aaudio: fix CTS for MMAP mode am: ec89b2e2f8 (5af194cf) · Commits · e / os / android_frameworks_av

media/libaaudio/src/binding/IAAudioService.cpp

+1 −2

Original line number	Diff line number	Diff line
		@@ -45,11 +45,9 @@ public:
		Parcel data, reply;
		// send command
		data.writeInterfaceToken(IAAudioService::getInterfaceDescriptor());
		ALOGV("BpAAudioService::client openStream --------------------");
		// request.dump();
		request.writeToParcel(&data);
		status_t err = remote()->transact(OPEN_STREAM, data, &reply);
		ALOGV("BpAAudioService::client openStream returned %d", err);
		if (err != NO_ERROR) {
		ALOGE("BpAAudioService::client openStream transact failed %d", err);
		return AAudioConvert_androidToAAudioResult(err);
		@@ -57,6 +55,7 @@ public:
		// parse reply
		aaudio_handle_t stream;
		err = reply.readInt32(&stream);
		ALOGD("BpAAudioService::client openStream returned stream = 0x%08x", stream);
		if (err != NO_ERROR) {
		ALOGE("BpAAudioService::client transact(OPEN_STREAM) readInt %d", err);
		return AAudioConvert_androidToAAudioResult(err);

media/libaaudio/src/client/AudioEndpoint.cpp

+4 −0

Original line number	Diff line number	Diff line
		@@ -246,3 +246,7 @@ int32_t AudioEndpoint::getBufferCapacityInFrames() const
		return (int32_t)mDataQueue->getBufferCapacityInFrames();
		}

		void AudioEndpoint::dump() const {
		ALOGD("AudioEndpoint: data readCounter = %lld", (long long) mDataQueue->getReadCounter());
		ALOGD("AudioEndpoint: data writeCounter = %lld", (long long) mDataQueue->getWriteCounter());
		}

media/libaaudio/src/client/AudioEndpoint.h

+2 −0

Original line number	Diff line number	Diff line
		@@ -91,6 +91,8 @@ public:

		int32_t getBufferCapacityInFrames() const;

		void dump() const;

		private:
		android::FifoBuffer *mUpCommandQueue;
		android::FifoBuffer *mDataQueue;

media/libaaudio/src/client/AudioStreamInternal.cpp

+54 −41

Original line number	Diff line number	Diff line
		@@ -63,8 +63,8 @@ AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterfa
		, mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
		, mFramesPerBurst(16)
		, mStreamVolume(1.0f)
		, mServiceInterface(serviceInterface)
		, mInService(inService) {
		, mInService(inService)
		, mServiceInterface(serviceInterface) {
		}

		AudioStreamInternal::~AudioStreamInternal() {
		@@ -154,13 +154,13 @@ aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
		if (getDataCallbackProc()) {
		mCallbackFrames = builder.getFramesPerDataCallback();
		if (mCallbackFrames > getBufferCapacity() / 2) {
		ALOGE("AudioStreamInternal.open(): framesPerCallback too large = %d, capacity = %d",
		ALOGE("AudioStreamInternal::open(): framesPerCallback too big = %d, capacity = %d",
		mCallbackFrames, getBufferCapacity());
		mServiceInterface.closeStream(mServiceStreamHandle);
		return AAUDIO_ERROR_OUT_OF_RANGE;

		} else if (mCallbackFrames < 0) {
		ALOGE("AudioStreamInternal.open(): framesPerCallback negative");
		ALOGE("AudioStreamInternal::open(): framesPerCallback negative");
		mServiceInterface.closeStream(mServiceStreamHandle);
		return AAUDIO_ERROR_OUT_OF_RANGE;

		@@ -185,7 +185,7 @@ aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
		}

		aaudio_result_t AudioStreamInternal::close() {
		ALOGD("AudioStreamInternal.close(): mServiceStreamHandle = 0x%08X",
		ALOGD("AudioStreamInternal::close(): mServiceStreamHandle = 0x%08X",
		mServiceStreamHandle);
		if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) {
		// Don't close a stream while it is running.
		@@ -201,12 +201,14 @@ aaudio_result_t AudioStreamInternal::close() {
		result, AAudio_convertResultToText(result));
		}
		}
		setState(AAUDIO_STREAM_STATE_CLOSING);
		aaudio_handle_t serviceStreamHandle = mServiceStreamHandle;
		mServiceStreamHandle = AAUDIO_HANDLE_INVALID;

		mServiceInterface.closeStream(serviceStreamHandle);
		delete[] mCallbackBuffer;
		mCallbackBuffer = nullptr;
		setState(AAUDIO_STREAM_STATE_CLOSED);
		return mEndPointParcelable.close();
		} else {
		return AAUDIO_ERROR_INVALID_HANDLE;
		@@ -228,16 +230,21 @@ static void aaudio_callback_thread_proc(void context)
		aaudio_result_t AudioStreamInternal::requestStart()
		{
		int64_t startTime;
		ALOGD("AudioStreamInternal(): start()");
		ALOGD("AudioStreamInternal()::requestStart()");
		if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
		return AAUDIO_ERROR_INVALID_STATE;
		}
		if (isActive()) {
		return AAUDIO_ERROR_INVALID_STATE;
		}
		aaudio_stream_state_t originalState = getState();

		startTime = AudioClock::getNanoseconds();
		mClockModel.start(startTime);
		setState(AAUDIO_STREAM_STATE_STARTING);
		aaudio_result_t result = AAudioConvert_androidToAAudioResult(startWithStatus());

		startTime = AudioClock::getNanoseconds();
		mClockModel.start(startTime);

		if (result == AAUDIO_OK && getDataCallbackProc() != nullptr) {
		// Launch the callback loop thread.
		int64_t periodNanos = mCallbackFrames
		@@ -246,6 +253,9 @@ aaudio_result_t AudioStreamInternal::requestStart()
		mCallbackEnabled.store(true);
		result = createThread(periodNanos, aaudio_callback_thread_proc, this);
		}
		if (result != AAUDIO_OK) {
		setState(originalState);
		}
		return result;
		}

		@@ -279,7 +289,7 @@ aaudio_result_t AudioStreamInternal::stopCallback()
		aaudio_result_t AudioStreamInternal::requestPauseInternal()
		{
		if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
		ALOGE("AudioStreamInternal(): requestPauseInternal() mServiceStreamHandle invalid = 0x%08X",
		ALOGE("AudioStreamInternal::requestPauseInternal() mServiceStreamHandle invalid = 0x%08X",
		mServiceStreamHandle);
		return AAUDIO_ERROR_INVALID_STATE;
		}
		@@ -301,7 +311,7 @@ aaudio_result_t AudioStreamInternal::requestPause()

		aaudio_result_t AudioStreamInternal::requestFlush() {
		if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
		ALOGE("AudioStreamInternal(): requestFlush() mServiceStreamHandle invalid = 0x%08X",
		ALOGE("AudioStreamInternal::requestFlush() mServiceStreamHandle invalid = 0x%08X",
		mServiceStreamHandle);
		return AAUDIO_ERROR_INVALID_STATE;
		}
		@@ -312,13 +322,14 @@ aaudio_result_t AudioStreamInternal::requestFlush() {

		// TODO for Play only
		void AudioStreamInternal::onFlushFromServer() {
		ALOGD("AudioStreamInternal(): onFlushFromServer()");
		int64_t readCounter = mAudioEndpoint.getDataReadCounter();
		int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();

		// Bump offset so caller does not see the retrograde motion in getFramesRead().
		int64_t framesFlushed = writeCounter - readCounter;
		mFramesOffsetFromService += framesFlushed;
		ALOGD("AudioStreamInternal::onFlushFromServer() readN = %lld, writeN = %lld, offset = %lld",
		(long long)readCounter, (long long)writeCounter, (long long)mFramesOffsetFromService);

		// Flush written frames by forcing writeCounter to readCounter.
		// This is because we cannot move the read counter in the hardware.
		@@ -328,7 +339,7 @@ void AudioStreamInternal::onFlushFromServer() {
		aaudio_result_t AudioStreamInternal::requestStopInternal()
		{
		if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
		ALOGE("AudioStreamInternal(): requestStopInternal() mServiceStreamHandle invalid = 0x%08X",
		ALOGE("AudioStreamInternal::requestStopInternal() mServiceStreamHandle invalid = 0x%08X",
		mServiceStreamHandle);
		return AAUDIO_ERROR_INVALID_STATE;
		}
		@@ -370,7 +381,7 @@ aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId,
		int64_t *timeNanoseconds) {
		// TODO Generate in server and pass to client. Return latest.
		int64_t time = AudioClock::getNanoseconds();
		*framePosition = mClockModel.convertTimeToPosition(time);
		*framePosition = mClockModel.convertTimeToPosition(time) + mFramesOffsetFromService;
		// TODO Get a more accurate timestamp from the service. This code just adds a fudge factor.
		timeNanoseconds = time + (6 AAUDIO_NANOS_PER_MILLISECOND);
		return AAUDIO_OK;
		@@ -383,31 +394,28 @@ aaudio_result_t AudioStreamInternal::updateStateWhileWaiting() {
		return processCommands();
		}

		#if LOG_TIMESTAMPS
		static void AudioStreamInternal_logTimestamp(AAudioServiceMessage &command) {
		void AudioStreamInternal::logTimestamp(AAudioServiceMessage &command) {
		static int64_t oldPosition = 0;
		static int64_t oldTime = 0;
		int64_t framePosition = command.timestamp.position;
		int64_t nanoTime = command.timestamp.timestamp;
		ALOGD("AudioStreamInternal() timestamp says framePosition = %08lld at nanoTime %lld",
		ALOGD("AudioStreamInternal: timestamp says framePosition = %08lld at nanoTime %lld",
		(long long) framePosition,
		(long long) nanoTime);
		int64_t nanosDelta = nanoTime - oldTime;
		if (nanosDelta > 0 && oldTime > 0) {
		int64_t framesDelta = framePosition - oldPosition;
		int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta;
		ALOGD("AudioStreamInternal() - framesDelta = %08lld", (long long) framesDelta);
		ALOGD("AudioStreamInternal() - nanosDelta = %08lld", (long long) nanosDelta);
		ALOGD("AudioStreamInternal() - measured rate = %lld", (long long) rate);
		ALOGD("AudioStreamInternal: framesDelta = %08lld, nanosDelta = %08lld, rate = %lld",
		(long long) framesDelta, (long long) nanosDelta, (long long) rate);
		}
		oldPosition = framePosition;
		oldTime = nanoTime;
		}
		#endif

		aaudio_result_t AudioStreamInternal::onTimestampFromServer(AAudioServiceMessage *message) {
		#if LOG_TIMESTAMPS
		AudioStreamInternal_logTimestamp(*message);
		logTimestamp(*message);
		#endif
		processTimestamp(message->timestamp.position, message->timestamp.timestamp);
		return AAUDIO_OK;
		@@ -417,47 +425,48 @@ aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *mes
		aaudio_result_t result = AAUDIO_OK;
		switch (message->event.event) {
		case AAUDIO_SERVICE_EVENT_STARTED:
		ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STARTED");
		ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_STARTED");
		if (getState() == AAUDIO_STREAM_STATE_STARTING) {
		setState(AAUDIO_STREAM_STATE_STARTED);
		}
		break;
		case AAUDIO_SERVICE_EVENT_PAUSED:
		ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_PAUSED");
		ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_PAUSED");
		if (getState() == AAUDIO_STREAM_STATE_PAUSING) {
		setState(AAUDIO_STREAM_STATE_PAUSED);
		}
		break;
		case AAUDIO_SERVICE_EVENT_STOPPED:
		ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STOPPED");
		ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_STOPPED");
		if (getState() == AAUDIO_STREAM_STATE_STOPPING) {
		setState(AAUDIO_STREAM_STATE_STOPPED);
		}
		break;
		case AAUDIO_SERVICE_EVENT_FLUSHED:
		ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_FLUSHED");
		ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_FLUSHED");
		if (getState() == AAUDIO_STREAM_STATE_FLUSHING) {
		setState(AAUDIO_STREAM_STATE_FLUSHED);
		onFlushFromServer();
		}
		break;
		case AAUDIO_SERVICE_EVENT_CLOSED:
		ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_CLOSED");
		ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_CLOSED");
		setState(AAUDIO_STREAM_STATE_CLOSED);
		break;
		case AAUDIO_SERVICE_EVENT_DISCONNECTED:
		result = AAUDIO_ERROR_DISCONNECTED;
		setState(AAUDIO_STREAM_STATE_DISCONNECTED);
		ALOGW("WARNING - processCommands() AAUDIO_SERVICE_EVENT_DISCONNECTED");
		ALOGW("WARNING - AudioStreamInternal::onEventFromServer()"
		" AAUDIO_SERVICE_EVENT_DISCONNECTED");
		break;
		case AAUDIO_SERVICE_EVENT_VOLUME:
		mStreamVolume = (float)message->event.dataDouble;
		doSetVolume();
		ALOGD("processCommands() AAUDIO_SERVICE_EVENT_VOLUME %lf",
		ALOGD("AudioStreamInternal::onEventFromServer() AAUDIO_SERVICE_EVENT_VOLUME %lf",
		message->event.dataDouble);
		break;
		default:
		ALOGW("WARNING - processCommands() Unrecognized event = %d",
		ALOGW("WARNING - AudioStreamInternal::onEventFromServer() Unrecognized event = %d",
		(int) message->event.event);
		break;
		}
		@@ -499,27 +508,27 @@ aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames
		{
		const char * traceName = (mInService) ? "aaWrtS" : "aaWrtC";
		ATRACE_BEGIN(traceName);
		aaudio_result_t result = AAUDIO_OK;
		int32_t loopCount = 0;
		uint8_t* audioData = (uint8_t*)buffer;
		int64_t currentTimeNanos = AudioClock::getNanoseconds();
		int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
		int32_t framesLeft = numFrames;

		int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
		if (ATRACE_ENABLED()) {
		const char * traceName = (mInService) ? "aaFullS" : "aaFullC";
		ATRACE_INT(traceName, fullFrames);
		}

		aaudio_result_t result = AAUDIO_OK;
		int32_t loopCount = 0;
		uint8_t* audioData = (uint8_t*)buffer;
		int64_t currentTimeNanos = AudioClock::getNanoseconds();
		const int64_t entryTimeNanos = currentTimeNanos;
		const int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
		int32_t framesLeft = numFrames;

		// Loop until all the data has been processed or until a timeout occurs.
		while (framesLeft > 0) {
		// The call to processDataNow() will not block. It will just read as much as it can.
		// The call to processDataNow() will not block. It will just process as much as it can.
		int64_t wakeTimeNanos = 0;
		aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft,
		currentTimeNanos, &wakeTimeNanos);
		if (framesProcessed < 0) {
		ALOGE("AudioStreamInternal::processData() loop: framesProcessed = %d", framesProcessed);
		result = framesProcessed;
		break;
		}
		@@ -537,12 +546,16 @@ aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames
		if (wakeTimeNanos > deadlineNanos) {
		// If we time out, just return the framesWritten so far.
		// TODO remove after we fix the deadline bug
		ALOGE("AudioStreamInternal::processData(): timed out after %lld nanos",
		ALOGW("AudioStreamInternal::processData(): entered at %lld nanos, currently %lld",
		(long long) entryTimeNanos, (long long) currentTimeNanos);
		ALOGW("AudioStreamInternal::processData(): timed out after %lld nanos",
		(long long) timeoutNanoseconds);
		ALOGE("AudioStreamInternal::processData(): wakeTime = %lld, deadline = %lld nanos",
		ALOGW("AudioStreamInternal::processData(): wakeTime = %lld, deadline = %lld nanos",
		(long long) wakeTimeNanos, (long long) deadlineNanos);
		ALOGE("AudioStreamInternal::processData(): past deadline by %d micros",
		ALOGW("AudioStreamInternal::processData(): past deadline by %d micros",
		(int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
		mClockModel.dump();
		mAudioEndpoint.dump();
		break;
		}

media/libaaudio/src/client/AudioStreamInternal.h

+5 −2

Original line number	Diff line number	Diff line
		@@ -125,6 +125,8 @@ protected:

		aaudio_result_t onTimestampFromServer(AAudioServiceMessage *message);

		void logTimestamp(AAudioServiceMessage &message);

		// Calculate timeout for an operation involving framesPerOperation.
		int64_t calculateReasonableTimeout(int32_t framesPerOperation);

		@@ -155,6 +157,9 @@ protected:
		uint8_t *mCallbackBuffer = nullptr;
		int32_t mCallbackFrames = 0;

		// The service uses this for SHARED mode.
		bool mInService = false; // Is this running in the client or the service?

		private:
		/*
		* Asynchronous write with data conversion.
		@@ -172,8 +177,6 @@ private:
		EndpointDescriptor mEndpointDescriptor; // buffer description with resolved addresses
		AAudioServiceInterface &mServiceInterface; // abstract interface to the service

		// The service uses this for SHARED mode.
		bool mInService = false; // Is this running in the client or the service?
		};

		} /* namespace aaudio */