diff options
Diffstat (limited to 'media/libstagefright')
-rw-r--r-- | media/libstagefright/ACodec.cpp | 121 | ||||
-rw-r--r-- | media/libstagefright/MediaDefs.cpp | 1 | ||||
-rw-r--r-- | media/libstagefright/OMXCodec.cpp | 93 | ||||
-rw-r--r-- | media/libstagefright/mpeg2ts/ATSParser.cpp | 6 | ||||
-rw-r--r-- | media/libstagefright/mpeg2ts/ATSParser.h | 4 | ||||
-rw-r--r-- | media/libstagefright/mpeg2ts/ESQueue.cpp | 190 | ||||
-rw-r--r-- | media/libstagefright/mpeg2ts/ESQueue.h | 2 |
7 files changed, 358 insertions, 59 deletions
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index f5fb622..08a3c7f 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -37,7 +37,9 @@ #include <media/hardware/HardwareAPI.h> +#include <OMX_AudioExt.h> #include <OMX_Component.h> +#include <OMX_IndexExt.h> #include "include/avc_utils.h" @@ -640,34 +642,18 @@ status_t ACodec::configureOutputBuffersFromNativeWindow( return err; } - // FIXME: assume that surface is controlled by app (native window - // returns the number for the case when surface is not controlled by app) - // FIXME2: This means that minUndeqeueudBufs can be 1 larger than reported - // For now, try to allocate 1 more buffer, but don't fail if unsuccessful - - // Use conservative allocation while also trying to reduce starvation - // - // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the - // minimum needed for the consumer to be able to work - // 2. try to allocate two (2) additional buffers to reduce starvation from - // the consumer - // plus an extra buffer to account for incorrect minUndequeuedBufs - for (OMX_U32 extraBuffers = 2 + 1; /* condition inside loop */; extraBuffers--) { - OMX_U32 newBufferCount = - def.nBufferCountMin + *minUndequeuedBuffers + extraBuffers; + // XXX: Is this the right logic to use? It's not clear to me what the OMX + // buffer counts refer to - how do they account for the renderer holding on + // to buffers? + if (def.nBufferCountActual < def.nBufferCountMin + *minUndequeuedBuffers) { + OMX_U32 newBufferCount = def.nBufferCountMin + *minUndequeuedBuffers; def.nBufferCountActual = newBufferCount; err = mOMX->setParameter( mNode, OMX_IndexParamPortDefinition, &def, sizeof(def)); - if (err == OK) { - *minUndequeuedBuffers += extraBuffers; - break; - } - - ALOGW("[%s] setting nBufferCountActual to %lu failed: %d", - mComponentName.c_str(), newBufferCount, err); - /* exit condition */ - if (extraBuffers == 0) { + if (err != OK) { + ALOGE("[%s] setting nBufferCountActual to %lu failed: %d", + mComponentName.c_str(), newBufferCount, err); return err; } } @@ -692,7 +678,6 @@ status_t ACodec::allocateOutputBuffersFromNativeWindow() { &bufferCount, &bufferSize, &minUndequeuedBuffers); if (err != 0) return err; - mNumUndequeuedBuffers = minUndequeuedBuffers; ALOGV("[%s] Allocating %lu buffers from a native window of size %lu on " "output port", @@ -758,7 +743,6 @@ status_t ACodec::allocateOutputMetaDataBuffers() { &bufferCount, &bufferSize, &minUndequeuedBuffers); if (err != 0) return err; - mNumUndequeuedBuffers = minUndequeuedBuffers; ALOGV("[%s] Allocating %lu meta buffers on output port", mComponentName.c_str(), bufferCount); @@ -999,6 +983,10 @@ 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 = @@ -1297,6 +1285,15 @@ 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) { @@ -1493,6 +1490,44 @@ 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) { @@ -2449,7 +2484,19 @@ void ACodec::waitUntilAllPossibleNativeWindowBuffersAreReturnedToUs() { return; } - while (countBuffersOwnedByNativeWindow() > mNumUndequeuedBuffers + int minUndequeuedBufs = 0; + status_t err = mNativeWindow->query( + mNativeWindow.get(), NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS, + &minUndequeuedBufs); + + if (err != OK) { + ALOGE("[%s] NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS query failed: %s (%d)", + mComponentName.c_str(), strerror(-err), -err); + + minUndequeuedBufs = 0; + } + + while (countBuffersOwnedByNativeWindow() > (size_t)minUndequeuedBufs && dequeueBufferFromNativeWindow() != NULL) { // these buffers will be submitted as regular buffers; account for this if (mStoreMetaDataInOutputBuffers && mMetaDataBuffersToSubmit > 0) { @@ -2575,7 +2622,7 @@ 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; @@ -2681,6 +2728,24 @@ 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(); } 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 450fb3b..96c5a32 100644 --- a/media/libstagefright/OMXCodec.cpp +++ b/media/libstagefright/OMXCodec.cpp @@ -42,7 +42,9 @@ #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" @@ -94,7 +96,6 @@ static sp<MediaSource> InstantiateSoftwareEncoder( #define CODEC_LOGI(x, ...) ALOGI("[%s] "x, mComponentName, ##__VA_ARGS__) #define CODEC_LOGV(x, ...) ALOGV("[%s] "x, mComponentName, ##__VA_ARGS__) -#define CODEC_LOGW(x, ...) ALOGW("[%s] "x, mComponentName, ##__VA_ARGS__) #define CODEC_LOGE(x, ...) ALOGE("[%s] "x, mComponentName, ##__VA_ARGS__) struct OMXCodecObserver : public BnOMXObserver { @@ -531,6 +532,17 @@ 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 @@ -1397,6 +1409,10 @@ 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 = @@ -1780,42 +1796,21 @@ status_t OMXCodec::allocateOutputBuffersFromNativeWindow() { strerror(-err), -err); return err; } - // FIXME: assume that surface is controlled by app (native window - // returns the number for the case when surface is not controlled by app) - // FIXME2: This means that minUndeqeueudBufs can be 1 larger than reported - // For now, try to allocate 1 more buffer, but don't fail if unsuccessful - - // Use conservative allocation while also trying to reduce starvation - // - // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the - // minimum needed for the consumer to be able to work - // 2. try to allocate two (2) additional buffers to reduce starvation from - // the consumer - // plus an extra buffer to account for incorrect minUndequeuedBufs - CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d+1", - def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs); - - for (OMX_U32 extraBuffers = 2 + 1; /* condition inside loop */; extraBuffers--) { - OMX_U32 newBufferCount = - def.nBufferCountMin + minUndequeuedBufs + extraBuffers; + + // XXX: Is this the right logic to use? It's not clear to me what the OMX + // buffer counts refer to - how do they account for the renderer holding on + // to buffers? + if (def.nBufferCountActual < def.nBufferCountMin + minUndequeuedBufs) { + OMX_U32 newBufferCount = def.nBufferCountMin + minUndequeuedBufs; def.nBufferCountActual = newBufferCount; err = mOMX->setParameter( mNode, OMX_IndexParamPortDefinition, &def, sizeof(def)); - - if (err == OK) { - minUndequeuedBufs += extraBuffers; - break; - } - - CODEC_LOGW("setting nBufferCountActual to %lu failed: %d", - newBufferCount, err); - /* exit condition */ - if (extraBuffers == 0) { + if (err != OK) { + CODEC_LOGE("setting nBufferCountActual to %lu failed: %d", + newBufferCount, err); return err; } } - CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d+1", - def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs); err = native_window_set_buffer_count( mNativeWindow.get(), def.nBufferCountActual); @@ -3513,6 +3508,31 @@ 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); @@ -4446,6 +4466,17 @@ 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/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp index 2c8cf8d..d1afd8b 100644 --- a/media/libstagefright/mpeg2ts/ATSParser.cpp +++ b/media/libstagefright/mpeg2ts/ATSParser.cpp @@ -508,6 +508,11 @@ ATSParser::Stream::Stream( ElementaryStreamQueue::PCM_AUDIO); break; + case STREAMTYPE_AC3: + mQueue = new ElementaryStreamQueue( + ElementaryStreamQueue::AC3); + break; + default: break; } @@ -616,6 +621,7 @@ 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 8a80069..86b025f 100644 --- a/media/libstagefright/mpeg2ts/ATSParser.h +++ b/media/libstagefright/mpeg2ts/ATSParser.h @@ -89,6 +89,10 @@ 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 1960b27..e9252cc 100644 --- a/media/libstagefright/mpeg2ts/ESQueue.cpp +++ b/media/libstagefright/mpeg2ts/ESQueue.cpp @@ -57,6 +57,122 @@ 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, 3, 4, 4, 5}; + 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. @@ -225,6 +341,33 @@ 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; @@ -329,6 +472,8 @@ sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnit() { return dequeueAccessUnitH264(); case AAC: return dequeueAccessUnitAAC(); + case AC3: + return dequeueAccessUnitAC3(); case MPEG_VIDEO: return dequeueAccessUnitMPEGVideo(); case MPEG4_VIDEO: @@ -341,6 +486,51 @@ 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(); |