Snap for 4496165 from 3700e162a5526ada762d081de34c9abdd085c5fe to pi-release (6c0057a9) · Commits · e / os / android_frameworks_av

media/libmedia/include/media/MediaMetadataRetrieverInterface.h

+1 −1

Original line number	Original line	Diff line number	Diff line
	@@ -49,7 +49,7 @@ public:
	int index, int colorFormat, bool metaOnly) = 0;		int index, int colorFormat, bool metaOnly) = 0;
	virtual status_t getFrameAtIndex(		virtual status_t getFrameAtIndex(
	std::vector<VideoFrame> frames,		std::vector<VideoFrame> frames,
	int frameIndex, int numFrames, int colorFormat, bool metaOnly);		int frameIndex, int numFrames, int colorFormat, bool metaOnly) = 0;
	virtual MediaAlbumArt* extractAlbumArt() = 0;		virtual MediaAlbumArt* extractAlbumArt() = 0;
	virtual const char* extractMetadata(int keyCode) = 0;		virtual const char* extractMetadata(int keyCode) = 0;
	};		};

services/audioflinger/Effects.cpp

+79 −84

Original line number	Original line	Diff line number	Diff line
	@@ -296,6 +296,43 @@ void AudioFlinger::EffectModule::process()
	const bool auxType =		const bool auxType =
	(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY;		(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY;

			// safeInputOutputSampleCount is 0 if the channel count between input and output
			// buffers do not match. This prevents automatic accumulation or copying between the
			// input and output effect buffers without an intermediary effect process.
			// TODO: consider implementing channel conversion.
			const size_t safeInputOutputSampleCount =
			inChannelCount != outChannelCount ? 0
			: outChannelCount * std::min(
			mConfig.inputCfg.buffer.frameCount,
			mConfig.outputCfg.buffer.frameCount);
			const auto accumulateInputToOutput = [this, safeInputOutputSampleCount]() {
			#ifdef FLOAT_EFFECT_CHAIN
			accumulate_float(
			mConfig.outputCfg.buffer.f32,
			mConfig.inputCfg.buffer.f32,
			safeInputOutputSampleCount);
			#else
			accumulate_i16(
			mConfig.outputCfg.buffer.s16,
			mConfig.inputCfg.buffer.s16,
			safeInputOutputSampleCount);
			#endif
			};
			const auto copyInputToOutput = [this, safeInputOutputSampleCount]() {
			#ifdef FLOAT_EFFECT_CHAIN
			memcpy(
			mConfig.outputCfg.buffer.f32,
			mConfig.inputCfg.buffer.f32,
			safeInputOutputSampleCount * sizeof(*mConfig.outputCfg.buffer.f32));

			#else
			memcpy(
			mConfig.outputCfg.buffer.s16,
			mConfig.inputCfg.buffer.s16,
			safeInputOutputSampleCount * sizeof(*mConfig.outputCfg.buffer.s16));
			#endif
			};

	if (isProcessEnabled()) {		if (isProcessEnabled()) {
	int ret;		int ret;
	if (isProcessImplemented()) {		if (isProcessImplemented()) {
	@@ -308,97 +345,69 @@ void AudioFlinger::EffectModule::process()
	static_assert(sizeof(float) <= sizeof(int32_t),		static_assert(sizeof(float) <= sizeof(int32_t),
	"in-place conversion requires sizeof(float) <= sizeof(int32_t)");		"in-place conversion requires sizeof(float) <= sizeof(int32_t)");

	const int32_t * const p32 = mConfig.inputCfg.buffer.s32;		memcpy_to_float_from_q4_27(
	float * const pFloat = mConfig.inputCfg.buffer.f32;		mConfig.inputCfg.buffer.f32,
	memcpy_to_float_from_q4_27(pFloat, p32, mConfig.inputCfg.buffer.frameCount);
	} else {
	memcpy_to_i16_from_q4_27(mConfig.inputCfg.buffer.s16,
	mConfig.inputCfg.buffer.s32,		mConfig.inputCfg.buffer.s32,
	mConfig.inputCfg.buffer.frameCount);		mConfig.inputCfg.buffer.frameCount);
	}		} else
	#else		#endif
	memcpy_to_i16_from_q4_27(mConfig.inputCfg.buffer.s16,		{
			memcpy_to_i16_from_q4_27(
			mConfig.inputCfg.buffer.s16,
	mConfig.inputCfg.buffer.s32,		mConfig.inputCfg.buffer.s32,
	mConfig.inputCfg.buffer.frameCount);		mConfig.inputCfg.buffer.frameCount);
	#endif		}
	}		}
	#ifdef FLOAT_EFFECT_CHAIN		#ifdef FLOAT_EFFECT_CHAIN
	if (mSupportsFloat) {		if (!mSupportsFloat) { // convert input to int16_t as effect doesn't support float.
	ret = mEffectInterface->process();
	} else {
	{ // convert input to int16_t as effect doesn't support float.
	if (!auxType) {		if (!auxType) {
	if (mInConversionBuffer.get() == nullptr) {		if (mInConversionBuffer.get() == nullptr) {
	ALOGW("%s: mInConversionBuffer is null, bypassing", __func__);		ALOGW("%s: mInConversionBuffer is null, bypassing", __func__);
	goto data_bypass;		goto data_bypass;
	}		}
	const float * const pIn = mInBuffer->audioBuffer()->f32;
	int16_t * const pIn16 = mInConversionBuffer->audioBuffer()->s16;
	memcpy_to_i16_from_float(		memcpy_to_i16_from_float(
	pIn16, pIn, inChannelCount * mConfig.inputCfg.buffer.frameCount);		mInConversionBuffer->audioBuffer()->s16,
			mInBuffer->audioBuffer()->f32,
			inChannelCount * mConfig.inputCfg.buffer.frameCount);
	}		}
	if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {		if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
	if (mOutConversionBuffer.get() == nullptr) {		if (mOutConversionBuffer.get() == nullptr) {
	ALOGW("%s: mOutConversionBuffer is null, bypassing", __func__);		ALOGW("%s: mOutConversionBuffer is null, bypassing", __func__);
	goto data_bypass;		goto data_bypass;
	}		}
	int16_t * const pOut16 = mOutConversionBuffer->audioBuffer()->s16;
	const float * const pOut = mOutBuffer->audioBuffer()->f32;
	memcpy_to_i16_from_float(		memcpy_to_i16_from_float(
	pOut16,		mOutConversionBuffer->audioBuffer()->s16,
	pOut,		mOutBuffer->audioBuffer()->f32,
	outChannelCount * mConfig.outputCfg.buffer.frameCount);		outChannelCount * mConfig.outputCfg.buffer.frameCount);
	}		}
	}		}
			#endif

	ret = mEffectInterface->process();		ret = mEffectInterface->process();

	{ // convert output back to float.		#ifdef FLOAT_EFFECT_CHAIN
	const int16_t * const pOut16 = mOutConversionBuffer->audioBuffer()->s16;		if (!mSupportsFloat) { // convert output int16_t back to float.
	float * const pOut = mOutBuffer->audioBuffer()->f32;
	memcpy_to_float_from_i16(		memcpy_to_float_from_i16(
	pOut, pOut16, outChannelCount * mConfig.outputCfg.buffer.frameCount);		mOutBuffer->audioBuffer()->f32,
	}		mOutConversionBuffer->audioBuffer()->s16,
			outChannelCount * mConfig.outputCfg.buffer.frameCount);
	}		}
	#else
	ret = mEffectInterface->process();
	#endif		#endif
	} else {		} else {
	#ifdef FLOAT_EFFECT_CHAIN		#ifdef FLOAT_EFFECT_CHAIN
	data_bypass:		data_bypass:
	#endif		#endif
	if (!auxType /* aux effects do not require data bypass */		if (!auxType /* aux effects do not require data bypass */
	&& mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw		&& mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
	&& inChannelCount == outChannelCount) {
	const size_t sampleCount = std::min(
	mConfig.inputCfg.buffer.frameCount,
	mConfig.outputCfg.buffer.frameCount) * outChannelCount;

	#ifdef FLOAT_EFFECT_CHAIN
	const float * const in = mConfig.inputCfg.buffer.f32;
	float * const out = mConfig.outputCfg.buffer.f32;

	if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
	accumulate_float(out, in, sampleCount);
	} else {
	memcpy(mConfig.outputCfg.buffer.f32, mConfig.inputCfg.buffer.f32,
	sampleCount * sizeof(*mConfig.outputCfg.buffer.f32));
	}

	#else
	const int16_t * const in = mConfig.inputCfg.buffer.s16;
	int16_t * const out = mConfig.outputCfg.buffer.s16;

	if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {		if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
	accumulate_i16(out, in, sampleCount);		accumulateInputToOutput();
	} else {		} else {
	memcpy(mConfig.outputCfg.buffer.s16, mConfig.inputCfg.buffer.s16,		copyInputToOutput();
	sampleCount * sizeof(*mConfig.outputCfg.buffer.s16));
	}		}
	#endif
	}		}
	ret = -ENODATA;		ret = -ENODATA;
	}		}

	// force transition to IDLE state when engine is ready		// force transition to IDLE state when engine is ready
	if (mState == STOPPED && ret == -ENODATA) {		if (mState == STOPPED && ret == -ENODATA) {
	mDisableWaitCnt = 1;		mDisableWaitCnt = 1;
	@@ -417,21 +426,8 @@ void AudioFlinger::EffectModule::process()
	// If an insert effect is idle and input buffer is different from output buffer,		// If an insert effect is idle and input buffer is different from output buffer,
	// accumulate input onto output		// accumulate input onto output
	sp<EffectChain> chain = mChain.promote();		sp<EffectChain> chain = mChain.promote();
	if (chain != 0		if (chain.get() != nullptr && chain->activeTrackCnt() != 0) {
	&& chain->activeTrackCnt() != 0		accumulateInputToOutput();
	&& inChannelCount == outChannelCount) {
	const size_t sampleCount = std::min(
	mConfig.inputCfg.buffer.frameCount,
	mConfig.outputCfg.buffer.frameCount) * outChannelCount;
	#ifdef FLOAT_EFFECT_CHAIN
	const float * const in = mConfig.inputCfg.buffer.f32;
	float * const out = mConfig.outputCfg.buffer.f32;
	accumulate_float(out, in, sampleCount);
	#else
	const int16_t * const in = mConfig.inputCfg.buffer.s16;
	int16_t * const out = mConfig.outputCfg.buffer.s16;
	accumulate_i16(out, in, sampleCount);
	#endif
	}		}
	}		}
	}		}
	@@ -927,7 +923,6 @@ void AudioFlinger::EffectModule::setInBuffer(const sp<EffectBufferHalInterface>&
	(void)EffectBufferHalInterface::allocate(size, &mInConversionBuffer);		(void)EffectBufferHalInterface::allocate(size, &mInConversionBuffer);
	}		}
	if (mInConversionBuffer.get() != nullptr) {		if (mInConversionBuffer.get() != nullptr) {
	// FIXME: confirm buffer has enough size.
	mInConversionBuffer->setFrameCount(inFrameCount);		mInConversionBuffer->setFrameCount(inFrameCount);
	mEffectInterface->setInBuffer(mInConversionBuffer);		mEffectInterface->setInBuffer(mInConversionBuffer);
	} else if (size > 0) {		} else if (size > 0) {

services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp

+4 −6

Original line number	Original line	Diff line number	Diff line
	@@ -27,14 +27,12 @@ namespace android {

	sp<AudioPort> AudioPortVector::findByTagName(const String8 &tagName) const		sp<AudioPort> AudioPortVector::findByTagName(const String8 &tagName) const
	{		{
	sp<AudioPort> port = 0;		for (const auto& port : *this) {
	for (size_t i = 0; i < size(); i++) {		if (port->getTagName() == tagName) {
	if (itemAt(i)->getTagName() == tagName) {		return port;
	port = itemAt(i);
	break;
	}		}
	}		}
	return port;		return nullptr;
	}		}

	status_t AudioRouteVector::dump(int fd, int spaces) const		status_t AudioRouteVector::dump(int fd, int spaces) const

services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp

+8 −10

Original line number	Original line	Diff line number	Diff line
	@@ -74,11 +74,11 @@ void AudioPort::toAudioPort(struct audio_port *port) const
	SortedVector<audio_format_t> flatenedFormats;		SortedVector<audio_format_t> flatenedFormats;
	SampleRateVector flatenedRates;		SampleRateVector flatenedRates;
	ChannelsVector flatenedChannels;		ChannelsVector flatenedChannels;
	for (size_t profileIndex = 0; profileIndex < mProfiles.size(); profileIndex++) {		for (const auto& profile : mProfiles) {
	if (mProfiles[profileIndex]->isValid()) {		if (profile->isValid()) {
	audio_format_t formatToExport = mProfiles[profileIndex]->getFormat();		audio_format_t formatToExport = profile->getFormat();
	const SampleRateVector &ratesToExport = mProfiles[profileIndex]->getSampleRates();		const SampleRateVector &ratesToExport = profile->getSampleRates();
	const ChannelsVector &channelsToExport = mProfiles[profileIndex]->getChannels();		const ChannelsVector &channelsToExport = profile->getChannels();

	if (flatenedFormats.indexOf(formatToExport) < 0) {		if (flatenedFormats.indexOf(formatToExport) < 0) {
	flatenedFormats.add(formatToExport);		flatenedFormats.add(formatToExport);
	@@ -130,14 +130,12 @@ void AudioPort::toAudioPort(struct audio_port *port) const

	void AudioPort::importAudioPort(const sp<AudioPort>& port, bool force __unused)		void AudioPort::importAudioPort(const sp<AudioPort>& port, bool force __unused)
	{		{
	size_t indexToImport;		for (const auto& profileToImport : port->mProfiles) {
	for (indexToImport = 0; indexToImport < port->mProfiles.size(); indexToImport++) {
	const sp<AudioProfile> &profileToImport = port->mProfiles[indexToImport];
	if (profileToImport->isValid()) {		if (profileToImport->isValid()) {
	// Import only valid port, i.e. valid format, non empty rates and channels masks		// Import only valid port, i.e. valid format, non empty rates and channels masks
	bool hasSameProfile = false;		bool hasSameProfile = false;
	for (size_t profileIndex = 0; profileIndex < mProfiles.size(); profileIndex++) {		for (const auto& profile : mProfiles) {
	if (mProfiles[profileIndex] == profileToImport) {		if (profile == profileToImport) {
	// never import a profile twice		// never import a profile twice
	hasSameProfile = true;		hasSameProfile = true;
	break;		break;

services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp

+1 −2

Original line number	Original line	Diff line number	Diff line
	@@ -233,8 +233,7 @@ status_t AudioProfileVector::checkExactProfile(uint32_t samplingRate,
	return NO_ERROR;		return NO_ERROR;
	}		}

	for (size_t i = 0; i < size(); i++) {		for (const auto& profile : *this) {
	const sp<AudioProfile> profile = itemAt(i);
	if (profile->checkExact(samplingRate, channelMask, format) == NO_ERROR) {		if (profile->checkExact(samplingRate, channelMask, format) == NO_ERROR) {
	return NO_ERROR;		return NO_ERROR;
	}		}