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author | Dave Burke <daveburke@google.com> | 2012-04-30 12:45:27 -0700 |
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committer | Android (Google) Code Review <android-gerrit@google.com> | 2012-04-30 12:45:27 -0700 |
commit | 4c9cd95b27a638701be2ffa1713273ba2f624770 (patch) | |
tree | c7f5f7820aa39c965706b7ea504d9b53f5147688 /media | |
parent | 0c0abd4ad26971b5fba94734137fe0bb1a590ab6 (diff) | |
parent | f60c660f048d5f5e2458cff243c20400d73757a7 (diff) | |
download | frameworks_av-4c9cd95b27a638701be2ffa1713273ba2f624770.zip frameworks_av-4c9cd95b27a638701be2ffa1713273ba2f624770.tar.gz frameworks_av-4c9cd95b27a638701be2ffa1713273ba2f624770.tar.bz2 |
Merge "Added support for HE-AAC recording" into jb-dev
Diffstat (limited to 'media')
-rw-r--r-- | media/libmedia/MediaProfiles.cpp | 3 | ||||
-rw-r--r-- | media/libmediaplayerservice/StagefrightRecorder.cpp | 12 | ||||
-rw-r--r-- | media/libstagefright/codecs/aacdec/SoftAAC2.cpp | 50 | ||||
-rw-r--r-- | media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp | 31 |
4 files changed, 66 insertions, 30 deletions
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp index c08f033..6929efa 100644 --- a/media/libmedia/MediaProfiles.cpp +++ b/media/libmedia/MediaProfiles.cpp @@ -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[] = { diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp index b676cc7..727fd0d 100644 --- a/media/libmediaplayerservice/StagefrightRecorder.cpp +++ b/media/libmediaplayerservice/StagefrightRecorder.cpp @@ -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; diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp index 547a554..bf7befd 100644 --- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp +++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp @@ -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, - outBuffer, - outHeader->nAllocLen, - AACDEC_FLUSH); + 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; @@ -352,23 +357,27 @@ 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, - inBuffer, - inBufferLength, - bytesValid); + int prevNumChannels = mStreamInfo->numChannels; - decoderErr = aacDecoder_DecodeFrame(mAACDecoder, - outBuffer, - outHeader->nAllocLen, - flags); + 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); + decoderErr = aacDecoder_DecodeFrame(mAACDecoder, + outBuffer, + 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. diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp b/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp index 4947fb2..7719435 100644 --- a/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp +++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder2.cpp @@ -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)); |