl2Legacy2Aidl) {
+ ExtraAudioDescriptor initial =
+ make_ExtraAudioDescriptor(std::get<0>(GetParam()), std::get<1>(GetParam()));
+ auto conv = aidl2legacy_ExtraAudioDescriptor_audio_extra_audio_descriptor(initial);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = legacy2aidl_audio_extra_audio_descriptor_ExtraAudioDescriptor(conv.value());
+ ASSERT_TRUE(convBack.ok());
+ EXPECT_EQ(initial, convBack.value());
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ ExtraAudioDescriptor, ExtraAudioDescriptorRoundTripTest,
+ testing::Values(std::make_tuple(AudioStandard::NONE, AudioEncapsulationType::NONE),
+ std::make_tuple(AudioStandard::EDID, AudioEncapsulationType::NONE),
+ std::make_tuple(AudioStandard::EDID, AudioEncapsulationType::IEC61937)));
+
+TEST(AudioPortSessionExtRoundTripTest, Aidl2Legacy2Aidl) {
+ const int32_t initial = 7;
+ auto conv = aidl2legacy_int32_t_audio_port_session_ext(initial);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = legacy2aidl_audio_port_session_ext_int32_t(conv.value());
+ ASSERT_TRUE(convBack.ok());
+ EXPECT_EQ(initial, convBack.value());
+}
+
+class AudioGainTest : public testing::TestWithParam<bool> {};
+TEST_P(AudioGainTest, Legacy2Aidl2Legacy) {
+ audio_port_v7 port;
+ port.num_gains = 2;
+ port.gains[0] = {.mode = AUDIO_GAIN_MODE_JOINT,
+ .channel_mask = AUDIO_CHANNEL_IN_STEREO,
+ .min_value = -3200,
+ .max_value = 600,
+ .default_value = 0,
+ .step_value = 100,
+ .min_ramp_ms = 10,
+ .max_ramp_ms = 20};
+ port.gains[1] = {.mode = AUDIO_GAIN_MODE_JOINT,
+ .channel_mask = AUDIO_CHANNEL_IN_MONO,
+ .min_value = -8800,
+ .max_value = 4000,
+ .default_value = 0,
+ .step_value = 100,
+ .min_ramp_ms = 192,
+ .max_ramp_ms = 224};
+
+ const auto isInput = GetParam();
+ for (int i = 0; i < port.num_gains; i++) {
+ auto initial = port.gains[i];
+ auto conv = legacy2aidl_audio_gain_AudioGain(initial, isInput);
+ ASSERT_TRUE(conv.ok());
+ auto convBack = aidl2legacy_AudioGain_audio_gain(conv.value(), isInput);
+ ASSERT_TRUE(convBack.ok());
+ EXPECT_EQ(initial.mode, convBack.value().mode);
+ EXPECT_EQ(initial.channel_mask, convBack.value().channel_mask);
+ EXPECT_EQ(initial.min_value, convBack.value().min_value);
+ EXPECT_EQ(initial.max_value, convBack.value().max_value);
+ EXPECT_EQ(initial.default_value, convBack.value().default_value);
+ EXPECT_EQ(initial.step_value, convBack.value().step_value);
+ EXPECT_EQ(initial.min_ramp_ms, convBack.value().min_ramp_ms);
+ EXPECT_EQ(initial.max_ramp_ms, convBack.value().max_ramp_ms);
+ }
+}
+INSTANTIATE_TEST_SUITE_P(AudioGain, AudioGainTest, testing::Values(true, false));
diff --git a/media/libaudioclient/tests/audio_aidl_status_tests.cpp b/media/libaudioclient/tests/audio_aidl_status_tests.cpp
index 5517091..8a7e6c1 100644
--- a/media/libaudioclient/tests/audio_aidl_status_tests.cpp
+++ b/media/libaudioclient/tests/audio_aidl_status_tests.cpp
@@ -37,25 +37,10 @@
// Special status values are preserved on round trip.
TEST(audio_aidl_status_tests, statusRoundTripSpecialValues) {
- for (status_t status : {
- OK,
- UNKNOWN_ERROR,
- NO_MEMORY,
- INVALID_OPERATION,
- BAD_VALUE,
- BAD_TYPE,
- NAME_NOT_FOUND,
- PERMISSION_DENIED,
- NO_INIT,
- ALREADY_EXISTS,
- DEAD_OBJECT,
- FAILED_TRANSACTION,
- BAD_INDEX,
- NOT_ENOUGH_DATA,
- WOULD_BLOCK,
- TIMED_OUT,
- UNKNOWN_TRANSACTION,
- FDS_NOT_ALLOWED}) {
+ for (status_t status :
+ {OK, UNKNOWN_ERROR, NO_MEMORY, INVALID_OPERATION, BAD_VALUE, BAD_TYPE, NAME_NOT_FOUND,
+ PERMISSION_DENIED, NO_INIT, ALREADY_EXISTS, DEAD_OBJECT, FAILED_TRANSACTION, BAD_INDEX,
+ NOT_ENOUGH_DATA, WOULD_BLOCK, TIMED_OUT, UNKNOWN_TRANSACTION, FDS_NOT_ALLOWED}) {
ASSERT_EQ(status, statusTFromBinderStatus(binderStatusFromStatusT(status)));
}
}
@@ -63,47 +48,29 @@
// Binder exceptions show as an error (not fixed at this time); these come fromExceptionCode().
TEST(audio_aidl_status_tests, binderStatusExceptions) {
for (int exceptionCode : {
- //Status::EX_NONE,
- Status::EX_SECURITY,
- Status::EX_BAD_PARCELABLE,
- Status::EX_ILLEGAL_ARGUMENT,
- Status::EX_NULL_POINTER,
- Status::EX_ILLEGAL_STATE,
- Status::EX_NETWORK_MAIN_THREAD,
- Status::EX_UNSUPPORTED_OPERATION,
- //Status::EX_SERVICE_SPECIFIC, -- tested fromServiceSpecificError()
- Status::EX_PARCELABLE,
- // This is special and Java specific; see Parcel.java.
- Status::EX_HAS_REPLY_HEADER,
- // This is special, and indicates to C++ binder proxies that the
- // transaction has failed at a low level.
- //Status::EX_TRANSACTION_FAILED, -- tested fromStatusT().
- }) {
+ // Status::EX_NONE,
+ Status::EX_SECURITY, Status::EX_BAD_PARCELABLE, Status::EX_ILLEGAL_ARGUMENT,
+ Status::EX_NULL_POINTER, Status::EX_ILLEGAL_STATE, Status::EX_NETWORK_MAIN_THREAD,
+ Status::EX_UNSUPPORTED_OPERATION,
+ // Status::EX_SERVICE_SPECIFIC, -- tested fromServiceSpecificError()
+ Status::EX_PARCELABLE,
+ // This is special and Java specific; see Parcel.java.
+ Status::EX_HAS_REPLY_HEADER,
+ // This is special, and indicates to C++ binder proxies that the
+ // transaction has failed at a low level.
+ // Status::EX_TRANSACTION_FAILED, -- tested fromStatusT().
+ }) {
ASSERT_NE(OK, statusTFromBinderStatus(Status::fromExceptionCode(exceptionCode)));
}
}
// Binder transaction errors show exactly in status_t; these come fromStatusT().
TEST(audio_aidl_status_tests, binderStatusTransactionError) {
- for (status_t status : {
- OK, // Note: fromStatusT does check if this is 0, so this is no error.
- UNKNOWN_ERROR,
- NO_MEMORY,
- INVALID_OPERATION,
- BAD_VALUE,
- BAD_TYPE,
- NAME_NOT_FOUND,
- PERMISSION_DENIED,
- NO_INIT,
- ALREADY_EXISTS,
- DEAD_OBJECT,
- FAILED_TRANSACTION,
- BAD_INDEX,
- NOT_ENOUGH_DATA,
- WOULD_BLOCK,
- TIMED_OUT,
- UNKNOWN_TRANSACTION,
- FDS_NOT_ALLOWED}) {
+ for (status_t status :
+ {OK, // Note: fromStatusT does check if this is 0, so this is no error.
+ UNKNOWN_ERROR, NO_MEMORY, INVALID_OPERATION, BAD_VALUE, BAD_TYPE, NAME_NOT_FOUND,
+ PERMISSION_DENIED, NO_INIT, ALREADY_EXISTS, DEAD_OBJECT, FAILED_TRANSACTION, BAD_INDEX,
+ NOT_ENOUGH_DATA, WOULD_BLOCK, TIMED_OUT, UNKNOWN_TRANSACTION, FDS_NOT_ALLOWED}) {
ASSERT_EQ(status, statusTFromBinderStatus(Status::fromStatusT(status)));
}
}
diff --git a/media/libaudioclient/tests/audio_test_template.xml b/media/libaudioclient/tests/audio_test_template.xml
new file mode 100644
index 0000000..ed0cb21
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_template.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2022 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="Unit test configuration for {MODULE}">
+ <target_preparer class="com.android.tradefed.targetprep.RootTargetPreparer" />
+
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push-file" key="{MODULE}" value="/data/local/tmp/{MODULE}" />
+
+ <!-- Files used for audio testing -->
+ <option name="push-file" key="bbb_1ch_8kHz_s16le.raw" value="/data/local/tmp/bbb_1ch_8kHz_s16le.raw" />
+ <option name="push-file" key="bbb_2ch_24kHz_s16le.raw" value="/data/local/tmp/bbb_2ch_24kHz_s16le.raw" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="{MODULE}" />
+ </test>
+</configuration>
diff --git a/media/libaudioclient/tests/audio_test_utils.cpp b/media/libaudioclient/tests/audio_test_utils.cpp
new file mode 100644
index 0000000..018d920
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_utils.cpp
@@ -0,0 +1,795 @@
+/*
+ * 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 "AudioTestUtils"
+
+#include <utils/Log.h>
+
+#include "audio_test_utils.h"
+
+// Generates a random string.
+void CreateRandomFile(int& fd) {
+ std::string filename = "/data/local/tmp/record-XXXXXX";
+ fd = mkstemp(filename.data());
+}
+
+void OnAudioDeviceUpdateNotifier::onAudioDeviceUpdate(audio_io_handle_t audioIo,
+ audio_port_handle_t deviceId) {
+ std::unique_lock<std::mutex> lock{mMutex};
+ ALOGD("%s audioIo=%d deviceId=%d", __func__, audioIo, deviceId);
+ mAudioIo = audioIo;
+ mDeviceId = deviceId;
+ mCondition.notify_all();
+}
+
+status_t OnAudioDeviceUpdateNotifier::waitForAudioDeviceCb() {
+ std::unique_lock<std::mutex> lock{mMutex};
+ if (mAudioIo == AUDIO_IO_HANDLE_NONE) {
+ mCondition.wait_for(lock, std::chrono::milliseconds(500));
+ if (mAudioIo == AUDIO_IO_HANDLE_NONE) return TIMED_OUT;
+ }
+ return OK;
+}
+
+AudioPlayback::AudioPlayback(uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask, audio_output_flags_t flags,
+ audio_session_t sessionId, AudioTrack::transfer_type transferType,
+ audio_attributes_t* attributes, audio_offload_info_t* info)
+ : mSampleRate(sampleRate),
+ mFormat(format),
+ mChannelMask(channelMask),
+ mFlags(flags),
+ mSessionId(sessionId),
+ mTransferType(transferType),
+ mAttributes(attributes),
+ mOffloadInfo(info) {
+ mStopPlaying = false;
+ mBytesUsedSoFar = 0;
+ mState = PLAY_NO_INIT;
+ mMemCapacity = 0;
+ mMemoryDealer = nullptr;
+ mMemory = nullptr;
+}
+
+AudioPlayback::~AudioPlayback() {
+ stop();
+}
+
+status_t AudioPlayback::create() {
+ if (mState != PLAY_NO_INIT) return INVALID_OPERATION;
+ std::string packageName{"AudioPlayback"};
+ AttributionSourceState attributionSource;
+ attributionSource.packageName = packageName;
+ attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+ attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+ attributionSource.token = sp<BBinder>::make();
+ if (mTransferType == AudioTrack::TRANSFER_OBTAIN) {
+ mTrack = new AudioTrack(attributionSource);
+ mTrack->set(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, 0 /* frameCount */,
+ mFlags, nullptr /* callback */, 0 /* notificationFrames */,
+ nullptr /* sharedBuffer */, false /*canCallJava */, mSessionId, mTransferType,
+ mOffloadInfo, attributionSource, mAttributes);
+ } else if (mTransferType == AudioTrack::TRANSFER_SHARED) {
+ mTrack = new AudioTrack(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, mMemory,
+ mFlags, wp<AudioTrack::IAudioTrackCallback>::fromExisting(this), 0,
+ mSessionId, mTransferType, nullptr, attributionSource, mAttributes);
+ } else {
+ ALOGE("Test application is not handling transfer type %s",
+ AudioTrack::convertTransferToText(mTransferType));
+ return INVALID_OPERATION;
+ }
+ mTrack->setCallerName(packageName);
+ status_t status = mTrack->initCheck();
+ if (NO_ERROR == status) mState = PLAY_READY;
+ return status;
+}
+
+status_t AudioPlayback::loadResource(const char* name) {
+ status_t status = OK;
+ FILE* fp = fopen(name, "rbe");
+ struct stat buf {};
+ if (fp && !fstat(fileno(fp), &buf)) {
+ mMemCapacity = buf.st_size;
+ mMemoryDealer = new MemoryDealer(mMemCapacity, "AudioPlayback");
+ if (nullptr == mMemoryDealer.get()) {
+ ALOGE("couldn't get MemoryDealer!");
+ fclose(fp);
+ return NO_MEMORY;
+ }
+ mMemory = mMemoryDealer->allocate(mMemCapacity);
+ if (nullptr == mMemory.get()) {
+ ALOGE("couldn't get IMemory!");
+ fclose(fp);
+ return NO_MEMORY;
+ }
+ uint8_t* ipBuffer = static_cast<uint8_t*>(static_cast<void*>(mMemory->unsecurePointer()));
+ fread(ipBuffer, sizeof(uint8_t), mMemCapacity, fp);
+ } else {
+ ALOGE("unable to open input file %s", name);
+ status = NAME_NOT_FOUND;
+ }
+ if (fp) fclose(fp);
+ return status;
+}
+
+sp<AudioTrack> AudioPlayback::getAudioTrackHandle() {
+ return (PLAY_NO_INIT != mState) ? mTrack : nullptr;
+}
+
+status_t AudioPlayback::start() {
+ status_t status;
+ if (PLAY_READY != mState) {
+ return INVALID_OPERATION;
+ } else {
+ status = mTrack->start();
+ if (OK == status) {
+ mState = PLAY_STARTED;
+ LOG_FATAL_IF(false != mTrack->stopped());
+ }
+ }
+ return status;
+}
+
+void AudioPlayback::onBufferEnd() {
+ std::unique_lock<std::mutex> lock{mMutex};
+ mStopPlaying = true;
+ mCondition.notify_all();
+}
+
+status_t AudioPlayback::fillBuffer() {
+ if (PLAY_STARTED != mState && PLAY_STOPPED != mState) return INVALID_OPERATION;
+ int retry = 25;
+ uint8_t* ipBuffer = static_cast<uint8_t*>(static_cast<void*>(mMemory->unsecurePointer()));
+ size_t nonContig = 0;
+ size_t bytesAvailable = mMemCapacity - mBytesUsedSoFar;
+ while (bytesAvailable > 0) {
+ AudioTrack::Buffer trackBuffer;
+ trackBuffer.frameCount = mTrack->frameCount() * 2;
+ status_t status = mTrack->obtainBuffer(&trackBuffer, retry, &nonContig);
+ if (OK == status) {
+ size_t bytesToCopy = std::min(bytesAvailable, trackBuffer.size());
+ if (bytesToCopy > 0) {
+ memcpy(trackBuffer.data(), ipBuffer + mBytesUsedSoFar, bytesToCopy);
+ }
+ mTrack->releaseBuffer(&trackBuffer);
+ mBytesUsedSoFar += bytesToCopy;
+ bytesAvailable = mMemCapacity - mBytesUsedSoFar;
+ if (bytesAvailable == 0) {
+ stop();
+ }
+ } else if (WOULD_BLOCK == status) {
+ if (mStopPlaying)
+ return OK;
+ else
+ return TIMED_OUT;
+ }
+ }
+ return OK;
+}
+
+status_t AudioPlayback::waitForConsumption(bool testSeek) {
+ if (PLAY_STARTED != mState) return INVALID_OPERATION;
+ // in static buffer mode, lets not play clips with duration > 30 sec
+ int retry = 30;
+ // Total number of frames in the input file.
+ size_t totalFrameCount = mMemCapacity / mTrack->frameSize();
+ while (!mStopPlaying && retry > 0) {
+ // Get the total numbers of frames played.
+ uint32_t currPosition;
+ mTrack->getPosition(&currPosition);
+ if (testSeek && (currPosition > totalFrameCount * 0.6)) {
+ testSeek = false;
+ if (!mTrack->hasStarted()) return BAD_VALUE;
+ mTrack->pauseAndWait(std::chrono::seconds(2));
+ if (mTrack->hasStarted()) return BAD_VALUE;
+ mTrack->reload();
+ mTrack->getPosition(&currPosition);
+ if (currPosition != 0) return BAD_VALUE;
+ mTrack->start();
+ while (currPosition < totalFrameCount * 0.3) {
+ mTrack->getPosition(&currPosition);
+ }
+ mTrack->pauseAndWait(std::chrono::seconds(2));
+ uint32_t setPosition = totalFrameCount * 0.9;
+ mTrack->setPosition(setPosition);
+ uint32_t bufferPosition;
+ mTrack->getBufferPosition(&bufferPosition);
+ if (bufferPosition != setPosition) return BAD_VALUE;
+ mTrack->start();
+ }
+ std::this_thread::sleep_for(std::chrono::milliseconds(300));
+ retry--;
+ }
+ if (!mStopPlaying) return TIMED_OUT;
+ return OK;
+}
+
+status_t AudioPlayback::onProcess(bool testSeek) {
+ if (mTransferType == AudioTrack::TRANSFER_SHARED)
+ return waitForConsumption(testSeek);
+ else if (mTransferType == AudioTrack::TRANSFER_OBTAIN)
+ return fillBuffer();
+ else
+ return INVALID_OPERATION;
+}
+
+void AudioPlayback::stop() {
+ std::unique_lock<std::mutex> lock{mMutex};
+ mStopPlaying = true;
+ if (mState != PLAY_STOPPED) {
+ int32_t msec = 0;
+ (void)mTrack->pendingDuration(&msec);
+ mTrack->stopAndJoinCallbacks();
+ LOG_FATAL_IF(true != mTrack->stopped());
+ mState = PLAY_STOPPED;
+ if (msec > 0) {
+ ALOGD("deleting recycled track, waiting for data drain (%d msec)", msec);
+ usleep(msec * 1000LL);
+ }
+ }
+}
+
+// hold pcm data sent by AudioRecord
+RawBuffer::RawBuffer(int64_t ptsPipeline, int64_t ptsManual, int32_t capacity)
+ : mData(capacity > 0 ? new uint8_t[capacity] : nullptr),
+ mPtsPipeline(ptsPipeline),
+ mPtsManual(ptsManual),
+ mCapacity(capacity) {}
+
+// Simple AudioCapture
+size_t AudioCapture::onMoreData(const AudioRecord::Buffer& buffer) {
+ if (mState != REC_STARTED) {
+ ALOGE("Unexpected Callback from audiorecord, not reading data");
+ return 0;
+ }
+
+ // no more frames to read
+ if (mNumFramesReceived > mNumFramesToRecord || mStopRecording) {
+ mStopRecording = true;
+ return 0;
+ }
+
+ int64_t timeUs = 0, position = 0, timeNs = 0;
+ ExtendedTimestamp ts;
+ ExtendedTimestamp::Location location;
+ const int32_t usPerSec = 1000000;
+
+ if (mRecord->getTimestamp(&ts) == OK &&
+ ts.getBestTimestamp(&position, &timeNs, ExtendedTimestamp::TIMEBASE_MONOTONIC, &location) ==
+ OK) {
+ // Use audio timestamp.
+ timeUs = timeNs / 1000 -
+ (position - mNumFramesReceived + mNumFramesLost) * usPerSec / mSampleRate;
+ } else {
+ // This should not happen in normal case.
+ ALOGW("Failed to get audio timestamp, fallback to use systemclock");
+ timeUs = systemTime() / 1000LL;
+ // Estimate the real sampling time of the 1st sample in this buffer
+ // from AudioRecord's latency. (Apply this adjustment first so that
+ // the start time logic is not affected.)
+ timeUs -= mRecord->latency() * 1000LL;
+ }
+
+ ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
+
+ const size_t frameSize = mRecord->frameSize();
+ uint64_t numLostBytes = (uint64_t)mRecord->getInputFramesLost() * frameSize;
+ if (numLostBytes > 0) {
+ ALOGW("Lost audio record data: %" PRIu64 " bytes", numLostBytes);
+ }
+ std::deque<RawBuffer> tmpQueue;
+ while (numLostBytes > 0) {
+ uint64_t bufferSize = numLostBytes;
+ if (numLostBytes > mMaxBytesPerCallback) {
+ numLostBytes -= mMaxBytesPerCallback;
+ bufferSize = mMaxBytesPerCallback;
+ } else {
+ numLostBytes = 0;
+ }
+ const int64_t timestampUs =
+ ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+ mRecord->getSampleRate();
+ RawBuffer emptyBuffer{timeUs, timestampUs, static_cast<int32_t>(bufferSize)};
+ memset(emptyBuffer.mData.get(), 0, bufferSize);
+ mNumFramesLost += bufferSize / frameSize;
+ mNumFramesReceived += bufferSize / frameSize;
+ tmpQueue.push_back(std::move(emptyBuffer));
+ }
+
+ if (buffer.size() == 0) {
+ ALOGW("Nothing is available from AudioRecord callback buffer");
+ } else {
+ const size_t bufferSize = buffer.size();
+ const int64_t timestampUs =
+ ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+ mRecord->getSampleRate();
+ RawBuffer audioBuffer{timeUs, timestampUs, static_cast<int32_t>(bufferSize)};
+ memcpy(audioBuffer.mData.get(), buffer.data(), bufferSize);
+ mNumFramesReceived += bufferSize / frameSize;
+ tmpQueue.push_back(std::move(audioBuffer));
+ }
+
+ if (tmpQueue.size() > 0) {
+ std::unique_lock<std::mutex> lock{mMutex};
+ for (auto it = tmpQueue.begin(); it != tmpQueue.end(); it++)
+ mBuffersReceived.push_back(std::move(*it));
+ mCondition.notify_all();
+ }
+ return buffer.size();
+}
+
+void AudioCapture::onOverrun() {
+ ALOGV("received event overrun");
+ mBufferOverrun = true;
+}
+
+void AudioCapture::onMarker(uint32_t markerPosition) {
+ ALOGV("received Callback at position %d", markerPosition);
+ mReceivedCbMarkerAtPosition = markerPosition;
+}
+
+void AudioCapture::onNewPos(uint32_t markerPosition) {
+ ALOGV("received Callback at position %d", markerPosition);
+ mReceivedCbMarkerCount++;
+}
+
+void AudioCapture::onNewIAudioRecord() {
+ ALOGV("IAudioRecord is re-created");
+}
+
+AudioCapture::AudioCapture(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask, audio_input_flags_t flags,
+ audio_session_t sessionId, AudioRecord::transfer_type transferType)
+ : mInputSource(inputSource),
+ mSampleRate(sampleRate),
+ mFormat(format),
+ mChannelMask(channelMask),
+ mFlags(flags),
+ mSessionId(sessionId),
+ mTransferType(transferType) {
+ mFrameCount = 0;
+ mNotificationFrames = 0;
+ mNumFramesToRecord = 0;
+ mNumFramesReceived = 0;
+ mNumFramesLost = 0;
+ mBufferOverrun = false;
+ mMarkerPosition = 0;
+ mMarkerPeriod = 0;
+ mReceivedCbMarkerAtPosition = -1;
+ mReceivedCbMarkerCount = 0;
+ mState = REC_NO_INIT;
+ mStopRecording = false;
+#if RECORD_TO_FILE
+ CreateRandomFile(mOutFileFd);
+#endif
+}
+
+AudioCapture::~AudioCapture() {
+ if (mOutFileFd > 0) close(mOutFileFd);
+ stop();
+}
+
+status_t AudioCapture::create() {
+ if (mState != REC_NO_INIT) return INVALID_OPERATION;
+ // get Min Frame Count
+ size_t minFrameCount;
+ status_t status =
+ AudioRecord::getMinFrameCount(&minFrameCount, mSampleRate, mFormat, mChannelMask);
+ if (NO_ERROR != status) return status;
+ // Limit notificationFrames basing on client bufferSize
+ const int samplesPerFrame = audio_channel_count_from_in_mask(mChannelMask);
+ const int bytesPerSample = audio_bytes_per_sample(mFormat);
+ mNotificationFrames = mMaxBytesPerCallback / (samplesPerFrame * bytesPerSample);
+ // select frameCount to be at least minFrameCount
+ mFrameCount = 2 * mNotificationFrames;
+ while (mFrameCount < minFrameCount) {
+ mFrameCount += mNotificationFrames;
+ }
+ if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+ ALOGW("Overriding all previous computations");
+ mFrameCount = 0;
+ mNotificationFrames = 0;
+ }
+ mNumFramesToRecord = (mSampleRate * 0.25); // record .25 sec
+ std::string packageName{"AudioCapture"};
+ AttributionSourceState attributionSource;
+ attributionSource.packageName = packageName;
+ attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+ attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+ attributionSource.token = sp<BBinder>::make();
+ if (mTransferType == AudioRecord::TRANSFER_OBTAIN) {
+ if (mSampleRate == 48000) { // test all available constructors
+ mRecord = new AudioRecord(mInputSource, mSampleRate, mFormat, mChannelMask,
+ attributionSource, mFrameCount, nullptr /* callback */,
+ mNotificationFrames, mSessionId, mTransferType, mFlags);
+ } else {
+ mRecord = new AudioRecord(attributionSource);
+ status = mRecord->set(mInputSource, mSampleRate, mFormat, mChannelMask, mFrameCount,
+ nullptr /* callback */, 0 /* notificationFrames */,
+ false /* canCallJava */, mSessionId, mTransferType, mFlags,
+ attributionSource.uid, attributionSource.pid);
+ }
+ if (NO_ERROR != status) return status;
+ } else if (mTransferType == AudioRecord::TRANSFER_CALLBACK) {
+ mRecord = new AudioRecord(mInputSource, mSampleRate, mFormat, mChannelMask,
+ attributionSource, mFrameCount, this, mNotificationFrames,
+ mSessionId, mTransferType, mFlags);
+ } else {
+ ALOGE("Test application is not handling transfer type %s",
+ AudioRecord::convertTransferToText(mTransferType));
+ return NO_INIT;
+ }
+ mRecord->setCallerName(packageName);
+ status = mRecord->initCheck();
+ if (NO_ERROR == status) mState = REC_READY;
+ if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+ mFrameCount = mRecord->frameCount();
+ mNotificationFrames = mRecord->getNotificationPeriodInFrames();
+ mMaxBytesPerCallback = mNotificationFrames * samplesPerFrame * bytesPerSample;
+ }
+ return status;
+}
+
+sp<AudioRecord> AudioCapture::getAudioRecordHandle() {
+ return (REC_NO_INIT == mState) ? nullptr : mRecord;
+}
+
+status_t AudioCapture::start(AudioSystem::sync_event_t event, audio_session_t triggerSession) {
+ status_t status;
+ if (REC_READY != mState) {
+ return INVALID_OPERATION;
+ } else {
+ status = mRecord->start(event, triggerSession);
+ if (OK == status) {
+ mState = REC_STARTED;
+ LOG_FATAL_IF(false != mRecord->stopped());
+ }
+ }
+ return status;
+}
+
+status_t AudioCapture::stop() {
+ status_t status = OK;
+ mStopRecording = true;
+ if (mState != REC_STOPPED) {
+ mRecord->stopAndJoinCallbacks();
+ mState = REC_STOPPED;
+ LOG_FATAL_IF(true != mRecord->stopped());
+ }
+ return status;
+}
+
+status_t AudioCapture::obtainBuffer(RawBuffer& buffer) {
+ if (REC_STARTED != mState && REC_STOPPED != mState) return INVALID_OPERATION;
+ int retry = 25;
+ AudioRecord::Buffer recordBuffer;
+ recordBuffer.frameCount = mNotificationFrames;
+ size_t nonContig = 0;
+ status_t status = mRecord->obtainBuffer(&recordBuffer, retry, &nonContig);
+ if (OK == status) {
+ const int64_t timestampUs =
+ ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+ mRecord->getSampleRate();
+ RawBuffer buff{-1, timestampUs, static_cast<int32_t>(recordBuffer.size())};
+ memcpy(buff.mData.get(), recordBuffer.data(), recordBuffer.size());
+ buffer = std::move(buff);
+ mNumFramesReceived += recordBuffer.size() / mRecord->frameSize();
+ mRecord->releaseBuffer(&recordBuffer);
+ if (mNumFramesReceived > mNumFramesToRecord) {
+ stop();
+ }
+ } else if (status == WOULD_BLOCK) {
+ if (mStopRecording)
+ return WOULD_BLOCK;
+ else
+ return TIMED_OUT;
+ }
+ return OK;
+}
+
+status_t AudioCapture::obtainBufferCb(RawBuffer& buffer) {
+ if (REC_STARTED != mState) return INVALID_OPERATION;
+ int retry = 10;
+ std::unique_lock<std::mutex> lock{mMutex};
+ while (mBuffersReceived.empty() && !mStopRecording && retry > 0) {
+ mCondition.wait_for(lock, std::chrono::milliseconds(100));
+ retry--;
+ }
+ if (!mBuffersReceived.empty()) {
+ auto it = mBuffersReceived.begin();
+ buffer = std::move(*it);
+ mBuffersReceived.erase(it);
+ } else {
+ if (retry == 0) return TIMED_OUT;
+ if (mStopRecording)
+ return WOULD_BLOCK;
+ else
+ return UNKNOWN_ERROR;
+ }
+ return OK;
+}
+
+status_t AudioCapture::audioProcess() {
+ RawBuffer buffer;
+ while (true) {
+ status_t status;
+ if (mTransferType == AudioRecord::TRANSFER_CALLBACK)
+ status = obtainBufferCb(buffer);
+ else
+ status = obtainBuffer(buffer);
+ switch (status) {
+ case OK:
+ if (mOutFileFd > 0) {
+ const char* ptr =
+ static_cast<const char*>(static_cast<void*>(buffer.mData.get()));
+ write(mOutFileFd, ptr, buffer.mCapacity);
+ }
+ break;
+ case WOULD_BLOCK:
+ return OK;
+ case TIMED_OUT: // "recorder application timed out from receiving buffers"
+ case NO_INIT: // "recorder not initialized"
+ case INVALID_OPERATION: // "recorder not started"
+ case UNKNOWN_ERROR: // "Unknown error"
+ default:
+ return status;
+ }
+ }
+}
+
+status_t listAudioPorts(std::vector<audio_port_v7>& portsVec) {
+ int attempts = 5;
+ status_t status;
+ unsigned int generation1, generation;
+ unsigned int numPorts = 0;
+ do {
+ if (attempts-- < 0) {
+ status = TIMED_OUT;
+ break;
+ }
+ status = AudioSystem::listAudioPorts(AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_NONE, &numPorts,
+ nullptr, &generation1);
+ if (status != NO_ERROR) {
+ ALOGE("AudioSystem::listAudioPorts returned error %d", status);
+ break;
+ }
+ portsVec.resize(numPorts);
+ status = AudioSystem::listAudioPorts(AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_NONE, &numPorts,
+ portsVec.data(), &generation);
+ } while (generation1 != generation && status == NO_ERROR);
+ if (status != NO_ERROR) {
+ numPorts = 0;
+ portsVec.clear();
+ }
+ return status;
+}
+
+status_t getPortById(const audio_port_handle_t portId, audio_port_v7& port) {
+ std::vector<struct audio_port_v7> ports;
+ status_t status = listAudioPorts(ports);
+ if (status != OK) return status;
+ for (auto i = 0; i < ports.size(); i++) {
+ if (ports[i].id == portId) {
+ port = ports[i];
+ return OK;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t getPortByAttributes(audio_port_role_t role, audio_port_type_t type,
+ audio_devices_t deviceType, audio_port_v7& port) {
+ std::vector<struct audio_port_v7> ports;
+ status_t status = listAudioPorts(ports);
+ if (status != OK) return status;
+ for (auto i = 0; i < ports.size(); i++) {
+ if (ports[i].role == role && ports[i].type == type &&
+ ports[i].ext.device.type == deviceType) {
+ port = ports[i];
+ return OK;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t listAudioPatches(std::vector<struct audio_patch>& patchesVec) {
+ int attempts = 5;
+ status_t status;
+ unsigned int generation1, generation;
+ unsigned int numPatches = 0;
+ do {
+ if (attempts-- < 0) {
+ status = TIMED_OUT;
+ break;
+ }
+ status = AudioSystem::listAudioPatches(&numPatches, nullptr, &generation1);
+ if (status != NO_ERROR) {
+ ALOGE("AudioSystem::listAudioPatches returned error %d", status);
+ break;
+ }
+ patchesVec.resize(numPatches);
+ status = AudioSystem::listAudioPatches(&numPatches, patchesVec.data(), &generation);
+ } while (generation1 != generation && status == NO_ERROR);
+ if (status != NO_ERROR) {
+ numPatches = 0;
+ patchesVec.clear();
+ }
+ return status;
+}
+
+status_t getPatchForOutputMix(audio_io_handle_t audioIo, audio_patch& patch) {
+ std::vector<struct audio_patch> patches;
+ status_t status = listAudioPatches(patches);
+ if (status != OK) return status;
+
+ for (auto i = 0; i < patches.size(); i++) {
+ for (auto j = 0; j < patches[i].num_sources; j++) {
+ if (patches[i].sources[j].type == AUDIO_PORT_TYPE_MIX &&
+ patches[i].sources[j].ext.mix.handle == audioIo) {
+ patch = patches[i];
+ return OK;
+ }
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t getPatchForInputMix(audio_io_handle_t audioIo, audio_patch& patch) {
+ std::vector<struct audio_patch> patches;
+ status_t status = listAudioPatches(patches);
+ if (status != OK) return status;
+
+ for (auto i = 0; i < patches.size(); i++) {
+ for (auto j = 0; j < patches[i].num_sinks; j++) {
+ if (patches[i].sinks[j].type == AUDIO_PORT_TYPE_MIX &&
+ patches[i].sinks[j].ext.mix.handle == audioIo) {
+ patch = patches[i];
+ return OK;
+ }
+ }
+ }
+ return BAD_VALUE;
+}
+
+bool patchContainsOutputDevice(audio_port_handle_t deviceId, audio_patch patch) {
+ for (auto j = 0; j < patch.num_sinks; j++) {
+ if (patch.sinks[j].type == AUDIO_PORT_TYPE_DEVICE && patch.sinks[j].id == deviceId) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool patchContainsInputDevice(audio_port_handle_t deviceId, audio_patch patch) {
+ for (auto j = 0; j < patch.num_sources; j++) {
+ if (patch.sources[j].type == AUDIO_PORT_TYPE_DEVICE && patch.sources[j].id == deviceId) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool checkPatchPlayback(audio_io_handle_t audioIo, audio_port_handle_t deviceId) {
+ struct audio_patch patch;
+ if (getPatchForOutputMix(audioIo, patch) == OK) {
+ return patchContainsOutputDevice(deviceId, patch);
+ }
+ return false;
+}
+
+bool checkPatchCapture(audio_io_handle_t audioIo, audio_port_handle_t deviceId) {
+ struct audio_patch patch;
+ if (getPatchForInputMix(audioIo, patch) == OK) {
+ return patchContainsInputDevice(deviceId, patch);
+ }
+ return false;
+}
+
+std::string dumpPortConfig(const audio_port_config& port) {
+ std::ostringstream result;
+ std::string deviceInfo;
+ if (port.type == AUDIO_PORT_TYPE_DEVICE) {
+ if (port.ext.device.type & AUDIO_DEVICE_BIT_IN) {
+ InputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+ } else {
+ OutputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+ }
+ deviceInfo += std::string(", address = ") + port.ext.device.address;
+ }
+ result << "audio_port_handle_t = " << port.id << ", "
+ << "Role = " << (port.role == AUDIO_PORT_ROLE_SOURCE ? "source" : "sink") << ", "
+ << "Type = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? "device" : "mix") << ", "
+ << "deviceInfo = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? deviceInfo : "") << ", "
+ << "config_mask = 0x" << std::hex << port.config_mask << std::dec << ", ";
+ if (port.config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ result << "sample rate = " << port.sample_rate << ", ";
+ }
+ if (port.config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ result << "channel mask = " << port.channel_mask << ", ";
+ }
+ if (port.config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ result << "format = " << port.format << ", ";
+ }
+ result << "input flags = " << port.flags.input << ", ";
+ result << "output flags = " << port.flags.output << ", ";
+ result << "mix io handle = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? 0 : port.ext.mix.handle)
+ << "\n";
+ return result.str();
+}
+
+std::string dumpPatch(const audio_patch& patch) {
+ std::ostringstream result;
+ result << "----------------- Dumping Patch ------------ \n";
+ result << "Patch Handle: " << patch.id << ", sources: " << patch.num_sources
+ << ", sink: " << patch.num_sinks << "\n";
+ audio_port_v7 port;
+ for (uint32_t i = 0; i < patch.num_sources; i++) {
+ result << "----------------- Dumping Source Port Config @ index " << i
+ << " ------------ \n";
+ result << dumpPortConfig(patch.sources[i]);
+ result << "----------------- Dumping Source Port for id " << patch.sources[i].id
+ << " ------------ \n";
+ getPortById(patch.sources[i].id, port);
+ result << dumpPort(port);
+ }
+ for (uint32_t i = 0; i < patch.num_sinks; i++) {
+ result << "----------------- Dumping Sink Port Config @ index " << i << " ------------ \n";
+ result << dumpPortConfig(patch.sinks[i]);
+ result << "----------------- Dumping Sink Port for id " << patch.sinks[i].id
+ << " ------------ \n";
+ getPortById(patch.sinks[i].id, port);
+ result << dumpPort(port);
+ }
+ return result.str();
+}
+
+std::string dumpPort(const audio_port_v7& port) {
+ std::ostringstream result;
+ std::string deviceInfo;
+ if (port.type == AUDIO_PORT_TYPE_DEVICE) {
+ if (port.ext.device.type & AUDIO_DEVICE_BIT_IN) {
+ InputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+ } else {
+ OutputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+ }
+ deviceInfo += std::string(", address = ") + port.ext.device.address;
+ }
+ result << "audio_port_handle_t = " << port.id << ", "
+ << "Role = " << (port.role == AUDIO_PORT_ROLE_SOURCE ? "source" : "sink") << ", "
+ << "Type = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? "device" : "mix") << ", "
+ << "deviceInfo = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? deviceInfo : "") << ", "
+ << "Name = " << port.name << ", "
+ << "num profiles = " << port.num_audio_profiles << ", "
+ << "mix io handle = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? 0 : port.ext.mix.handle)
+ << ", ";
+ for (int i = 0; i < port.num_audio_profiles; i++) {
+ result << "AudioProfile = " << i << " {";
+ result << "format = " << port.audio_profiles[i].format << ", ";
+ result << "samplerates = ";
+ for (int j = 0; j < port.audio_profiles[i].num_sample_rates; j++) {
+ result << port.audio_profiles[i].sample_rates[j] << ", ";
+ }
+ result << "channelmasks = ";
+ for (int j = 0; j < port.audio_profiles[i].num_channel_masks; j++) {
+ result << "0x" << std::hex << port.audio_profiles[i].channel_masks[j] << std::dec
+ << ", ";
+ }
+ result << "} ";
+ }
+ result << dumpPortConfig(port.active_config);
+ return result.str();
+}
diff --git a/media/libaudioclient/tests/audio_test_utils.h b/media/libaudioclient/tests/audio_test_utils.h
new file mode 100644
index 0000000..526d5c4
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_utils.h
@@ -0,0 +1,188 @@
+/*
+ * 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.
+ */
+
+#ifndef AUDIO_TEST_UTILS_H_
+#define AUDIO_TEST_UTILS_H_
+
+#include <sys/stat.h>
+#include <unistd.h>
+#include <atomic>
+#include <chrono>
+#include <cinttypes>
+#include <deque>
+#include <memory>
+#include <mutex>
+#include <thread>
+
+#include <binder/MemoryDealer.h>
+#include <media/AidlConversion.h>
+#include <media/AudioRecord.h>
+#include <media/AudioTrack.h>
+
+#define RECORD_TO_FILE 0
+
+using namespace android;
+
+void CreateRandomFile(int& fd);
+status_t listAudioPorts(std::vector<audio_port_v7>& portsVec);
+status_t listAudioPatches(std::vector<struct audio_patch>& patchesVec);
+status_t getPortByAttributes(audio_port_role_t role, audio_port_type_t type,
+ audio_devices_t deviceType, audio_port_v7& port);
+status_t getPatchForOutputMix(audio_io_handle_t audioIo, audio_patch& patch);
+status_t getPatchForInputMix(audio_io_handle_t audioIo, audio_patch& patch);
+bool patchContainsOutputDevice(audio_port_handle_t deviceId, audio_patch patch);
+bool patchContainsInputDevice(audio_port_handle_t deviceId, audio_patch patch);
+bool checkPatchPlayback(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+bool checkPatchCapture(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+std::string dumpPort(const audio_port_v7& port);
+std::string dumpPortConfig(const audio_port_config& port);
+std::string dumpPatch(const audio_patch& patch);
+
+class OnAudioDeviceUpdateNotifier : public AudioSystem::AudioDeviceCallback {
+ public:
+ audio_io_handle_t mAudioIo = AUDIO_IO_HANDLE_NONE;
+ audio_port_handle_t mDeviceId = AUDIO_PORT_HANDLE_NONE;
+ std::mutex mMutex;
+ std::condition_variable mCondition;
+
+ void onAudioDeviceUpdate(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+ status_t waitForAudioDeviceCb();
+};
+
+// Simple AudioPlayback class.
+class AudioPlayback : public AudioTrack::IAudioTrackCallback {
+ friend sp<AudioPlayback>;
+ AudioPlayback(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask,
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ audio_session_t sessionId = AUDIO_SESSION_NONE,
+ AudioTrack::transfer_type transferType = AudioTrack::TRANSFER_SHARED,
+ audio_attributes_t* attributes = nullptr, audio_offload_info_t* info = nullptr);
+
+ public:
+ status_t loadResource(const char* name);
+ status_t create();
+ sp<AudioTrack> getAudioTrackHandle();
+ status_t start();
+ status_t waitForConsumption(bool testSeek = false);
+ status_t fillBuffer();
+ status_t onProcess(bool testSeek = false);
+ virtual void onBufferEnd() override;
+ void stop();
+
+ bool mStopPlaying;
+ std::mutex mMutex;
+ std::condition_variable mCondition;
+
+ enum State {
+ PLAY_NO_INIT,
+ PLAY_READY,
+ PLAY_STARTED,
+ PLAY_STOPPED,
+ };
+
+ private:
+ ~AudioPlayback();
+ const uint32_t mSampleRate;
+ const audio_format_t mFormat;
+ const audio_channel_mask_t mChannelMask;
+ const audio_output_flags_t mFlags;
+ const audio_session_t mSessionId;
+ const AudioTrack::transfer_type mTransferType;
+ const audio_attributes_t* mAttributes;
+ const audio_offload_info_t* mOffloadInfo;
+
+ size_t mBytesUsedSoFar;
+ State mState;
+ size_t mMemCapacity;
+ sp<MemoryDealer> mMemoryDealer;
+ sp<IMemory> mMemory;
+
+ sp<AudioTrack> mTrack;
+};
+
+// hold pcm data sent by AudioRecord
+class RawBuffer {
+ public:
+ RawBuffer(int64_t ptsPipeline = -1, int64_t ptsManual = -1, int32_t capacity = 0);
+
+ std::unique_ptr<uint8_t[]> mData;
+ int64_t mPtsPipeline;
+ int64_t mPtsManual;
+ int32_t mCapacity;
+};
+
+// Simple AudioCapture
+class AudioCapture : public AudioRecord::IAudioRecordCallback {
+ public:
+ AudioCapture(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
+ audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
+ AudioRecord::transfer_type transferType = AudioRecord::TRANSFER_CALLBACK);
+ ~AudioCapture();
+ size_t onMoreData(const AudioRecord::Buffer& buffer) override;
+ void onOverrun() override;
+ void onMarker(uint32_t markerPosition) override;
+ void onNewPos(uint32_t newPos) override;
+ void onNewIAudioRecord() override;
+ status_t create();
+ sp<AudioRecord> getAudioRecordHandle();
+ status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+ audio_session_t triggerSession = AUDIO_SESSION_NONE);
+ status_t obtainBufferCb(RawBuffer& buffer);
+ status_t obtainBuffer(RawBuffer& buffer);
+ status_t audioProcess();
+ status_t stop();
+
+ uint32_t mFrameCount;
+ uint32_t mNotificationFrames;
+ int64_t mNumFramesToRecord;
+ int64_t mNumFramesReceived;
+ int64_t mNumFramesLost;
+ uint32_t mMarkerPosition;
+ uint32_t mMarkerPeriod;
+ uint32_t mReceivedCbMarkerAtPosition;
+ uint32_t mReceivedCbMarkerCount;
+ bool mBufferOverrun;
+
+ enum State {
+ REC_NO_INIT,
+ REC_READY,
+ REC_STARTED,
+ REC_STOPPED,
+ };
+
+ private:
+ const audio_source_t mInputSource;
+ const uint32_t mSampleRate;
+ const audio_format_t mFormat;
+ const audio_channel_mask_t mChannelMask;
+ const audio_input_flags_t mFlags;
+ const audio_session_t mSessionId;
+ const AudioRecord::transfer_type mTransferType;
+
+ size_t mMaxBytesPerCallback = 2048;
+ sp<AudioRecord> mRecord;
+ State mState;
+ bool mStopRecording;
+ int mOutFileFd = -1;
+
+ std::mutex mMutex;
+ std::condition_variable mCondition;
+ std::deque<RawBuffer> mBuffersReceived;
+};
+
+#endif // AUDIO_TEST_UTILS_H_
diff --git a/media/libaudioclient/tests/audioclient_serialization_tests.cpp b/media/libaudioclient/tests/audioclient_serialization_tests.cpp
new file mode 100644
index 0000000..93baefd6
--- /dev/null
+++ b/media/libaudioclient/tests/audioclient_serialization_tests.cpp
@@ -0,0 +1,310 @@
+/*
+ * 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 "AudioClientSerializationUnitTests"
+
+#include <cstdint>
+#include <cstdlib>
+#include <ctime>
+
+#include <gtest/gtest.h>
+
+#include <android_audio_policy_configuration_V7_0-enums.h>
+#include <xsdc/XsdcSupport.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+namespace xsd {
+using namespace ::android::audio::policy::configuration::V7_0;
+}
+
+template <typename T, typename X, typename FUNC>
+std::vector<T> getFlags(const xsdc_enum_range<X>& range, const FUNC& func,
+ const std::string& findString = {}) {
+ std::vector<T> vec;
+ for (const auto& xsdEnumVal : range) {
+ T enumVal;
+ std::string enumString = toString(xsdEnumVal);
+ if (enumString.find(findString) != std::string::npos &&
+ func(enumString.c_str(), &enumVal)) {
+ vec.push_back(enumVal);
+ }
+ }
+ return vec;
+}
+
+static const std::vector<audio_usage_t> kUsages =
+ getFlags<audio_usage_t, xsd::AudioUsage, decltype(audio_usage_from_string)>(
+ xsdc_enum_range<xsd::AudioUsage>{}, audio_usage_from_string);
+
+static const std::vector<audio_content_type_t> kContentType =
+ getFlags<audio_content_type_t, xsd::AudioContentType,
+ decltype(audio_content_type_from_string)>(xsdc_enum_range<xsd::AudioContentType>{},
+ audio_content_type_from_string);
+
+static const std::vector<audio_source_t> kInputSources =
+ getFlags<audio_source_t, xsd::AudioSource, decltype(audio_source_from_string)>(
+ xsdc_enum_range<xsd::AudioSource>{}, audio_source_from_string);
+
+static const std::vector<audio_stream_type_t> kStreamtypes =
+ getFlags<audio_stream_type_t, xsd::AudioStreamType,
+ decltype(audio_stream_type_from_string)>(xsdc_enum_range<xsd::AudioStreamType>{},
+ audio_stream_type_from_string);
+
+static const std::vector<uint32_t> kMixMatchRules = {
+ RULE_MATCH_ATTRIBUTE_USAGE,
+ RULE_EXCLUDE_ATTRIBUTE_USAGE,
+ RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET,
+ RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET,
+ RULE_MATCH_UID,
+ RULE_EXCLUDE_UID,
+ RULE_MATCH_USERID,
+ RULE_EXCLUDE_USERID,
+};
+
+// Generates a random string.
+std::string CreateRandomString(size_t n) {
+ std::string data =
+ "abcdefghijklmnopqrstuvwxyz"
+ "ABCDEFGHIJKLMNOPQRSTUVWXYZ"
+ "0123456789";
+ srand(static_cast<unsigned int>(time(0)));
+ std::string s(n, ' ');
+ for (size_t i = 0; i < n; ++i) {
+ s[i] = data[rand() % data.size()];
+ }
+ return s;
+}
+
+class FillAudioAttributes {
+ public:
+ void fillAudioAttributes(audio_attributes_t& attr);
+
+ unsigned int mSeed;
+};
+
+void FillAudioAttributes::fillAudioAttributes(audio_attributes_t& attr) {
+ attr.content_type = kContentType[rand() % kContentType.size()];
+ attr.usage = kUsages[rand() % kUsages.size()];
+ attr.source = kInputSources[rand() % kInputSources.size()];
+ // attr.flags -> [0, (1 << (CAPTURE_PRIVATE + 1) - 1)]
+ attr.flags = static_cast<audio_flags_mask_t>(rand() & 0x3fff);
+ sprintf(attr.tags, "%s",
+ CreateRandomString((int)rand() % (AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1)).c_str());
+}
+
+class SerializationTest : public FillAudioAttributes, public ::testing::Test {
+ void SetUp() override {
+ mSeed = static_cast<unsigned int>(time(0));
+ srand(mSeed);
+ }
+};
+
+// UNIT TESTS
+TEST_F(SerializationTest, AudioProductStrategyBinderization) {
+ for (int j = 0; j < 512; j++) {
+ const std::string name{"Test APSBinderization for seed::" + std::to_string(mSeed)};
+ std::vector<AudioAttributes> audioattributesvector;
+ for (auto i = 0; i < 16; i++) {
+ audio_attributes_t attributes;
+ fillAudioAttributes(attributes);
+ AudioAttributes audioattributes{static_cast<volume_group_t>(rand()),
+ kStreamtypes[rand() % kStreamtypes.size()], attributes};
+ audioattributesvector.push_back(audioattributes);
+ }
+ product_strategy_t psId = static_cast<product_strategy_t>(rand());
+ AudioProductStrategy aps{name, audioattributesvector, psId};
+
+ Parcel p;
+ EXPECT_EQ(NO_ERROR, aps.writeToParcel(&p)) << name;
+
+ AudioProductStrategy apsCopy;
+ p.setDataPosition(0);
+ EXPECT_EQ(NO_ERROR, apsCopy.readFromParcel(&p)) << name;
+ EXPECT_EQ(apsCopy.getName(), name) << name;
+ EXPECT_EQ(apsCopy.getId(), psId) << name;
+ auto avec = apsCopy.getAudioAttributes();
+ EXPECT_EQ(avec.size(), audioattributesvector.size()) << name;
+ for (int i = 0; i < audioattributesvector.size(); i++) {
+ EXPECT_EQ(avec[i].getGroupId(), audioattributesvector[i].getGroupId()) << name;
+ EXPECT_EQ(avec[i].getStreamType(), audioattributesvector[i].getStreamType()) << name;
+ EXPECT_TRUE(avec[i].getAttributes() == audioattributesvector[i].getAttributes())
+ << name;
+ }
+ }
+}
+
+TEST_F(SerializationTest, AudioVolumeGroupBinderization) {
+ for (int j = 0; j < 512; j++) {
+ const std::string name{"Test AVGBinderization for seed::" + std::to_string(mSeed)};
+ volume_group_t groupId = static_cast<volume_group_t>(rand());
+ std::vector<audio_attributes_t> attributesvector;
+ for (auto i = 0; i < 16; i++) {
+ audio_attributes_t attributes;
+ fillAudioAttributes(attributes);
+ attributesvector.push_back(attributes);
+ }
+ std::vector<audio_stream_type_t> streamsvector;
+ for (auto i = 0; i < 8; i++) {
+ streamsvector.push_back(kStreamtypes[rand() % kStreamtypes.size()]);
+ }
+ AudioVolumeGroup avg{name, groupId, attributesvector, streamsvector};
+
+ Parcel p;
+ EXPECT_EQ(NO_ERROR, avg.writeToParcel(&p));
+
+ AudioVolumeGroup avgCopy;
+ p.setDataPosition(0);
+ EXPECT_EQ(NO_ERROR, avgCopy.readFromParcel(&p)) << name;
+ EXPECT_EQ(avgCopy.getName(), name) << name;
+ EXPECT_EQ(avgCopy.getId(), groupId) << name;
+ auto avec = avgCopy.getAudioAttributes();
+ EXPECT_EQ(avec.size(), attributesvector.size()) << name;
+ for (int i = 0; i < avec.size(); i++) {
+ EXPECT_TRUE(avec[i] == attributesvector[i]) << name;
+ }
+ StreamTypeVector svec = avgCopy.getStreamTypes();
+ EXPECT_EQ(svec.size(), streamsvector.size()) << name;
+ for (int i = 0; i < svec.size(); i++) {
+ EXPECT_EQ(svec[i], streamsvector[i]) << name;
+ }
+ }
+}
+
+TEST_F(SerializationTest, AudioMixBinderization) {
+ for (int j = 0; j < 512; j++) {
+ const std::string msg{"Test AMBinderization for seed::" + std::to_string(mSeed)};
+ Vector<AudioMixMatchCriterion> criteria;
+ for (int i = 0; i < 16; i++) {
+ AudioMixMatchCriterion ammc{kUsages[rand() % kUsages.size()],
+ kInputSources[rand() % kInputSources.size()],
+ kMixMatchRules[rand() % kMixMatchRules.size()]};
+ criteria.add(ammc);
+ }
+ audio_config_t config{};
+ config.sample_rate = 48000;
+ config.channel_mask = AUDIO_CHANNEL_IN_MONO;
+ config.format = AUDIO_FORMAT_PCM_16_BIT;
+ config.offload_info = AUDIO_INFO_INITIALIZER;
+ config.frame_count = 4800;
+ AudioMix am{criteria,
+ static_cast<uint32_t>(rand()),
+ config,
+ static_cast<uint32_t>(rand()),
+ String8(msg.c_str()),
+ static_cast<uint32_t>(rand())};
+
+ Parcel p;
+ EXPECT_EQ(NO_ERROR, am.writeToParcel(&p)) << msg;
+
+ AudioMix amCopy;
+ p.setDataPosition(0);
+ EXPECT_EQ(NO_ERROR, amCopy.readFromParcel(&p)) << msg;
+ EXPECT_EQ(amCopy.mMixType, am.mMixType) << msg;
+ EXPECT_EQ(amCopy.mFormat.sample_rate, am.mFormat.sample_rate) << msg;
+ EXPECT_EQ(amCopy.mFormat.channel_mask, am.mFormat.channel_mask) << msg;
+ EXPECT_EQ(amCopy.mFormat.format, am.mFormat.format) << msg;
+ EXPECT_EQ(amCopy.mRouteFlags, am.mRouteFlags) << msg;
+ EXPECT_EQ(amCopy.mDeviceAddress, am.mDeviceAddress) << msg;
+ EXPECT_EQ(amCopy.mCbFlags, am.mCbFlags) << msg;
+ EXPECT_EQ(amCopy.mCriteria.size(), am.mCriteria.size()) << msg;
+ for (auto i = 0; i < amCopy.mCriteria.size(); i++) {
+ EXPECT_EQ(amCopy.mCriteria[i].mRule, am.mCriteria[i].mRule) << msg;
+ EXPECT_EQ(amCopy.mCriteria[i].mValue.mUserId, am.mCriteria[i].mValue.mUserId) << msg;
+ }
+ }
+}
+
+using MMCTestParams = std::tuple<audio_usage_t, audio_source_t, uint32_t>;
+
+class MMCParameterizedTest : public FillAudioAttributes,
+ public ::testing::TestWithParam<MMCTestParams> {
+ public:
+ MMCParameterizedTest()
+ : mAudioUsage(std::get<0>(GetParam())),
+ mAudioSource(std::get<1>(GetParam())),
+ mAudioMixMatchRules(std::get<2>(GetParam())){};
+
+ const audio_usage_t mAudioUsage;
+ const audio_source_t mAudioSource;
+ const uint32_t mAudioMixMatchRules;
+
+ void SetUp() override {
+ mSeed = static_cast<unsigned int>(time(0));
+ srand(mSeed);
+ }
+};
+
+TEST_P(MMCParameterizedTest, AudioMixMatchCriterionBinderization) {
+ const std::string msg{"Test AMMCBinderization for seed::" + std::to_string(mSeed)};
+ AudioMixMatchCriterion ammc{mAudioUsage, mAudioSource, mAudioMixMatchRules};
+
+ Parcel p;
+ EXPECT_EQ(NO_ERROR, ammc.writeToParcel(&p)) << msg;
+
+ AudioMixMatchCriterion ammcCopy;
+ p.setDataPosition(0);
+ EXPECT_EQ(NO_ERROR, ammcCopy.readFromParcel(&p)) << msg;
+ EXPECT_EQ(ammcCopy.mRule, ammc.mRule) << msg;
+ EXPECT_EQ(ammcCopy.mValue.mUserId, ammc.mValue.mUserId) << msg;
+}
+
+// audioUsage, audioSource, audioMixMatchRules
+INSTANTIATE_TEST_SUITE_P(SerializationParameterizedTests, MMCParameterizedTest,
+ ::testing::Combine(testing::ValuesIn(kUsages),
+ testing::ValuesIn(kInputSources),
+ testing::ValuesIn(kMixMatchRules)));
+
+using AudioAttributesTestParams = std::tuple<audio_stream_type_t>;
+
+class AudioAttributesParameterizedTest
+ : public FillAudioAttributes,
+ public ::testing::TestWithParam<AudioAttributesTestParams> {
+ public:
+ AudioAttributesParameterizedTest() : mAudioStream(std::get<0>(GetParam())){};
+
+ const audio_stream_type_t mAudioStream;
+
+ void SetUp() override {
+ mSeed = static_cast<unsigned int>(time(0));
+ srand(mSeed);
+ }
+};
+
+TEST_P(AudioAttributesParameterizedTest, AudioAttributesBinderization) {
+ const std::string msg{"Test AABinderization for seed::" + std::to_string(mSeed)};
+ volume_group_t groupId = static_cast<volume_group_t>(rand());
+ audio_stream_type_t stream = mAudioStream;
+ audio_attributes_t attributes;
+ fillAudioAttributes(attributes);
+ AudioAttributes audioattributes{groupId, stream, attributes};
+
+ Parcel p;
+ EXPECT_EQ(NO_ERROR, audioattributes.writeToParcel(&p)) << msg;
+
+ AudioAttributes audioattributesCopy;
+ p.setDataPosition(0);
+ EXPECT_EQ(NO_ERROR, audioattributesCopy.readFromParcel(&p)) << msg;
+ EXPECT_EQ(audioattributesCopy.getGroupId(), audioattributes.getGroupId()) << msg;
+ EXPECT_EQ(audioattributesCopy.getStreamType(), audioattributes.getStreamType()) << msg;
+ EXPECT_TRUE(audioattributesCopy.getAttributes() == attributes) << msg;
+}
+
+// audioStream
+INSTANTIATE_TEST_SUITE_P(SerializationParameterizedTests, AudioAttributesParameterizedTest,
+ ::testing::Combine(testing::ValuesIn(kStreamtypes)));
diff --git a/media/libaudioclient/tests/audioeffect_tests.cpp b/media/libaudioclient/tests/audioeffect_tests.cpp
new file mode 100644
index 0000000..93fe306
--- /dev/null
+++ b/media/libaudioclient/tests/audioeffect_tests.cpp
@@ -0,0 +1,335 @@
+/*
+ * 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 "AudioEffectUnitTests"
+
+#include <gtest/gtest.h>
+#include <media/AudioEffect.h>
+#include <system/audio_effects/effect_visualizer.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+static constexpr int kDefaultInputEffectPriority = -1;
+static constexpr int kDefaultOutputEffectPriority = 0;
+
+static const char* gPackageName = "AudioEffectTest";
+
+bool isEffectExistsOnAudioSession(const effect_uuid_t* type, int priority,
+ audio_session_t sessionId) {
+ std::string packageName{gPackageName};
+ AttributionSourceState attributionSource;
+ attributionSource.packageName = packageName;
+ attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+ attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+ attributionSource.token = sp<BBinder>::make();
+ sp<AudioEffect> effect = new AudioEffect(attributionSource);
+ effect->set(type, nullptr /* uid */, priority, nullptr /* callback */, sessionId);
+ return effect->initCheck() == ALREADY_EXISTS;
+}
+
+bool isEffectDefaultOnRecord(const effect_uuid_t* type, const sp<AudioRecord>& audioRecord) {
+ effect_descriptor_t descriptors[AudioEffect::kMaxPreProcessing];
+ uint32_t numEffects = AudioEffect::kMaxPreProcessing;
+ status_t ret = AudioEffect::queryDefaultPreProcessing(audioRecord->getSessionId(), descriptors,
+ &numEffects);
+ if (ret != OK) {
+ return false;
+ }
+ for (int i = 0; i < numEffects; i++) {
+ if (memcmp(&descriptors[i].type, type, sizeof(effect_uuid_t)) == 0) {
+ return true;
+ }
+ }
+ return false;
+}
+
+void listEffectsAvailable(std::vector<effect_descriptor_t>& descriptors) {
+ uint32_t numEffects = 0;
+ if (NO_ERROR == AudioEffect::queryNumberEffects(&numEffects)) {
+ for (auto i = 0; i < numEffects; i++) {
+ effect_descriptor_t des;
+ if (NO_ERROR == AudioEffect::queryEffect(i, &des)) descriptors.push_back(des);
+ }
+ }
+}
+
+bool isPreprocessing(effect_descriptor_t& descriptor) {
+ return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC);
+}
+
+bool isInsert(effect_descriptor_t& descriptor) {
+ return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT);
+}
+
+bool isAux(effect_descriptor_t& descriptor) {
+ return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY);
+}
+
+bool isFastCompatible(effect_descriptor_t& descriptor) {
+ return !(((descriptor.flags & EFFECT_FLAG_HW_ACC_MASK) == 0) &&
+ ((descriptor.flags & EFFECT_FLAG_NO_PROCESS) == 0));
+}
+
+// UNIT TESTS
+TEST(AudioEffectTest, getEffectDescriptor) {
+ effect_uuid_t randomType = {
+ 0x81781c08, 0x93dd, 0x11ec, 0xb909, {0x02, 0x42, 0xac, 0x12, 0x00, 0x02}};
+ effect_uuid_t randomUuid = {
+ 0x653730e1, 0x1be1, 0x438e, 0xa35a, {0xfc, 0x9b, 0xa1, 0x2a, 0x5e, 0xc9}};
+ effect_uuid_t empty = EFFECT_UUID_INITIALIZER;
+
+ effect_descriptor_t descriptor;
+ EXPECT_EQ(NAME_NOT_FOUND, AudioEffect::getEffectDescriptor(&randomUuid, &randomType,
+ EFFECT_FLAG_TYPE_MASK, &descriptor));
+
+ std::vector<effect_descriptor_t> descriptors;
+ listEffectsAvailable(descriptors);
+
+ for (auto i = 0; i < descriptors.size(); i++) {
+ EXPECT_EQ(NO_ERROR,
+ AudioEffect::getEffectDescriptor(&descriptors[i].uuid, &descriptors[i].type,
+ EFFECT_FLAG_TYPE_MASK, &descriptor));
+ EXPECT_EQ(0, memcmp(&descriptor, &descriptors[i], sizeof(effect_uuid_t)));
+ }
+ // negative tests
+ if (descriptors.size() > 0) {
+ EXPECT_EQ(BAD_VALUE,
+ AudioEffect::getEffectDescriptor(&descriptors[0].uuid, &descriptors[0].type,
+ EFFECT_FLAG_TYPE_MASK, nullptr));
+ }
+ EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(nullptr, nullptr,
+ EFFECT_FLAG_TYPE_PRE_PROC, &descriptor));
+ EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(&empty, &randomType,
+ EFFECT_FLAG_TYPE_MASK, nullptr));
+ EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(nullptr, &randomType,
+ EFFECT_FLAG_TYPE_POST_PROC, &descriptor));
+ EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(&randomUuid, nullptr,
+ EFFECT_FLAG_TYPE_INSERT, &descriptor));
+}
+
+TEST(AudioEffectTest, DISABLED_GetSetParameterForEffect) {
+ std::string packageName{gPackageName};
+ AttributionSourceState attributionSource;
+ attributionSource.packageName = packageName;
+ attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+ attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+ attributionSource.token = sp<BBinder>::make();
+ sp<AudioEffect> visualizer = new AudioEffect(attributionSource);
+ ASSERT_NE(visualizer, nullptr) << "effect not created";
+ visualizer->set(SL_IID_VISUALIZATION);
+ status_t status = visualizer->initCheck();
+ ASSERT_TRUE(status == NO_ERROR || status == ALREADY_EXISTS) << "Init check error";
+ ASSERT_EQ(NO_ERROR, visualizer->setEnabled(true)) << "visualizer not enabled";
+
+ uint32_t buf32[3][sizeof(effect_param_t) / sizeof(uint32_t) + 2];
+ effect_param_t* vis_none = (effect_param_t*)(buf32[0]);
+ effect_param_t* vis_rms = (effect_param_t*)(buf32[1]);
+ effect_param_t* vis_tmp = (effect_param_t*)(buf32[2]);
+
+ // Visualizer::setMeasurementMode()
+ vis_none->psize = sizeof(uint32_t);
+ vis_none->vsize = sizeof(uint32_t);
+ *(int32_t*)vis_none->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+ *((int32_t*)vis_none->data + 1) = MEASUREMENT_MODE_NONE;
+ EXPECT_EQ(NO_ERROR, visualizer->setParameter(vis_none))
+ << "setMeasurementMode doesn't report success";
+
+ // Visualizer::getMeasurementMode()
+ vis_tmp->psize = sizeof(uint32_t);
+ vis_tmp->vsize = sizeof(uint32_t);
+ *(int32_t*)vis_tmp->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+ *((int32_t*)vis_tmp->data + 1) = 23;
+ EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+ << "getMeasurementMode doesn't report success";
+ EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_none->data + 1))
+ << "target mode does not match set mode";
+
+ // Visualizer::setMeasurementModeDeferred()
+ vis_rms->psize = sizeof(uint32_t);
+ vis_rms->vsize = sizeof(uint32_t);
+ *(int32_t*)vis_rms->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+ *((int32_t*)vis_rms->data + 1) = MEASUREMENT_MODE_PEAK_RMS;
+ EXPECT_EQ(NO_ERROR, visualizer->setParameterDeferred(vis_rms))
+ << "setMeasurementModeDeferred doesn't report success";
+
+ *((int32_t*)vis_tmp->data + 1) = 23;
+ EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+ << "getMeasurementMode doesn't report success";
+ EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_none->data + 1))
+ << "target mode does not match set mode";
+
+ // setParameterCommit
+ EXPECT_EQ(NO_ERROR, visualizer->setParameterCommit())
+ << "setMeasurementModeCommit does not report success";
+
+ // validate Params
+ *((int32_t*)vis_tmp->data + 1) = 23;
+ EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+ << "getMeasurementMode doesn't report success";
+ EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_rms->data + 1))
+ << "target mode does not match set mode";
+}
+
+TEST(AudioEffectTest, ManageSourceDefaultEffects) {
+ int32_t selectedEffect = -1;
+
+ const uint32_t sampleRate = 44100;
+ const audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
+ const audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_STEREO;
+ sp<AudioCapture> capture = nullptr;
+
+ std::vector<effect_descriptor_t> descriptors;
+ listEffectsAvailable(descriptors);
+ for (auto i = 0; i < descriptors.size(); i++) {
+ if (isPreprocessing(descriptors[i])) {
+ capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+ ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+ EXPECT_EQ(NO_ERROR, capture->create());
+ EXPECT_EQ(NO_ERROR, capture->start());
+ if (!isEffectDefaultOnRecord(&descriptors[i].type, capture->getAudioRecordHandle())) {
+ selectedEffect = i;
+ break;
+ }
+ }
+ }
+ if (selectedEffect == -1) GTEST_SKIP() << " expected at least one preprocessing effect";
+ effect_uuid_t selectedEffectType = descriptors[selectedEffect].type;
+
+ char type[512];
+ AudioEffect::guidToString(&selectedEffectType, type, sizeof(type));
+
+ capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+ ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+ EXPECT_EQ(NO_ERROR, capture->create());
+ EXPECT_EQ(NO_ERROR, capture->start());
+ EXPECT_FALSE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+ << "Effect should not have been default on record. " << type;
+ EXPECT_FALSE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+ capture->getAudioRecordHandle()->getSessionId()))
+ << "Effect should not have been added. " << type;
+ EXPECT_EQ(OK, capture->audioProcess());
+ EXPECT_EQ(OK, capture->stop());
+
+ String16 name{gPackageName};
+ audio_unique_id_t effectId;
+ status_t status = AudioEffect::addSourceDefaultEffect(
+ type, name, nullptr, kDefaultInputEffectPriority, AUDIO_SOURCE_MIC, &effectId);
+ EXPECT_EQ(NO_ERROR, status) << "Adding default effect failed: " << type;
+
+ capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+ ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+ EXPECT_EQ(NO_ERROR, capture->create());
+ EXPECT_EQ(NO_ERROR, capture->start());
+ EXPECT_TRUE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+ << "Effect should have been default on record. " << type;
+ EXPECT_TRUE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+ capture->getAudioRecordHandle()->getSessionId()))
+ << "Effect should have been added. " << type;
+ EXPECT_EQ(OK, capture->audioProcess());
+ EXPECT_EQ(OK, capture->stop());
+
+ status = AudioEffect::removeSourceDefaultEffect(effectId);
+ EXPECT_EQ(NO_ERROR, status);
+ capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+ ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+ EXPECT_EQ(NO_ERROR, capture->create());
+ EXPECT_EQ(NO_ERROR, capture->start());
+ EXPECT_FALSE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+ << "Effect should not have been default on record. " << type;
+ EXPECT_FALSE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+ capture->getAudioRecordHandle()->getSessionId()))
+ << "Effect should not have been added. " << type;
+ EXPECT_EQ(OK, capture->audioProcess());
+ EXPECT_EQ(OK, capture->stop());
+}
+
+TEST(AudioEffectTest, ManageStreamDefaultEffects) {
+ int32_t selectedEffect = -1;
+
+ std::vector<effect_descriptor_t> descriptors;
+ listEffectsAvailable(descriptors);
+ for (auto i = 0; i < descriptors.size(); i++) {
+ if (isAux(descriptors[i])) {
+ selectedEffect = i;
+ break;
+ }
+ }
+ if (selectedEffect == -1) GTEST_SKIP() << " expected at least one Aux effect";
+ effect_uuid_t* selectedEffectType = &descriptors[selectedEffect].type;
+
+ char type[512];
+ AudioEffect::guidToString(selectedEffectType, type, sizeof(type));
+ // create track
+ audio_attributes_t attributes;
+ attributes.usage = AUDIO_USAGE_MEDIA;
+ attributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+ auto playback = sp<AudioPlayback>::make(
+ 44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+ ASSERT_NE(nullptr, playback);
+ ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+ EXPECT_EQ(NO_ERROR, playback->create());
+ EXPECT_EQ(NO_ERROR, playback->start());
+ EXPECT_FALSE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+ playback->getAudioTrackHandle()->getSessionId()))
+ << "Effect should not have been added. " << type;
+ EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+ playback->stop();
+ playback.clear();
+
+ String16 name{gPackageName};
+ audio_unique_id_t id;
+ status_t status = AudioEffect::addStreamDefaultEffect(
+ type, name, nullptr, kDefaultOutputEffectPriority, AUDIO_USAGE_MEDIA, &id);
+ EXPECT_EQ(NO_ERROR, status) << "Adding default effect failed: " << type;
+
+ playback = sp<AudioPlayback>::make(
+ 44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+ ASSERT_NE(nullptr, playback);
+ ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+ EXPECT_EQ(NO_ERROR, playback->create());
+ float level = 0.2f, levelGot;
+ playback->getAudioTrackHandle()->setAuxEffectSendLevel(level);
+ EXPECT_EQ(NO_ERROR, playback->start());
+ EXPECT_TRUE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+ playback->getAudioTrackHandle()->getSessionId()))
+ << "Effect should have been added. " << type;
+ EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+ playback->getAudioTrackHandle()->getAuxEffectSendLevel(&levelGot);
+ EXPECT_EQ(level, levelGot);
+ playback->stop();
+ playback.clear();
+
+ status = AudioEffect::removeStreamDefaultEffect(id);
+ EXPECT_EQ(NO_ERROR, status);
+ playback = sp<AudioPlayback>::make(
+ 44100 /*sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+ ASSERT_NE(nullptr, playback);
+ ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+ EXPECT_EQ(NO_ERROR, playback->create());
+ EXPECT_EQ(NO_ERROR, playback->start());
+ EXPECT_FALSE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+ playback->getAudioTrackHandle()->getSessionId()))
+ << "Effect should not have been added. " << type;
+ EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+ playback->stop();
+ playback.clear();
+}
diff --git a/media/libaudioclient/tests/audiorecord_tests.cpp b/media/libaudioclient/tests/audiorecord_tests.cpp
new file mode 100644
index 0000000..754e6cc
--- /dev/null
+++ b/media/libaudioclient/tests/audiorecord_tests.cpp
@@ -0,0 +1,235 @@
+/*
+ * 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 "AudioRecordTest"
+
+#include <gtest/gtest.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+class AudioRecordTest : public ::testing::Test {
+ public:
+ virtual void SetUp() override {
+ mAC = new AudioCapture(AUDIO_SOURCE_DEFAULT, 44100, AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_IN_FRONT);
+ ASSERT_NE(nullptr, mAC);
+ ASSERT_EQ(OK, mAC->create()) << "record creation failed";
+ }
+
+ virtual void TearDown() override {
+ if (mAC) ASSERT_EQ(OK, mAC->stop());
+ }
+
+ sp<AudioCapture> mAC;
+};
+
+class AudioRecordCreateTest
+ : public ::testing::TestWithParam<
+ std::tuple<uint32_t, audio_format_t, audio_channel_mask_t, audio_input_flags_t,
+ audio_session_t, audio_source_t>> {
+ public:
+ AudioRecordCreateTest()
+ : mSampleRate(std::get<0>(GetParam())),
+ mFormat(std::get<1>(GetParam())),
+ mChannelMask(std::get<2>(GetParam())),
+ mFlags(std::get<3>(GetParam())),
+ mSessionId(std::get<4>(GetParam())),
+ mInputSource(std::get<5>(GetParam())){};
+
+ const uint32_t mSampleRate;
+ const audio_format_t mFormat;
+ const audio_channel_mask_t mChannelMask;
+ const audio_input_flags_t mFlags;
+ const audio_session_t mSessionId;
+ const audio_source_t mInputSource;
+ const AudioRecord::transfer_type mTransferType = AudioRecord::TRANSFER_OBTAIN;
+
+ sp<AudioCapture> mAC;
+
+ virtual void SetUp() override {
+ mAC = new AudioCapture(mInputSource, mSampleRate, mFormat, mChannelMask, mFlags, mSessionId,
+ mTransferType);
+ ASSERT_NE(nullptr, mAC);
+ ASSERT_EQ(OK, mAC->create()) << "record creation failed";
+ }
+
+ virtual void TearDown() override {
+ if (mAC) ASSERT_EQ(OK, mAC->stop());
+ }
+};
+
+TEST_F(AudioRecordTest, TestSimpleRecord) {
+ EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+ EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestAudioCbNotifier) {
+ EXPECT_EQ(BAD_VALUE, mAC->getAudioRecordHandle()->addAudioDeviceCallback(nullptr));
+ sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+ sp<OnAudioDeviceUpdateNotifier> cbOld = new OnAudioDeviceUpdateNotifier();
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cbOld));
+ EXPECT_EQ(INVALID_OPERATION, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cbOld));
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cb));
+ EXPECT_EQ(OK, mAC->start()) << "record creation failed";
+ EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+ EXPECT_EQ(AUDIO_IO_HANDLE_NONE, cbOld->mAudioIo);
+ EXPECT_EQ(AUDIO_PORT_HANDLE_NONE, cbOld->mDeviceId);
+ EXPECT_NE(AUDIO_IO_HANDLE_NONE, cb->mAudioIo);
+ EXPECT_NE(AUDIO_PORT_HANDLE_NONE, cb->mDeviceId);
+ EXPECT_EQ(BAD_VALUE, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(nullptr));
+ EXPECT_EQ(INVALID_OPERATION, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(cbOld));
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(cb));
+ mAC->stop();
+}
+
+TEST_F(AudioRecordTest, TestEventRecordTrackPause) {
+ const auto playback = sp<AudioPlayback>::make(
+ 8000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO);
+ ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_1ch_8kHz_s16le.raw"))
+ << "Unable to open Resource";
+ EXPECT_EQ(OK, playback->create()) << "AudioTrack Creation failed";
+ audio_session_t audioTrackSession = playback->getAudioTrackHandle()->getSessionId();
+ EXPECT_EQ(OK, mAC->start(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE, audioTrackSession))
+ << "record creation failed";
+ EXPECT_EQ(OK, playback->start());
+ RawBuffer buffer;
+ status_t status = mAC->obtainBufferCb(buffer);
+ EXPECT_EQ(status, TIMED_OUT) << "Not expecting any callbacks until track sends Sync event";
+ playback->getAudioTrackHandle()->pause();
+ EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+ playback->stop();
+}
+
+TEST_F(AudioRecordTest, TestEventRecordTrackStop) {
+ const auto playback = sp<AudioPlayback>::make(
+ 8000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO);
+ ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_1ch_8kHz_s16le.raw"))
+ << "Unable to open Resource";
+ EXPECT_EQ(OK, playback->create()) << "AudioTrack Creation failed";
+ audio_session_t audioTrackSession = playback->getAudioTrackHandle()->getSessionId();
+ EXPECT_EQ(OK, mAC->start(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE, audioTrackSession))
+ << "record creation failed";
+ EXPECT_EQ(OK, playback->start());
+ RawBuffer buffer;
+ status_t status = mAC->obtainBufferCb(buffer);
+ EXPECT_EQ(status, TIMED_OUT) << "Not expecting any callbacks until track sends Sync event";
+ playback->stop();
+ EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestGetSetMarker) {
+ mAC->mMarkerPosition = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setMarkerPosition(mAC->mMarkerPosition))
+ << "setMarkerPosition() failed";
+ uint32_t marker;
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getMarkerPosition(&marker))
+ << "getMarkerPosition() failed";
+ EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+ EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+ EXPECT_EQ(marker, mAC->mMarkerPosition)
+ << "configured marker and received marker are different";
+ EXPECT_EQ(mAC->mReceivedCbMarkerAtPosition, mAC->mMarkerPosition)
+ << "configured marker and received cb marker are different";
+}
+
+TEST_F(AudioRecordTest, TestGetSetMarkerPeriodical) {
+ mAC->mMarkerPeriod = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setPositionUpdatePeriod(mAC->mMarkerPeriod))
+ << "setPositionUpdatePeriod() failed";
+ uint32_t marker;
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPositionUpdatePeriod(&marker))
+ << "getPositionUpdatePeriod() failed";
+ EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+ EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+ EXPECT_EQ(marker, mAC->mMarkerPeriod) << "configured marker and received marker are different";
+ EXPECT_EQ(mAC->mReceivedCbMarkerCount, mAC->mNumFramesToRecord / mAC->mMarkerPeriod)
+ << "configured marker and received cb marker are different";
+}
+
+TEST_F(AudioRecordTest, TestGetPosition) {
+ uint32_t position;
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPosition(&position)) << "getPosition() failed";
+ EXPECT_EQ(0, position);
+ EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+ EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+ EXPECT_EQ(OK, mAC->stop());
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPosition(&position)) << "getPosition() failed";
+}
+
+// TODO: Add checkPatchCapture(), verify the information of patch via dumpPort() and dumpPatch()
+TEST_P(AudioRecordCreateTest, TestCreateRecord) {
+ EXPECT_EQ(mFormat, mAC->getAudioRecordHandle()->format());
+ EXPECT_EQ(audio_channel_count_from_in_mask(mChannelMask),
+ mAC->getAudioRecordHandle()->channelCount());
+ if (mAC->mFrameCount != 0)
+ EXPECT_LE(mAC->mFrameCount, mAC->getAudioRecordHandle()->frameCount());
+ EXPECT_EQ(mInputSource, mAC->getAudioRecordHandle()->inputSource());
+ if (mSampleRate != 0) EXPECT_EQ(mSampleRate, mAC->getAudioRecordHandle()->getSampleRate());
+ if (mSessionId != AUDIO_SESSION_NONE)
+ EXPECT_EQ(mSessionId, mAC->getAudioRecordHandle()->getSessionId());
+ if (mTransferType != AudioRecord::TRANSFER_CALLBACK) {
+ uint32_t marker;
+ mAC->mMarkerPosition = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+ EXPECT_EQ(INVALID_OPERATION,
+ mAC->getAudioRecordHandle()->setMarkerPosition(mAC->mMarkerPosition));
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getMarkerPosition(&marker));
+ EXPECT_EQ(INVALID_OPERATION,
+ mAC->getAudioRecordHandle()->setPositionUpdatePeriod(mAC->mMarkerPosition));
+ EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPositionUpdatePeriod(&marker));
+ }
+ EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+ EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+// for port primary input
+INSTANTIATE_TEST_SUITE_P(AudioRecordPrimaryInput, AudioRecordCreateTest,
+ ::testing::Combine(::testing::Values(8000, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000),
+ ::testing::Values(AUDIO_FORMAT_PCM_8_24_BIT),
+ ::testing::Values(AUDIO_CHANNEL_IN_MONO,
+ AUDIO_CHANNEL_IN_STEREO,
+ AUDIO_CHANNEL_IN_FRONT_BACK),
+ ::testing::Values(AUDIO_INPUT_FLAG_NONE),
+ ::testing::Values(AUDIO_SESSION_NONE),
+ ::testing::Values(AUDIO_SOURCE_DEFAULT)));
+
+// for port fast input
+INSTANTIATE_TEST_SUITE_P(AudioRecordFastInput, AudioRecordCreateTest,
+ ::testing::Combine(::testing::Values(8000, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000),
+ ::testing::Values(AUDIO_FORMAT_PCM_8_24_BIT),
+ ::testing::Values(AUDIO_CHANNEL_IN_MONO,
+ AUDIO_CHANNEL_IN_STEREO,
+ AUDIO_CHANNEL_IN_FRONT_BACK),
+ ::testing::Values(AUDIO_INPUT_FLAG_FAST),
+ ::testing::Values(AUDIO_SESSION_NONE),
+ ::testing::Values(AUDIO_SOURCE_DEFAULT)));
+
+// misc
+INSTANTIATE_TEST_SUITE_P(AudioRecordMiscInput, AudioRecordCreateTest,
+ ::testing::Combine(::testing::Values(48000),
+ ::testing::Values(AUDIO_FORMAT_PCM_16_BIT),
+ ::testing::Values(AUDIO_CHANNEL_IN_MONO),
+ ::testing::Values(AUDIO_INPUT_FLAG_NONE),
+ ::testing::Values(AUDIO_SESSION_NONE),
+ ::testing::Values(AUDIO_SOURCE_MIC,
+ AUDIO_SOURCE_CAMCORDER,
+ AUDIO_SOURCE_VOICE_RECOGNITION,
+ AUDIO_SOURCE_VOICE_COMMUNICATION,
+ AUDIO_SOURCE_UNPROCESSED)));
diff --git a/media/libaudioclient/tests/audiorouting_tests.cpp b/media/libaudioclient/tests/audiorouting_tests.cpp
new file mode 100644
index 0000000..32ba597
--- /dev/null
+++ b/media/libaudioclient/tests/audiorouting_tests.cpp
@@ -0,0 +1,253 @@
+/*
+ * 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
+
+#include <cutils/properties.h>
+#include <gtest/gtest.h>
+#include <libxml/parser.h>
+#include <libxml/xinclude.h>
+#include <string.h>
+#include <system/audio_config.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+template <class T>
+constexpr void (*xmlDeleter)(T* t);
+template <>
+constexpr auto xmlDeleter<xmlDoc> = xmlFreeDoc;
+template <>
+constexpr auto xmlDeleter<xmlChar> = [](xmlChar* s) { xmlFree(s); };
+
+/** @return a unique_ptr with the correct deleter for the libxml2 object. */
+template <class T>
+constexpr auto make_xmlUnique(T* t) {
+ // Wrap deleter in lambda to enable empty base optimization
+ auto deleter = [](T* t) { xmlDeleter<T>(t); };
+ return std::unique_ptr<T, decltype(deleter)>{t, deleter};
+}
+
+std::string getXmlAttribute(const xmlNode* cur, const char* attribute) {
+ auto charPtr = make_xmlUnique(xmlGetProp(cur, reinterpret_cast<const xmlChar*>(attribute)));
+ if (charPtr == NULL) {
+ return "";
+ }
+ std::string value(reinterpret_cast<const char*>(charPtr.get()));
+ return value;
+}
+
+struct MixPort {
+ std::string name;
+ std::string role;
+ std::string flags;
+};
+
+struct Route {
+ std::string name;
+ std::string sources;
+ std::string sink;
+};
+
+status_t parse_audio_policy_configuration_xml(std::vector<std::string>& attachedDevices,
+ std::vector<MixPort>& mixPorts,
+ std::vector<Route>& routes) {
+ std::string path = audio_find_readable_configuration_file("audio_policy_configuration.xml");
+ if (path.length() == 0) return UNKNOWN_ERROR;
+ auto doc = make_xmlUnique(xmlParseFile(path.c_str()));
+ if (doc == nullptr) return UNKNOWN_ERROR;
+ xmlNode* root = xmlDocGetRootElement(doc.get());
+ if (root == nullptr) return UNKNOWN_ERROR;
+ if (xmlXIncludeProcess(doc.get()) < 0) return UNKNOWN_ERROR;
+ mixPorts.clear();
+ if (!xmlStrcmp(root->name, reinterpret_cast<const xmlChar*>("audioPolicyConfiguration"))) {
+ std::string raw{getXmlAttribute(root, "version")};
+ for (auto* child = root->xmlChildrenNode; child != nullptr; child = child->next) {
+ if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>("modules"))) {
+ xmlNode* root = child;
+ for (auto* child = root->xmlChildrenNode; child != nullptr; child = child->next) {
+ if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>("module"))) {
+ xmlNode* root = child;
+ for (auto* child = root->xmlChildrenNode; child != nullptr;
+ child = child->next) {
+ if (!xmlStrcmp(child->name,
+ reinterpret_cast<const xmlChar*>("mixPorts"))) {
+ xmlNode* root = child;
+ for (auto* child = root->xmlChildrenNode; child != nullptr;
+ child = child->next) {
+ if (!xmlStrcmp(child->name,
+ reinterpret_cast<const xmlChar*>("mixPort"))) {
+ MixPort mixPort;
+ xmlNode* root = child;
+ mixPort.name = getXmlAttribute(root, "name");
+ mixPort.role = getXmlAttribute(root, "role");
+ mixPort.flags = getXmlAttribute(root, "flags");
+ if (mixPort.role == "source") mixPorts.push_back(mixPort);
+ }
+ }
+ } else if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>(
+ "attachedDevices"))) {
+ xmlNode* root = child;
+ for (auto* child = root->xmlChildrenNode; child != nullptr;
+ child = child->next) {
+ if (!xmlStrcmp(child->name,
+ reinterpret_cast<const xmlChar*>("item"))) {
+ auto xmlValue = make_xmlUnique(xmlNodeListGetString(
+ child->doc, child->xmlChildrenNode, 1));
+ if (xmlValue == nullptr) {
+ raw = "";
+ } else {
+ raw = reinterpret_cast<const char*>(xmlValue.get());
+ }
+ std::string& value = raw;
+ attachedDevices.push_back(std::move(value));
+ }
+ }
+ } else if (!xmlStrcmp(child->name,
+ reinterpret_cast<const xmlChar*>("routes"))) {
+ xmlNode* root = child;
+ for (auto* child = root->xmlChildrenNode; child != nullptr;
+ child = child->next) {
+ if (!xmlStrcmp(child->name,
+ reinterpret_cast<const xmlChar*>("route"))) {
+ Route route;
+ xmlNode* root = child;
+ route.name = getXmlAttribute(root, "name");
+ route.sources = getXmlAttribute(root, "sources");
+ route.sink = getXmlAttribute(root, "sink");
+ routes.push_back(route);
+ }
+ }
+ }
+ }
+ }
+ }
+ }
+ }
+ }
+ return OK;
+}
+
+// UNIT TEST
+TEST(AudioTrackTest, TestPerformanceMode) {
+ std::vector<std::string> attachedDevices;
+ std::vector<MixPort> mixPorts;
+ std::vector<Route> routes;
+ EXPECT_EQ(OK, parse_audio_policy_configuration_xml(attachedDevices, mixPorts, routes));
+ std::string output_flags_string[] = {"AUDIO_OUTPUT_FLAG_FAST", "AUDIO_OUTPUT_FLAG_DEEP_BUFFER"};
+ audio_output_flags_t output_flags[] = {AUDIO_OUTPUT_FLAG_FAST, AUDIO_OUTPUT_FLAG_DEEP_BUFFER};
+ audio_flags_mask_t flags[] = {AUDIO_FLAG_LOW_LATENCY, AUDIO_FLAG_DEEP_BUFFER};
+ bool hasFlag = false;
+ for (int i = 0; i < sizeof(flags) / sizeof(flags[0]); i++) {
+ hasFlag = false;
+ for (int j = 0; j < mixPorts.size() && !hasFlag; j++) {
+ MixPort port = mixPorts[j];
+ if (port.role == "source" && port.flags.find(output_flags_string[i]) != -1) {
+ for (int k = 0; k < routes.size() && !hasFlag; k++) {
+ if (routes[k].sources.find(port.name) != -1 &&
+ std::find(attachedDevices.begin(), attachedDevices.end(), routes[k].sink) !=
+ attachedDevices.end()) {
+ hasFlag = true;
+ std::cerr << "found port with flag " << output_flags_string[i] << "@ "
+ << " port :: name : " << port.name << " role : " << port.role
+ << " port :: flags : " << port.flags
+ << " connected via route name : " << routes[k].name
+ << " route sources : " << routes[k].sources
+ << " route sink : " << routes[k].sink << std::endl;
+ }
+ }
+ }
+ }
+ if (!hasFlag) continue;
+ audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
+ attributes.usage = AUDIO_USAGE_MEDIA;
+ attributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+ attributes.flags = flags[i];
+ sp<AudioPlayback> ap = sp<AudioPlayback>::make(
+ 0 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_OBTAIN,
+ &attributes);
+ ASSERT_NE(nullptr, ap);
+ ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+ << "Unable to open Resource";
+ EXPECT_EQ(OK, ap->create()) << "track creation failed";
+ sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+ EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cb));
+ EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+ EXPECT_EQ(OK, ap->onProcess());
+ EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+ EXPECT_TRUE(checkPatchPlayback(cb->mAudioIo, cb->mDeviceId));
+ EXPECT_NE(0, ap->getAudioTrackHandle()->getFlags() & output_flags[i]);
+ audio_patch patch;
+ EXPECT_EQ(OK, getPatchForOutputMix(cb->mAudioIo, patch));
+ for (auto j = 0; j < patch.num_sources; j++) {
+ if (patch.sources[j].type == AUDIO_PORT_TYPE_MIX &&
+ patch.sources[j].ext.mix.handle == cb->mAudioIo) {
+ if ((patch.sources[j].flags.output & output_flags[i]) == 0) {
+ ADD_FAILURE() << "expected output flag " << output_flags[i] << " is absent";
+ std::cerr << dumpPortConfig(patch.sources[j]);
+ }
+ }
+ }
+ ap->stop();
+ ap->getAudioTrackHandle()->removeAudioDeviceCallback(cb);
+ }
+}
+
+TEST(AudioTrackTest, TestRemoteSubmix) {
+ std::vector<std::string> attachedDevices;
+ std::vector<MixPort> mixPorts;
+ std::vector<Route> routes;
+ EXPECT_EQ(OK, parse_audio_policy_configuration_xml(attachedDevices, mixPorts, routes));
+ bool hasFlag = false;
+ for (int j = 0; j < attachedDevices.size() && !hasFlag; j++) {
+ if (attachedDevices[j].find("Remote Submix") != -1) hasFlag = true;
+ }
+ if (!hasFlag) GTEST_SKIP() << " Device does not have Remote Submix port.";
+ sp<AudioCapture> capture = new AudioCapture(AUDIO_SOURCE_REMOTE_SUBMIX, 48000,
+ AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO);
+ ASSERT_NE(nullptr, capture);
+ ASSERT_EQ(OK, capture->create()) << "record creation failed";
+
+ sp<AudioPlayback> playback = sp<AudioPlayback>::make(
+ 48000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE);
+ ASSERT_NE(nullptr, playback);
+ ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+ << "Unable to open Resource";
+ ASSERT_EQ(OK, playback->create()) << "track creation failed";
+
+ audio_port_v7 port;
+ status_t status = getPortByAttributes(AUDIO_PORT_ROLE_SOURCE, AUDIO_PORT_TYPE_DEVICE,
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, port);
+ EXPECT_EQ(OK, status) << "Could not find port";
+
+ EXPECT_EQ(OK, capture->start()) << "start recording failed";
+ EXPECT_EQ(port.id, capture->getAudioRecordHandle()->getRoutedDeviceId())
+ << "Capture NOT routed on expected port";
+
+ status = getPortByAttributes(AUDIO_PORT_ROLE_SINK, AUDIO_PORT_TYPE_DEVICE,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, port);
+ EXPECT_EQ(OK, status) << "Could not find port";
+
+ EXPECT_EQ(OK, playback->start()) << "audio track start failed";
+ EXPECT_EQ(OK, playback->onProcess());
+ ASSERT_EQ(port.id, playback->getAudioTrackHandle()->getRoutedDeviceId())
+ << "Playback NOT routed on expected port";
+ capture->stop();
+ playback->stop();
+}
diff --git a/media/libaudioclient/tests/audiotrack_tests.cpp b/media/libaudioclient/tests/audiotrack_tests.cpp
new file mode 100644
index 0000000..1b42a49
--- /dev/null
+++ b/media/libaudioclient/tests/audiotrack_tests.cpp
@@ -0,0 +1,211 @@
+/*
+ * 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
+
+#include <gtest/gtest.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+TEST(AudioTrackTest, TestPlayTrack) {
+ const auto ap = sp<AudioPlayback>::make(44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_OUT_STEREO, AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_SESSION_NONE, AudioTrack::TRANSFER_OBTAIN);
+ ASSERT_NE(nullptr, ap);
+ ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+ << "Unable to open Resource";
+ EXPECT_EQ(OK, ap->create()) << "track creation failed";
+ EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+ EXPECT_EQ(OK, ap->onProcess());
+ ap->stop();
+}
+
+TEST(AudioTrackTest, TestSeek) {
+ const auto ap = sp<AudioPlayback>::make(
+ 44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO);
+ ASSERT_NE(nullptr, ap);
+ ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+ << "Unable to open Resource";
+ EXPECT_EQ(OK, ap->create()) << "track creation failed";
+ EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+ EXPECT_EQ(OK, ap->onProcess(true));
+ ap->stop();
+}
+
+TEST(AudioTrackTest, OffloadOrDirectPlayback) {
+ audio_offload_info_t info = AUDIO_INFO_INITIALIZER;
+ info.sample_rate = 44100;
+ info.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ info.format = AUDIO_FORMAT_MP3;
+ info.stream_type = AUDIO_STREAM_MUSIC;
+ info.bit_rate = 192;
+ info.duration_us = 120 * 1000000; // 120 sec
+
+ audio_config_base_t config = {/* .sample_rate = */ info.sample_rate,
+ /* .channel_mask = */ info.channel_mask,
+ /* .format = */ AUDIO_FORMAT_PCM_16_BIT};
+ audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
+ attributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+ attributes.usage = AUDIO_USAGE_MEDIA;
+ attributes.flags = AUDIO_FLAG_NONE;
+
+ if (!AudioTrack::isDirectOutputSupported(config, attributes) &&
+ AUDIO_OFFLOAD_NOT_SUPPORTED == AudioSystem::getOffloadSupport(info)) {
+ GTEST_SKIP() << "offload or direct playback is not supported";
+ }
+ sp<AudioPlayback> ap = nullptr;
+ if (AUDIO_OFFLOAD_NOT_SUPPORTED != AudioSystem::getOffloadSupport(info)) {
+ ap = sp<AudioPlayback>::make(info.sample_rate, info.format, info.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, AUDIO_SESSION_NONE,
+ AudioTrack::TRANSFER_OBTAIN, nullptr, &info);
+ } else {
+ ap = sp<AudioPlayback>::make(config.sample_rate, config.format, config.channel_mask,
+ AUDIO_OUTPUT_FLAG_DIRECT, AUDIO_SESSION_NONE,
+ AudioTrack::TRANSFER_OBTAIN);
+ }
+ ASSERT_NE(nullptr, ap);
+ EXPECT_EQ(OK, ap->create()) << "track creation failed";
+ audio_dual_mono_mode_t mode;
+ if (OK != ap->getAudioTrackHandle()->getDualMonoMode(&mode)) {
+ std::cerr << "no dual mono presentation is available" << std::endl;
+ }
+ if (OK != ap->getAudioTrackHandle()->setDualMonoMode(AUDIO_DUAL_MONO_MODE_LR)) {
+ std::cerr << "no dual mono presentation is available" << std::endl;
+ } else {
+ EXPECT_EQ(OK, ap->getAudioTrackHandle()->getDualMonoMode(&mode));
+ EXPECT_EQ(AUDIO_DUAL_MONO_MODE_LR, mode);
+ }
+ float leveldB;
+ if (OK != ap->getAudioTrackHandle()->getAudioDescriptionMixLevel(&leveldB)) {
+ std::cerr << "Audio Description mixing is unavailable" << std::endl;
+ }
+ if (OK != ap->getAudioTrackHandle()->setAudioDescriptionMixLevel(3.14f)) {
+ std::cerr << "Audio Description mixing is unavailable" << std::endl;
+ } else {
+ EXPECT_EQ(OK, ap->getAudioTrackHandle()->getAudioDescriptionMixLevel(&leveldB));
+ EXPECT_EQ(3.14f, leveldB);
+ }
+ AudioPlaybackRate audioRate;
+ audioRate = ap->getAudioTrackHandle()->getPlaybackRate();
+ std::cerr << "playback speed :: " << audioRate.mSpeed << std::endl
+ << "playback pitch :: " << audioRate.mPitch << std::endl;
+ audioRate.mSpeed = 2.0f;
+ audioRate.mPitch = 2.0f;
+ audioRate.mStretchMode = AUDIO_TIMESTRETCH_STRETCH_VOICE;
+ audioRate.mFallbackMode = AUDIO_TIMESTRETCH_FALLBACK_MUTE;
+ EXPECT_TRUE(isAudioPlaybackRateValid(audioRate));
+ if (OK != ap->getAudioTrackHandle()->setPlaybackRate(audioRate)) {
+ std::cerr << "unable to set playback rate parameters" << std::endl;
+ } else {
+ AudioPlaybackRate audioRateLocal;
+ audioRateLocal = ap->getAudioTrackHandle()->getPlaybackRate();
+ EXPECT_TRUE(isAudioPlaybackRateEqual(audioRate, audioRateLocal));
+ }
+ ap->stop();
+}
+
+TEST(AudioTrackTest, TestAudioCbNotifier) {
+ const auto ap = sp<AudioPlayback>::make(0 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_OUT_STEREO, AUDIO_OUTPUT_FLAG_FAST,
+ AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED);
+ ASSERT_NE(nullptr, ap);
+ ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+ << "Unable to open Resource";
+ EXPECT_EQ(OK, ap->create()) << "track creation failed";
+ EXPECT_EQ(BAD_VALUE, ap->getAudioTrackHandle()->addAudioDeviceCallback(nullptr));
+ sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+ sp<OnAudioDeviceUpdateNotifier> cbOld = new OnAudioDeviceUpdateNotifier();
+ EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cbOld));
+ EXPECT_EQ(INVALID_OPERATION, ap->getAudioTrackHandle()->addAudioDeviceCallback(cbOld));
+ EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cb));
+ EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+ EXPECT_EQ(OK, ap->onProcess());
+ EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+ EXPECT_EQ(AUDIO_IO_HANDLE_NONE, cbOld->mAudioIo);
+ EXPECT_EQ(AUDIO_PORT_HANDLE_NONE, cbOld->mDeviceId);
+ EXPECT_NE(AUDIO_IO_HANDLE_NONE, cb->mAudioIo);
+ EXPECT_NE(AUDIO_PORT_HANDLE_NONE, cb->mDeviceId);
+ EXPECT_EQ(cb->mAudioIo, ap->getAudioTrackHandle()->getOutput());
+ EXPECT_EQ(cb->mDeviceId, ap->getAudioTrackHandle()->getRoutedDeviceId());
+ String8 keys;
+ keys = ap->getAudioTrackHandle()->getParameters(keys);
+ if (!keys.isEmpty()) {
+ std::cerr << "track parameters :: " << keys << std::endl;
+ }
+ EXPECT_TRUE(checkPatchPlayback(cb->mAudioIo, cb->mDeviceId));
+ EXPECT_EQ(BAD_VALUE, ap->getAudioTrackHandle()->removeAudioDeviceCallback(nullptr));
+ EXPECT_EQ(INVALID_OPERATION, ap->getAudioTrackHandle()->removeAudioDeviceCallback(cbOld));
+ EXPECT_EQ(OK, ap->getAudioTrackHandle()->removeAudioDeviceCallback(cb));
+ ap->stop();
+}
+
+class AudioTrackCreateTest
+ : public ::testing::TestWithParam<std::tuple<uint32_t, audio_format_t, audio_channel_mask_t,
+ audio_output_flags_t, audio_session_t>> {
+ public:
+ AudioTrackCreateTest()
+ : mSampleRate(std::get<0>(GetParam())),
+ mFormat(std::get<1>(GetParam())),
+ mChannelMask(std::get<2>(GetParam())),
+ mFlags(std::get<3>(GetParam())),
+ mSessionId(std::get<4>(GetParam())){};
+
+ const uint32_t mSampleRate;
+ const audio_format_t mFormat;
+ const audio_channel_mask_t mChannelMask;
+ const audio_output_flags_t mFlags;
+ const audio_session_t mSessionId;
+
+ sp<AudioPlayback> mAP;
+
+ virtual void SetUp() override {
+ mAP = sp<AudioPlayback>::make(mSampleRate, mFormat, mChannelMask, mFlags,
+ mSessionId);
+ ASSERT_NE(nullptr, mAP);
+ ASSERT_EQ(OK, mAP->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+ << "Unable to open Resource";
+ ASSERT_EQ(OK, mAP->create()) << "track creation failed";
+ }
+
+ virtual void TearDown() override {
+ if (mAP) mAP->stop();
+ }
+};
+
+TEST_P(AudioTrackCreateTest, TestCreateTrack) {
+ EXPECT_EQ(mFormat, mAP->getAudioTrackHandle()->format());
+ EXPECT_EQ(audio_channel_count_from_out_mask(mChannelMask),
+ mAP->getAudioTrackHandle()->channelCount());
+ if (mSampleRate != 0) EXPECT_EQ(mSampleRate, mAP->getAudioTrackHandle()->getSampleRate());
+ if (mSessionId != AUDIO_SESSION_NONE)
+ EXPECT_EQ(mSessionId, mAP->getAudioTrackHandle()->getSessionId());
+ EXPECT_EQ(mSampleRate, mAP->getAudioTrackHandle()->getOriginalSampleRate());
+ EXPECT_EQ(OK, mAP->start()) << "audio track start failed";
+ EXPECT_EQ(OK, mAP->onProcess());
+}
+
+// sampleRate, format, channelMask, flags, sessionId
+INSTANTIATE_TEST_SUITE_P(
+ AudioTrackParameterizedTest, AudioTrackCreateTest,
+ ::testing::Combine(::testing::Values(48000), ::testing::Values(AUDIO_FORMAT_PCM_16_BIT),
+ ::testing::Values(AUDIO_CHANNEL_OUT_STEREO),
+ ::testing::Values(AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_OUTPUT_FLAG_PRIMARY | AUDIO_OUTPUT_FLAG_FAST,
+ AUDIO_OUTPUT_FLAG_RAW | AUDIO_OUTPUT_FLAG_FAST,
+ AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+ ::testing::Values(AUDIO_SESSION_NONE)));
diff --git a/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw b/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw
new file mode 100644
index 0000000..2d1e4bf
--- /dev/null
+++ b/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw
Binary files differ
diff --git a/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw b/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw
new file mode 100644
index 0000000..c8ac5f7
--- /dev/null
+++ b/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw
Binary files differ
diff --git a/media/libaudioclient/tests/test_create_audiorecord.cpp b/media/libaudioclient/tests/test_create_audiorecord.cpp
index 2e0883b..277110b 100644
--- a/media/libaudioclient/tests/test_create_audiorecord.cpp
+++ b/media/libaudioclient/tests/test_create_audiorecord.cpp
@@ -29,14 +29,13 @@
#define NUM_ARGUMENTS 8
#define VERSION_VALUE "1.0"
-#define PACKAGE_NAME "AudioRecord test"
+#define PACKAGE_NAME "AudioRecord test"
namespace android {
using android::content::AttributionSourceState;
-int testRecord(FILE *inputFile, int outputFileFd)
-{
+int testRecord(FILE* inputFile, int outputFileFd) {
char line[MAX_INPUT_FILE_LINE_LENGTH];
uint32_t testCount = 0;
Vector<String16> args;
@@ -47,11 +46,9 @@
attributionSource.token = sp<BBinder>::make();
if (inputFile == nullptr) {
- sp<AudioRecord> record = new AudioRecord(AUDIO_SOURCE_DEFAULT,
- 0 /* sampleRate */,
- AUDIO_FORMAT_DEFAULT,
- AUDIO_CHANNEL_IN_MONO,
- attributionSource);
+ sp<AudioRecord> record =
+ new AudioRecord(AUDIO_SOURCE_DEFAULT, 0 /* sampleRate */, AUDIO_FORMAT_DEFAULT,
+ AUDIO_CHANNEL_IN_MONO, attributionSource);
if (record == 0 || record->initCheck() != NO_ERROR) {
write(outputFileFd, "Error creating AudioRecord\n",
sizeof("Error creating AudioRecord\n"));
@@ -80,11 +77,10 @@
char statusStr[MAX_OUTPUT_FILE_LINE_LENGTH];
bool fast = false;
- if (sscanf(line, " %u %x %x %zu %d %x %u %u",
- &sampleRate, &format, &channelMask,
- &frameCount, ¬ificationFrames,
- &flags, &sessionId, &inputSource) != NUM_ARGUMENTS) {
- fprintf(stderr, "Malformed line for test #%u in input file\n", testCount+1);
+ if (sscanf(line, " %u %x %x %zu %d %x %u %u", &sampleRate, &format, &channelMask,
+ &frameCount, ¬ificationFrames, &flags, &sessionId,
+ &inputSource) != NUM_ARGUMENTS) {
+ fprintf(stderr, "Malformed line for test #%u in input file\n", testCount + 1);
ret = 1;
continue;
}
@@ -100,21 +96,10 @@
sp<AudioRecord> record = new AudioRecord(attributionSource);
const auto emptyCallback = sp<AudioRecord::IAudioRecordCallback>::make();
- record->set(AUDIO_SOURCE_DEFAULT,
- sampleRate,
- format,
- channelMask,
- frameCount,
- fast ? emptyCallback : nullptr,
- notificationFrames,
- false,
- sessionId,
- fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT,
- flags,
- getuid(),
- getpid(),
- &attributes,
- AUDIO_PORT_HANDLE_NONE);
+ record->set(AUDIO_SOURCE_DEFAULT, sampleRate, format, channelMask, frameCount,
+ fast ? emptyCallback : nullptr, notificationFrames, false, sessionId,
+ fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT, flags,
+ getuid(), getpid(), &attributes, AUDIO_PORT_HANDLE_NONE);
status = record->initCheck();
sprintf(statusStr, "\n#### Test %u status %d\n", testCount, status);
write(outputFileFd, statusStr, strlen(statusStr));
@@ -126,11 +111,8 @@
return ret;
}
-}; // namespace android
+}; // namespace android
-
-int main(int argc, char **argv)
-{
+int main(int argc, char** argv) {
return android::main(argc, argv, android::testRecord);
}
-
diff --git a/media/libaudioclient/tests/test_create_audiotrack.cpp b/media/libaudioclient/tests/test_create_audiotrack.cpp
index e7231d3..4e09e21 100644
--- a/media/libaudioclient/tests/test_create_audiotrack.cpp
+++ b/media/libaudioclient/tests/test_create_audiotrack.cpp
@@ -32,18 +32,15 @@
namespace android {
-int testTrack(FILE *inputFile, int outputFileFd)
-{
+int testTrack(FILE* inputFile, int outputFileFd) {
char line[MAX_INPUT_FILE_LINE_LENGTH];
uint32_t testCount = 0;
Vector<String16> args;
int ret = 0;
if (inputFile == nullptr) {
- sp<AudioTrack> track = new AudioTrack(AUDIO_STREAM_DEFAULT,
- 0 /* sampleRate */,
- AUDIO_FORMAT_DEFAULT,
- AUDIO_CHANNEL_OUT_STEREO);
+ sp<AudioTrack> track = new AudioTrack(AUDIO_STREAM_DEFAULT, 0 /* sampleRate */,
+ AUDIO_FORMAT_DEFAULT, AUDIO_CHANNEL_OUT_STEREO);
if (track == 0 || track->initCheck() != NO_ERROR) {
write(outputFileFd, "Error creating AudioTrack\n",
sizeof("Error creating AudioTrack\n"));
@@ -78,11 +75,10 @@
bool offload = false;
bool fast = false;
- if (sscanf(line, " %u %x %x %zu %d %u %x %u %u %u",
- &sampleRate, &format, &channelMask,
- &frameCount, ¬ificationFrames, &useSharedBuffer,
- &flags, &sessionId, &usage, &contentType) != NUM_ARGUMENTS) {
- fprintf(stderr, "Malformed line for test #%u in input file\n", testCount+1);
+ if (sscanf(line, " %u %x %x %zu %d %u %x %u %u %u", &sampleRate, &format, &channelMask,
+ &frameCount, ¬ificationFrames, &useSharedBuffer, &flags, &sessionId, &usage,
+ &contentType) != NUM_ARGUMENTS) {
+ fprintf(stderr, "Malformed line for test #%u in input file\n", testCount + 1);
ret = 1;
continue;
}
@@ -90,7 +86,7 @@
if (useSharedBuffer != 0) {
size_t heapSize = audio_channel_count_from_out_mask(channelMask) *
- audio_bytes_per_sample(format) * frameCount;
+ audio_bytes_per_sample(format) * frameCount;
heap = new MemoryDealer(heapSize, "AudioTrack Heap Base");
sharedBuffer = heap->allocate(heapSize);
frameCount = 0;
@@ -111,25 +107,13 @@
attributes.usage = usage;
sp<AudioTrack> track = new AudioTrack();
const auto emptyCallback = sp<AudioTrack::IAudioTrackCallback>::make();
- track->set(AUDIO_STREAM_DEFAULT,
- sampleRate,
- format,
- channelMask,
- frameCount,
- flags,
- (fast || offload) ? emptyCallback : nullptr,
- notificationFrames,
- sharedBuffer,
- false,
- sessionId,
- ((fast && sharedBuffer == 0) || offload) ?
- AudioTrack::TRANSFER_CALLBACK : AudioTrack::TRANSFER_DEFAULT,
- offload ? &offloadInfo : nullptr,
- AttributionSourceState(),
- &attributes,
- false,
- 1.0f,
- AUDIO_PORT_HANDLE_NONE);
+ track->set(AUDIO_STREAM_DEFAULT, sampleRate, format, channelMask, frameCount, flags,
+ (fast || offload) ? emptyCallback : nullptr, notificationFrames, sharedBuffer,
+ false, sessionId,
+ ((fast && sharedBuffer == 0) || offload) ? AudioTrack::TRANSFER_CALLBACK
+ : AudioTrack::TRANSFER_DEFAULT,
+ offload ? &offloadInfo : nullptr, AttributionSourceState(), &attributes, false,
+ 1.0f, AUDIO_PORT_HANDLE_NONE);
status = track->initCheck();
sprintf(statusStr, "\n#### Test %u status %d\n", testCount, status);
write(outputFileFd, statusStr, strlen(statusStr));
@@ -141,11 +125,8 @@
return ret;
}
-}; // namespace android
+}; // namespace android
-
-int main(int argc, char **argv)
-{
+int main(int argc, char** argv) {
return android::main(argc, argv, android::testTrack);
}
-
diff --git a/media/libaudioclient/tests/test_create_utils.cpp b/media/libaudioclient/tests/test_create_utils.cpp
index caf5227..c2c2e8b 100644
--- a/media/libaudioclient/tests/test_create_utils.cpp
+++ b/media/libaudioclient/tests/test_create_utils.cpp
@@ -23,10 +23,10 @@
namespace android {
-int readLine(FILE *inputFile, char *line, int size) {
+int readLine(FILE* inputFile, char* line, int size) {
int ret = 0;
while (true) {
- char *str = fgets(line, size, inputFile);
+ char* str = fgets(line, size, inputFile);
if (str == nullptr) {
ret = -1;
break;
@@ -42,8 +42,7 @@
return ret;
}
-bool checkVersion(FILE *inputFile, const char *version)
-{
+bool checkVersion(FILE* inputFile, const char* version) {
char line[MAX_INPUT_FILE_LINE_LENGTH];
char versionKey[MAX_INPUT_FILE_LINE_LENGTH];
char versionValue[MAX_INPUT_FILE_LINE_LENGTH];
@@ -68,9 +67,8 @@
return true;
}
-int main(int argc, char **argv, test_func_t testFunc)
-{
- FILE *inputFile = nullptr;
+int main(int argc, char** argv, test_func_t testFunc) {
+ FILE* inputFile = nullptr;
int outputFileFd = STDOUT_FILENO;
mode_t mode = S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH;
int ret = 0;
@@ -96,7 +94,7 @@
if (strcmp(*argv, "-o") == 0) {
argv++;
if (*argv) {
- outputFileFd = open(*argv, O_WRONLY|O_CREAT, mode);
+ outputFileFd = open(*argv, O_WRONLY | O_CREAT, mode);
if (outputFileFd < 0) {
ret = 1;
}
@@ -126,5 +124,4 @@
return ret;
}
-}; // namespace android
-
+}; // namespace android
diff --git a/media/libaudioclient/tests/test_create_utils.h b/media/libaudioclient/tests/test_create_utils.h
index 9a6f9fa..110baf7 100644
--- a/media/libaudioclient/tests/test_create_utils.h
+++ b/media/libaudioclient/tests/test_create_utils.h
@@ -27,13 +27,12 @@
namespace android {
-int readLine(FILE *inputFile, char *line, int size);
+int readLine(FILE* inputFile, char* line, int size);
-bool checkVersion(FILE *inputFile, const char *version);
+bool checkVersion(FILE* inputFile, const char* version);
+typedef int (*test_func_t)(FILE* inputFile, int outputFileFd);
-typedef int (*test_func_t)(FILE *inputFile, int outputFileFd);
+int main(int argc, char** argv, test_func_t testFunc);
-int main(int argc, char **argv, test_func_t testFunc);
-
-}; // namespace android
+}; // namespace android
diff --git a/media/libaudioclient/tests/trackplayerbase_tests.cpp b/media/libaudioclient/tests/trackplayerbase_tests.cpp
new file mode 100644
index 0000000..c9b704d
--- /dev/null
+++ b/media/libaudioclient/tests/trackplayerbase_tests.cpp
@@ -0,0 +1,161 @@
+/*
+ * 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_TAG "TrackPlayerBaseTest"
+
+#include <gtest/gtest.h>
+
+#include <media/TrackPlayerBase.h>
+
+using namespace android;
+using namespace android::media;
+
+class TrackPlayer : public TrackPlayerBase, public AudioTrack::IAudioTrackCallback {
+ public:
+ // methods protected in base class
+ using TrackPlayerBase::playerPause;
+ using TrackPlayerBase::playerSetVolume;
+ using TrackPlayerBase::playerStart;
+ using TrackPlayerBase::playerStop;
+};
+
+class TrackPlayerBaseTest
+ : public ::testing::TestWithParam<std::tuple<double, double, uint32_t, uint32_t>> {
+ public:
+ TrackPlayerBaseTest()
+ : mDuration(std::get<0>(GetParam())),
+ mPulseFreq(std::get<1>(GetParam())),
+ mChannelCount(std::get<2>(GetParam())),
+ mSampleRate(std::get<3>(GetParam())){};
+
+ virtual void SetUp() override {
+ mFrameCount = mDuration * mSampleRate;
+ audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(mChannelCount);
+ sp<AudioTrack> track = new AudioTrack(mStreamType, mSampleRate, mFormat, channelMask,
+ mFrameCount, mFlags, nullptr /* callback */,
+ 0 /* notificationFrames */, AUDIO_SESSION_NONE);
+ ASSERT_EQ(track->initCheck(), NO_ERROR);
+
+ mPlayer = new TrackPlayer();
+ mPlayer->init(track.get(), mPlayer, PLAYER_TYPE_AAUDIO, AUDIO_USAGE_MEDIA,
+ AUDIO_SESSION_NONE);
+ sp<AudioTrack> playerTrack = mPlayer->mAudioTrack;
+ ASSERT_EQ(playerTrack->initCheck(), NO_ERROR);
+
+ mBufferSize = mFrameCount * playerTrack->frameSize();
+ mBuffer.resize(mBufferSize, 0);
+
+ // populate buffer
+ ASSERT_NE(mPulseFreq, 0);
+ int32_t nPulseSamples = mSampleRate / mPulseFreq;
+ int32_t pulseSize = nPulseSamples * playerTrack->frameSize();
+
+ int32_t marker = 0;
+ while (marker + pulseSize <= mBufferSize) {
+ memset(mBuffer.data() + marker, 127, pulseSize / 2);
+ marker += pulseSize;
+ }
+ }
+
+ void playBuffer() {
+ bool blocking = true;
+ ssize_t nbytes = mPlayer->mAudioTrack->write(mBuffer.data(), mBufferSize, blocking);
+ EXPECT_EQ(nbytes, mBufferSize) << "Did not write all data in blocking mode";
+ }
+
+ const double mDuration; // seconds
+ sp<TrackPlayer> mPlayer;
+
+ private:
+ const double mPulseFreq;
+ const uint32_t mChannelCount;
+ const uint32_t mSampleRate;
+
+ const audio_format_t mFormat = AUDIO_FORMAT_PCM_16_BIT;
+ const audio_output_flags_t mFlags = AUDIO_OUTPUT_FLAG_NONE;
+ const audio_stream_type_t mStreamType = AUDIO_STREAM_MUSIC;
+
+ int32_t mBufferSize;
+ int32_t mFrameCount;
+ std::vector<uint8_t> mBuffer;
+};
+
+class PlaybackTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(PlaybackTestParam, PlaybackTest) {
+ // no-op implementation
+ EXPECT_TRUE(mPlayer->setStartDelayMs(0).isOk());
+
+ ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+ ASSERT_NO_FATAL_FAILURE(playBuffer());
+ EXPECT_EQ(mPlayer->playerStop(), NO_ERROR);
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, PlaybackTestParam,
+ ::testing::Values(std::make_tuple(2.5, 25.0, 2, 48000)));
+
+class ChangeVolumeTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(ChangeVolumeTestParam, ChangeVolumeTest) {
+ float volume = 1.0f;
+ (void)mPlayer->setPlayerVolume(volume / 2, volume);
+
+ ASSERT_TRUE(mPlayer->start().isOk());
+ ASSERT_EQ(mPlayer->playerSetVolume(), NO_ERROR);
+
+ ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+ EXPECT_TRUE(mPlayer->stop().isOk());
+
+ std::vector<float> setVol = {0.95f, 0.05f, 0.5f, 0.25f, -1.0f, 1.0f, 1.0f};
+ std::vector<float> setPan = {0.0f, 0.0f, 1.0f, -1.0f, -1.0f, 0.5f, -0.5f};
+
+ ASSERT_TRUE(mPlayer->start().isOk());
+
+ for (int32_t i = 0; i < setVol.size(); i++) {
+ EXPECT_TRUE(mPlayer->setVolume(setVol[i]).isOk());
+ EXPECT_TRUE(mPlayer->setPan(setPan[i]).isOk());
+ ASSERT_NO_FATAL_FAILURE(playBuffer());
+ }
+ EXPECT_TRUE(mPlayer->stop().isOk());
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, ChangeVolumeTestParam,
+ ::testing::Values(std::make_tuple(1.0, 100.0, 1, 24000)));
+
+class PauseTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(PauseTestParam, PauseTest) {
+ ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+ ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+ ASSERT_EQ(mPlayer->playerPause(), NO_ERROR);
+ ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+
+ ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+ EXPECT_EQ(mPlayer->playerStop(), NO_ERROR);
+
+ for (int32_t i = 0; i < 5; i++) {
+ ASSERT_TRUE(mPlayer->start().isOk());
+ ASSERT_NO_FATAL_FAILURE(playBuffer());
+ ASSERT_TRUE(mPlayer->pause().isOk());
+ }
+ EXPECT_TRUE(mPlayer->stop().isOk());
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, PauseTestParam,
+ ::testing::Values(std::make_tuple(1.0, 75.0, 2, 24000)));
diff --git a/media/libaudiofoundation/include/media/AudioPort.h b/media/libaudiofoundation/include/media/AudioPort.h
index d6a098f..b1235f5 100644
--- a/media/libaudiofoundation/include/media/AudioPort.h
+++ b/media/libaudiofoundation/include/media/AudioPort.h
@@ -72,7 +72,7 @@
AudioProfileVector &getAudioProfiles() { return mProfiles; }
void setExtraAudioDescriptors(
- const std::vector<media::audio::common::ExtraAudioDescriptor> extraAudioDescriptors) {
+ const std::vector<media::audio::common::ExtraAudioDescriptor>& extraAudioDescriptors) {
mExtraAudioDescriptors = extraAudioDescriptors;
}
std::vector<media::audio::common::ExtraAudioDescriptor> &getExtraAudioDescriptors() {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 8546a7a..9351499 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -140,8 +140,6 @@
status_t AudioPolicyManager::setDeviceConnectionStateInt(
audio_policy_dev_state_t state, const android::media::audio::common::AudioPort& port,
audio_format_t encodedFormat) {
- // TODO: b/211601178 Forward 'port' to Audio HAL via mHwModules. For now, only device_type,
- // device_address and device_name are forwarded.
if (port.ext.getTag() != AudioPortExt::device) {
return BAD_VALUE;
}
@@ -160,7 +158,13 @@
sp<DeviceDescriptor> device = mHwModules.getDeviceDescriptor(
device_type, device_address.c_str(), device_name, encodedFormat,
state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
- return device ? setDeviceConnectionStateInt(device, state) : INVALID_OPERATION;
+ if (device == nullptr) {
+ return INVALID_OPERATION;
+ }
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ device->setExtraAudioDescriptors(port.extraAudioDescriptors);
+ }
+ return setDeviceConnectionStateInt(device, state);
}
status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t deviceType,
diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
index 70fdfcb..c7a60c2 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -127,7 +127,8 @@
attributionSource.packageName = "android";
attributionSource.token = sp<BBinder>::make();
sp<AudioEffect> fx = new AudioEffect(attributionSource);
- fx->set(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
+ fx->set(nullptr /*type */, &effect->mUuid, -1 /* priority */, nullptr /* callback */,
+ audioSession, input);
status_t status = fx->initCheck();
if (status != NO_ERROR && status != ALREADY_EXISTS) {
ALOGW("addInputEffects(): failed to create Fx %s on source %d",
@@ -279,7 +280,8 @@
attributionSource.packageName = "android";
attributionSource.token = sp<BBinder>::make();
sp<AudioEffect> fx = new AudioEffect(attributionSource);
- fx->set(NULL, &effect->mUuid, 0, 0, 0, audioSession, output);
+ fx->set(nullptr /* type */, &effect->mUuid, 0 /* priority */, nullptr /* callback */,
+ audioSession, output);
status_t status = fx->initCheck();
if (status != NO_ERROR && status != ALREADY_EXISTS) {
ALOGE("addOutputSessionEffects(): failed to create Fx %s on session %d",
@@ -984,8 +986,8 @@
attributionSource.packageName = "android";
attributionSource.token = sp<BBinder>::make();
sp<AudioEffect> fx = new AudioEffect(attributionSource);
- fx->set(EFFECT_UUID_NULL, &effectDesc->mUuid, 0, nullptr,
- nullptr, AUDIO_SESSION_DEVICE, AUDIO_IO_HANDLE_NONE,
+ fx->set(EFFECT_UUID_NULL, &effectDesc->mUuid, 0 /* priority */, nullptr /* callback */,
+ AUDIO_SESSION_DEVICE, AUDIO_IO_HANDLE_NONE,
AudioDeviceTypeAddr{deviceEffects->getDeviceType(),
deviceEffects->getDeviceAddress()});
status_t status = fx->initCheck();
diff --git a/services/audiopolicy/tests/Android.bp b/services/audiopolicy/tests/Android.bp
index 2e220bc..e887798 100644
--- a/services/audiopolicy/tests/Android.bp
+++ b/services/audiopolicy/tests/Android.bp
@@ -30,6 +30,8 @@
],
static_libs: [
+ "android.media.audio.common.types-V1-cpp",
+ "audioclient-types-aidl-cpp",
"libaudiopolicycomponents",
"libgmock",
],
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index 057fa58..96f58d2 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -103,8 +103,12 @@
++mAudioPortListUpdateCount;
}
- status_t setDeviceConnectedState(
- const struct audio_port_v7 *port __unused, bool connected __unused) override {
+ status_t setDeviceConnectedState(const struct audio_port_v7 *port, bool connected) override {
+ if (connected) {
+ mConnectedDevicePorts.push_back(*port);
+ } else {
+ mDisconnectedDevicePorts.push_back(*port);
+ }
return NO_ERROR;
}
@@ -150,6 +154,30 @@
return NO_ERROR;
}
+ size_t getConnectedDevicePortCount() const {
+ return mConnectedDevicePorts.size();
+ }
+
+ const struct audio_port_v7 *getLastConnectedDevicePort() const {
+ if (mConnectedDevicePorts.empty()) {
+ return nullptr;
+ }
+ auto it = --mConnectedDevicePorts.end();
+ return &(*it);
+ }
+
+ size_t getDisconnectedDevicePortCount() const {
+ return mDisconnectedDevicePorts.size();
+ }
+
+ const struct audio_port_v7 *getLastDisconnectedDevicePort() const {
+ if (mDisconnectedDevicePorts.empty()) {
+ return nullptr;
+ }
+ auto it = --mDisconnectedDevicePorts.end();
+ return &(*it);
+ }
+
private:
audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
@@ -158,6 +186,8 @@
std::set<std::string> mAllowedModuleNames;
size_t mAudioPortListUpdateCount = 0;
size_t mRoutingUpdatedUpdateCount = 0;
+ std::vector<struct audio_port_v7> mConnectedDevicePorts;
+ std::vector<struct audio_port_v7> mDisconnectedDevicePorts;
};
} // namespace android
diff --git a/services/audiopolicy/tests/AudioPolicyTestManager.h b/services/audiopolicy/tests/AudioPolicyTestManager.h
index 7441f20..2a7a060 100644
--- a/services/audiopolicy/tests/AudioPolicyTestManager.h
+++ b/services/audiopolicy/tests/AudioPolicyTestManager.h
@@ -37,6 +37,7 @@
using AudioPolicyManager::getDirectProfilesForAttributes;
using AudioPolicyManager::setDeviceConnectionState;
using AudioPolicyManager::deviceToAudioPort;
+ using AudioPolicyManager::handleDeviceConfigChange;
uint32_t getAudioPortGeneration() const { return mAudioPortGeneration; }
};
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 43b1a2a..bb00c48 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -919,6 +919,30 @@
EXPECT_TRUE(foundVoipTx);
}
+TEST_F(AudioPolicyManagerTestWithConfigurationFile, HandleDeviceConfigChange) {
+ {
+ const auto prevCounter = mClient->getRoutingUpdatedCounter();
+
+ EXPECT_EQ(NO_ERROR, mManager->setDeviceConnectionState(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ "", "", AUDIO_FORMAT_LDAC));
+ const auto currCounter = mClient->getRoutingUpdatedCounter();
+ EXPECT_GT(currCounter, prevCounter);
+ }
+ {
+ const auto prevCounter = mClient->getRoutingUpdatedCounter();
+ // Update device configuration
+ EXPECT_EQ(NO_ERROR, mManager->handleDeviceConfigChange(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+ "" /*address*/, "" /*name*/,
+ AUDIO_FORMAT_AAC));
+
+ // As mClient marks isReconfigA2dpSupported to false, device state needs to be toggled for
+ // config changes to take effect
+ const auto currCounter = mClient->getRoutingUpdatedCounter();
+ EXPECT_GT(currCounter, prevCounter);
+ }
+}
+
using PolicyMixTuple = std::tuple<audio_usage_t, audio_source_t, uint32_t>;
class AudioPolicyManagerTestDynamicPolicy : public AudioPolicyManagerTestWithConfigurationFile {
@@ -1700,6 +1724,45 @@
address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
}
+android::media::audio::common::ExtraAudioDescriptor make_ExtraAudioDescriptor(
+ android::media::audio::common::AudioStandard audioStandard,
+ android::media::audio::common::AudioEncapsulationType audioEncapsulationType) {
+ android::media::audio::common::ExtraAudioDescriptor result;
+ result.standard = audioStandard;
+ result.audioDescriptor = {0xb4, 0xaf, 0x98, 0x1a};
+ result.encapsulationType = audioEncapsulationType;
+ return result;
+}
+
+TEST_P(AudioPolicyManagerTestDeviceConnection, PassingExtraAudioDescriptors) {
+ const audio_devices_t type = std::get<0>(GetParam());
+ if (!audio_device_is_digital(type)) {
+ // EADs are used only for HDMI devices.
+ GTEST_SKIP() << "Not a digital device type: " << audio_device_to_string(type);
+ }
+ const std::string name = std::get<1>(GetParam());
+ const std::string address = std::get<2>(GetParam());
+ android::media::AudioPort audioPort;
+ ASSERT_EQ(NO_ERROR,
+ mManager->deviceToAudioPort(type, address.c_str(), name.c_str(), &audioPort));
+ android::media::audio::common::AudioPort& port = audioPort.hal;
+ port.extraAudioDescriptors.push_back(make_ExtraAudioDescriptor(
+ android::media::audio::common::AudioStandard::EDID,
+ android::media::audio::common::AudioEncapsulationType::IEC61937));
+ const size_t lastConnectedDevicePortCount = mClient->getConnectedDevicePortCount();
+ const size_t lastDisconnectedDevicePortCount = mClient->getDisconnectedDevicePortCount();
+ EXPECT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE, port, AUDIO_FORMAT_DEFAULT));
+ EXPECT_EQ(lastConnectedDevicePortCount + 1, mClient->getConnectedDevicePortCount());
+ EXPECT_EQ(lastDisconnectedDevicePortCount, mClient->getDisconnectedDevicePortCount());
+ const audio_port_v7* devicePort = mClient->getLastConnectedDevicePort();
+ EXPECT_EQ(port.extraAudioDescriptors.size(), devicePort->num_extra_audio_descriptors);
+ EXPECT_EQ(AUDIO_STANDARD_EDID, devicePort->extra_audio_descriptors[0].standard);
+ EXPECT_EQ(AUDIO_ENCAPSULATION_TYPE_IEC61937,
+ devicePort->extra_audio_descriptors[0].encapsulation_type);
+ EXPECT_NE(0, devicePort->extra_audio_descriptors[0].descriptor[0]);
+}
+
INSTANTIATE_TEST_CASE_P(
DeviceConnectionState,
AudioPolicyManagerTestDeviceConnection,
diff --git a/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
index 5e1822a..d342aea 100644
--- a/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
+++ b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
@@ -71,6 +71,9 @@
<devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET"
role="source" address="hfp_client_in">
</devicePort>
+ <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
+ encodedFormats="AUDIO_FORMAT_LDAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_AAC AUDIO_FORMAT_SBC">
+ </devicePort>
</devicePorts>
<routes>
<route type="mix" sink="Speaker"
@@ -85,6 +88,8 @@
sources="mixport_bt_hfp_output,voip_rx"/>
<route type="mix" sink="mixport_bt_hfp_input"
sources="BT SCO Headset Mic"/>
+ <route type="mix" sink="BT A2DP Out"
+ sources="primary output"/>
</routes>
</module>