am df9cce64: Merge "Allow multichannel FLAC files" (bc9f3fc7) · Commits · e / os / android_frameworks_av

media/libstagefright/FLACExtractor.cpp

+44 −17

Original line number	Diff line number	Diff line
		@@ -122,7 +122,7 @@ private:
		// media buffers
		size_t mMaxBufferSize;
		MediaBufferGroup *mGroup;
		void (mCopy)(short dst, const int const src, unsigned nSamples);
		void (mCopy)(short dst, const int const src, unsigned nSamples, unsigned nChannels);

		// handle to underlying libFLAC parser
		FLAC__StreamDecoder *mDecoder;
		@@ -380,14 +380,14 @@ void FLACParser::errorCallback(FLAC__StreamDecoderErrorStatus status)
		// Copy samples from FLAC native 32-bit non-interleaved to 16-bit interleaved.
		// These are candidates for optimization if needed.

		static void copyMono8(short dst, const int const *src, unsigned nSamples)
		static void copyMono8(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		*dst++ = src[0][i] << 8;
		}
		}

		static void copyStereo8(short dst, const int const *src, unsigned nSamples)
		static void copyStereo8(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		*dst++ = src[0][i] << 8;
		@@ -395,14 +395,23 @@ static void copyStereo8(short dst, const int const *src, unsigned nSamples)
		}
		}

		static void copyMono16(short dst, const int const *src, unsigned nSamples)
		static void copyMultiCh8(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		for (unsigned c = 0; c < nChannels; ++c) {
		*dst++ = src[c][i] << 8;
		}
		}
		}

		static void copyMono16(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		*dst++ = src[0][i];
		}
		}

		static void copyStereo16(short dst, const int const *src, unsigned nSamples)
		static void copyStereo16(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		*dst++ = src[0][i];
		@@ -410,16 +419,25 @@ static void copyStereo16(short dst, const int const *src, unsigned nSamples)
		}
		}

		static void copyMultiCh16(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		for (unsigned c = 0; c < nChannels; ++c) {
		*dst++ = src[c][i];
		}
		}
		}

		// 24-bit versions should do dithering or noise-shaping, here or in AudioFlinger

		static void copyMono24(short dst, const int const *src, unsigned nSamples)
		static void copyMono24(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		*dst++ = src[0][i] >> 8;
		}
		}

		static void copyStereo24(short dst, const int const *src, unsigned nSamples)
		static void copyStereo24(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		*dst++ = src[0][i] >> 8;
		@@ -427,7 +445,16 @@ static void copyStereo24(short dst, const int const *src, unsigned nSamples)
		}
		}

		static void copyTrespass(short dst, const int const *src, unsigned nSamples)
		static void copyMultiCh24(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		for (unsigned i = 0; i < nSamples; ++i) {
		for (unsigned c = 0; c < nChannels; ++c) {
		*dst++ = src[c][i] >> 8;
		}
		}
		}

		static void copyTrespass(short dst, const int const *src, unsigned nSamples, unsigned nChannels)
		{
		TRESPASS();
		}
		@@ -507,11 +534,7 @@ status_t FLACParser::init()
		}
		if (mStreamInfoValid) {
		// check channel count
		switch (getChannels()) {
		case 1:
		case 2:
		break;
		default:
		if (getChannels() == 0 \|\| getChannels() > 8) {
		ALOGE("unsupported channel count %u", getChannels());
		return NO_INIT;
		}
		@@ -536,9 +559,10 @@ status_t FLACParser::init()
		case 32000:
		case 44100:
		case 48000:
		case 88200:
		case 96000:
		break;
		default:
		// 96000 would require a proper downsampler in AudioFlinger
		ALOGE("unsupported sample rate %u", getSampleRate());
		return NO_INIT;
		}
		@@ -546,17 +570,20 @@ status_t FLACParser::init()
		static const struct {
		unsigned mChannels;
		unsigned mBitsPerSample;
		void (mCopy)(short dst, const int const src, unsigned nSamples);
		void (mCopy)(short dst, const int const src, unsigned nSamples, unsigned nChannels);
		} table[] = {
		{ 1, 8, copyMono8 },
		{ 2, 8, copyStereo8 },
		{ 8, 8, copyMultiCh8 },
		{ 1, 16, copyMono16 },
		{ 2, 16, copyStereo16 },
		{ 8, 16, copyMultiCh16 },
		{ 1, 24, copyMono24 },
		{ 2, 24, copyStereo24 },
		{ 8, 24, copyMultiCh24 },
		};
		for (unsigned i = 0; i < sizeof(table)/sizeof(table[0]); ++i) {
		if (table[i].mChannels == getChannels() &&
		if (table[i].mChannels >= getChannels() &&
		table[i].mBitsPerSample == getBitsPerSample()) {
		mCopy = table[i].mCopy;
		break;
		@@ -640,7 +667,7 @@ MediaBuffer *FLACParser::readBuffer(bool doSeek, FLAC__uint64 sample)
		short data = (short ) buffer->data();
		buffer->set_range(0, bufferSize);
		// copy PCM from FLAC write buffer to our media buffer, with interleaving
		(*mCopy)(data, mWriteBuffer, blocksize);
		(*mCopy)(data, mWriteBuffer, blocksize, getChannels());
		// fill in buffer metadata
		CHECK(mWriteHeader.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER);
		FLAC__uint64 sampleNumber = mWriteHeader.number.sample_number;