Snap for 11704135 from f37f18e399674f792a084947844635ec652a5d4d to 24Q3-release (97f78f36) · Commits · e / os / android_frameworks_av

media/libaudioclient/AudioRecord.cpp

+2 −15

Original line number	Diff line number	Diff line
		@@ -130,11 +130,7 @@ status_t AudioRecord::getMetrics(mediametrics::Item * &item)
		}

		AudioRecord::AudioRecord(const AttributionSourceState &client)
		: mActive(false), mStatus(NO_INIT), mClientAttributionSource(client),
		mSessionId(AUDIO_SESSION_ALLOCATE), mPreviousPriority(ANDROID_PRIORITY_NORMAL),
		mPreviousSchedulingGroup(SP_DEFAULT), mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
		mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE), mSelectedMicDirection(MIC_DIRECTION_UNSPECIFIED),
		mSelectedMicFieldDimension(MIC_FIELD_DIMENSION_DEFAULT)
		: mClientAttributionSource(client)
		{
		}

		@@ -154,13 +150,7 @@ AudioRecord::AudioRecord(
		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(nullptr)
		: mClientAttributionSource(client)
		{
		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));
		@@ -199,9 +189,6 @@ AudioRecord::~AudioRecord()
		}

		void AudioRecord::stopAndJoinCallbacks() {
		// Prevent nullptr crash if it did not open properly.
		if (mStatus != NO_ERROR) return;

		// Make sure that callback function exits in the case where
		// it is looping on buffer empty condition in obtainBuffer().
		// Otherwise the callback thread will never exit.

media/libaudioclient/include/media/AudioRecord.h

+16 −13

Original line number	Diff line number	Diff line
		@@ -681,7 +681,7 @@ private:

		// Current client state: false = stopped, true = active. Protected by mLock. If more states
		// are added, consider changing this to enum State { ... } mState as in AudioTrack.
		bool mActive;
		bool mActive = false;

		// for client callback handler

		@@ -708,7 +708,7 @@ private:
		Modulo<uint32_t> mNewPosition; // in frames
		uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS

		status_t mStatus;
		status_t mStatus = NO_INIT;

		android::content::AttributionSourceState mClientAttributionSource; // Owner's attribution source

		@@ -736,8 +736,8 @@ private:
		// held to read or write those bits reliably.
		audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const

		audio_session_t mSessionId;
		audio_port_handle_t mPortId; // Id from Audio Policy Manager
		audio_session_t mSessionId = AUDIO_SESSION_ALLOCATE;
		audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE;

		/**
		* mLogSessionId is a string identifying this AudioRecord for the metrics service.
		@@ -756,9 +756,9 @@ private:
		sp<IMemory> mBufferMemory;
		audio_io_handle_t mInput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getInputforAttr()

		int mPreviousPriority; // before start()
		SchedPolicy mPreviousSchedulingGroup;
		bool mAwaitBoost; // thread should wait for priority boost before running
		int mPreviousPriority = ANDROID_PRIORITY_NORMAL; // before start()
		SchedPolicy mPreviousSchedulingGroup = SP_DEFAULT;
		bool mAwaitBoost = false; // thread should wait for priority boost before running

		// The proxy should only be referenced while a lock is held because the proxy isn't
		// multi-thread safe.
		@@ -799,14 +799,17 @@ private:

		// For Device Selection API
		// a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing.
		audio_port_handle_t mSelectedDeviceId; // Device requested by the application.
		audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager:
		// May not match the app selection depending on other
		// activity and connected devices

		// Device requested by the application.
		audio_port_handle_t mSelectedDeviceId = AUDIO_PORT_HANDLE_NONE;
		// Device actually selected by AudioPolicyManager: This may not match the app
		// selection depending on other activity and connected devices
		audio_port_handle_t mRoutedDeviceId = AUDIO_PORT_HANDLE_NONE;

		wp<AudioSystem::AudioDeviceCallback> mDeviceCallback;

		audio_microphone_direction_t mSelectedMicDirection;
		float mSelectedMicFieldDimension;
		audio_microphone_direction_t mSelectedMicDirection = MIC_DIRECTION_UNSPECIFIED;
		float mSelectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT;

		int32_t mMaxSharedAudioHistoryMs = 0;
		std::string mSharedAudioPackageName = {};

media/libaudioclient/tests/audiorecord_tests.cpp

+19 −0

Original line number	Diff line number	Diff line
		@@ -28,6 +28,25 @@

		using namespace android;

		// Test that the basic constructor returns an object that doesn't crash
		// on stop() or destruction.

		TEST(AudioRecordTestBasic, EmptyAudioRecord) {
		AttributionSourceState attributionSource;
		attributionSource.packageName = "AudioRecordTest";
		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();
		const auto ar = sp<AudioRecord>::make(attributionSource);

		// test key commands on an unset AudioRecord.
		EXPECT_EQ(NO_INIT, ar->initCheck());
		EXPECT_EQ(true, ar->stopped());

		// just don't crash.
		ar->stop();
		}

		class AudioRecordTest : public ::testing::Test {
		public:
		void SetUp() override {

media/libaudiohal/impl/StreamHalAidl.cpp

+8 −4

Original line number	Diff line number	Diff line
		@@ -415,8 +415,10 @@ status_t StreamHalAidl::exit() {

		void StreamHalAidl::onAsyncTransferReady() {
		if (auto state = getState(); state == StreamDescriptor::State::TRANSFERRING) {
		// Retrieve the current state together with position counters.
		updateCountersIfNeeded();
		// Retrieve the current state together with position counters unconditionally
		// to ensure that the state on our side gets updated.
		sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
		nullptr, true /safeFromNonWorkerThread /);
		} else {
		ALOGW("%s: unexpected onTransferReady in the state %s", __func__, toString(state).c_str());
		}
		@@ -424,8 +426,10 @@ void StreamHalAidl::onAsyncTransferReady() {

		void StreamHalAidl::onAsyncDrainReady() {
		if (auto state = getState(); state == StreamDescriptor::State::DRAINING) {
		// Retrieve the current state together with position counters.
		updateCountersIfNeeded();
		// Retrieve the current state together with position counters unconditionally
		// to ensure that the state on our side gets updated.
		sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
		nullptr, true /safeFromNonWorkerThread /);
		} else {
		ALOGW("%s: unexpected onDrainReady in the state %s", __func__, toString(state).c_str());
		}

media/libeffects/lvm/wrapper/Reverb/aidl/ReverbContext.cpp

+46 −29

Original line number	Diff line number	Diff line
		@@ -353,46 +353,63 @@ IEffect::Status ReverbContext::process(float* in, float* out, int samples) {
		return status;
		}

		std::vector<float> inFrames(samples);
		std::vector<float> outFrames(frameCount * FCC_2);
		std::vector<float> inputSamples;
		std::vector<float> outputSamples(frameCount * FCC_2);

		if (isPreset() && mNextPreset != mPreset) {
		loadPreset();
		}

		if (isAuxiliary()) {
		inFrames.assign(in, in + samples);
		inputSamples.resize(samples);
		inputSamples.assign(in, in + samples);
		} else {
		// mono input is duplicated
		// Resizing to stereo is required to duplicate mono input
		inputSamples.resize(frameCount * FCC_2);
		if (channels >= FCC_2) {
		for (int i = 0; i < frameCount; i++) {
		inFrames[FCC_2 * i] = in[channels * i] * kSendLevel;
		inFrames[FCC_2 * i + 1] = in[channels * i + 1] * kSendLevel;
		inputSamples[FCC_2 * i] = in[channels * i] * kSendLevel;
		inputSamples[FCC_2 * i + 1] = in[channels * i + 1] * kSendLevel;
		}
		} else {
		for (int i = 0; i < frameCount; i++) {
		inFrames[FCC_2 * i] = inFrames[FCC_2 * i + 1] = in[i] * kSendLevel;
		inputSamples[FCC_2 * i] = inputSamples[FCC_2 * i + 1] = in[i] * kSendLevel;
		}
		}
		}

		if (isPreset() && mPreset == PresetReverb::Presets::NONE) {
		std::fill(outFrames.begin(), outFrames.end(), 0); // always stereo here
		std::fill(outputSamples.begin(), outputSamples.end(), 0); // always stereo here
		} else {
		if (!mEnabled && mSamplesToExitCount > 0) {
		std::fill(outFrames.begin(), outFrames.end(), 0);
		std::fill(outputSamples.begin(), outputSamples.end(), 0);
		}
		int inputBufferIndex = 0;
		int outputBufferIndex = 0;

		// LVREV library supports max of int16_t frames at a time
		constexpr int kMaxBlockFrames = std::numeric_limits<int16_t>::max();
		const auto inputFrameSize = getInputFrameSize();
		const auto outputFrameSize = getOutputFrameSize();

		/* Process the samples, producing a stereo output */
		for (int fc = frameCount; fc > 0;) {
		int processFrames = std::min(fc, kMaxBlockFrames);
		LVREV_ReturnStatus_en lvrevStatus =
		LVREV_Process(mInstance, /* Instance handle */
		inFrames.data(), /* Input buffer */
		outFrames.data(), /* Output buffer */
		frameCount); /* Number of samples to read */
		inputSamples.data() + inputBufferIndex, /* Input buffer */
		outputSamples.data() + outputBufferIndex, /* Output buffer */
		processFrames); /* Number of samples to process */
		if (lvrevStatus != LVREV_SUCCESS) {
		LOG(ERROR) << __func__ << " LVREV_Process error: " << lvrevStatus;
		return {EX_UNSUPPORTED_OPERATION, 0, 0};
		}

		fc -= processFrames;

		inputBufferIndex += processFrames * inputFrameSize / sizeof(float);
		outputBufferIndex += processFrames * outputFrameSize / sizeof(float);
		}
		}
		// Convert to 16 bits
		if (isAuxiliary()) {
		@@ -401,14 +418,14 @@ IEffect::Status ReverbContext::process(float* in, float* out, int samples) {
		if (channels >= FCC_2) {
		for (int i = 0; i < frameCount; i++) {
		// Mix with dry input
		outFrames[FCC_2 * i] += in[channels * i];
		outFrames[FCC_2 * i + 1] += in[channels * i + 1];
		outputSamples[FCC_2 * i] += in[channels * i];
		outputSamples[FCC_2 * i + 1] += in[channels * i + 1];
		}
		} else {
		for (int i = 0; i < frameCount; i++) {
		// Mix with dry input
		outFrames[FCC_2 * i] += in[i];
		outFrames[FCC_2 * i + 1] += in[i];
		outputSamples[FCC_2 * i] += in[i];
		outputSamples[FCC_2 * i + 1] += in[i];
		}
		}

		@@ -420,8 +437,8 @@ IEffect::Status ReverbContext::process(float* in, float* out, int samples) {
		float incr = (mVolume.right - vr) / frameCount;

		for (int i = 0; i < frameCount; i++) {
		outFrames[FCC_2 * i] *= vl;
		outFrames[FCC_2 * i + 1] *= vr;
		outputSamples[FCC_2 * i] *= vl;
		outputSamples[FCC_2 * i + 1] *= vr;

		vl += incl;
		vr += incr;
		@@ -430,8 +447,8 @@ IEffect::Status ReverbContext::process(float* in, float* out, int samples) {
		} else if (volumeMode != VOLUME_OFF) {
		if (mVolume.left != kUnitVolume \|\| mVolume.right != kUnitVolume) {
		for (int i = 0; i < frameCount; i++) {
		outFrames[FCC_2 * i] *= mVolume.left;
		outFrames[FCC_2 * i + 1] *= mVolume.right;
		outputSamples[FCC_2 * i] *= mVolume.left;
		outputSamples[FCC_2 * i + 1] *= mVolume.right;
		}
		}
		mPrevVolume = mVolume;
		@@ -441,8 +458,8 @@ IEffect::Status ReverbContext::process(float* in, float* out, int samples) {

		if (outChannels > 2) {
		for (int i = 0; i < frameCount; i++) {
		out[outChannels * i] = outFrames[FCC_2 * i];
		out[outChannels * i + 1] = outFrames[FCC_2 * i + 1];
		out[outChannels * i] = outputSamples[FCC_2 * i];
		out[outChannels * i + 1] = outputSamples[FCC_2 * i + 1];
		}
		if (!isAuxiliary()) {
		for (int i = 0; i < frameCount; i++) {
		@@ -454,10 +471,10 @@ IEffect::Status ReverbContext::process(float* in, float* out, int samples) {
		}
		} else {
		if (outChannels == FCC_1) {
		From2iToMono_Float(outFrames.data(), out, frameCount);
		From2iToMono_Float(outputSamples.data(), out, frameCount);
		} else {
		for (int i = 0; i < frameCount * FCC_2; i++) {
		out[i] = outFrames[i];
		out[i] = outputSamples[i];
		}
		}
		}