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Commit 4c9cd95b authored by Dave Burke's avatar Dave Burke Committed by Android (Google) Code Review
Browse files

Merge "Added support for HE-AAC recording" into jb-dev

parents 0c0abd4a f60c660f

include/media/mediarecorder.h

+2 −3
Original line number Diff line number Diff line
@@ -75,9 +75,8 @@ enum audio_encoder { 
    AUDIO_ENCODER_AMR_NB = 1,

    AUDIO_ENCODER_AMR_WB = 2,

    AUDIO_ENCODER_AAC = 3,

    AUDIO_ENCODER_AAC_PLUS = 4,

    AUDIO_ENCODER_EAAC_PLUS = 5,

    AUDIO_ENCODER_AAC_ELD = 6,

    AUDIO_ENCODER_HE_AAC = 4,

    AUDIO_ENCODER_AAC_ELD = 5,



    AUDIO_ENCODER_LIST_END // must be the last - used to validate the audio encoder type

};

media/libmedia/MediaProfiles.cpp

+2 −1
Original line number Diff line number Diff line
@@ -44,7 +44,8 @@ const MediaProfiles::NameToTagMap MediaProfiles::sAudioEncoderNameMap[] = { 
    {"amrnb",  AUDIO_ENCODER_AMR_NB},

    {"amrwb",  AUDIO_ENCODER_AMR_WB},

    {"aac",    AUDIO_ENCODER_AAC},

    {"aaceld", AUDIO_ENCODER_AAC_ELD},

    {"heaac",  AUDIO_ENCODER_HE_AAC},

    {"aaceld", AUDIO_ENCODER_AAC_ELD}

};



const MediaProfiles::NameToTagMap MediaProfiles::sFileFormatMap[] = {

media/libmediaplayerservice/StagefrightRecorder.cpp

+10 −2
Original line number Diff line number Diff line
@@ -820,10 +820,15 @@ sp<MediaSource> StagefrightRecorder::createAudioSource() { 
            mime = MEDIA_MIMETYPE_AUDIO_AAC;

            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectLC);

            break;

        case AUDIO_ENCODER_HE_AAC:

            mime = MEDIA_MIMETYPE_AUDIO_AAC;

            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectHE);

            break;

        case AUDIO_ENCODER_AAC_ELD:

            mime = MEDIA_MIMETYPE_AUDIO_AAC;

            encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectELD);

            break;



        default:

            ALOGE("Unknown audio encoder: %d", mAudioEncoder);

            return NULL;
@@ -844,7 +849,6 @@ sp<MediaSource> StagefrightRecorder::createAudioSource() { 


    OMXClient client;

    CHECK_EQ(client.connect(), (status_t)OK);



    sp<MediaSource> audioEncoder =

        OMXCodec::Create(client.interface(), encMeta,

                         true /* createEncoder */, audioSource);
@@ -859,6 +863,7 @@ status_t StagefrightRecorder::startAACRecording() { 
    CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_AAC_ADTS);



    CHECK(mAudioEncoder == AUDIO_ENCODER_AAC ||

          mAudioEncoder == AUDIO_ENCODER_HE_AAC ||

          mAudioEncoder == AUDIO_ENCODER_AAC_ELD);

    CHECK(mAudioSource != AUDIO_SOURCE_CNT);
@@ -977,7 +982,9 @@ status_t StagefrightRecorder::startMPEG2TSRecording() { 
    sp<MediaWriter> writer = new MPEG2TSWriter(mOutputFd);



    if (mAudioSource != AUDIO_SOURCE_CNT) {

        if (mAudioEncoder != AUDIO_ENCODER_AAC) {

        if (mAudioEncoder != AUDIO_ENCODER_AAC &&

            mAudioEncoder != AUDIO_ENCODER_HE_AAC &&

            mAudioEncoder != AUDIO_ENCODER_AAC_ELD) {

            return ERROR_UNSUPPORTED;

        }
@@ -1442,6 +1449,7 @@ status_t StagefrightRecorder::setupAudioEncoder(const sp<MediaWriter>& writer) { 
        case AUDIO_ENCODER_AMR_NB:

        case AUDIO_ENCODER_AMR_WB:

        case AUDIO_ENCODER_AAC:

        case AUDIO_ENCODER_HE_AAC:

        case AUDIO_ENCODER_AAC_ELD:

            break;

media/libstagefright/codecs/aacdec/SoftAAC2.cpp

+30 −20
Original line number Diff line number Diff line
@@ -142,9 +142,9 @@ OMX_ERRORTYPE SoftAAC2::internalGetParameter( 
                aacParams->nSampleRate = 44100;

                aacParams->nFrameLength = 0;

            } else {

                aacParams->nChannels = mStreamInfo->channelConfig;

                aacParams->nSampleRate = mStreamInfo->aacSampleRate;

                aacParams->nFrameLength = mStreamInfo->aacSamplesPerFrame;

                aacParams->nChannels = mStreamInfo->numChannels;

                aacParams->nSampleRate = mStreamInfo->sampleRate;

                aacParams->nFrameLength = mStreamInfo->frameSize;

            }



            return OMX_ErrorNone;
@@ -175,7 +175,7 @@ OMX_ERRORTYPE SoftAAC2::internalGetParameter( 
                pcmParams->nChannels = 1;

                pcmParams->nSamplingRate = 44100;

            } else {

                pcmParams->nChannels = mStreamInfo->channelConfig;

                pcmParams->nChannels = mStreamInfo->numChannels;

                pcmParams->nSamplingRate = mStreamInfo->sampleRate;

            }
@@ -185,6 +185,7 @@ OMX_ERRORTYPE SoftAAC2::internalGetParameter( 
        default:

            return SimpleSoftOMXComponent::internalGetParameter(index, params);

    }



}



OMX_ERRORTYPE SoftAAC2::internalSetParameter(
@@ -254,7 +255,6 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { 
    UCHAR* inBuffer[FILEREAD_MAX_LAYERS];

    UINT inBufferLength[FILEREAD_MAX_LAYERS] = {0};

    UINT bytesValid[FILEREAD_MAX_LAYERS] = {0};

    AAC_DECODER_ERROR decoderErr;



    List<BufferInfo *> &inQueue = getPortQueue(0);

    List<BufferInfo *> &outQueue = getPortQueue(1);
@@ -277,7 +277,6 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { 
            notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);

            return;

        }



        inQueue.erase(inQueue.begin());

        info->mOwnedByUs = false;

        notifyEmptyBufferDone(header);
@@ -303,10 +302,16 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { 
            // the AACDEC_FLUSH flag set

            INT_PCM *outBuffer =

                    reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);

            decoderErr = aacDecoder_DecodeFrame(mAACDecoder,

            AAC_DECODER_ERROR decoderErr = aacDecoder_DecodeFrame(mAACDecoder,

                                                                  outBuffer,

                                                                  outHeader->nAllocLen,

                                                                  AACDEC_FLUSH);

            if (decoderErr != AAC_DEC_OK) {

                mSignalledError = true;

                notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);

                return;

            }



            outHeader->nFilledLen =

                    mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;

            outHeader->nFlags = OMX_BUFFERFLAG_EOS;
@@ -352,14 +357,17 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { 
            inBufferLength[0] = inHeader->nFilledLen;

        }





        // Fill and decode

        INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);

        bytesValid[0] = inBufferLength[0];



        int flags = mInputDiscontinuity ? AACDEC_INTR : 0;

        int prevSampleRate = mStreamInfo->sampleRate;

        decoderErr = aacDecoder_Fill(mAACDecoder,

        int prevNumChannels = mStreamInfo->numChannels;



        AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS;

        while (bytesValid[0] > 0 && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {

            aacDecoder_Fill(mAACDecoder,

                            inBuffer,

                            inBufferLength,

                            bytesValid);
@@ -369,6 +377,7 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { 
                                                outHeader->nAllocLen,

                                                flags);



        }

        mInputDiscontinuity = false;



        /*
@@ -386,7 +395,8 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { 
         * AAC+/eAAC+ until the first data frame is decoded.

         */

        if (mInputBufferCount <= 2) {

            if (mStreamInfo->sampleRate != prevSampleRate) {

            if (mStreamInfo->sampleRate != prevSampleRate ||

                mStreamInfo->numChannels != prevNumChannels) {

                // We're going to want to revisit this input buffer, but

                // may have already advanced the offset. Undo that if

                // necessary.

media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp

+24 −7
Original line number Diff line number Diff line
@@ -239,7 +239,6 @@ OMX_ERRORTYPE SoftAACEncoder2::internalSetParameter( 
            mBitRate = aacParams->nBitRate;

            mNumChannels = aacParams->nChannels;

            mSampleRate = aacParams->nSampleRate;



            if (aacParams->eAACProfile != OMX_AUDIO_AACObjectNull) {

                mAACProfile = aacParams->eAACProfile;

            }
@@ -262,7 +261,6 @@ OMX_ERRORTYPE SoftAACEncoder2::internalSetParameter( 


            mNumChannels = pcmParams->nChannels;

            mSampleRate = pcmParams->nSamplingRate;



            if (setAudioParams() != OK) {

                return OMX_ErrorUndefined;

            }
@@ -275,7 +273,7 @@ OMX_ERRORTYPE SoftAACEncoder2::internalSetParameter( 
    }

}



CHANNEL_MODE getChannelMode(OMX_U32 nChannels) {

static CHANNEL_MODE getChannelMode(OMX_U32 nChannels) {

    CHANNEL_MODE chMode = MODE_INVALID;

    switch (nChannels) {

        case 1: chMode = MODE_1; break;
@@ -289,6 +287,19 @@ CHANNEL_MODE getChannelMode(OMX_U32 nChannels) { 
    return chMode;

}



static AUDIO_OBJECT_TYPE getAOTFromProfile(OMX_U32 profile) {

    if (profile == OMX_AUDIO_AACObjectLC) {

        return AOT_AAC_LC;

    } else if (profile == OMX_AUDIO_AACObjectHE) {

        return AOT_SBR;

    } else if (profile == OMX_AUDIO_AACObjectELD) {

        return AOT_ER_AAC_ELD;

    } else {

        ALOGW("Unsupported AAC profile - defaulting to AAC-LC");

        return AOT_AAC_LC;

    }

}



status_t SoftAACEncoder2::setAudioParams() {

    // We call this whenever sample rate, number of channels or bitrate change

    // in reponse to setParameter calls.
@@ -297,7 +308,7 @@ status_t SoftAACEncoder2::setAudioParams() { 
         mSampleRate, mNumChannels, mBitRate);



    if (AACENC_OK != aacEncoder_SetParam(mAACEncoder, AACENC_AOT,

            mAACProfile == OMX_AUDIO_AACObjectELD ? AOT_ER_AAC_ELD : AOT_AAC_LC)) {

            getAOTFromProfile(mAACProfile))) {

        ALOGE("Failed to set AAC encoder parameters");

        return UNKNOWN_ERROR;

    }
@@ -341,12 +352,17 @@ void SoftAACEncoder2::onQueueFilled(OMX_U32 portIndex) { 
        }



        if (AACENC_OK != aacEncEncode(mAACEncoder, NULL, NULL, NULL, NULL)) {

            ALOGE("Failed to initialize AAC encoder");

            ALOGE("Unable to initialize encoder for profile / sample-rate / bit-rate / channels");

            notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);

            mSignalledError = true;

            return;

        }



        OMX_U32 actualBitRate  = aacEncoder_GetParam(mAACEncoder, AACENC_BITRATE);

        if (mBitRate != actualBitRate) {

            ALOGW("Requested bitrate %lu unsupported, using %lu", mBitRate, actualBitRate);

        }



        AACENC_InfoStruct encInfo;

        if (AACENC_OK != aacEncInfo(mAACEncoder, &encInfo)) {

            ALOGE("Failed to get AAC encoder info");
@@ -373,7 +389,7 @@ void SoftAACEncoder2::onQueueFilled(OMX_U32 portIndex) { 
    size_t numBytesPerInputFrame =

        mNumChannels * kNumSamplesPerFrame * sizeof(int16_t);



    // BUGBUG: Fraunhofer's decoder chokes on large chunks of AAC-ELD

    // Limit input size so we only get one ELD frame

    if (mAACProfile == OMX_AUDIO_AACObjectELD && numBytesPerInputFrame > 512) {

        numBytesPerInputFrame = 512;

    }
@@ -402,7 +418,7 @@ void SoftAACEncoder2::onQueueFilled(OMX_U32 portIndex) { 
            }



            if (mInputFrame == NULL) {

                mInputFrame = new int16_t[kNumSamplesPerFrame * mNumChannels];

                mInputFrame = new int16_t[numBytesPerInputFrame / sizeof(int16_t)];

            }



            if (mInputSize == 0) {
@@ -490,6 +506,7 @@ void SoftAACEncoder2::onQueueFilled(OMX_U32 portIndex) { 
        // Encode the mInputFrame, which is treated as a modulo buffer

        AACENC_ERROR encoderErr = AACENC_OK;

        size_t nOutputBytes = 0;



        do {

            memset(&outargs, 0, sizeof(outargs));