diff options
Diffstat (limited to 'media/libstagefright')
20 files changed, 691 insertions, 292 deletions
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index 5d5220f..eb274a8 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -35,7 +35,9 @@ #include <media/hardware/HardwareAPI.h> +#include <OMX_AudioExt.h> #include <OMX_Component.h> +#include <OMX_IndexExt.h> #include "include/avc_utils.h" @@ -977,6 +979,10 @@ status_t ACodec::setComponentRole( "audio_decoder.flac", "audio_encoder.flac" }, { MEDIA_MIMETYPE_AUDIO_MSGSM, "audio_decoder.gsm", "audio_encoder.gsm" }, + { MEDIA_MIMETYPE_VIDEO_MPEG2, + "video_decoder.mpeg2", "video_encoder.mpeg2" }, + { MEDIA_MIMETYPE_AUDIO_AC3, + "audio_decoder.ac3", "audio_encoder.ac3" }, }; static const size_t kNumMimeToRole = @@ -1268,6 +1274,15 @@ status_t ACodec::configureCodec( } else { err = setupRawAudioFormat(kPortIndexInput, sampleRate, numChannels); } + } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AC3)) { + int32_t numChannels; + int32_t sampleRate; + if (!msg->findInt32("channel-count", &numChannels) + || !msg->findInt32("sample-rate", &sampleRate)) { + err = INVALID_OPERATION; + } else { + err = setupAC3Codec(encoder, numChannels, sampleRate); + } } if (err != OK) { @@ -1464,6 +1479,44 @@ status_t ACodec::setupAACCodec( mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile)); } +status_t ACodec::setupAC3Codec( + bool encoder, int32_t numChannels, int32_t sampleRate) { + status_t err = setupRawAudioFormat( + encoder ? kPortIndexInput : kPortIndexOutput, sampleRate, numChannels); + + if (err != OK) { + return err; + } + + if (encoder) { + ALOGW("AC3 encoding is not supported."); + return INVALID_OPERATION; + } + + OMX_AUDIO_PARAM_ANDROID_AC3TYPE def; + InitOMXParams(&def); + def.nPortIndex = kPortIndexInput; + + err = mOMX->getParameter( + mNode, + (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3, + &def, + sizeof(def)); + + if (err != OK) { + return err; + } + + def.nChannels = numChannels; + def.nSampleRate = sampleRate; + + return mOMX->setParameter( + mNode, + (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3, + &def, + sizeof(def)); +} + static OMX_AUDIO_AMRBANDMODETYPE pickModeFromBitRate( bool isAMRWB, int32_t bps) { if (isAMRWB) { @@ -2558,7 +2611,7 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { { OMX_AUDIO_PORTDEFINITIONTYPE *audioDef = &def.format.audio; - switch (audioDef->eEncoding) { + switch ((int)audioDef->eEncoding) { case OMX_AUDIO_CodingPCM: { OMX_AUDIO_PARAM_PCMMODETYPE params; @@ -2664,6 +2717,24 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { break; } + case OMX_AUDIO_CodingAndroidAC3: + { + OMX_AUDIO_PARAM_ANDROID_AC3TYPE params; + InitOMXParams(¶ms); + params.nPortIndex = kPortIndexOutput; + + CHECK_EQ((status_t)OK, mOMX->getParameter( + mNode, + (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3, + ¶ms, + sizeof(params))); + + notify->setString("mime", MEDIA_MIMETYPE_AUDIO_AC3); + notify->setInt32("channel-count", params.nChannels); + notify->setInt32("sample-rate", params.nSampleRate); + break; + } + default: TRESPASS(); } @@ -3342,7 +3413,7 @@ bool ACodec::BaseState::onOMXFillBufferDone( sp<AMessage> reply = new AMessage(kWhatOutputBufferDrained, mCodec->id()); - if (!mCodec->mSentFormat) { + if (!mCodec->mSentFormat && rangeLength > 0) { mCodec->sendFormatChange(reply); } diff --git a/media/libstagefright/CameraSourceTimeLapse.cpp b/media/libstagefright/CameraSourceTimeLapse.cpp index 5772316..86844b8 100644 --- a/media/libstagefright/CameraSourceTimeLapse.cpp +++ b/media/libstagefright/CameraSourceTimeLapse.cpp @@ -85,7 +85,8 @@ CameraSourceTimeLapse::CameraSourceTimeLapse( mVideoWidth = videoSize.width; mVideoHeight = videoSize.height; - if (!trySettingVideoSize(videoSize.width, videoSize.height)) { + if (OK == mInitCheck && !trySettingVideoSize(videoSize.width, videoSize.height)) { + releaseCamera(); mInitCheck = NO_INIT; } diff --git a/media/libstagefright/MediaDefs.cpp b/media/libstagefright/MediaDefs.cpp index b5d4e44..340cba7 100644 --- a/media/libstagefright/MediaDefs.cpp +++ b/media/libstagefright/MediaDefs.cpp @@ -42,6 +42,7 @@ const char *MEDIA_MIMETYPE_AUDIO_RAW = "audio/raw"; const char *MEDIA_MIMETYPE_AUDIO_FLAC = "audio/flac"; const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS = "audio/aac-adts"; const char *MEDIA_MIMETYPE_AUDIO_MSGSM = "audio/gsm"; +const char *MEDIA_MIMETYPE_AUDIO_AC3 = "audio/ac3"; const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4"; const char *MEDIA_MIMETYPE_CONTAINER_WAV = "audio/x-wav"; diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp index 43736ad..625922f 100644 --- a/media/libstagefright/OMXCodec.cpp +++ b/media/libstagefright/OMXCodec.cpp @@ -40,7 +40,9 @@ #include <utils/Vector.h> #include <OMX_Audio.h> +#include <OMX_AudioExt.h> #include <OMX_Component.h> +#include <OMX_IndexExt.h> #include "include/avc_utils.h" @@ -528,6 +530,17 @@ status_t OMXCodec::configureCodec(const sp<MetaData> &meta) { sampleRate, numChannels); } + } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_AC3, mMIME)) { + int32_t numChannels; + int32_t sampleRate; + CHECK(meta->findInt32(kKeyChannelCount, &numChannels)); + CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); + + status_t err = setAC3Format(numChannels, sampleRate); + if (err != OK) { + CODEC_LOGE("setAC3Format() failed (err = %d)", err); + return err; + } } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_G711_ALAW, mMIME) || !strcasecmp(MEDIA_MIMETYPE_AUDIO_G711_MLAW, mMIME)) { // These are PCM-like formats with a fixed sample rate but @@ -1394,6 +1407,10 @@ void OMXCodec::setComponentRole( "audio_decoder.flac", "audio_encoder.flac" }, { MEDIA_MIMETYPE_AUDIO_MSGSM, "audio_decoder.gsm", "audio_encoder.gsm" }, + { MEDIA_MIMETYPE_VIDEO_MPEG2, + "video_decoder.mpeg2", "video_encoder.mpeg2" }, + { MEDIA_MIMETYPE_AUDIO_AC3, + "audio_decoder.ac3", "audio_encoder.ac3" }, }; static const size_t kNumMimeToRole = @@ -3489,6 +3506,31 @@ status_t OMXCodec::setAACFormat( return OK; } +status_t OMXCodec::setAC3Format(int32_t numChannels, int32_t sampleRate) { + OMX_AUDIO_PARAM_ANDROID_AC3TYPE def; + InitOMXParams(&def); + def.nPortIndex = kPortIndexInput; + + status_t err = mOMX->getParameter( + mNode, + (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3, + &def, + sizeof(def)); + + if (err != OK) { + return err; + } + + def.nChannels = numChannels; + def.nSampleRate = sampleRate; + + return mOMX->setParameter( + mNode, + (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3, + &def, + sizeof(def)); +} + void OMXCodec::setG711Format(int32_t numChannels) { CHECK(!mIsEncoder); setRawAudioFormat(kPortIndexInput, 8000, numChannels); @@ -4422,6 +4464,17 @@ void OMXCodec::initOutputFormat(const sp<MetaData> &inputFormat) { mOutputFormat->setInt32(kKeyChannelCount, numChannels); mOutputFormat->setInt32(kKeySampleRate, sampleRate); mOutputFormat->setInt32(kKeyBitRate, bitRate); + } else if (audio_def->eEncoding == + (OMX_AUDIO_CODINGTYPE)OMX_AUDIO_CodingAndroidAC3) { + mOutputFormat->setCString( + kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3); + int32_t numChannels, sampleRate, bitRate; + inputFormat->findInt32(kKeyChannelCount, &numChannels); + inputFormat->findInt32(kKeySampleRate, &sampleRate); + inputFormat->findInt32(kKeyBitRate, &bitRate); + mOutputFormat->setInt32(kKeyChannelCount, numChannels); + mOutputFormat->setInt32(kKeySampleRate, sampleRate); + mOutputFormat->setInt32(kKeyBitRate, bitRate); } else { CHECK(!"Should not be here. Unknown audio encoding."); } diff --git a/media/libstagefright/SkipCutBuffer.cpp b/media/libstagefright/SkipCutBuffer.cpp index 773854f..e2e6d79 100644 --- a/media/libstagefright/SkipCutBuffer.cpp +++ b/media/libstagefright/SkipCutBuffer.cpp @@ -25,7 +25,7 @@ namespace android { SkipCutBuffer::SkipCutBuffer(int32_t skip, int32_t cut) { - mFrontPadding = skip; + mFrontPadding = mSkip = skip; mBackPadding = cut; mWriteHead = 0; mReadHead = 0; @@ -94,6 +94,7 @@ void SkipCutBuffer::submit(const sp<ABuffer>& buffer) { void SkipCutBuffer::clear() { mWriteHead = mReadHead = 0; + mFrontPadding = mSkip; } void SkipCutBuffer::write(const char *src, size_t num) { diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp index 1b20cbb..f842e27 100644 --- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp +++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp @@ -58,6 +58,8 @@ SoftAAC2::SoftAAC2( mIsADTS(false), mInputBufferCount(0), mSignalledError(false), + mSawInputEos(false), + mSignalledOutputEos(false), mAnchorTimeUs(0), mNumSamplesOutput(0), mOutputPortSettingsChange(NONE) { @@ -350,115 +352,83 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { return; } - while (!inQueue.empty() && !outQueue.empty()) { - BufferInfo *inInfo = *inQueue.begin(); - OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) { + BufferInfo *inInfo = NULL; + OMX_BUFFERHEADERTYPE *inHeader = NULL; + if (!inQueue.empty()) { + inInfo = *inQueue.begin(); + inHeader = inInfo->mHeader; + } BufferInfo *outInfo = *outQueue.begin(); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; + outHeader->nFlags = 0; - if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { - inQueue.erase(inQueue.begin()); - inInfo->mOwnedByUs = false; - notifyEmptyBufferDone(inHeader); - - if (mDecoderHasData) { - // flush out the decoder's delayed data by calling DecodeFrame - // one more time, with the AACDEC_FLUSH flag set - INT_PCM *outBuffer = - reinterpret_cast<INT_PCM *>( - outHeader->pBuffer + outHeader->nOffset); - - AAC_DECODER_ERROR decoderErr = - aacDecoder_DecodeFrame(mAACDecoder, - outBuffer, - outHeader->nAllocLen, - AACDEC_FLUSH); - mDecoderHasData = false; - - if (decoderErr != AAC_DEC_OK) { - mSignalledError = true; - - notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, - NULL); - - return; - } - - outHeader->nFilledLen = - mStreamInfo->frameSize - * sizeof(int16_t) - * mStreamInfo->numChannels; - } else { - // we never submitted any data to the decoder, so there's nothing to flush out - outHeader->nFilledLen = 0; + if (inHeader) { + if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + mSawInputEos = true; } - outHeader->nFlags = OMX_BUFFERFLAG_EOS; - - outQueue.erase(outQueue.begin()); - outInfo->mOwnedByUs = false; - notifyFillBufferDone(outHeader); - return; - } - - if (inHeader->nOffset == 0) { - mAnchorTimeUs = inHeader->nTimeStamp; - mNumSamplesOutput = 0; - } + if (inHeader->nOffset == 0 && inHeader->nFilledLen) { + mAnchorTimeUs = inHeader->nTimeStamp; + mNumSamplesOutput = 0; + } - size_t adtsHeaderSize = 0; - if (mIsADTS) { - // skip 30 bits, aac_frame_length follows. - // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll????? + if (mIsADTS) { + size_t adtsHeaderSize = 0; + // skip 30 bits, aac_frame_length follows. + // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll????? - const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset; + const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset; - bool signalError = false; - if (inHeader->nFilledLen < 7) { - ALOGE("Audio data too short to contain even the ADTS header. " - "Got %ld bytes.", inHeader->nFilledLen); - hexdump(adtsHeader, inHeader->nFilledLen); - signalError = true; - } else { - bool protectionAbsent = (adtsHeader[1] & 1); - - unsigned aac_frame_length = - ((adtsHeader[3] & 3) << 11) - | (adtsHeader[4] << 3) - | (adtsHeader[5] >> 5); - - if (inHeader->nFilledLen < aac_frame_length) { - ALOGE("Not enough audio data for the complete frame. " - "Got %ld bytes, frame size according to the ADTS " - "header is %u bytes.", - inHeader->nFilledLen, aac_frame_length); + bool signalError = false; + if (inHeader->nFilledLen < 7) { + ALOGE("Audio data too short to contain even the ADTS header. " + "Got %ld bytes.", inHeader->nFilledLen); hexdump(adtsHeader, inHeader->nFilledLen); signalError = true; } else { - adtsHeaderSize = (protectionAbsent ? 7 : 9); - - inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize; - inBufferLength[0] = aac_frame_length - adtsHeaderSize; - - inHeader->nOffset += adtsHeaderSize; - inHeader->nFilledLen -= adtsHeaderSize; + bool protectionAbsent = (adtsHeader[1] & 1); + + unsigned aac_frame_length = + ((adtsHeader[3] & 3) << 11) + | (adtsHeader[4] << 3) + | (adtsHeader[5] >> 5); + + if (inHeader->nFilledLen < aac_frame_length) { + ALOGE("Not enough audio data for the complete frame. " + "Got %ld bytes, frame size according to the ADTS " + "header is %u bytes.", + inHeader->nFilledLen, aac_frame_length); + hexdump(adtsHeader, inHeader->nFilledLen); + signalError = true; + } else { + adtsHeaderSize = (protectionAbsent ? 7 : 9); + + inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize; + inBufferLength[0] = aac_frame_length - adtsHeaderSize; + + inHeader->nOffset += adtsHeaderSize; + inHeader->nFilledLen -= adtsHeaderSize; + } } - } - if (signalError) { - mSignalledError = true; + if (signalError) { + mSignalledError = true; - notify(OMX_EventError, - OMX_ErrorStreamCorrupt, - ERROR_MALFORMED, - NULL); + notify(OMX_EventError, + OMX_ErrorStreamCorrupt, + ERROR_MALFORMED, + NULL); - return; + return; + } + } else { + inBuffer[0] = inHeader->pBuffer + inHeader->nOffset; + inBufferLength[0] = inHeader->nFilledLen; } } else { - inBuffer[0] = inHeader->pBuffer + inHeader->nOffset; - inBufferLength[0] = inHeader->nFilledLen; + inBufferLength[0] = 0; } // Fill and decode @@ -471,50 +441,66 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { int prevNumChannels = mStreamInfo->numChannels; AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS; - while (bytesValid[0] > 0 && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) { + while ((bytesValid[0] > 0 || mSawInputEos) && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) { + mDecoderHasData |= (bytesValid[0] > 0); aacDecoder_Fill(mAACDecoder, inBuffer, inBufferLength, bytesValid); - mDecoderHasData = true; decoderErr = aacDecoder_DecodeFrame(mAACDecoder, outBuffer, outHeader->nAllocLen, 0 /* flags */); - if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) { - ALOGW("Not enough bits, bytesValid %d", bytesValid[0]); + if (mSawInputEos && bytesValid[0] <= 0) { + if (mDecoderHasData) { + // flush out the decoder's delayed data by calling DecodeFrame + // one more time, with the AACDEC_FLUSH flag set + decoderErr = aacDecoder_DecodeFrame(mAACDecoder, + outBuffer, + outHeader->nAllocLen, + AACDEC_FLUSH); + mDecoderHasData = false; + } + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + mSignalledOutputEos = true; + break; + } else { + ALOGW("Not enough bits, bytesValid %d", bytesValid[0]); + } } } size_t numOutBytes = mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels; - if (decoderErr == AAC_DEC_OK) { - UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0]; - inHeader->nFilledLen -= inBufferUsedLength; - inHeader->nOffset += inBufferUsedLength; - } else { - ALOGW("AAC decoder returned error %d, substituting silence", - decoderErr); + if (inHeader) { + if (decoderErr == AAC_DEC_OK) { + UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0]; + inHeader->nFilledLen -= inBufferUsedLength; + inHeader->nOffset += inBufferUsedLength; + } else { + ALOGW("AAC decoder returned error %d, substituting silence", + decoderErr); - memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes); + memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes); - // Discard input buffer. - inHeader->nFilledLen = 0; + // Discard input buffer. + inHeader->nFilledLen = 0; - aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1); + aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1); - // fall through - } + // fall through + } - if (inHeader->nFilledLen == 0) { - inInfo->mOwnedByUs = false; - inQueue.erase(inQueue.begin()); - inInfo = NULL; - notifyEmptyBufferDone(inHeader); - inHeader = NULL; + if (inHeader->nFilledLen == 0) { + inInfo->mOwnedByUs = false; + inQueue.erase(inQueue.begin()); + inInfo = NULL; + notifyEmptyBufferDone(inHeader); + inHeader = NULL; + } } /* @@ -555,7 +541,6 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { // we've previously decoded valid data, in the latter case // (decode failed) we'll output a silent frame. outHeader->nFilledLen = numOutBytes; - outHeader->nFlags = 0; outHeader->nTimeStamp = mAnchorTimeUs @@ -582,6 +567,12 @@ void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) { // depend on fragments from the last one decoded. // drain all existing data drainDecoder(); + // force decoder loop to drop the first decoded buffer by resetting these state variables, + // but only if initialization has already happened. + if (mInputBufferCount != 0) { + mInputBufferCount = 1; + mStreamInfo->sampleRate = 0; + } } } @@ -606,6 +597,8 @@ void SoftAAC2::onReset() { mStreamInfo->sampleRate = 0; mSignalledError = false; + mSawInputEos = false; + mSignalledOutputEos = false; mOutputPortSettingsChange = NONE; } diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h index 2d960ab..a7ea1e2 100644 --- a/media/libstagefright/codecs/aacdec/SoftAAC2.h +++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h @@ -55,6 +55,8 @@ private: bool mDecoderHasData; size_t mInputBufferCount; bool mSignalledError; + bool mSawInputEos; + bool mSignalledOutputEos; int64_t mAnchorTimeUs; int64_t mNumSamplesOutput; diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp index 7c382fb..877e3cb 100644 --- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp +++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp @@ -49,6 +49,8 @@ SoftMP3::SoftMP3( mNumChannels(2), mSamplingRate(44100), mSignalledError(false), + mSawInputEos(false), + mSignalledOutputEos(false), mOutputPortSettingsChange(NONE) { initPorts(); initDecoder(); @@ -194,48 +196,36 @@ void SoftMP3::onQueueFilled(OMX_U32 portIndex) { List<BufferInfo *> &inQueue = getPortQueue(0); List<BufferInfo *> &outQueue = getPortQueue(1); - while (!inQueue.empty() && !outQueue.empty()) { - BufferInfo *inInfo = *inQueue.begin(); - OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) { + BufferInfo *inInfo = NULL; + OMX_BUFFERHEADERTYPE *inHeader = NULL; + if (!inQueue.empty()) { + inInfo = *inQueue.begin(); + inHeader = inInfo->mHeader; + } BufferInfo *outInfo = *outQueue.begin(); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; + outHeader->nFlags = 0; - if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { - inQueue.erase(inQueue.begin()); - inInfo->mOwnedByUs = false; - notifyEmptyBufferDone(inHeader); - - if (!mIsFirst) { - // pad the end of the stream with 529 samples, since that many samples - // were trimmed off the beginning when decoding started - outHeader->nFilledLen = - kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t); + if (inHeader) { + if (inHeader->nOffset == 0 && inHeader->nFilledLen) { + mAnchorTimeUs = inHeader->nTimeStamp; + mNumFramesOutput = 0; + } - memset(outHeader->pBuffer, 0, outHeader->nFilledLen); - } else { - // Since we never discarded frames from the start, we won't have - // to add any padding at the end either. - outHeader->nFilledLen = 0; + if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + mSawInputEos = true; } - outHeader->nFlags = OMX_BUFFERFLAG_EOS; + mConfig->pInputBuffer = + inHeader->pBuffer + inHeader->nOffset; - outQueue.erase(outQueue.begin()); - outInfo->mOwnedByUs = false; - notifyFillBufferDone(outHeader); - return; - } - - if (inHeader->nOffset == 0) { - mAnchorTimeUs = inHeader->nTimeStamp; - mNumFramesOutput = 0; + mConfig->inputBufferCurrentLength = inHeader->nFilledLen; + } else { + mConfig->pInputBuffer = NULL; + mConfig->inputBufferCurrentLength = 0; } - - mConfig->pInputBuffer = - inHeader->pBuffer + inHeader->nOffset; - - mConfig->inputBufferCurrentLength = inHeader->nFilledLen; mConfig->inputBufferMaxLength = 0; mConfig->inputBufferUsedLength = 0; @@ -262,13 +252,28 @@ void SoftMP3::onQueueFilled(OMX_U32 portIndex) { mConfig->outputFrameSize = kOutputBufferSize / sizeof(int16_t); } - // This is recoverable, just ignore the current frame and - // play silence instead. - memset(outHeader->pBuffer, - 0, - mConfig->outputFrameSize * sizeof(int16_t)); - - mConfig->inputBufferUsedLength = inHeader->nFilledLen; + if (decoderErr == NO_ENOUGH_MAIN_DATA_ERROR && mSawInputEos) { + if (!mIsFirst) { + // pad the end of the stream with 529 samples, since that many samples + // were trimmed off the beginning when decoding started + outHeader->nOffset = 0; + outHeader->nFilledLen = kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t); + + memset(outHeader->pBuffer, 0, outHeader->nFilledLen); + } + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + mSignalledOutputEos = true; + } else { + // This is recoverable, just ignore the current frame and + // play silence instead. + memset(outHeader->pBuffer, + 0, + mConfig->outputFrameSize * sizeof(int16_t)); + + if (inHeader) { + mConfig->inputBufferUsedLength = inHeader->nFilledLen; + } + } } else if (mConfig->samplingRate != mSamplingRate || mConfig->num_channels != mNumChannels) { mSamplingRate = mConfig->samplingRate; @@ -289,7 +294,7 @@ void SoftMP3::onQueueFilled(OMX_U32 portIndex) { outHeader->nFilledLen = mConfig->outputFrameSize * sizeof(int16_t) - outHeader->nOffset; - } else { + } else if (!mSignalledOutputEos) { outHeader->nOffset = 0; outHeader->nFilledLen = mConfig->outputFrameSize * sizeof(int16_t); } @@ -298,23 +303,24 @@ void SoftMP3::onQueueFilled(OMX_U32 portIndex) { mAnchorTimeUs + (mNumFramesOutput * 1000000ll) / mConfig->samplingRate; - outHeader->nFlags = 0; - - CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength); + if (inHeader) { + CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength); - inHeader->nOffset += mConfig->inputBufferUsedLength; - inHeader->nFilledLen -= mConfig->inputBufferUsedLength; + inHeader->nOffset += mConfig->inputBufferUsedLength; + inHeader->nFilledLen -= mConfig->inputBufferUsedLength; - mNumFramesOutput += mConfig->outputFrameSize / mNumChannels; - if (inHeader->nFilledLen == 0) { - inInfo->mOwnedByUs = false; - inQueue.erase(inQueue.begin()); - inInfo = NULL; - notifyEmptyBufferDone(inHeader); - inHeader = NULL; + if (inHeader->nFilledLen == 0) { + inInfo->mOwnedByUs = false; + inQueue.erase(inQueue.begin()); + inInfo = NULL; + notifyEmptyBufferDone(inHeader); + inHeader = NULL; + } } + mNumFramesOutput += mConfig->outputFrameSize / mNumChannels; + outInfo->mOwnedByUs = false; outQueue.erase(outQueue.begin()); outInfo = NULL; @@ -362,6 +368,8 @@ void SoftMP3::onReset() { pvmp3_InitDecoder(mConfig, mDecoderBuf); mIsFirst = true; mSignalledError = false; + mSawInputEos = false; + mSignalledOutputEos = false; mOutputPortSettingsChange = NONE; } diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.h b/media/libstagefright/codecs/mp3dec/SoftMP3.h index 4af91ea..f9e7b53 100644 --- a/media/libstagefright/codecs/mp3dec/SoftMP3.h +++ b/media/libstagefright/codecs/mp3dec/SoftMP3.h @@ -61,6 +61,8 @@ private: bool mIsFirst; bool mSignalledError; + bool mSawInputEos; + bool mSignalledOutputEos; enum { NONE, diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp index 51bb958..515e4d3 100644 --- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp +++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp @@ -54,6 +54,8 @@ SoftVorbis::SoftVorbis( mAnchorTimeUs(0), mNumFramesOutput(0), mNumFramesLeftOnPage(-1), + mSawInputEos(false), + mSignalledOutputEos(false), mOutputPortSettingsChange(NONE) { initPorts(); CHECK_EQ(initDecoder(), (status_t)OK); @@ -290,48 +292,47 @@ void SoftVorbis::onQueueFilled(OMX_U32 portIndex) { return; } - while (!inQueue.empty() && !outQueue.empty()) { - BufferInfo *inInfo = *inQueue.begin(); - OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) { + BufferInfo *inInfo = NULL; + OMX_BUFFERHEADERTYPE *inHeader = NULL; + if (!inQueue.empty()) { + inInfo = *inQueue.begin(); + inHeader = inInfo->mHeader; + } BufferInfo *outInfo = *outQueue.begin(); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; - if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { - inQueue.erase(inQueue.begin()); - inInfo->mOwnedByUs = false; - notifyEmptyBufferDone(inHeader); + int32_t numPageSamples = 0; - outHeader->nFilledLen = 0; - outHeader->nFlags = OMX_BUFFERFLAG_EOS; + if (inHeader) { + if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + mSawInputEos = true; + } - outQueue.erase(outQueue.begin()); - outInfo->mOwnedByUs = false; - notifyFillBufferDone(outHeader); - return; - } + if (inHeader->nFilledLen || !mSawInputEos) { + CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples)); + memcpy(&numPageSamples, + inHeader->pBuffer + + inHeader->nOffset + inHeader->nFilledLen - 4, + sizeof(numPageSamples)); - int32_t numPageSamples; - CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples)); - memcpy(&numPageSamples, - inHeader->pBuffer - + inHeader->nOffset + inHeader->nFilledLen - 4, - sizeof(numPageSamples)); + if (inHeader->nOffset == 0) { + mAnchorTimeUs = inHeader->nTimeStamp; + mNumFramesOutput = 0; + } - if (numPageSamples >= 0) { - mNumFramesLeftOnPage = numPageSamples; + inHeader->nFilledLen -= sizeof(numPageSamples);; + } } - if (inHeader->nOffset == 0) { - mAnchorTimeUs = inHeader->nTimeStamp; - mNumFramesOutput = 0; + if (numPageSamples >= 0) { + mNumFramesLeftOnPage = numPageSamples; } - inHeader->nFilledLen -= sizeof(numPageSamples);; - ogg_buffer buf; - buf.data = inHeader->pBuffer + inHeader->nOffset; - buf.size = inHeader->nFilledLen; + buf.data = inHeader ? inHeader->pBuffer + inHeader->nOffset : NULL; + buf.size = inHeader ? inHeader->nFilledLen : 0; buf.refcount = 1; buf.ptr.owner = NULL; @@ -351,6 +352,7 @@ void SoftVorbis::onQueueFilled(OMX_U32 portIndex) { int numFrames = 0; + outHeader->nFlags = 0; int err = vorbis_dsp_synthesis(mState, &pack, 1); if (err != 0) { ALOGW("vorbis_dsp_synthesis returned %d", err); @@ -370,13 +372,16 @@ void SoftVorbis::onQueueFilled(OMX_U32 portIndex) { ALOGV("discarding %d frames at end of page", numFrames - mNumFramesLeftOnPage); numFrames = mNumFramesLeftOnPage; + if (mSawInputEos) { + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + mSignalledOutputEos = true; + } } mNumFramesLeftOnPage -= numFrames; } outHeader->nFilledLen = numFrames * sizeof(int16_t) * mVi->channels; outHeader->nOffset = 0; - outHeader->nFlags = 0; outHeader->nTimeStamp = mAnchorTimeUs @@ -384,11 +389,13 @@ void SoftVorbis::onQueueFilled(OMX_U32 portIndex) { mNumFramesOutput += numFrames; - inInfo->mOwnedByUs = false; - inQueue.erase(inQueue.begin()); - inInfo = NULL; - notifyEmptyBufferDone(inHeader); - inHeader = NULL; + if (inHeader) { + inInfo->mOwnedByUs = false; + inQueue.erase(inQueue.begin()); + inInfo = NULL; + notifyEmptyBufferDone(inHeader); + inHeader = NULL; + } outInfo->mOwnedByUs = false; outQueue.erase(outQueue.begin()); @@ -425,6 +432,8 @@ void SoftVorbis::onReset() { mVi = NULL; } + mSawInputEos = false; + mSignalledOutputEos = false; mOutputPortSettingsChange = NONE; } diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h index cb628a0..1d00816 100644 --- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h +++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h @@ -59,6 +59,8 @@ private: int64_t mAnchorTimeUs; int64_t mNumFramesOutput; int32_t mNumFramesLeftOnPage; + bool mSawInputEos; + bool mSignalledOutputEos; enum { NONE, diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index bd12ddc..233db44 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -632,9 +632,6 @@ sp<M3UParser> LiveSession::fetchPlaylist( // playlist unchanged *unchanged = true; - ALOGV("Playlist unchanged, refresh state is now %d", - (int)mRefreshState); - return NULL; } diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 973b779..1754bf2 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -47,6 +47,7 @@ namespace android { // static const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll; +const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll; PlaylistFetcher::PlaylistFetcher( const sp<AMessage> ¬ify, @@ -61,6 +62,7 @@ PlaylistFetcher::PlaylistFetcher( mSeqNumber(-1), mNumRetries(0), mStartup(true), + mPrepared(false), mNextPTSTimeUs(-1ll), mMonitorQueueGeneration(0), mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY), @@ -103,10 +105,16 @@ int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const { return segmentStartUs; } -bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const { - if (mPlaylist == NULL) { +int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const { + int64_t nowUs = ALooper::GetNowUs(); + + if (mPlaylist == NULL || mLastPlaylistFetchTimeUs < 0ll) { CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY); - return true; + return 0ll; + } + + if (mPlaylist->isComplete()) { + return (~0llu >> 1); } int32_t targetDurationSecs; @@ -157,7 +165,8 @@ bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const { break; } - return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs; + int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs; + return delayUs > 0ll ? delayUs : 0ll; } status_t PlaylistFetcher::decryptBuffer( @@ -274,7 +283,15 @@ status_t PlaylistFetcher::decryptBuffer( return OK; } -void PlaylistFetcher::postMonitorQueue(int64_t delayUs) { +void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) { + int64_t maxDelayUs = delayUsToRefreshPlaylist(); + if (maxDelayUs < minDelayUs) { + maxDelayUs = minDelayUs; + } + if (delayUs > maxDelayUs) { + ALOGV("Need to refresh playlist in %lld", maxDelayUs); + delayUs = maxDelayUs; + } sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id()); msg->setInt32("generation", mMonitorQueueGeneration); msg->post(delayUs); @@ -415,6 +432,7 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { if (mStartTimeUs >= 0ll) { mSeqNumber = -1; mStartup = true; + mPrepared = false; } postMonitorQueue(); @@ -456,40 +474,62 @@ void PlaylistFetcher::queueDiscontinuity( void PlaylistFetcher::onMonitorQueue() { bool downloadMore = false; + refreshPlaylist(); + + int32_t targetDurationSecs; + int64_t targetDurationUs = kMinBufferedDurationUs; + if (mPlaylist != NULL) { + CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); + targetDurationUs = targetDurationSecs * 1000000ll; + } - status_t finalResult; + // buffer at least 3 times the target duration, or up to 10 seconds + int64_t durationToBufferUs = targetDurationUs * 3; + if (durationToBufferUs > kMinBufferedDurationUs) { + durationToBufferUs = kMinBufferedDurationUs; + } + + int64_t bufferedDurationUs = 0ll; + status_t finalResult = NOT_ENOUGH_DATA; if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) { sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); - int64_t bufferedDurationUs = + bufferedDurationUs = packetSource->getBufferedDurationUs(&finalResult); - - downloadMore = (bufferedDurationUs < kMinBufferedDurationUs); finalResult = OK; } else { bool first = true; - int64_t minBufferedDurationUs = 0ll; for (size_t i = 0; i < mPacketSources.size(); ++i) { if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) { continue; } - int64_t bufferedDurationUs = + int64_t bufferedStreamDurationUs = mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); - - if (first || bufferedDurationUs < minBufferedDurationUs) { - minBufferedDurationUs = bufferedDurationUs; + if (first || bufferedStreamDurationUs < bufferedDurationUs) { + bufferedDurationUs = bufferedStreamDurationUs; first = false; } } + } + downloadMore = (bufferedDurationUs < durationToBufferUs); + + // signal start if buffered up at least the target size + if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) { + mPrepared = true; - downloadMore = - !first && (minBufferedDurationUs < kMinBufferedDurationUs); + ALOGV("prepared, buffered=%lld > %lld", + bufferedDurationUs, targetDurationUs); + sp<AMessage> msg = mNotify->dup(); + msg->setInt32("what", kWhatTemporarilyDoneFetching); + msg->post(); } if (finalResult == OK && downloadMore) { + ALOGV("monitoring, buffered=%lld < %lld", + bufferedDurationUs, durationToBufferUs); onDownloadNext(); } else { // Nothing to do yet, try again in a second. @@ -498,15 +538,17 @@ void PlaylistFetcher::onMonitorQueue() { msg->setInt32("what", kWhatTemporarilyDoneFetching); msg->post(); - postMonitorQueue(1000000ll); + int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2; + ALOGV("pausing for %lld, buffered=%lld > %lld", + delayUs, bufferedDurationUs, durationToBufferUs); + // :TRICKY: need to enforce minimum delay because the delay to + // refresh the playlist will become 0 + postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0); } } -void PlaylistFetcher::onDownloadNext() { - int64_t nowUs = ALooper::GetNowUs(); - - if (mLastPlaylistFetchTimeUs < 0ll - || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) { +status_t PlaylistFetcher::refreshPlaylist() { + if (delayUsToRefreshPlaylist() <= 0) { bool unchanged; sp<M3UParser> playlist = mSession->fetchPlaylist( mURI.c_str(), mPlaylistHash, &unchanged); @@ -522,7 +564,7 @@ void PlaylistFetcher::onDownloadNext() { } else { ALOGE("failed to load playlist at url '%s'", mURI.c_str()); notifyError(ERROR_IO); - return; + return ERROR_IO; } } else { mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY; @@ -535,6 +577,13 @@ void PlaylistFetcher::onDownloadNext() { mLastPlaylistFetchTimeUs = ALooper::GetNowUs(); } + return OK; +} + +void PlaylistFetcher::onDownloadNext() { + if (refreshPlaylist() != OK) { + return; + } int32_t firstSeqNumberInPlaylist; if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( @@ -553,12 +602,18 @@ void PlaylistFetcher::onDownloadNext() { if (mPlaylist->isComplete() || mPlaylist->isEvent()) { mSeqNumber = getSeqNumberForTime(mStartTimeUs); + ALOGV("Initial sequence number for time %lld is %ld from (%ld .. %ld)", + mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist, + lastSeqNumberInPlaylist); } else { // If this is a live session, start 3 segments from the end. mSeqNumber = lastSeqNumberInPlaylist - 3; if (mSeqNumber < firstSeqNumberInPlaylist) { mSeqNumber = firstSeqNumberInPlaylist; } + ALOGV("Initial sequence number for live event %ld from (%ld .. %ld)", + mSeqNumber, firstSeqNumberInPlaylist, + lastSeqNumberInPlaylist); } mStartTimeUs = -1ll; @@ -570,16 +625,34 @@ void PlaylistFetcher::onDownloadNext() { ++mNumRetries; if (mSeqNumber > lastSeqNumberInPlaylist) { - mLastPlaylistFetchTimeUs = -1; - postMonitorQueue(3000000ll); + // refresh in increasing fraction (1/2, 1/3, ...) of the + // playlist's target duration or 3 seconds, whichever is less + int32_t targetDurationSecs; + CHECK(mPlaylist->meta()->findInt32( + "target-duration", &targetDurationSecs)); + int64_t delayUs = mPlaylist->size() * targetDurationSecs * + 1000000ll / (1 + mNumRetries); + if (delayUs > kMaxMonitorDelayUs) { + delayUs = kMaxMonitorDelayUs; + } + ALOGV("sequence number high: %ld from (%ld .. %ld), monitor in %lld (retry=%d)", + mSeqNumber, firstSeqNumberInPlaylist, + lastSeqNumberInPlaylist, delayUs, mNumRetries); + postMonitorQueue(delayUs); return; } // we've missed the boat, let's start from the lowest sequence // number available and signal a discontinuity. - ALOGI("We've missed the boat, restarting playback."); - mSeqNumber = lastSeqNumberInPlaylist; + ALOGI("We've missed the boat, restarting playback." + " mStartup=%d, was looking for %d in %d-%d", + mStartup, mSeqNumber, firstSeqNumberInPlaylist, + lastSeqNumberInPlaylist); + mSeqNumber = lastSeqNumberInPlaylist - 3; + if (mSeqNumber < firstSeqNumberInPlaylist) { + mSeqNumber = firstSeqNumberInPlaylist; + } explicitDiscontinuity = true; // fall through diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index 1648e02..78dea20 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -79,6 +79,7 @@ private: }; static const int64_t kMinBufferedDurationUs; + static const int64_t kMaxMonitorDelayUs; sp<AMessage> mNotify; sp<LiveSession> mSession; @@ -97,6 +98,7 @@ private: int32_t mSeqNumber; int32_t mNumRetries; bool mStartup; + bool mPrepared; int64_t mNextPTSTimeUs; int32_t mMonitorQueueGeneration; @@ -120,10 +122,11 @@ private: status_t decryptBuffer( size_t playlistIndex, const sp<ABuffer> &buffer); - void postMonitorQueue(int64_t delayUs = 0); + void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0); void cancelMonitorQueue(); - bool timeToRefreshPlaylist(int64_t nowUs) const; + int64_t delayUsToRefreshPlaylist() const; + status_t refreshPlaylist(); // Returns the media time in us of the segment specified by seqNumber. // This is computed by summing the durations of all segments before it. diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp index 34d671a..a486522 100644 --- a/media/libstagefright/id3/ID3.cpp +++ b/media/libstagefright/id3/ID3.cpp @@ -468,49 +468,6 @@ void ID3::Iterator::getID(String8 *id) const { } } -static void convertISO8859ToString8( - const uint8_t *data, size_t size, - String8 *s) { - size_t utf8len = 0; - for (size_t i = 0; i < size; ++i) { - if (data[i] == '\0') { - size = i; - break; - } else if (data[i] < 0x80) { - ++utf8len; - } else { - utf8len += 2; - } - } - - if (utf8len == size) { - // Only ASCII characters present. - - s->setTo((const char *)data, size); - return; - } - - char *tmp = new char[utf8len]; - char *ptr = tmp; - for (size_t i = 0; i < size; ++i) { - if (data[i] == '\0') { - break; - } else if (data[i] < 0x80) { - *ptr++ = data[i]; - } else if (data[i] < 0xc0) { - *ptr++ = 0xc2; - *ptr++ = data[i]; - } else { - *ptr++ = 0xc3; - *ptr++ = data[i] - 64; - } - } - - s->setTo(tmp, utf8len); - - delete[] tmp; - tmp = NULL; -} // the 2nd argument is used to get the data following the \0 in a comment field void ID3::Iterator::getString(String8 *id, String8 *comment) const { @@ -543,7 +500,9 @@ void ID3::Iterator::getstring(String8 *id, bool otherdata) const { return; } - convertISO8859ToString8(frameData, mFrameSize, id); + // this is supposed to be ISO-8859-1, but pass it up as-is to the caller, who will figure + // out the real encoding + id->setTo((const char*)frameData, mFrameSize); return; } @@ -561,13 +520,13 @@ void ID3::Iterator::getstring(String8 *id, bool otherdata) const { } if (encoding == 0x00) { - // ISO 8859-1 - convertISO8859ToString8(frameData + 1, n, id); + // supposedly ISO 8859-1 + id->setTo((const char*)frameData + 1, n); } else if (encoding == 0x03) { - // UTF-8 + // supposedly UTF-8 id->setTo((const char *)(frameData + 1), n); } else if (encoding == 0x02) { - // UTF-16 BE, no byte order mark. + // supposedly UTF-16 BE, no byte order mark. // API wants number of characters, not number of bytes... int len = n / 2; const char16_t *framedata = (const char16_t *) (frameData + 1); @@ -583,7 +542,7 @@ void ID3::Iterator::getstring(String8 *id, bool otherdata) const { if (framedatacopy != NULL) { delete[] framedatacopy; } - } else { + } else if (encoding == 0x01) { // UCS-2 // API wants number of characters, not number of bytes... int len = n / 2; @@ -602,7 +561,27 @@ void ID3::Iterator::getstring(String8 *id, bool otherdata) const { framedata++; len--; } - id->setTo(framedata, len); + + // check if the resulting data consists entirely of 8-bit values + bool eightBit = true; + for (int i = 0; i < len; i++) { + if (framedata[i] > 0xff) { + eightBit = false; + break; + } + } + if (eightBit) { + // collapse to 8 bit, then let the media scanner client figure out the real encoding + char *frame8 = new char[len]; + for (int i = 0; i < len; i++) { + frame8[i] = framedata[i]; + } + id->setTo(frame8, len); + delete [] frame8; + } else { + id->setTo(framedata, len); + } + if (framedatacopy != NULL) { delete[] framedatacopy; } diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp index 175a263..cb57a2f 100644 --- a/media/libstagefright/mpeg2ts/ATSParser.cpp +++ b/media/libstagefright/mpeg2ts/ATSParser.cpp @@ -506,6 +506,11 @@ ATSParser::Stream::Stream( ElementaryStreamQueue::PCM_AUDIO); break; + case STREAMTYPE_AC3: + mQueue = new ElementaryStreamQueue( + ElementaryStreamQueue::AC3); + break; + default: break; } @@ -614,6 +619,7 @@ bool ATSParser::Stream::isAudio() const { case STREAMTYPE_MPEG2_AUDIO: case STREAMTYPE_MPEG2_AUDIO_ADTS: case STREAMTYPE_PCM_AUDIO: + case STREAMTYPE_AC3: return true; default: diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h index a10edc9..d4e30b4 100644 --- a/media/libstagefright/mpeg2ts/ATSParser.h +++ b/media/libstagefright/mpeg2ts/ATSParser.h @@ -88,6 +88,10 @@ struct ATSParser : public RefBase { STREAMTYPE_MPEG2_AUDIO_ADTS = 0x0f, STREAMTYPE_MPEG4_VIDEO = 0x10, STREAMTYPE_H264 = 0x1b, + + // From ATSC A/53 Part 3:2009, 6.7.1 + STREAMTYPE_AC3 = 0x81, + STREAMTYPE_PCM_AUDIO = 0x83, }; diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp index 8f9c9c8..ea79885 100644 --- a/media/libstagefright/mpeg2ts/ESQueue.cpp +++ b/media/libstagefright/mpeg2ts/ESQueue.cpp @@ -56,6 +56,122 @@ void ElementaryStreamQueue::clear(bool clearFormat) { } } +// Parse AC3 header assuming the current ptr is start position of syncframe, +// update metadata only applicable, and return the payload size +static unsigned parseAC3SyncFrame( + const uint8_t *ptr, size_t size, sp<MetaData> *metaData) { + static const unsigned channelCountTable[] = {2, 1, 2, 3, 4, 4, 5, 6}; + static const unsigned samplingRateTable[] = {48000, 44100, 32000}; + static const unsigned rates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, + 320, 384, 448, 512, 576, 640}; + + static const unsigned frameSizeTable[19][3] = { + { 64, 69, 96 }, + { 80, 87, 120 }, + { 96, 104, 144 }, + { 112, 121, 168 }, + { 128, 139, 192 }, + { 160, 174, 240 }, + { 192, 208, 288 }, + { 224, 243, 336 }, + { 256, 278, 384 }, + { 320, 348, 480 }, + { 384, 417, 576 }, + { 448, 487, 672 }, + { 512, 557, 768 }, + { 640, 696, 960 }, + { 768, 835, 1152 }, + { 896, 975, 1344 }, + { 1024, 1114, 1536 }, + { 1152, 1253, 1728 }, + { 1280, 1393, 1920 }, + }; + + ABitReader bits(ptr, size); + unsigned syncStartPos = 0; // in bytes + if (bits.numBitsLeft() < 16) { + return 0; + } + if (bits.getBits(16) != 0x0B77) { + return 0; + } + + if (bits.numBitsLeft() < 16 + 2 + 6 + 5 + 3 + 3) { + ALOGV("Not enough bits left for further parsing"); + return 0; + } + bits.skipBits(16); // crc1 + + unsigned fscod = bits.getBits(2); + if (fscod == 3) { + ALOGW("Incorrect fscod in AC3 header"); + return 0; + } + + unsigned frmsizecod = bits.getBits(6); + if (frmsizecod > 37) { + ALOGW("Incorrect frmsizecod in AC3 header"); + return 0; + } + + unsigned bsid = bits.getBits(5); + if (bsid > 8) { + ALOGW("Incorrect bsid in AC3 header. Possibly E-AC-3?"); + return 0; + } + + unsigned bsmod = bits.getBits(3); + unsigned acmod = bits.getBits(3); + unsigned cmixlev = 0; + unsigned surmixlev = 0; + unsigned dsurmod = 0; + + if ((acmod & 1) > 0 && acmod != 1) { + if (bits.numBitsLeft() < 2) { + return 0; + } + cmixlev = bits.getBits(2); + } + if ((acmod & 4) > 0) { + if (bits.numBitsLeft() < 2) { + return 0; + } + surmixlev = bits.getBits(2); + } + if (acmod == 2) { + if (bits.numBitsLeft() < 2) { + return 0; + } + dsurmod = bits.getBits(2); + } + + if (bits.numBitsLeft() < 1) { + return 0; + } + unsigned lfeon = bits.getBits(1); + + unsigned samplingRate = samplingRateTable[fscod]; + unsigned payloadSize = frameSizeTable[frmsizecod >> 1][fscod]; + if (fscod == 1) { + payloadSize += frmsizecod & 1; + } + payloadSize <<= 1; // convert from 16-bit words to bytes + + unsigned channelCount = channelCountTable[acmod] + lfeon; + + if (metaData != NULL) { + (*metaData)->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3); + (*metaData)->setInt32(kKeyChannelCount, channelCount); + (*metaData)->setInt32(kKeySampleRate, samplingRate); + } + + return payloadSize; +} + +static bool IsSeeminglyValidAC3Header(const uint8_t *ptr, size_t size) { + return parseAC3SyncFrame(ptr, size, NULL) > 0; +} + static bool IsSeeminglyValidADTSHeader(const uint8_t *ptr, size_t size) { if (size < 3) { // Not enough data to verify header. @@ -224,6 +340,33 @@ status_t ElementaryStreamQueue::appendData( break; } + case AC3: + { + uint8_t *ptr = (uint8_t *)data; + + ssize_t startOffset = -1; + for (size_t i = 0; i < size; ++i) { + if (IsSeeminglyValidAC3Header(&ptr[i], size - i)) { + startOffset = i; + break; + } + } + + if (startOffset < 0) { + return ERROR_MALFORMED; + } + + if (startOffset > 0) { + ALOGI("found something resembling an AC3 syncword at " + "offset %d", + startOffset); + } + + data = &ptr[startOffset]; + size -= startOffset; + break; + } + case MPEG_AUDIO: { uint8_t *ptr = (uint8_t *)data; @@ -328,6 +471,8 @@ sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnit() { return dequeueAccessUnitH264(); case AAC: return dequeueAccessUnitAAC(); + case AC3: + return dequeueAccessUnitAC3(); case MPEG_VIDEO: return dequeueAccessUnitMPEGVideo(); case MPEG4_VIDEO: @@ -340,6 +485,51 @@ sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnit() { } } +sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAC3() { + unsigned syncStartPos = 0; // in bytes + unsigned payloadSize = 0; + sp<MetaData> format = new MetaData; + while (true) { + if (syncStartPos + 2 >= mBuffer->size()) { + return NULL; + } + + payloadSize = parseAC3SyncFrame( + mBuffer->data() + syncStartPos, + mBuffer->size() - syncStartPos, + &format); + if (payloadSize > 0) { + break; + } + ++syncStartPos; + } + + if (mBuffer->size() < syncStartPos + payloadSize) { + ALOGV("Not enough buffer size for AC3"); + return NULL; + } + + if (mFormat == NULL) { + mFormat = format; + } + + sp<ABuffer> accessUnit = new ABuffer(syncStartPos + payloadSize); + memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + payloadSize); + + int64_t timeUs = fetchTimestamp(syncStartPos + payloadSize); + CHECK_GE(timeUs, 0ll); + accessUnit->meta()->setInt64("timeUs", timeUs); + + memmove( + mBuffer->data(), + mBuffer->data() + syncStartPos + payloadSize, + mBuffer->size() - syncStartPos - payloadSize); + + mBuffer->setRange(0, mBuffer->size() - syncStartPos - payloadSize); + + return accessUnit; +} + sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitPCMAudio() { if (mBuffer->size() < 4) { return NULL; diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h index 66a8087..a2cca77 100644 --- a/media/libstagefright/mpeg2ts/ESQueue.h +++ b/media/libstagefright/mpeg2ts/ESQueue.h @@ -32,6 +32,7 @@ struct ElementaryStreamQueue { enum Mode { H264, AAC, + AC3, MPEG_AUDIO, MPEG_VIDEO, MPEG4_VIDEO, @@ -67,6 +68,7 @@ private: sp<ABuffer> dequeueAccessUnitH264(); sp<ABuffer> dequeueAccessUnitAAC(); + sp<ABuffer> dequeueAccessUnitAC3(); sp<ABuffer> dequeueAccessUnitMPEGAudio(); sp<ABuffer> dequeueAccessUnitMPEGVideo(); sp<ABuffer> dequeueAccessUnitMPEG4Video(); diff --git a/media/libstagefright/timedtext/test/Android.mk b/media/libstagefright/timedtext/test/Android.mk index a5e7ba2..9a9fde2 100644 --- a/media/libstagefright/timedtext/test/Android.mk +++ b/media/libstagefright/timedtext/test/Android.mk @@ -2,7 +2,6 @@ LOCAL_PATH:= $(call my-dir) # ================================================================ # Unit tests for libstagefright_timedtext -# See also /development/testrunner/test_defs.xml # ================================================================ # ================================================================ @@ -18,10 +17,13 @@ LOCAL_SRC_FILES := TimedTextSRTSource_test.cpp LOCAL_C_INCLUDES := \ $(TOP)/external/expat/lib \ - $(TOP)/frameworks/base/media/libstagefright/timedtext + $(TOP)/frameworks/av/media/libstagefright/timedtext LOCAL_SHARED_LIBRARIES := \ + libbinder \ libexpat \ - libstagefright + libstagefright \ + libstagefright_foundation \ + libutils include $(BUILD_NATIVE_TEST) |