diff options
Diffstat (limited to 'media')
21 files changed, 227 insertions, 76 deletions
diff --git a/media/libeffects/downmix/Android.mk b/media/libeffects/downmix/Android.mk index 2bb6dbe..e0ca8af 100644 --- a/media/libeffects/downmix/Android.mk +++ b/media/libeffects/downmix/Android.mk @@ -15,16 +15,10 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE_RELATIVE_PATH := soundfx -ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true) -LOCAL_LDLIBS += -ldl -endif - LOCAL_C_INCLUDES := \ $(call include-path-for, audio-effects) \ $(call include-path-for, audio-utils) -LOCAL_PRELINK_MODULE := false - LOCAL_CFLAGS += -fvisibility=hidden include $(BUILD_SHARED_LIBRARY) diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk index 9e8cb83..ea3c59d 100644 --- a/media/libeffects/preprocessing/Android.mk +++ b/media/libeffects/preprocessing/Android.mk @@ -24,12 +24,7 @@ LOCAL_SHARED_LIBRARIES := \ libutils \ liblog -ifeq ($(TARGET_SIMULATOR),true) -LOCAL_LDLIBS += -ldl -else LOCAL_SHARED_LIBRARIES += libdl -endif - LOCAL_CFLAGS += -fvisibility=hidden include $(BUILD_SHARED_LIBRARY) diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 2c8605c..97ab8f8 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -484,6 +484,8 @@ status_t AudioRecord::openRecord_l(size_t epoch) size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, // but we will still need the original value also int originalSessionId = mSessionId; + sp<IMemory> iMem; // for cblk + sp<IMemory> bufferMem; sp<IAudioRecord> record = audioFlinger->openRecord(input, mSampleRate, mFormat, mChannelMask, @@ -491,6 +493,8 @@ status_t AudioRecord::openRecord_l(size_t epoch) &trackFlags, tid, &mSessionId, + iMem, + bufferMem, &status); ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, "session ID changed from %d to %d", originalSessionId, mSessionId); @@ -504,7 +508,6 @@ status_t AudioRecord::openRecord_l(size_t epoch) // AudioFlinger now owns the reference to the I/O handle, // so we are no longer responsible for releasing it. - sp<IMemory> iMem = record->getCblk(); if (iMem == 0) { ALOGE("Could not get control block"); return NO_INIT; @@ -514,6 +517,22 @@ status_t AudioRecord::openRecord_l(size_t epoch) ALOGE("Could not get control block pointer"); return NO_INIT; } + audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); + + // Starting address of buffers in shared memory. + // The buffers are either immediately after the control block, + // or in a separate area at discretion of server. + void *buffers; + if (bufferMem == 0) { + buffers = cblk + 1; + } else { + buffers = bufferMem->pointer(); + if (buffers == NULL) { + ALOGE("Could not get buffer pointer"); + return NO_INIT; + } + } + // invariant that mAudioRecord != 0 is true only after set() returns successfully if (mAudioRecord != 0) { mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); @@ -522,7 +541,7 @@ status_t AudioRecord::openRecord_l(size_t epoch) mAudioRecord = record; mCblkMemory = iMem; - audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); + mBufferMemory = bufferMem; mCblk = cblk; // note that temp is the (possibly revised) value of frameCount if (temp < frameCount || (frameCount == 0 && temp == 0)) { @@ -552,11 +571,6 @@ status_t AudioRecord::openRecord_l(size_t epoch) mInput = input; mRefreshRemaining = true; - // Starting address of buffers in shared memory, immediately after the control block. This - // address is for the mapping within client address space. AudioFlinger::TrackBase::mBuffer - // is for the server address space. - void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); - mFrameCount = frameCount; // If IAudioRecord is re-created, don't let the requested frameCount // decrease. This can confuse clients that cache frameCount(). @@ -631,6 +645,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *r // keep them from going away if another thread re-creates the track during obtainBuffer() sp<AudioRecordClientProxy> proxy; sp<IMemory> iMem; + sp<IMemory> bufferMem; { // start of lock scope AutoMutex lock(mLock); @@ -654,6 +669,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *r // Keep the extra references proxy = mProxy; iMem = mCblkMemory; + bufferMem = mBufferMemory; // Non-blocking if track is stopped if (!mActive) { @@ -986,7 +1002,7 @@ status_t AudioRecord::restoreRecord_l(const char *from) status_t result; // if the new IAudioRecord is created, openRecord_l() will modify the - // following member variables: mAudioRecord, mCblkMemory and mCblk. + // following member variables: mAudioRecord, mCblkMemory, mCblk, mBufferMemory. // It will also delete the strong references on previous IAudioRecord and IMemory size_t position = mProxy->getPosition(); mNewPosition = position + mUpdatePeriod; diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index dc4f90e..aaaa3f1 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -315,12 +315,20 @@ status_t AudioTrack::set( flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); } - if (audio_is_linear_pcm(format)) { - mFrameSize = channelCount * audio_bytes_per_sample(format); - mFrameSizeAF = channelCount * sizeof(int16_t); + if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { + if (audio_is_linear_pcm(format)) { + mFrameSize = channelCount * audio_bytes_per_sample(format); + } else { + mFrameSize = sizeof(uint8_t); + } + mFrameSizeAF = mFrameSize; } else { - mFrameSize = sizeof(uint8_t); - mFrameSizeAF = sizeof(uint8_t); + ALOG_ASSERT(audio_is_linear_pcm(format)); + mFrameSize = channelCount * audio_bytes_per_sample(format); + mFrameSizeAF = channelCount * audio_bytes_per_sample( + format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); + // createTrack will return an error if PCM format is not supported by server, + // so no need to check for specific PCM formats here } // Make copy of input parameter offloadInfo so that in the future: @@ -931,7 +939,11 @@ status_t AudioTrack::createTrack_l(size_t epoch) // Ensure that buffer alignment matches channel count // 8-bit data in shared memory is not currently supported by AudioFlinger - size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; + size_t alignment = audio_bytes_per_sample( + mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); + if (alignment & 1) { + alignment = 1; + } if (mChannelCount > 1) { // More than 2 channels does not require stronger alignment than stereo alignment <<= 1; @@ -947,7 +959,7 @@ status_t AudioTrack::createTrack_l(size_t epoch) // there's no frameCount parameter. // But when initializing a shared buffer AudioTrack via set(), // there _is_ a frameCount parameter. We silently ignore it. - frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t); + frameCount = mSharedBuffer->size() / mFrameSizeAF; } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp index 58c9fc1..323b675 100644 --- a/media/libmedia/AudioTrackShared.cpp +++ b/media/libmedia/AudioTrackShared.cpp @@ -621,7 +621,7 @@ void ServerProxy::releaseBuffer(Buffer* buffer) android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear); } - mCblk->mServer += stepCount; + cblk->mServer += stepCount; size_t half = mFrameCount / 2; if (half == 0) { @@ -679,10 +679,11 @@ size_t AudioTrackServerProxy::framesReady() } bool AudioTrackServerProxy::setStreamEndDone() { + audio_track_cblk_t* cblk = mCblk; bool old = - (android_atomic_or(CBLK_STREAM_END_DONE, &mCblk->mFlags) & CBLK_STREAM_END_DONE) != 0; + (android_atomic_or(CBLK_STREAM_END_DONE, &cblk->mFlags) & CBLK_STREAM_END_DONE) != 0; if (!old) { - (void) __futex_syscall3(&mCblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, + (void) __futex_syscall3(&cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1); } return old; @@ -690,10 +691,11 @@ bool AudioTrackServerProxy::setStreamEndDone() { void AudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount) { - mCblk->u.mStreaming.mUnderrunFrames += frameCount; + audio_track_cblk_t* cblk = mCblk; + cblk->u.mStreaming.mUnderrunFrames += frameCount; // FIXME also wake futex so that underrun is noticed more quickly - (void) android_atomic_or(CBLK_UNDERRUN, &mCblk->mFlags); + (void) android_atomic_or(CBLK_UNDERRUN, &cblk->mFlags); } // --------------------------------------------------------------------------- diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index 1940fe7..0e2463e 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -169,6 +169,8 @@ public: track_flags_t *flags, pid_t tid, int *sessionId, + sp<IMemory>& cblk, + sp<IMemory>& buffers, status_t *status) { Parcel data, reply; @@ -188,6 +190,8 @@ public: lSessionId = *sessionId; } data.writeInt32(lSessionId); + cblk.clear(); + buffers.clear(); status_t lStatus = remote()->transact(OPEN_RECORD, data, &reply); if (lStatus != NO_ERROR) { ALOGE("openRecord error: %s", strerror(-lStatus)); @@ -206,17 +210,34 @@ public: } lStatus = reply.readInt32(); record = interface_cast<IAudioRecord>(reply.readStrongBinder()); + cblk = interface_cast<IMemory>(reply.readStrongBinder()); + if (cblk != 0 && cblk->pointer() == NULL) { + cblk.clear(); + } + buffers = interface_cast<IMemory>(reply.readStrongBinder()); + if (buffers != 0 && buffers->pointer() == NULL) { + buffers.clear(); + } if (lStatus == NO_ERROR) { if (record == 0) { ALOGE("openRecord should have returned an IAudioRecord"); lStatus = UNKNOWN_ERROR; + } else if (cblk == 0) { + ALOGE("openRecord should have returned a cblk"); + lStatus = NO_MEMORY; } + // buffers is permitted to be 0 } else { - if (record != 0) { - ALOGE("openRecord returned an IAudioRecord but with status %d", lStatus); - record.clear(); + if (record != 0 || cblk != 0 || buffers != 0) { + ALOGE("openRecord returned an IAudioRecord, cblk, " + "or buffers but with status %d", lStatus); } } + if (lStatus != NO_ERROR) { + record.clear(); + cblk.clear(); + buffers.clear(); + } } if (status != NULL) { *status = lStatus; @@ -838,15 +859,20 @@ status_t BnAudioFlinger::onTransact( track_flags_t flags = (track_flags_t) data.readInt32(); pid_t tid = (pid_t) data.readInt32(); int sessionId = data.readInt32(); + sp<IMemory> cblk; + sp<IMemory> buffers; status_t status; sp<IAudioRecord> record = openRecord(input, - sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId, &status); + sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId, + cblk, buffers, &status); LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR)); reply->writeInt64(frameCount); reply->writeInt32(flags); reply->writeInt32(sessionId); reply->writeInt32(status); reply->writeStrongBinder(record->asBinder()); + reply->writeStrongBinder(cblk->asBinder()); + reply->writeStrongBinder(buffers->asBinder()); return NO_ERROR; } break; case SAMPLE_RATE: { diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp index 9866d70..8a4a383 100644 --- a/media/libmedia/IAudioRecord.cpp +++ b/media/libmedia/IAudioRecord.cpp @@ -29,7 +29,7 @@ namespace android { enum { - GET_CBLK = IBinder::FIRST_CALL_TRANSACTION, + UNUSED_WAS_GET_CBLK = IBinder::FIRST_CALL_TRANSACTION, START, STOP }; @@ -42,21 +42,6 @@ public: { } - virtual sp<IMemory> getCblk() const - { - Parcel data, reply; - sp<IMemory> cblk; - data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor()); - status_t status = remote()->transact(GET_CBLK, data, &reply); - if (status == NO_ERROR) { - cblk = interface_cast<IMemory>(reply.readStrongBinder()); - if (cblk != 0 && cblk->pointer() == NULL) { - cblk.clear(); - } - } - return cblk; - } - virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { Parcel data, reply; @@ -89,11 +74,6 @@ status_t BnAudioRecord::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { switch (code) { - case GET_CBLK: { - CHECK_INTERFACE(IAudioRecord, data, reply); - reply->writeStrongBinder(getCblk()->asBinder()); - return NO_ERROR; - } break; case START: { CHECK_INTERFACE(IAudioRecord, data, reply); int /*AudioSystem::sync_event_t*/ event = data.readInt32(); diff --git a/media/libnbaio/AudioStreamInSource.cpp b/media/libnbaio/AudioStreamInSource.cpp index 80bf61a..af8f365 100644 --- a/media/libnbaio/AudioStreamInSource.cpp +++ b/media/libnbaio/AudioStreamInSource.cpp @@ -63,7 +63,7 @@ size_t AudioStreamInSource::framesOverrun() return mFramesOverrun; } -ssize_t AudioStreamInSource::read(void *buffer, size_t count) +ssize_t AudioStreamInSource::read(void *buffer, size_t count, int64_t readPTS __unused) { if (CC_UNLIKELY(!Format_isValid(mFormat))) { return NEGOTIATE; diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp index 28a034c..6e0ec8c 100644 --- a/media/libnbaio/Pipe.cpp +++ b/media/libnbaio/Pipe.cpp @@ -25,19 +25,22 @@ namespace android { -Pipe::Pipe(size_t maxFrames, const NBAIO_Format& format) : +Pipe::Pipe(size_t maxFrames, const NBAIO_Format& format, void *buffer) : NBAIO_Sink(format), mMaxFrames(roundup(maxFrames)), - mBuffer(malloc(mMaxFrames * Format_frameSize(format))), + mBuffer(buffer == NULL ? malloc(mMaxFrames * Format_frameSize(format)) : buffer), mRear(0), - mReaders(0) + mReaders(0), + mFreeBufferInDestructor(buffer == NULL) { } Pipe::~Pipe() { ALOG_ASSERT(android_atomic_acquire_load(&mReaders) == 0); - free(mBuffer); + if (mFreeBufferInDestructor) { + free(mBuffer); + } } ssize_t Pipe::write(const void *buffer, size_t count) diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index 537d9de..0a3a3b6 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -3670,7 +3670,28 @@ void ACodec::BaseState::onOutputBufferDrained(const sp<AMessage> &msg) { ATRACE_NAME("render"); // The client wants this buffer to be rendered. + int64_t timestampNs = 0; + if (!msg->findInt64("timestampNs", ×tampNs)) { + // TODO: it seems like we should use the timestamp + // in the (media)buffer as it potentially came from + // an input surface, but we did not propagate it prior to + // API 20. Perhaps check for target SDK version. +#if 0 + if (info->mData->meta()->findInt64("timeUs", ×tampNs)) { + ALOGI("using buffer PTS of %" PRId64, timestampNs); + timestampNs *= 1000; + } +#endif + } + status_t err; + err = native_window_set_buffers_timestamp(mCodec->mNativeWindow.get(), timestampNs); + if (err != OK) { + ALOGW("failed to set buffer timestamp: %d", err); + } else { + ALOGI("set PTS to %" PRId64, timestampNs); + } + if ((err = mCodec->mNativeWindow->queueBuffer( mCodec->mNativeWindow.get(), info->mGraphicBuffer.get(), -1)) == OK) { diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp index 8d3032b..d679be1 100644 --- a/media/libstagefright/AwesomePlayer.cpp +++ b/media/libstagefright/AwesomePlayer.cpp @@ -106,12 +106,15 @@ struct AwesomeLocalRenderer : public AwesomeRenderer { } virtual void render(MediaBuffer *buffer) { + int64_t timeUs; + CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs)); + render((const uint8_t *)buffer->data() + buffer->range_offset(), - buffer->range_length()); + buffer->range_length(), timeUs * 1000); } - void render(const void *data, size_t size) { - mTarget->render(data, size, NULL); + void render(const void *data, size_t size, int64_t timestampNs) { + mTarget->render(data, size, timestampNs, NULL); } protected: diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp index 2a3fa04..e07b6aa 100644 --- a/media/libstagefright/MPEG4Extractor.cpp +++ b/media/libstagefright/MPEG4Extractor.cpp @@ -3540,7 +3540,7 @@ status_t MPEG4Source::read( off64_t offset; size_t size; - uint32_t cts; + uint32_t cts, stts; bool isSyncSample; bool newBuffer = false; if (mBuffer == NULL) { @@ -3548,7 +3548,7 @@ status_t MPEG4Source::read( status_t err = mSampleTable->getMetaDataForSample( - mCurrentSampleIndex, &offset, &size, &cts, &isSyncSample); + mCurrentSampleIndex, &offset, &size, &cts, &isSyncSample, &stts); if (err != OK) { return err; @@ -3579,6 +3579,8 @@ status_t MPEG4Source::read( mBuffer->meta_data()->clear(); mBuffer->meta_data()->setInt64( kKeyTime, ((int64_t)cts * 1000000) / mTimescale); + mBuffer->meta_data()->setInt64( + kKeyDuration, ((int64_t)stts * 1000000) / mTimescale); if (targetSampleTimeUs >= 0) { mBuffer->meta_data()->setInt64( @@ -3701,6 +3703,8 @@ status_t MPEG4Source::read( mBuffer->meta_data()->clear(); mBuffer->meta_data()->setInt64( kKeyTime, ((int64_t)cts * 1000000) / mTimescale); + mBuffer->meta_data()->setInt64( + kKeyDuration, ((int64_t)stts * 1000000) / mTimescale); if (targetSampleTimeUs >= 0) { mBuffer->meta_data()->setInt64( @@ -3850,6 +3854,8 @@ status_t MPEG4Source::fragmentedRead( mBuffer->set_range(0, size); mBuffer->meta_data()->setInt64( kKeyTime, ((int64_t)cts * 1000000) / mTimescale); + mBuffer->meta_data()->setInt64( + kKeyDuration, ((int64_t)smpl->duration * 1000000) / mTimescale); if (targetSampleTimeUs >= 0) { mBuffer->meta_data()->setInt64( @@ -3973,6 +3979,8 @@ status_t MPEG4Source::fragmentedRead( mBuffer->meta_data()->setInt64( kKeyTime, ((int64_t)cts * 1000000) / mTimescale); + mBuffer->meta_data()->setInt64( + kKeyDuration, ((int64_t)smpl->duration * 1000000) / mTimescale); if (targetSampleTimeUs >= 0) { mBuffer->meta_data()->setInt64( diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp index 601dccf..5b525f2 100644 --- a/media/libstagefright/MediaCodec.cpp +++ b/media/libstagefright/MediaCodec.cpp @@ -17,6 +17,7 @@ //#define LOG_NDEBUG 0 #define LOG_TAG "MediaCodec" #include <utils/Log.h> +#include <inttypes.h> #include <media/stagefright/MediaCodec.h> @@ -323,6 +324,16 @@ status_t MediaCodec::renderOutputBufferAndRelease(size_t index) { return PostAndAwaitResponse(msg, &response); } +status_t MediaCodec::renderOutputBufferAndRelease(size_t index, int64_t timestampNs) { + sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id()); + msg->setSize("index", index); + msg->setInt32("render", true); + msg->setInt64("timestampNs", timestampNs); + + sp<AMessage> response; + return PostAndAwaitResponse(msg, &response); +} + status_t MediaCodec::releaseOutputBuffer(size_t index) { sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id()); msg->setSize("index", index); @@ -1707,9 +1718,25 @@ status_t MediaCodec::onReleaseOutputBuffer(const sp<AMessage> &msg) { if (render && info->mData != NULL && info->mData->size() != 0) { info->mNotify->setInt32("render", true); + int64_t timestampNs = 0; + if (msg->findInt64("timestampNs", ×tampNs)) { + info->mNotify->setInt64("timestampNs", timestampNs); + } else { + // TODO: it seems like we should use the timestamp + // in the (media)buffer as it potentially came from + // an input surface, but we did not propagate it prior to + // API 20. Perhaps check for target SDK version. +#if 0 + if (info->mData->meta()->findInt64("timeUs", ×tampNs)) { + ALOGI("using buffer PTS of %" PRId64, timestampNs); + timestampNs *= 1000; + } +#endif + } + if (mSoftRenderer != NULL) { mSoftRenderer->render( - info->mData->data(), info->mData->size(), NULL); + info->mData->data(), info->mData->size(), timestampNs, NULL); } } diff --git a/media/libstagefright/SampleIterator.cpp b/media/libstagefright/SampleIterator.cpp index eae721b..2748349 100644 --- a/media/libstagefright/SampleIterator.cpp +++ b/media/libstagefright/SampleIterator.cpp @@ -133,7 +133,8 @@ status_t SampleIterator::seekTo(uint32_t sampleIndex) { } status_t err; - if ((err = findSampleTime(sampleIndex, &mCurrentSampleTime)) != OK) { + if ((err = findSampleTimeAndDuration( + sampleIndex, &mCurrentSampleTime, &mCurrentSampleDuration)) != OK) { ALOGE("findSampleTime return error"); return err; } @@ -285,8 +286,8 @@ status_t SampleIterator::getSampleSizeDirect( return OK; } -status_t SampleIterator::findSampleTime( - uint32_t sampleIndex, uint32_t *time) { +status_t SampleIterator::findSampleTimeAndDuration( + uint32_t sampleIndex, uint32_t *time, uint32_t *duration) { if (sampleIndex >= mTable->mNumSampleSizes) { return ERROR_OUT_OF_RANGE; } @@ -309,6 +310,8 @@ status_t SampleIterator::findSampleTime( *time += mTable->getCompositionTimeOffset(sampleIndex); + *duration = mTTSDuration; + return OK; } diff --git a/media/libstagefright/SampleTable.cpp b/media/libstagefright/SampleTable.cpp index d9858d7..9a92805 100644 --- a/media/libstagefright/SampleTable.cpp +++ b/media/libstagefright/SampleTable.cpp @@ -778,7 +778,8 @@ status_t SampleTable::getMetaDataForSample( off64_t *offset, size_t *size, uint32_t *compositionTime, - bool *isSyncSample) { + bool *isSyncSample, + uint32_t *sampleDuration) { Mutex::Autolock autoLock(mLock); status_t err; @@ -820,6 +821,10 @@ status_t SampleTable::getMetaDataForSample( } } + if (sampleDuration) { + *sampleDuration = mSampleIterator->getSampleDuration(); + } + return OK; } diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp index 77f21b7..67dfcd2 100644 --- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp +++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp @@ -138,7 +138,7 @@ static int ALIGN(int x, int y) { } void SoftwareRenderer::render( - const void *data, size_t size, void *platformPrivate) { + const void *data, size_t size, int64_t timestampNs, void *platformPrivate) { ANativeWindowBuffer *buf; int err; if ((err = native_window_dequeue_buffer_and_wait(mNativeWindow.get(), @@ -230,6 +230,11 @@ void SoftwareRenderer::render( CHECK_EQ(0, mapper.unlock(buf->handle)); + if ((err = native_window_set_buffers_timestamp(mNativeWindow.get(), + timestampNs)) != 0) { + ALOGW("Surface::set_buffers_timestamp returned error %d", err); + } + if ((err = mNativeWindow->queueBuffer(mNativeWindow.get(), buf, -1)) != 0) { ALOGW("Surface::queueBuffer returned error %d", err); diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index c34f3cb..326d85b 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -69,6 +69,7 @@ PlaylistFetcher::PlaylistFetcher( mNumRetries(0), mStartup(true), mPrepared(false), + mSkipToFirstIDRAfterConnect(false), mNextPTSTimeUs(-1ll), mMonitorQueueGeneration(0), mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY), @@ -1097,12 +1098,30 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu continue; } + if (stream == LiveSession::STREAMTYPE_VIDEO && mVideoMime.empty()) { + const char *mime; + if (source->getFormat()->findCString(kKeyMIMEType, &mime)) { + mVideoMime.setTo(mime); + if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)) { + mSkipToFirstIDRAfterConnect = true; + } + } + } + int64_t timeUs; sp<ABuffer> accessUnit; status_t finalResult; while (source->hasBufferAvailable(&finalResult) && source->dequeueAccessUnit(&accessUnit) == OK) { + if (stream == LiveSession::STREAMTYPE_VIDEO && mSkipToFirstIDRAfterConnect) { + if (!IsIDR(accessUnit)) { + continue; + } else { + mSkipToFirstIDRAfterConnect = false; + } + } + CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); if (mMinStartTimeUs > 0) { if (timeUs < mMinStartTimeUs) { @@ -1183,9 +1202,35 @@ status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &bu return OK; } +/* static */ +bool PlaylistFetcher::bufferStartsWithWebVTTMagicSequence( + const sp<ABuffer> &buffer) { + size_t pos = 0; + + // skip possible BOM + if (buffer->size() >= pos + 3 && + !memcmp("\xef\xbb\xbf", buffer->data() + pos, 3)) { + pos += 3; + } + + // accept WEBVTT followed by SPACE, TAB or (CR) LF + if (buffer->size() < pos + 6 || + memcmp("WEBVTT", buffer->data() + pos, 6)) { + return false; + } + pos += 6; + + if (buffer->size() == pos) { + return true; + } + + uint8_t sep = buffer->data()[pos]; + return sep == ' ' || sep == '\t' || sep == '\n' || sep == '\r'; +} + status_t PlaylistFetcher::extractAndQueueAccessUnits( const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta) { - if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) { + if (bufferStartsWithWebVTTMagicSequence(buffer)) { if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) { ALOGE("This stream only contains subtitles."); return ERROR_MALFORMED; diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index 7e21523..e4fdbff 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -91,6 +91,7 @@ private: static const int32_t kNumSkipFrames; static bool bufferStartsWithTsSyncByte(const sp<ABuffer>& buffer); + static bool bufferStartsWithWebVTTMagicSequence(const sp<ABuffer>& buffer); // notifications to mSession sp<AMessage> mNotify; @@ -98,6 +99,7 @@ private: sp<LiveSession> mSession; AString mURI; + AString mVideoMime; uint32_t mStreamTypeMask; int64_t mStartTimeUs; @@ -115,6 +117,7 @@ private: int32_t mNumRetries; bool mStartup; bool mPrepared; + bool mSkipToFirstIDRAfterConnect; int64_t mNextPTSTimeUs; int32_t mMonitorQueueGeneration; diff --git a/media/libstagefright/include/SampleIterator.h b/media/libstagefright/include/SampleIterator.h index b5a043c..60c9e7e 100644 --- a/media/libstagefright/include/SampleIterator.h +++ b/media/libstagefright/include/SampleIterator.h @@ -30,6 +30,7 @@ struct SampleIterator { off64_t getSampleOffset() const { return mCurrentSampleOffset; } size_t getSampleSize() const { return mCurrentSampleSize; } uint32_t getSampleTime() const { return mCurrentSampleTime; } + uint32_t getSampleDuration() const { return mCurrentSampleDuration; } status_t getSampleSizeDirect( uint32_t sampleIndex, size_t *size); @@ -61,11 +62,12 @@ private: off64_t mCurrentSampleOffset; size_t mCurrentSampleSize; uint32_t mCurrentSampleTime; + uint32_t mCurrentSampleDuration; void reset(); status_t findChunkRange(uint32_t sampleIndex); status_t getChunkOffset(uint32_t chunk, off64_t *offset); - status_t findSampleTime(uint32_t sampleIndex, uint32_t *time); + status_t findSampleTimeAndDuration(uint32_t sampleIndex, uint32_t *time, uint32_t *duration); SampleIterator(const SampleIterator &); SampleIterator &operator=(const SampleIterator &); diff --git a/media/libstagefright/include/SampleTable.h b/media/libstagefright/include/SampleTable.h index 847dff7..fe146f2 100644 --- a/media/libstagefright/include/SampleTable.h +++ b/media/libstagefright/include/SampleTable.h @@ -66,7 +66,8 @@ public: off64_t *offset, size_t *size, uint32_t *compositionTime, - bool *isSyncSample = NULL); + bool *isSyncSample = NULL, + uint32_t *sampleDuration = NULL); enum { kFlagBefore, diff --git a/media/libstagefright/include/SoftwareRenderer.h b/media/libstagefright/include/SoftwareRenderer.h index 7ab0042..0ba670c 100644 --- a/media/libstagefright/include/SoftwareRenderer.h +++ b/media/libstagefright/include/SoftwareRenderer.h @@ -34,7 +34,7 @@ public: ~SoftwareRenderer(); void render( - const void *data, size_t size, void *platformPrivate); + const void *data, size_t size, int64_t timestampNs, void *platformPrivate); private: enum YUVMode { |