diff options
| -rw-r--r-- | media/libstagefright/FLACExtractor.cpp | 61 |
1 files changed, 44 insertions, 17 deletions
diff --git a/media/libstagefright/FLACExtractor.cpp b/media/libstagefright/FLACExtractor.cpp index 29bb056..098fcf9 100644 --- a/media/libstagefright/FLACExtractor.cpp +++ b/media/libstagefright/FLACExtractor.cpp @@ -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; |