Merge "Always set AudioTrack callback if it is offload playback." into main (1352139e) · Commits · e / os / android_frameworks_av

media/libaaudio/examples/utils/AAudioSimplePlayer.h

+6 −1

Original line number	Diff line number	Diff line
		@@ -119,7 +119,8 @@ public:
		aaudio_result_t open(const AAudioParameters &parameters,
		AAudioStream_dataCallback dataCallback = nullptr,
		AAudioStream_errorCallback errorCallback = nullptr,
		void *userContext = nullptr) {
		void *userContext = nullptr,
		AAudioStream_presentationEndCallback presentationEndCallback = nullptr) {
		aaudio_result_t result = AAUDIO_OK;

		// Use an AAudioStreamBuilder to contain requested parameters.
		@@ -137,6 +138,10 @@ public:
		if (errorCallback != nullptr) {
		AAudioStreamBuilder_setErrorCallback(builder, errorCallback, userContext);
		}
		if (presentationEndCallback != nullptr) {
		AAudioStreamBuilder_setPresentationEndCallback(
		builder, presentationEndCallback, userContext);
		}
		//AAudioStreamBuilder_setFramesPerDataCallback(builder, CALLBACK_SIZE_FRAMES);
		//AAudioStreamBuilder_setBufferCapacityInFrames(builder, 48 * 8);

media/libaaudio/src/legacy/AudioStreamTrack.cpp

+3 −0

Original line number	Diff line number	Diff line
		@@ -130,6 +130,9 @@ aaudio_result_t AudioStreamTrack::open(const AudioStreamBuilder& builder)
		// that is some multiple of the burst size.
		notificationFrames = 0 - DEFAULT_BURSTS_PER_BUFFER_CAPACITY;
		}
		} else if (getPerformanceMode() == AAUDIO_PERFORMANCE_MODE_POWER_SAVING_OFFLOADED) {
		streamTransferType = AudioTrack::transfer_type::TRANSFER_SYNC_NOTIF_CALLBACK;
		callback = wp<AudioTrack::IAudioTrackCallback>::fromExisting(this);
		}
		mCallbackBufferSize = builder.getFramesPerDataCallback();

media/libaaudio/tests/test_pcm_offload.cpp

+94 −21

Original line number	Diff line number	Diff line
		@@ -16,6 +16,7 @@

		// PCM offload

		#include <memory>
		#include <stdio.h>
		#include <stdlib.h>
		#include <vector>
		@@ -35,17 +36,23 @@ aaudio_data_callback_result_t MyDatacallback(AAudioStream* stream,

		void MyErrorCallback(AAudioStream* /stream/, void* /userData/, aaudio_result_t error);

		void MyPresentationEndCallback(AAudioStream* /stream/, void* userData);

		class OffloadPlayer : public AAudioSimplePlayer {
		public:
		OffloadPlayer(AAudioArgsParser& argParser, int delay, int padding, int streamFrames)
		: mArgParser(argParser), mDelay(delay), mPadding(padding), mStreamFrames(streamFrames) {
		OffloadPlayer(AAudioArgsParser& argParser, int delay, int padding, int streamFrames,
		bool useDataCallback)
		: mArgParser(argParser), mDelay(delay), mPadding(padding), mStreamFrames(streamFrames),
		mUseDataCallback(useDataCallback) {
		}

		aaudio_result_t open() {
		aaudio_result_t result = AAudioSimplePlayer::open(mArgParser,
		&MyDatacallback,
		aaudio_result_t result = AAudioSimplePlayer::open(
		mArgParser,
		mUseDataCallback ? &MyDatacallback : nullptr,
		&MyErrorCallback,
		this);
		this,
		&MyPresentationEndCallback);
		if (result != AAUDIO_OK) {
		return result;
		}
		@@ -62,20 +69,7 @@ public:
		void* audioData,
		int32_t numFrames) {
		// Just handle PCM_16 and PCM_FLOAT for testing
		switch (AAudioStream_getFormat(stream)) {
		case AAUDIO_FORMAT_PCM_I16: {
		int16_t audioBuffer = static_cast<int16_t >(audioData);
		for (int i = 0; i < mChannelCount; ++i) {
		mSines[i].render(&audioBuffer[i], mChannelCount, numFrames);
		}
		} break;
		case AAUDIO_FORMAT_PCM_FLOAT: {
		float audioBuffer = static_cast<float >(audioData);
		for (int i = 0; i < mChannelCount; ++i) {
		mSines[i].render(&audioBuffer[i], mChannelCount, numFrames);
		}
		} break;
		default:
		if (!fillData(stream, audioData, numFrames)) {
		return AAUDIO_CALLBACK_RESULT_STOP;
		}
		mFramesWritten += numFrames;
		@@ -90,11 +84,78 @@ public:
		return AAUDIO_CALLBACK_RESULT_CONTINUE;
		}

		void presentationEnd(AAudioStream* stream) {
		printf("Presentation end\n");
		if (!mUseDataCallback) {
		writeAllStreamData(stream);
		}
		}

		void writeData() {
		writeAllStreamData(getStream());
		}

		private:
		void writeAllStreamData(AAudioStream* stream) {
		int bytesPerFrame = mChannelCount;
		std::shared_ptr<uint8_t[]> data;
		switch (AAudioStream_getFormat(stream)) {
		case AAUDIO_FORMAT_PCM_I16: {
		bytesPerFrame *= 2;
		} break;
		case AAUDIO_FORMAT_PCM_FLOAT: {
		bytesPerFrame *= 4;
		} break;
		default:
		printf("Unsupported format %d\n", AAudioStream_getFormat(stream));
		return;
		}
		data = std::make_shared<uint8_t[]>(bytesPerFrame * mStreamFrames);
		fillData(stream, static_cast<void*>(data.get()), mStreamFrames);
		int bytesWritten = 0;
		int framesLeft = mStreamFrames;
		while (framesLeft > 0) {
		auto framesWritten = AAudioStream_write(
		stream, static_cast<void *>(&data[bytesWritten]),
		framesLeft, NANOS_PER_SECOND);
		if (framesWritten < 0) {
		printf("Failed to write data %d\n", framesWritten);
		return;
		}
		printf("Write data succeed, frames=%d\n", framesWritten);
		framesLeft -= framesWritten;
		bytesWritten += framesWritten * bytesPerFrame;
		}
		if (auto result = setOffloadEndOfStream(); result != AAUDIO_OK) {
		printf("Failed to set offload end of stream, result=%d\n", result);
		}
		}

		bool fillData(AAudioStream* stream, void* data, int numFrames) {
		switch (AAudioStream_getFormat(stream)) {
		case AAUDIO_FORMAT_PCM_I16: {
		int16_t audioBuffer = static_cast<int16_t >(data);
		for (int i = 0; i < mChannelCount; ++i) {
		mSines[i].render(&audioBuffer[i], mChannelCount, numFrames);
		}
		} break;
		case AAUDIO_FORMAT_PCM_FLOAT: {
		float audioBuffer = static_cast<float >(data);
		for (int i = 0; i < mChannelCount; ++i) {
		mSines[i].render(&audioBuffer[i], mChannelCount, numFrames);
		}
		} break;
		default:
		return false;
		}
		return true;
		}

		const AAudioArgsParser mArgParser;
		const int mDelay;
		const int mPadding;
		const int mStreamFrames;
		const bool mUseDataCallback;

		int mChannelCount;
		std::vector<SineGenerator> mSines;
		@@ -113,12 +174,18 @@ void MyErrorCallback(AAudioStream* /stream/, void* /userData/, aaudio_result
		printf("Error callback, error=%d\n", error);
		}

		void MyPresentationEndCallback(AAudioStream* stream, void* userData) {
		OffloadPlayer* player = static_cast<OffloadPlayer*>(userData);
		return player->presentationEnd(stream);
		}

		static void usage() {
		AAudioArgsParser::usage();
		printf(" -D{delay} offload delay in frames\n");
		printf(" -P{padding} offload padding in frames\n");
		printf(" -E{frames} frames to notify end of stream\n");
		printf(" -T{seconds} time to run the test\n");
		printf(" -B use blocking write instead of data callback\n");
		}

		int main(int argc, char **argv) {
		@@ -127,6 +194,7 @@ int main(int argc, char **argv) {
		int padding = 0;
		int streamFrames = 0;
		int timeToRun = DEFAULT_TIME_TO_RUN_IN_SECOND;
		bool useDataCallback = true;
		for (int i = 1; i < argc; ++i) {
		const char *arg = argv[i];
		if (argParser.parseArg(arg)) {
		@@ -145,6 +213,9 @@ int main(int argc, char **argv) {
		case 'T':
		timeToRun = atoi(&arg[2]);
		break;
		case 'B':
		useDataCallback = false;
		break;
		default:
		usage();
		exit(EXIT_FAILURE);
		@@ -159,7 +230,7 @@ int main(int argc, char **argv) {
		// Force to use offload mode
		argParser.setPerformanceMode(AAUDIO_PERFORMANCE_MODE_POWER_SAVING_OFFLOADED);

		OffloadPlayer player(argParser, delay, padding, streamFrames);
		OffloadPlayer player(argParser, delay, padding, streamFrames, useDataCallback);
		if (auto result = player.open(); result != AAUDIO_OK) {
		printf("Failed to open stream, error=%d\n", result);
		exit(EXIT_FAILURE);
		@@ -172,6 +243,8 @@ int main(int argc, char **argv) {
		if (auto result = player.start(); result != AAUDIO_OK) {
		printf("Failed to start stream, error=%d", result);
		exit(EXIT_FAILURE);
		} else if (!useDataCallback) {
		player.writeData();
		}

		sleep(timeToRun);