Merge changes I3200e32e,I9ac3db0c,I5530dfe4,I317e2cbc,I501a5956, ...
* changes:
AudioAidlLegacyConversionTest: Added test for Aidl2Legacy2Aidl.
AudioAidlLegacyConversionTest: Added test for Aidl2Legacy2Aidl.
Updates to audiotrack tests
Update AudioEffect callers to new interface
Add tests to libaudioclient pre- and post-submit
Remove legacy callback interfaces
Update audiorecord tests
Skip test if chosen effect is not available on device
Add audio routing tests
Add unit tests for trackPlayerBase
Add serialization tests for libaudioclient
Add audioeffect unit tests
Add audiotrack unit tests
Add audiorecord unit tests
Enable clang format hook for media/libaudioclient/tests
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/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();