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	Diff line number	Diff line
		@@ -49,7 +49,7 @@ public:
		int index, int colorFormat, bool metaOnly) = 0;
		virtual status_t getFrameAtIndex(
		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 const char* extractMetadata(int keyCode) = 0;
		};

services/audioflinger/Effects.cpp

+79 −84

Original line number	Diff line number	Diff line
		@@ -296,6 +296,43 @@ void AudioFlinger::EffectModule::process()
		const bool auxType =
		(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()) {
		int ret;
		if (isProcessImplemented()) {
		@@ -308,97 +345,69 @@ void AudioFlinger::EffectModule::process()
		static_assert(sizeof(float) <= sizeof(int32_t),
		"in-place conversion requires sizeof(float) <= sizeof(int32_t)");

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

		ret = mEffectInterface->process();

		{ // convert output back to float.
		const int16_t * const pOut16 = mOutConversionBuffer->audioBuffer()->s16;
		float * const pOut = mOutBuffer->audioBuffer()->f32;
		#ifdef FLOAT_EFFECT_CHAIN
		if (!mSupportsFloat) { // convert output int16_t back to float.
		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
		} else {
		#ifdef FLOAT_EFFECT_CHAIN
		data_bypass:
		#endif
		if (!auxType /* aux effects do not require data bypass */
		&& 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;

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

		// force transition to IDLE state when engine is ready
		if (mState == STOPPED && ret == -ENODATA) {
		mDisableWaitCnt = 1;
		@@ -417,21 +426,8 @@ void AudioFlinger::EffectModule::process()
		// If an insert effect is idle and input buffer is different from output buffer,
		// accumulate input onto output
		sp<EffectChain> chain = mChain.promote();
		if (chain != 0
		&& chain->activeTrackCnt() != 0
		&& 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
		if (chain.get() != nullptr && chain->activeTrackCnt() != 0) {
		accumulateInputToOutput();
		}
		}
		}
		@@ -927,7 +923,6 @@ void AudioFlinger::EffectModule::setInBuffer(const sp<EffectBufferHalInterface>&
		(void)EffectBufferHalInterface::allocate(size, &mInConversionBuffer);
		}
		if (mInConversionBuffer.get() != nullptr) {
		// FIXME: confirm buffer has enough size.
		mInConversionBuffer->setFrameCount(inFrameCount);
		mEffectInterface->setInBuffer(mInConversionBuffer);
		} else if (size > 0) {

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

+4 −6

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

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

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

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

+8 −10

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

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

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

+1 −2

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

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