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Commit 91de11b0 authored by Patrik2 Carlsson's avatar Patrik2 Carlsson Committed by Zoran Jovanovic
Browse files

Allow multichannel FLAC files 

FLACExtractor had artificial limits which did not allow multichannel
audio, 88.2 or 96 kHz samplerate, these were removed since it is up
to the audio subsystem to handle.

Change-Id: Iadc20cdf4724814b5c980827a528a1e1b4bd6a82
parent f0e3acd0

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;