/* vim:set ts=2 sw=2 sts=2 et cindent: */ /* ***** BEGIN LICENSE BLOCK ***** * Version: ML 1.1/GPL 2.0/LGPL 2.1 * * The contents of this file are subject to the Mozilla Public License Version * 1.1 (the "License"); you may not use this file except in compliance with * the License. You may obtain a copy of the License at * http://www.mozilla.org/MPL/ * * Software distributed under the License is distributed on an "AS IS" basis, * WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License * for the specific language governing rights and limitations under the * License. * * The Original Code is Mozilla code. * * The Initial Developer of the Original Code is the Mozilla Corporation. * Portions created by the Initial Developer are Copyright (C) 2007 * the Initial Developer. All Rights Reserved. * * Contributor(s): * Chris Double * Chris Pearce * * Alternatively, the contents of this file may be used under the terms of * either the GNU General Public License Version 2 or later (the "GPL"), or * the GNU Lesser General Public License Version 2.1 or later (the "LGPL"), * in which case the provisions of the GPL or the LGPL are applicable instead * of those above. If you wish to allow use of your version of this file only * under the terms of either the GPL or the LGPL, and not to allow others to * use your version of this file under the terms of the MPL, indicate your * decision by deleting the provisions above and replace them with the notice * and other provisions required by the GPL or the LGPL. If you do not delete * the provisions above, a recipient may use your version of this file under * the terms of any one of the MPL, the GPL or the LGPL. * * ***** END LICENSE BLOCK ***** */ #include #include "nsAudioStream.h" #include "nsTArray.h" #include "nsBuiltinDecoder.h" #include "nsBuiltinDecoderReader.h" #include "nsBuiltinDecoderStateMachine.h" #include "mozilla/mozalloc.h" #include "VideoUtils.h" #include "nsTimeRanges.h" using namespace mozilla; using namespace mozilla::layers; #ifdef PR_LOGGING extern PRLogModuleInfo* gBuiltinDecoderLog; #define LOG(type, msg) PR_LOG(gBuiltinDecoderLog, type, msg) #else #define LOG(type, msg) #endif // Wait this number of seconds when buffering, then leave and play // as best as we can if the required amount of data hasn't been // retrieved. #define BUFFERING_WAIT 30 // The amount of data to retrieve during buffering is computed based // on the download rate. BUFFERING_MIN_RATE is the minimum download // rate to be used in that calculation to help avoid constant buffering // attempts at a time when the average download rate has not stabilised. #define BUFFERING_MIN_RATE 50000 #define BUFFERING_RATE(x) ((x)< BUFFERING_MIN_RATE ? BUFFERING_MIN_RATE : (x)) // If audio queue has less than this many usecs of decoded audio, we won't risk // trying to decode the video, we'll skip decoding video up to the next // keyframe. We may increase this value for an individual decoder if we // encounter video frames which take a long time to decode. static const PRUint32 LOW_AUDIO_USECS = 300000; // If more than this many usecs of decoded audio is queued, we'll hold off // decoding more audio. If we increase the low audio threshold (see // LOW_AUDIO_USECS above) we'll also increase this value to ensure it's not // less than the low audio threshold. const PRInt64 AMPLE_AUDIO_USECS = 1000000; // Maximum number of bytes we'll allocate and write at once to the audio // hardware when the audio stream contains missing samples and we're // writing silence in order to fill the gap. We limit our silence-writes // to 32KB in order to avoid allocating an impossibly large chunk of // memory if we encounter a large chunk of silence. const PRUint32 SILENCE_BYTES_CHUNK = 32 * 1024; // If we have fewer than LOW_VIDEO_FRAMES decoded frames, and // we're not "pumping video", we'll skip the video up to the next keyframe // which is at or after the current playback position. static const PRUint32 LOW_VIDEO_FRAMES = 1; // If we've got more than AMPLE_VIDEO_FRAMES decoded video frames waiting in // the video queue, we will not decode any more video frames until some have // been consumed by the play state machine thread. static const PRUint32 AMPLE_VIDEO_FRAMES = 10; // Arbitrary "frame duration" when playing only audio. static const int AUDIO_DURATION_USECS = 40000; // If we increase our "low audio threshold" (see LOW_AUDIO_USECS above), we // use this as a factor in all our calculations. Increasing this will cause // us to be more likely to increase our low audio threshold, and to // increase it by more. static const int THRESHOLD_FACTOR = 2; // If we have less than this much undecoded data available, we'll consider // ourselves to be running low on undecoded data. We determine how much // undecoded data we have remaining using the reader's GetBuffered() // implementation. static const PRInt64 LOW_DATA_THRESHOLD_USECS = 5000000; // LOW_DATA_THRESHOLD_USECS needs to be greater than AMPLE_AUDIO_USECS, otherwise // the skip-to-keyframe logic can activate when we're running low on data. PR_STATIC_ASSERT(LOW_DATA_THRESHOLD_USECS > AMPLE_AUDIO_USECS); // Amount of excess usecs of data to add in to the "should we buffer" calculation. static const PRUint32 EXHAUSTED_DATA_MARGIN_USECS = 60000; // If we enter buffering within QUICK_BUFFER_THRESHOLD_USECS seconds of starting // decoding, we'll enter "quick buffering" mode, which exits a lot sooner than // normal buffering mode. This exists so that if the decode-ahead exhausts the // downloaded data while decode/playback is just starting up (for example // after a seek while the media is still playing, or when playing a media // as soon as it's load started), we won't necessarily stop for 30s and wait // for buffering. We may actually be able to playback in this case, so exit // buffering early and try to play. If it turns out we can't play, we'll fall // back to buffering normally. static const PRUint32 QUICK_BUFFER_THRESHOLD_USECS = 2000000; // If we're quick buffering, we'll remain in buffering mode while we have less than // QUICK_BUFFERING_LOW_DATA_USECS of decoded data available. static const PRUint32 QUICK_BUFFERING_LOW_DATA_USECS = 1000000; // If QUICK_BUFFERING_LOW_DATA_USECS is > AMPLE_AUDIO_USECS, we won't exit // quick buffering in a timely fashion, as the decode pauses when it // reaches AMPLE_AUDIO_USECS decoded data, and thus we'll never reach // QUICK_BUFFERING_LOW_DATA_USECS. PR_STATIC_ASSERT(QUICK_BUFFERING_LOW_DATA_USECS <= AMPLE_AUDIO_USECS); static TimeDuration UsecsToDuration(PRInt64 aUsecs) { return TimeDuration::FromMilliseconds(static_cast(aUsecs) / USECS_PER_MS); } static PRInt64 DurationToUsecs(TimeDuration aDuration) { return static_cast(aDuration.ToSeconds() * USECS_PER_S); } class nsAudioMetadataEventRunner : public nsRunnable { private: nsCOMPtr mDecoder; public: nsAudioMetadataEventRunner(nsBuiltinDecoder* aDecoder, PRUint32 aChannels, PRUint32 aRate) : mDecoder(aDecoder), mChannels(aChannels), mRate(aRate) { } NS_IMETHOD Run() { mDecoder->MetadataLoaded(mChannels, mRate); return NS_OK; } const PRUint32 mChannels; const PRUint32 mRate; }; nsBuiltinDecoderStateMachine::nsBuiltinDecoderStateMachine(nsBuiltinDecoder* aDecoder, nsBuiltinDecoderReader* aReader) : mDecoder(aDecoder), mState(DECODER_STATE_DECODING_METADATA), mAudioReentrantMonitor("media.audiostream"), mCbCrSize(0), mPlayDuration(0), mStartTime(-1), mEndTime(-1), mSeekTime(0), mReader(aReader), mCurrentFrameTime(0), mAudioStartTime(-1), mAudioEndTime(-1), mVideoFrameEndTime(-1), mVolume(1.0), mSeekable(PR_TRUE), mPositionChangeQueued(PR_FALSE), mAudioCompleted(PR_FALSE), mGotDurationFromMetaData(PR_FALSE), mStopDecodeThreads(PR_TRUE), mQuickBuffering(PR_FALSE), mEventManager(aDecoder) { MOZ_COUNT_CTOR(nsBuiltinDecoderStateMachine); } nsBuiltinDecoderStateMachine::~nsBuiltinDecoderStateMachine() { MOZ_COUNT_DTOR(nsBuiltinDecoderStateMachine); } PRBool nsBuiltinDecoderStateMachine::HasFutureAudio() const { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); NS_ASSERTION(HasAudio(), "Should only call HasFutureAudio() when we have audio"); // We've got audio ready to play if: // 1. We've not completed playback of audio, and // 2. we either have more than the threshold of decoded audio available, or // we've completely decoded all audio (but not finished playing it yet // as per 1). return !mAudioCompleted && (AudioDecodedUsecs() > LOW_AUDIO_USECS || mReader->mAudioQueue.IsFinished()); } PRBool nsBuiltinDecoderStateMachine::HaveNextFrameData() const { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); return (!HasAudio() || HasFutureAudio()) && (!HasVideo() || mReader->mVideoQueue.GetSize() > 0); } PRInt64 nsBuiltinDecoderStateMachine::GetDecodedAudioDuration() { NS_ASSERTION(OnDecodeThread(), "Should be on decode thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); PRInt64 audioDecoded = mReader->mAudioQueue.Duration(); if (mAudioEndTime != -1) { audioDecoded += mAudioEndTime - GetMediaTime(); } return audioDecoded; } void nsBuiltinDecoderStateMachine::DecodeLoop() { NS_ASSERTION(OnDecodeThread(), "Should be on decode thread."); // We want to "pump" the decode until we've got a few frames/samples decoded // before we consider whether decode is falling behind. PRBool audioPump = PR_TRUE; PRBool videoPump = PR_TRUE; // If the video decode is falling behind the audio, we'll start dropping the // inter-frames up until the next keyframe which is at or before the current // playback position. skipToNextKeyframe is PR_TRUE if we're currently // skipping up to the next keyframe. PRBool skipToNextKeyframe = PR_FALSE; // Once we've decoded more than videoPumpThreshold video frames, we'll // no longer be considered to be "pumping video". const unsigned videoPumpThreshold = AMPLE_VIDEO_FRAMES / 2; // After the audio decode fills with more than audioPumpThreshold usecs // of decoded audio, we'll start to check whether the audio or video decode // is falling behind. const unsigned audioPumpThreshold = LOW_AUDIO_USECS * 2; // Our local low audio threshold. We may increase this if we're slow to // decode video frames, in order to reduce the chance of audio underruns. PRInt64 lowAudioThreshold = LOW_AUDIO_USECS; // Our local ample audio threshold. If we increase lowAudioThreshold, we'll // also increase this too appropriately (we don't want lowAudioThreshold to // be greater than ampleAudioThreshold, else we'd stop decoding!). PRInt64 ampleAudioThreshold = AMPLE_AUDIO_USECS; MediaQueue& videoQueue = mReader->mVideoQueue; MediaQueue& audioQueue = mReader->mAudioQueue; ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); PRBool videoPlaying = HasVideo(); PRBool audioPlaying = HasAudio(); // Main decode loop. while (mState != DECODER_STATE_SHUTDOWN && !mStopDecodeThreads && (videoPlaying || audioPlaying)) { // We don't want to consider skipping to the next keyframe if we've // only just started up the decode loop, so wait until we've decoded // some frames before enabling the keyframe skip logic on video. if (videoPump && static_cast(videoQueue.GetSize()) >= videoPumpThreshold) { videoPump = PR_FALSE; } // We don't want to consider skipping to the next keyframe if we've // only just started up the decode loop, so wait until we've decoded // some audio data before enabling the keyframe skip logic on audio. if (audioPump && GetDecodedAudioDuration() >= audioPumpThreshold) { audioPump = PR_FALSE; } // We'll skip the video decode to the nearest keyframe if we're low on // audio, or if we're low on video, provided we're not running low on // data to decode. If we're running low on downloaded data to decode, // we won't start keyframe skipping, as we'll be pausing playback to buffer // soon anyway and we'll want to be able to display frames immediately // after buffering finishes. if (mState == DECODER_STATE_DECODING && !skipToNextKeyframe && videoPlaying && ((!audioPump && audioPlaying && GetDecodedAudioDuration() < lowAudioThreshold) || (!videoPump && videoPlaying && static_cast(videoQueue.GetSize()) < LOW_VIDEO_FRAMES)) && !HasLowUndecodedData()) { skipToNextKeyframe = PR_TRUE; LOG(PR_LOG_DEBUG, ("%p Skipping video decode to the next keyframe", mDecoder)); } // Video decode. if (videoPlaying && static_cast(videoQueue.GetSize()) < AMPLE_VIDEO_FRAMES) { // Time the video decode, so that if it's slow, we can increase our low // audio threshold to reduce the chance of an audio underrun while we're // waiting for a video decode to complete. TimeDuration decodeTime; { PRInt64 currentTime = GetMediaTime(); ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); TimeStamp start = TimeStamp::Now(); videoPlaying = mReader->DecodeVideoFrame(skipToNextKeyframe, currentTime); decodeTime = TimeStamp::Now() - start; } if (THRESHOLD_FACTOR * DurationToUsecs(decodeTime) > lowAudioThreshold && !HasLowUndecodedData()) { lowAudioThreshold = NS_MIN(THRESHOLD_FACTOR * DurationToUsecs(decodeTime), AMPLE_AUDIO_USECS); ampleAudioThreshold = NS_MAX(THRESHOLD_FACTOR * lowAudioThreshold, ampleAudioThreshold); LOG(PR_LOG_DEBUG, ("Slow video decode, set lowAudioThreshold=%lld ampleAudioThreshold=%lld", lowAudioThreshold, ampleAudioThreshold)); } } // Audio decode. if (audioPlaying && (GetDecodedAudioDuration() < ampleAudioThreshold || audioQueue.GetSize() == 0)) { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); audioPlaying = mReader->DecodeAudioData(); } // Notify to ensure that the AudioLoop() is not waiting, in case it was // waiting for more audio to be decoded. mDecoder->GetReentrantMonitor().NotifyAll(); if (!IsPlaying()) { // Update the ready state, so that the play DOM events fire. We only // need to do this if we're not playing; if we're playing the playback // code will do an update whenever it advances a frame. UpdateReadyState(); } if (mState != DECODER_STATE_SHUTDOWN && !mStopDecodeThreads && (!audioPlaying || (GetDecodedAudioDuration() >= ampleAudioThreshold && audioQueue.GetSize() > 0)) && (!videoPlaying || static_cast(videoQueue.GetSize()) >= AMPLE_VIDEO_FRAMES)) { // All active bitstreams' decode is well ahead of the playback // position, we may as well wait for the playback to catch up. Note the // audio push thread acquires and notifies the decoder monitor every time // it pops SoundData off the audio queue. So if the audio push thread pops // the last SoundData off the audio queue right after that queue reported // it was non-empty here, we'll receive a notification on the decoder // monitor which will wake us up shortly after we sleep, thus preventing // both the decode and audio push threads waiting at the same time. // See bug 620326. mon.Wait(); } } // End decode loop. if (!mStopDecodeThreads && mState != DECODER_STATE_SHUTDOWN && mState != DECODER_STATE_SEEKING) { mState = DECODER_STATE_COMPLETED; mDecoder->GetReentrantMonitor().NotifyAll(); } LOG(PR_LOG_DEBUG, ("%p Shutting down DecodeLoop this=%p", mDecoder, this)); } PRBool nsBuiltinDecoderStateMachine::IsPlaying() { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); return !mPlayStartTime.IsNull(); } void nsBuiltinDecoderStateMachine::AudioLoop() { NS_ASSERTION(OnAudioThread(), "Should be on audio thread."); LOG(PR_LOG_DEBUG, ("%p Begun audio thread/loop", mDecoder)); PRInt64 audioDuration = 0; PRInt64 audioStartTime = -1; PRUint32 channels, rate; double volume = -1; PRBool setVolume; PRInt32 minWriteSamples = -1; PRInt64 samplesAtLastSleep = 0; { ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); mAudioCompleted = PR_FALSE; audioStartTime = mAudioStartTime; channels = mInfo.mAudioChannels; rate = mInfo.mAudioRate; NS_ASSERTION(audioStartTime != -1, "Should have audio start time by now"); } while (1) { // Wait while we're not playing, and we're not shutting down, or we're // playing and we've got no audio to play. { ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); NS_ASSERTION(mState != DECODER_STATE_DECODING_METADATA, "Should have meta data before audio started playing."); while (mState != DECODER_STATE_SHUTDOWN && !mStopDecodeThreads && (!IsPlaying() || mState == DECODER_STATE_BUFFERING || (mReader->mAudioQueue.GetSize() == 0 && !mReader->mAudioQueue.AtEndOfStream()))) { samplesAtLastSleep = audioDuration; mon.Wait(); } // If we're shutting down, break out and exit the audio thread. if (mState == DECODER_STATE_SHUTDOWN || mStopDecodeThreads || mReader->mAudioQueue.AtEndOfStream()) { break; } // We only want to go to the expense of taking the audio monitor and // changing the volume if it's the first time we've entered the loop // (as we must sync the volume in case it's changed since the // nsAudioStream was created) or if the volume has changed. setVolume = volume != mVolume; volume = mVolume; } if (setVolume || minWriteSamples == -1) { ReentrantMonitorAutoEnter audioMon(mAudioReentrantMonitor); if (mAudioStream) { if (setVolume) { mAudioStream->SetVolume(volume); } if (minWriteSamples == -1) { minWriteSamples = mAudioStream->GetMinWriteSamples(); } } } NS_ASSERTION(mReader->mAudioQueue.GetSize() > 0, "Should have data to play"); // See if there's missing samples in the audio stream. If there is, push // silence into the audio hardware, so we can play across the gap. const SoundData* s = mReader->mAudioQueue.PeekFront(); // Calculate the number of samples that have been pushed onto the audio // hardware. PRInt64 playedSamples = 0; if (!UsecsToSamples(audioStartTime, rate, playedSamples)) { NS_WARNING("Int overflow converting playedSamples"); break; } if (!AddOverflow(playedSamples, audioDuration, playedSamples)) { NS_WARNING("Int overflow adding playedSamples"); break; } // Calculate the timestamp of the next chunk of audio in numbers of // samples. PRInt64 sampleTime = 0; if (!UsecsToSamples(s->mTime, rate, sampleTime)) { NS_WARNING("Int overflow converting sampleTime"); break; } PRInt64 missingSamples = 0; if (!AddOverflow(sampleTime, -playedSamples, missingSamples)) { NS_WARNING("Int overflow adding missingSamples"); break; } if (missingSamples > 0) { // The next sound chunk begins some time after the end of the last chunk // we pushed to the sound hardware. We must push silence into the audio // hardware so that the next sound chunk begins playback at the correct // time. missingSamples = NS_MIN(static_cast(PR_UINT32_MAX), missingSamples); audioDuration += PlaySilence(static_cast(missingSamples), channels, playedSamples); } else { audioDuration += PlayFromAudioQueue(sampleTime, channels); } { ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); PRInt64 playedUsecs; if (!SamplesToUsecs(audioDuration, rate, playedUsecs)) { NS_WARNING("Int overflow calculating playedUsecs"); break; } if (!AddOverflow(audioStartTime, playedUsecs, mAudioEndTime)) { NS_WARNING("Int overflow calculating audio end time"); break; } PRInt64 audioAhead = mAudioEndTime - GetMediaTime(); if (audioAhead > AMPLE_AUDIO_USECS && audioDuration - samplesAtLastSleep > minWriteSamples) { samplesAtLastSleep = audioDuration; // We've pushed enough audio onto the hardware that we've queued up a // significant amount ahead of the playback position. The decode // thread will be going to sleep, so we won't get any new samples // anyway, so sleep until we need to push to the hardware again. Wait(AMPLE_AUDIO_USECS / 2); // Kick the decode thread; since above we only do a NotifyAll when // we pop an audio chunk of the queue, the decoder won't wake up if // we've got no more decoded chunks to push to the hardware. We can // hit this condition if the last sample in the stream doesn't have // it's EOS flag set, and the decode thread sleeps just after decoding // that packet, but before realising there's no more packets. mon.NotifyAll(); } } } if (mReader->mAudioQueue.AtEndOfStream() && mState != DECODER_STATE_SHUTDOWN && !mStopDecodeThreads) { // Last sample pushed to audio hardware, wait for the audio to finish, // before the audio thread terminates. ReentrantMonitorAutoEnter audioMon(mAudioReentrantMonitor); if (mAudioStream) { PRBool seeking = PR_FALSE; PRInt64 oldPosition = -1; { ReentrantMonitorAutoExit audioExit(mAudioReentrantMonitor); ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); PRInt64 position = GetMediaTime(); while (oldPosition != position && mAudioEndTime - position > 0 && mState != DECODER_STATE_SEEKING && mState != DECODER_STATE_SHUTDOWN) { const PRInt64 DRAIN_BLOCK_USECS = 100000; Wait(NS_MIN(mAudioEndTime - position, DRAIN_BLOCK_USECS)); oldPosition = position; position = GetMediaTime(); } if (mState == DECODER_STATE_SEEKING) { seeking = PR_TRUE; } } if (!seeking && mAudioStream && !mAudioStream->IsPaused()) { mAudioStream->Drain(); // Fire one last event for any extra samples that didn't fill a framebuffer. mEventManager.Drain(mAudioEndTime); } } LOG(PR_LOG_DEBUG, ("%p Reached audio stream end.", mDecoder)); } { ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); mAudioCompleted = PR_TRUE; UpdateReadyState(); // Kick the decode and state machine threads; they may be sleeping waiting // for this to finish. mDecoder->GetReentrantMonitor().NotifyAll(); } LOG(PR_LOG_DEBUG, ("%p Audio stream finished playing, audio thread exit", mDecoder)); } PRUint32 nsBuiltinDecoderStateMachine::PlaySilence(PRUint32 aSamples, PRUint32 aChannels, PRUint64 aSampleOffset) { ReentrantMonitorAutoEnter audioMon(mAudioReentrantMonitor); if (!mAudioStream || mAudioStream->IsPaused()) { // The state machine has paused since we've released the decoder // monitor and acquired the audio monitor. Don't write any audio. return 0; } PRUint32 maxSamples = SILENCE_BYTES_CHUNK / aChannels; PRUint32 samples = NS_MIN(aSamples, maxSamples); PRUint32 numValues = samples * aChannels; nsAutoArrayPtr buf(new SoundDataValue[numValues]); memset(buf.get(), 0, sizeof(SoundDataValue) * numValues); mAudioStream->Write(buf, numValues, PR_TRUE); // Dispatch events to the DOM for the audio just written. mEventManager.QueueWrittenAudioData(buf.get(), numValues, (aSampleOffset + samples) * aChannels); return samples; } PRUint32 nsBuiltinDecoderStateMachine::PlayFromAudioQueue(PRUint64 aSampleOffset, PRUint32 aChannels) { nsAutoPtr sound(mReader->mAudioQueue.PopFront()); { ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); NS_WARN_IF_FALSE(IsPlaying(), "Should be playing"); // Awaken the decode loop if it's waiting for space to free up in the // audio queue. mDecoder->GetReentrantMonitor().NotifyAll(); } PRInt64 offset = -1; PRUint32 samples = 0; { ReentrantMonitorAutoEnter audioMon(mAudioReentrantMonitor); if (!mAudioStream) { return 0; } // The state machine could have paused since we've released the decoder // monitor and acquired the audio monitor. Rather than acquire both // monitors, the audio stream also maintains whether its paused or not. // This prevents us from doing a blocking write while holding the audio // monitor while paused; we would block, and the state machine won't be // able to acquire the audio monitor in order to resume or destroy the // audio stream. if (!mAudioStream->IsPaused()) { mAudioStream->Write(sound->mAudioData, sound->AudioDataLength(), PR_TRUE); offset = sound->mOffset; samples = sound->mSamples; // Dispatch events to the DOM for the audio just written. mEventManager.QueueWrittenAudioData(sound->mAudioData.get(), sound->AudioDataLength(), (aSampleOffset + samples) * aChannels); } else { mReader->mAudioQueue.PushFront(sound); sound.forget(); } } if (offset != -1) { mDecoder->UpdatePlaybackOffset(offset); } return samples; } nsresult nsBuiltinDecoderStateMachine::Init(nsDecoderStateMachine* aCloneDonor) { nsBuiltinDecoderReader* cloneReader = nsnull; if (aCloneDonor) { cloneReader = static_cast(aCloneDonor)->mReader; } return mReader->Init(cloneReader); } void nsBuiltinDecoderStateMachine::StopPlayback(eStopMode aMode) { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); mDecoder->mPlaybackStatistics.Stop(TimeStamp::Now()); // Reset mPlayStartTime before we pause/shutdown the nsAudioStream. This is // so that if the audio loop is about to write audio, it will have the chance // to check to see if we're paused and not write the audio. If not, the // audio thread can block in the write, and we deadlock trying to acquire // the audio monitor upon resume playback. if (IsPlaying()) { mPlayDuration += DurationToUsecs(TimeStamp::Now() - mPlayStartTime); mPlayStartTime = TimeStamp(); } if (HasAudio()) { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); ReentrantMonitorAutoEnter audioMon(mAudioReentrantMonitor); if (mAudioStream) { if (aMode == AUDIO_PAUSE) { mAudioStream->Pause(); } else if (aMode == AUDIO_SHUTDOWN) { mAudioStream->Shutdown(); mAudioStream = nsnull; mEventManager.Clear(); } } } } void nsBuiltinDecoderStateMachine::StartPlayback() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); NS_ASSERTION(!IsPlaying(), "Shouldn't be playing when StartPlayback() is called"); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); LOG(PR_LOG_DEBUG, ("%p StartPlayback", mDecoder)); mDecoder->mPlaybackStatistics.Start(TimeStamp::Now()); if (HasAudio()) { PRInt32 rate = mInfo.mAudioRate; PRInt32 channels = mInfo.mAudioChannels; { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); ReentrantMonitorAutoEnter audioMon(mAudioReentrantMonitor); if (mAudioStream) { // We have an audiostream, so it must have been paused the last time // StopPlayback() was called. mAudioStream->Resume(); } else { // No audiostream, create one. mAudioStream = nsAudioStream::AllocateStream(); mAudioStream->Init(channels, rate, MOZ_SOUND_DATA_FORMAT); mAudioStream->SetVolume(mVolume); } } } mPlayStartTime = TimeStamp::Now(); mDecoder->GetReentrantMonitor().NotifyAll(); } void nsBuiltinDecoderStateMachine::UpdatePlaybackPositionInternal(PRInt64 aTime) { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); NS_ASSERTION(mStartTime >= 0, "Should have positive mStartTime"); mCurrentFrameTime = aTime - mStartTime; NS_ASSERTION(mCurrentFrameTime >= 0, "CurrentTime should be positive!"); if (aTime > mEndTime) { NS_ASSERTION(mCurrentFrameTime > GetDuration(), "CurrentTime must be after duration if aTime > endTime!"); mEndTime = aTime; nsCOMPtr event = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::DurationChanged); NS_DispatchToMainThread(event, NS_DISPATCH_NORMAL); } } void nsBuiltinDecoderStateMachine::UpdatePlaybackPosition(PRInt64 aTime) { UpdatePlaybackPositionInternal(aTime); if (!mPositionChangeQueued) { mPositionChangeQueued = PR_TRUE; nsCOMPtr event = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::PlaybackPositionChanged); NS_DispatchToMainThread(event, NS_DISPATCH_NORMAL); } // Notify DOM of any queued up audioavailable events mEventManager.DispatchPendingEvents(GetMediaTime()); } void nsBuiltinDecoderStateMachine::ClearPositionChangeFlag() { NS_ASSERTION(NS_IsMainThread(), "Should be on main thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); mPositionChangeQueued = PR_FALSE; } nsHTMLMediaElement::NextFrameStatus nsBuiltinDecoderStateMachine::GetNextFrameStatus() { ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); if (IsBuffering() || IsSeeking()) { return nsHTMLMediaElement::NEXT_FRAME_UNAVAILABLE_BUFFERING; } else if (HaveNextFrameData()) { return nsHTMLMediaElement::NEXT_FRAME_AVAILABLE; } return nsHTMLMediaElement::NEXT_FRAME_UNAVAILABLE; } void nsBuiltinDecoderStateMachine::SetVolume(double volume) { NS_ASSERTION(NS_IsMainThread(), "Should be on main thread."); ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); mVolume = volume; } double nsBuiltinDecoderStateMachine::GetCurrentTime() const { NS_ASSERTION(NS_IsMainThread() || mDecoder->OnStateMachineThread() || OnDecodeThread(), "Should be on main, decode, or state machine thread."); return static_cast(mCurrentFrameTime) / static_cast(USECS_PER_S); } PRInt64 nsBuiltinDecoderStateMachine::GetDuration() { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); if (mEndTime == -1 || mStartTime == -1) return -1; return mEndTime - mStartTime; } void nsBuiltinDecoderStateMachine::SetDuration(PRInt64 aDuration) { NS_ASSERTION(NS_IsMainThread() || mDecoder->OnStateMachineThread(), "Should be on main or state machine thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); if (aDuration == -1) { return; } if (mStartTime != -1) { mEndTime = mStartTime + aDuration; } else { mStartTime = 0; mEndTime = aDuration; } } void nsBuiltinDecoderStateMachine::SetEndTime(PRInt64 aEndTime) { NS_ASSERTION(OnStateMachineThread(), "Should be on state machine thread"); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); mEndTime = aEndTime; } void nsBuiltinDecoderStateMachine::SetSeekable(PRBool aSeekable) { NS_ASSERTION(NS_IsMainThread(), "Should be on main thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); mSeekable = aSeekable; } void nsBuiltinDecoderStateMachine::Shutdown() { NS_ASSERTION(NS_IsMainThread(), "Should be on main thread."); // Once we've entered the shutdown state here there's no going back. ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); // Change state before issuing shutdown request to threads so those // threads can start exiting cleanly during the Shutdown call. LOG(PR_LOG_DEBUG, ("%p Changed state to SHUTDOWN", mDecoder)); mState = DECODER_STATE_SHUTDOWN; mDecoder->GetReentrantMonitor().NotifyAll(); } void nsBuiltinDecoderStateMachine::StartDecoding() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); if (mState != DECODER_STATE_DECODING) { mDecodeStartTime = TimeStamp::Now(); } mState = DECODER_STATE_DECODING; } void nsBuiltinDecoderStateMachine::Play() { NS_ASSERTION(NS_IsMainThread(), "Should be on main thread."); // When asked to play, switch to decoding state only if // we are currently buffering. In other cases, we'll start playing anyway // when the state machine notices the decoder's state change to PLAYING. ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); if (mState == DECODER_STATE_BUFFERING) { LOG(PR_LOG_DEBUG, ("%p Changed state from BUFFERING to DECODING", mDecoder)); mState = DECODER_STATE_DECODING; mDecodeStartTime = TimeStamp::Now(); mDecoder->GetReentrantMonitor().NotifyAll(); } } void nsBuiltinDecoderStateMachine::ResetPlayback() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mVideoFrameEndTime = -1; mAudioStartTime = -1; mAudioEndTime = -1; mAudioCompleted = PR_FALSE; } void nsBuiltinDecoderStateMachine::Seek(double aTime) { NS_ASSERTION(NS_IsMainThread(), "Should be on main thread."); ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); // nsBuiltinDecoder::mPlayState should be SEEKING while we seek, and // in that case nsBuiltinDecoder shouldn't be calling us. NS_ASSERTION(mState != DECODER_STATE_SEEKING, "We shouldn't already be seeking"); NS_ASSERTION(mState >= DECODER_STATE_DECODING, "We should have loaded metadata"); double t = aTime * static_cast(USECS_PER_S); if (t > PR_INT64_MAX) { // Prevent integer overflow. return; } mSeekTime = static_cast(t) + mStartTime; NS_ASSERTION(mSeekTime >= mStartTime && mSeekTime <= mEndTime, "Can only seek in range [0,duration]"); // Bound the seek time to be inside the media range. NS_ASSERTION(mStartTime != -1, "Should know start time by now"); NS_ASSERTION(mEndTime != -1, "Should know end time by now"); mSeekTime = NS_MIN(mSeekTime, mEndTime); mSeekTime = NS_MAX(mStartTime, mSeekTime); LOG(PR_LOG_DEBUG, ("%p Changed state to SEEKING (to %f)", mDecoder, aTime)); mState = DECODER_STATE_SEEKING; } void nsBuiltinDecoderStateMachine::StopDecodeThreads() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); mStopDecodeThreads = PR_TRUE; mDecoder->GetReentrantMonitor().NotifyAll(); if (mDecodeThread) { { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); mDecodeThread->Shutdown(); } mDecodeThread = nsnull; } if (mAudioThread) { { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); mAudioThread->Shutdown(); } mAudioThread = nsnull; } } nsresult nsBuiltinDecoderStateMachine::StartDecodeThreads() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); mStopDecodeThreads = PR_FALSE; if (!mDecodeThread && mState < DECODER_STATE_COMPLETED) { nsresult rv = NS_NewThread(getter_AddRefs(mDecodeThread)); if (NS_FAILED(rv)) { mState = DECODER_STATE_SHUTDOWN; return rv; } nsCOMPtr event = NS_NewRunnableMethod(this, &nsBuiltinDecoderStateMachine::DecodeLoop); mDecodeThread->Dispatch(event, NS_DISPATCH_NORMAL); } if (HasAudio() && !mAudioThread) { nsresult rv = NS_NewThread(getter_AddRefs(mAudioThread)); if (NS_FAILED(rv)) { mState = DECODER_STATE_SHUTDOWN; return rv; } nsCOMPtr event = NS_NewRunnableMethod(this, &nsBuiltinDecoderStateMachine::AudioLoop); mAudioThread->Dispatch(event, NS_DISPATCH_NORMAL); } return NS_OK; } PRInt64 nsBuiltinDecoderStateMachine::AudioDecodedUsecs() const { NS_ASSERTION(HasAudio(), "Should only call AudioDecodedUsecs() when we have audio"); // The amount of audio we have decoded is the amount of audio data we've // already decoded and pushed to the hardware, plus the amount of audio // data waiting to be pushed to the hardware. PRInt64 pushed = (mAudioEndTime != -1) ? (mAudioEndTime - GetMediaTime()) : 0; return pushed + mReader->mAudioQueue.Duration(); } PRBool nsBuiltinDecoderStateMachine::HasLowDecodedData(PRInt64 aAudioUsecs) const { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); // We consider ourselves low on decoded data if we're low on audio, // provided we've not decoded to the end of the audio stream, or // if we're only playing video and we're low on video frames, provided // we've not decoded to the end of the video stream. return ((HasAudio() && !mReader->mAudioQueue.IsFinished() && AudioDecodedUsecs() < aAudioUsecs) || (!HasAudio() && HasVideo() && !mReader->mVideoQueue.IsFinished() && static_cast(mReader->mVideoQueue.GetSize()) < LOW_VIDEO_FRAMES)); } PRBool nsBuiltinDecoderStateMachine::HasLowUndecodedData() const { return GetUndecodedData() < LOW_DATA_THRESHOLD_USECS; } PRInt64 nsBuiltinDecoderStateMachine::GetUndecodedData() const { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); NS_ASSERTION(mState > DECODER_STATE_DECODING_METADATA, "Must have loaded metadata for GetBuffered() to work"); nsTimeRanges buffered; nsresult res = mDecoder->GetBuffered(&buffered); NS_ENSURE_SUCCESS(res, 0); double currentTime = GetCurrentTime(); nsIDOMTimeRanges* r = static_cast(&buffered); PRUint32 length = 0; res = r->GetLength(&length); NS_ENSURE_SUCCESS(res, 0); for (PRUint32 index = 0; index < length; ++index) { double start, end; res = r->Start(index, &start); NS_ENSURE_SUCCESS(res, 0); res = r->End(index, &end); NS_ENSURE_SUCCESS(res, 0); if (start <= currentTime && end >= currentTime) { return static_cast((end - currentTime) * 1000); } } return 0; } void nsBuiltinDecoderStateMachine::SetFrameBufferLength(PRUint32 aLength) { NS_ASSERTION(aLength >= 512 && aLength <= 16384, "The length must be between 512 and 16384"); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); mEventManager.SetSignalBufferLength(aLength); } nsresult nsBuiltinDecoderStateMachine::Run() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); nsMediaStream* stream = mDecoder->GetCurrentStream(); NS_ENSURE_TRUE(stream, NS_ERROR_NULL_POINTER); while (PR_TRUE) { ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor()); switch (mState) { case DECODER_STATE_SHUTDOWN: if (IsPlaying()) { StopPlayback(AUDIO_SHUTDOWN); } StopDecodeThreads(); NS_ASSERTION(mState == DECODER_STATE_SHUTDOWN, "How did we escape from the shutdown state???"); return NS_OK; case DECODER_STATE_DECODING_METADATA: { LoadMetadata(); if (mState == DECODER_STATE_SHUTDOWN) { continue; } VideoData* videoData = FindStartTime(); if (videoData) { nsIntSize display = mInfo.mDisplay; float aspect = mInfo.mPixelAspectRatio; { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); RenderVideoFrame(videoData, TimeStamp::Now(), display, aspect); } } // Start the decode threads, so that we can pre buffer the streams. // and calculate the start time in order to determine the duration. if (NS_FAILED(StartDecodeThreads())) { continue; } NS_ASSERTION(mStartTime != -1, "Must have start time"); NS_ASSERTION((!HasVideo() && !HasAudio()) || !mSeekable || mEndTime != -1, "Active seekable media should have end time"); NS_ASSERTION(!mSeekable || GetDuration() != -1, "Seekable media should have duration"); LOG(PR_LOG_DEBUG, ("%p Media goes from %lld to %lld (duration %lld) seekable=%d", mDecoder, mStartTime, mEndTime, GetDuration(), mSeekable)); if (mState == DECODER_STATE_SHUTDOWN) continue; // Inform the element that we've loaded the metadata and the first frame, // setting the default framebuffer size for audioavailable events. Also, // if there is audio, let the MozAudioAvailable event manager know about // the metadata. if (HasAudio()) { mEventManager.Init(mInfo.mAudioChannels, mInfo.mAudioRate); // Set the buffer length at the decoder level to be able, to be able // to retrive the value via media element method. The RequestFrameBufferLength // will call the nsBuiltinDecoderStateMachine::SetFrameBufferLength(). PRUint32 frameBufferLength = mInfo.mAudioChannels * FRAMEBUFFER_LENGTH_PER_CHANNEL; mDecoder->RequestFrameBufferLength(frameBufferLength); } nsCOMPtr metadataLoadedEvent = new nsAudioMetadataEventRunner(mDecoder, mInfo.mAudioChannels, mInfo.mAudioRate); NS_DispatchToMainThread(metadataLoadedEvent, NS_DISPATCH_NORMAL); if (mState == DECODER_STATE_DECODING_METADATA) { LOG(PR_LOG_DEBUG, ("%p Changed state from DECODING_METADATA to DECODING", mDecoder)); StartDecoding(); } // Start playback. if (mDecoder->GetState() == nsBuiltinDecoder::PLAY_STATE_PLAYING) { if (!IsPlaying()) { StartPlayback(); } } } break; case DECODER_STATE_DECODING: { if (NS_FAILED(StartDecodeThreads())) { continue; } AdvanceFrame(); if (mState != DECODER_STATE_DECODING) continue; } break; case DECODER_STATE_SEEKING: { // During the seek, don't have a lock on the decoder state, // otherwise long seek operations can block the main thread. // The events dispatched to the main thread are SYNC calls. // These calls are made outside of the decode monitor lock so // it is safe for the main thread to makes calls that acquire // the lock since it won't deadlock. We check the state when // acquiring the lock again in case shutdown has occurred // during the time when we didn't have the lock. PRInt64 seekTime = mSeekTime; mDecoder->StopProgressUpdates(); PRBool currentTimeChanged = false; PRInt64 mediaTime = GetMediaTime(); if (mediaTime != seekTime) { currentTimeChanged = true; UpdatePlaybackPositionInternal(seekTime); } // SeekingStarted will do a UpdateReadyStateForData which will // inform the element and its users that we have no frames // to display { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); nsCOMPtr startEvent = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::SeekingStarted); NS_DispatchToMainThread(startEvent, NS_DISPATCH_SYNC); } if (currentTimeChanged) { // The seek target is different than the current playback position, // we'll need to seek the playback position, so shutdown our decode // and audio threads. StopPlayback(AUDIO_SHUTDOWN); StopDecodeThreads(); ResetPlayback(); nsresult res; { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); // Now perform the seek. We must not hold the state machine monitor // while we seek, since the seek decodes. res = mReader->Seek(seekTime, mStartTime, mEndTime, mediaTime); } if (NS_SUCCEEDED(res)){ SoundData* audio = HasAudio() ? mReader->mAudioQueue.PeekFront() : nsnull; NS_ASSERTION(!audio || (audio->mTime <= seekTime && seekTime <= audio->mTime + audio->mDuration), "Seek target should lie inside the first audio block after seek"); PRInt64 startTime = (audio && audio->mTime < seekTime) ? audio->mTime : seekTime; mAudioStartTime = startTime; mPlayDuration = startTime - mStartTime; if (HasVideo()) { nsAutoPtr video(mReader->mVideoQueue.PeekFront()); if (video) { NS_ASSERTION(video->mTime <= seekTime && seekTime <= video->mEndTime, "Seek target should lie inside the first frame after seek"); nsIntSize display = mInfo.mDisplay; float aspect = mInfo.mPixelAspectRatio; { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); RenderVideoFrame(video, TimeStamp::Now(), display, aspect); } mReader->mVideoQueue.PopFront(); nsCOMPtr event = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::Invalidate); NS_DispatchToMainThread(event, NS_DISPATCH_NORMAL); } } } } mDecoder->StartProgressUpdates(); if (mState == DECODER_STATE_SHUTDOWN) continue; // Try to decode another frame to detect if we're at the end... LOG(PR_LOG_DEBUG, ("%p Seek completed, mCurrentFrameTime=%lld\n", mDecoder, mCurrentFrameTime)); // Change state to DECODING or COMPLETED now. SeekingStopped will // call nsBuiltinDecoderStateMachine::Seek to reset our state to SEEKING // if we need to seek again. nsCOMPtr stopEvent; if (GetMediaTime() == mEndTime) { LOG(PR_LOG_DEBUG, ("%p Changed state from SEEKING (to %lld) to COMPLETED", mDecoder, seekTime)); stopEvent = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::SeekingStoppedAtEnd); mState = DECODER_STATE_COMPLETED; } else { LOG(PR_LOG_DEBUG, ("%p Changed state from SEEKING (to %lld) to DECODING", mDecoder, seekTime)); stopEvent = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::SeekingStopped); StartDecoding(); } mDecoder->GetReentrantMonitor().NotifyAll(); { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); NS_DispatchToMainThread(stopEvent, NS_DISPATCH_SYNC); } // Reset quick buffering status. This ensures that if we began the // seek while quick-buffering, we won't bypass quick buffering mode // if we need to buffer after the seek. mQuickBuffering = PR_FALSE; } break; case DECODER_STATE_BUFFERING: { if (IsPlaying()) { StopPlayback(AUDIO_PAUSE); mDecoder->GetReentrantMonitor().NotifyAll(); } TimeStamp now = TimeStamp::Now(); NS_ASSERTION(!mBufferingStart.IsNull(), "Must know buffering start time."); // We will remain in the buffering state if we've not decoded enough // data to begin playback, or if we've not downloaded a reasonable // amount of data inside our buffering time. TimeDuration elapsed = now - mBufferingStart; PRBool isLiveStream = mDecoder->GetCurrentStream()->GetLength() == -1; if ((isLiveStream || !mDecoder->CanPlayThrough()) && elapsed < TimeDuration::FromSeconds(BUFFERING_WAIT) && (mQuickBuffering ? HasLowDecodedData(QUICK_BUFFERING_LOW_DATA_USECS) : (GetUndecodedData() < BUFFERING_WAIT * USECS_PER_S)) && !stream->IsDataCachedToEndOfStream(mDecoder->mDecoderPosition) && !stream->IsSuspended()) { LOG(PR_LOG_DEBUG, ("Buffering: %.3lfs/%ds, timeout in %.3lfs %s", GetUndecodedData() / static_cast(USECS_PER_S), BUFFERING_WAIT, BUFFERING_WAIT - elapsed.ToSeconds(), (mQuickBuffering ? "(quick exit)" : ""))); Wait(USECS_PER_S); if (mState == DECODER_STATE_SHUTDOWN) continue; } else { LOG(PR_LOG_DEBUG, ("%p Changed state from BUFFERING to DECODING", mDecoder)); LOG(PR_LOG_DEBUG, ("%p Buffered for %.3lfs", mDecoder, (now - mBufferingStart).ToSeconds())); StartDecoding(); } if (mState != DECODER_STATE_BUFFERING) { // Notify to allow blocked decoder thread to continue mDecoder->GetReentrantMonitor().NotifyAll(); UpdateReadyState(); if (mDecoder->GetState() == nsBuiltinDecoder::PLAY_STATE_PLAYING) { if (!IsPlaying()) { StartPlayback(); } } } break; } case DECODER_STATE_COMPLETED: { if (NS_FAILED(StartDecodeThreads())) { continue; } // Play the remaining media. We want to run AdvanceFrame() at least // once to ensure the current playback position is advanced to the // end of the media, and so that we update the readyState. do { AdvanceFrame(); } while (mState == DECODER_STATE_COMPLETED && (mReader->mVideoQueue.GetSize() > 0 || (HasAudio() && !mAudioCompleted))); if (mAudioStream) { // Close the audio stream so that next time audio is used a new stream // is created. The StopPlayback call also resets the IsPlaying() state // so audio is restarted correctly. StopPlayback(AUDIO_SHUTDOWN); } if (mState != DECODER_STATE_COMPLETED) continue; LOG(PR_LOG_DEBUG, ("%p Shutting down the state machine thread", mDecoder)); StopDecodeThreads(); if (mDecoder->GetState() == nsBuiltinDecoder::PLAY_STATE_PLAYING) { PRInt64 videoTime = HasVideo() ? mVideoFrameEndTime : 0; PRInt64 clockTime = NS_MAX(mEndTime, NS_MAX(videoTime, GetAudioClock())); UpdatePlaybackPosition(clockTime); { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); nsCOMPtr event = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::PlaybackEnded); NS_DispatchToMainThread(event, NS_DISPATCH_SYNC); } } if (mState == DECODER_STATE_COMPLETED) { // We've finished playback. Shutdown the state machine thread, // in order to save memory on thread stacks, particuarly on Linux. nsCOMPtr event = new ShutdownThreadEvent(mDecoder->mStateMachineThread); NS_DispatchToMainThread(event, NS_DISPATCH_NORMAL); mDecoder->mStateMachineThread = nsnull; return NS_OK; } } break; } } return NS_OK; } void nsBuiltinDecoderStateMachine::RenderVideoFrame(VideoData* aData, TimeStamp aTarget, nsIntSize aDisplaySize, float aAspectRatio) { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mDecoder->GetReentrantMonitor().AssertNotCurrentThreadIn(); if (aData->mDuplicate) { return; } nsRefPtr image = aData->mImage; if (image) { mDecoder->SetVideoData(gfxIntSize(aDisplaySize.width, aDisplaySize.height), aAspectRatio, image, aTarget); } } PRInt64 nsBuiltinDecoderStateMachine::GetAudioClock() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); if (!mAudioStream || !HasAudio()) return -1; PRInt64 t = mAudioStream->GetPosition(); return (t == -1) ? -1 : t + mAudioStartTime; } void nsBuiltinDecoderStateMachine::AdvanceFrame() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); // When it's time to display a frame, decode the frame and display it. if (mDecoder->GetState() == nsBuiltinDecoder::PLAY_STATE_PLAYING) { if (HasAudio() && mAudioStartTime == -1 && !mAudioCompleted) { // We've got audio (so we should sync off the audio clock), but we've not // played a sample on the audio thread, so we can't get a time from the // audio clock. Just wait and then return, to give the audio clock time // to tick. This should really wait for a specific signal from the audio // thread rather than polling after a sleep. See bug 568431 comment 4. Wait(AUDIO_DURATION_USECS); return; } // Determine the clock time. If we've got audio, and we've not reached // the end of the audio, use the audio clock. However if we've finished // audio, or don't have audio, use the system clock. PRInt64 clock_time = -1; if (!IsPlaying()) { clock_time = mPlayDuration + mStartTime; } else { PRInt64 audio_time = GetAudioClock(); if (HasAudio() && !mAudioCompleted && audio_time != -1) { clock_time = audio_time; // Resync against the audio clock, while we're trusting the // audio clock. This ensures no "drift", particularly on Linux. mPlayDuration = clock_time - mStartTime; mPlayStartTime = TimeStamp::Now(); } else { // Sound is disabled on this system. Sync to the system clock. clock_time = DurationToUsecs(TimeStamp::Now() - mPlayStartTime) + mPlayDuration; // Ensure the clock can never go backwards. NS_ASSERTION(mCurrentFrameTime <= clock_time, "Clock should go forwards"); clock_time = NS_MAX(mCurrentFrameTime, clock_time) + mStartTime; } } // Skip frames up to the frame at the playback position, and figure out // the time remaining until it's time to display the next frame. PRInt64 remainingTime = AUDIO_DURATION_USECS; NS_ASSERTION(clock_time >= mStartTime, "Should have positive clock time."); nsAutoPtr currentFrame; if (mReader->mVideoQueue.GetSize() > 0) { VideoData* frame = mReader->mVideoQueue.PeekFront(); while (clock_time >= frame->mTime) { mVideoFrameEndTime = frame->mEndTime; currentFrame = frame; mReader->mVideoQueue.PopFront(); mDecoder->UpdatePlaybackOffset(frame->mOffset); if (mReader->mVideoQueue.GetSize() == 0) break; frame = mReader->mVideoQueue.PeekFront(); } // Current frame has already been presented, wait until it's time to // present the next frame. if (frame && !currentFrame) { PRInt64 now = IsPlaying() ? (DurationToUsecs(TimeStamp::Now() - mPlayStartTime) + mPlayDuration) : mPlayDuration; remainingTime = frame->mTime - mStartTime - now; } } // Check to see if we don't have enough data to play up to the next frame. // If we don't, switch to buffering mode. nsMediaStream* stream = mDecoder->GetCurrentStream(); if (mState == DECODER_STATE_DECODING && mDecoder->GetState() == nsBuiltinDecoder::PLAY_STATE_PLAYING && HasLowDecodedData(remainingTime + EXHAUSTED_DATA_MARGIN_USECS) && !stream->IsDataCachedToEndOfStream(mDecoder->mDecoderPosition) && !stream->IsSuspended() && (JustExitedQuickBuffering() || HasLowUndecodedData())) { if (currentFrame) { mReader->mVideoQueue.PushFront(currentFrame.forget()); } StartBuffering(); return; } // We've got enough data to keep playing until at least the next frame. // Start playing now if need be. if (!IsPlaying()) { StartPlayback(); mDecoder->GetReentrantMonitor().NotifyAll(); } if (currentFrame) { // Decode one frame and display it. TimeStamp presTime = mPlayStartTime - UsecsToDuration(mPlayDuration) + UsecsToDuration(currentFrame->mTime - mStartTime); NS_ASSERTION(currentFrame->mTime >= mStartTime, "Should have positive frame time"); { nsIntSize display = mInfo.mDisplay; float aspect = mInfo.mPixelAspectRatio; { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); // If we have video, we want to increment the clock in steps of the frame // duration. RenderVideoFrame(currentFrame, presTime, display, aspect); } } mDecoder->GetFrameStatistics().NotifyPresentedFrame(); PRInt64 now = DurationToUsecs(TimeStamp::Now() - mPlayStartTime) + mPlayDuration; remainingTime = currentFrame->mEndTime - mStartTime - now; currentFrame = nsnull; } // Kick the decode thread in case it filled its buffers and put itself // to sleep. mDecoder->GetReentrantMonitor().NotifyAll(); // Cap the current time to the larger of the audio and video end time. // This ensures that if we're running off the system clock, we don't // advance the clock to after the media end time. if (mVideoFrameEndTime != -1 || mAudioEndTime != -1) { // These will be non -1 if we've displayed a video frame, or played an audio sample. clock_time = NS_MIN(clock_time, NS_MAX(mVideoFrameEndTime, mAudioEndTime)); if (clock_time > GetMediaTime()) { // Only update the playback position if the clock time is greater // than the previous playback position. The audio clock can // sometimes report a time less than its previously reported in // some situations, and we need to gracefully handle that. UpdatePlaybackPosition(clock_time); } } // If the number of audio/video samples queued has changed, either by // this function popping and playing a video sample, or by the audio // thread popping and playing an audio sample, we may need to update our // ready state. Post an update to do so. UpdateReadyState(); if (remainingTime > 0) { Wait(remainingTime); } } else { if (IsPlaying()) { StopPlayback(AUDIO_PAUSE); mDecoder->GetReentrantMonitor().NotifyAll(); } if (mState == DECODER_STATE_DECODING || mState == DECODER_STATE_COMPLETED) { mDecoder->GetReentrantMonitor().Wait(); } } } void nsBuiltinDecoderStateMachine::Wait(PRInt64 aUsecs) { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); TimeStamp end = TimeStamp::Now() + UsecsToDuration(aUsecs); TimeStamp now; while ((now = TimeStamp::Now()) < end && mState != DECODER_STATE_SHUTDOWN && mState != DECODER_STATE_SEEKING) { PRInt64 ms = static_cast(NS_round((end - now).ToSeconds() * 1000)); if (ms == 0 || ms > PR_UINT32_MAX) { break; } mDecoder->GetReentrantMonitor().Wait(PR_MillisecondsToInterval(static_cast(ms))); } } VideoData* nsBuiltinDecoderStateMachine::FindStartTime() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); PRInt64 startTime = 0; mStartTime = 0; VideoData* v = nsnull; { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); v = mReader->FindStartTime(startTime); } if (startTime != 0) { mStartTime = startTime; if (mGotDurationFromMetaData) { NS_ASSERTION(mEndTime != -1, "We should have mEndTime as supplied duration here"); // We were specified a duration from a Content-Duration HTTP header. // Adjust mEndTime so that mEndTime-mStartTime matches the specified // duration. mEndTime = mStartTime + mEndTime; } } // Set the audio start time to be start of media. If this lies before the // first acutal audio sample we have, we'll inject silence during playback // to ensure the audio starts at the correct time. mAudioStartTime = mStartTime; LOG(PR_LOG_DEBUG, ("%p Media start time is %lld", mDecoder, mStartTime)); return v; } void nsBuiltinDecoderStateMachine::UpdateReadyState() { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); nsCOMPtr event; switch (GetNextFrameStatus()) { case nsHTMLMediaElement::NEXT_FRAME_UNAVAILABLE_BUFFERING: event = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::NextFrameUnavailableBuffering); break; case nsHTMLMediaElement::NEXT_FRAME_AVAILABLE: event = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::NextFrameAvailable); break; case nsHTMLMediaElement::NEXT_FRAME_UNAVAILABLE: event = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::NextFrameUnavailable); break; default: PR_NOT_REACHED("unhandled frame state"); } NS_DispatchToMainThread(event, NS_DISPATCH_NORMAL); } void nsBuiltinDecoderStateMachine::LoadMetadata() { NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread."); mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); LOG(PR_LOG_DEBUG, ("%p Loading Media Headers", mDecoder)); nsresult res; nsVideoInfo info; { ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor()); res = mReader->ReadMetadata(&info); } mInfo = info; if (NS_FAILED(res) || (!info.mHasVideo && !info.mHasAudio)) { mState = DECODER_STATE_SHUTDOWN; nsCOMPtr event = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::DecodeError); NS_DispatchToMainThread(event, NS_DISPATCH_NORMAL); return; } mDecoder->StartProgressUpdates(); mGotDurationFromMetaData = (GetDuration() != -1); } PRBool nsBuiltinDecoderStateMachine::JustExitedQuickBuffering() { return !mDecodeStartTime.IsNull() && mQuickBuffering && (TimeStamp::Now() - mDecodeStartTime) < TimeDuration::FromSeconds(QUICK_BUFFER_THRESHOLD_USECS); } void nsBuiltinDecoderStateMachine::StartBuffering() { mDecoder->GetReentrantMonitor().AssertCurrentThreadIn(); TimeDuration decodeDuration = TimeStamp::Now() - mDecodeStartTime; // Go into quick buffering mode provided we've not just left buffering using // a "quick exit". This stops us flip-flopping between playing and buffering // when the download speed is similar to the decode speed. mQuickBuffering = !JustExitedQuickBuffering() && decodeDuration < UsecsToDuration(QUICK_BUFFER_THRESHOLD_USECS); mBufferingStart = TimeStamp::Now(); // We need to tell the element that buffering has started. // We can't just directly send an asynchronous runnable that // eventually fires the "waiting" event. The problem is that // there might be pending main-thread events, such as "data // received" notifications, that mean we're not actually still // buffering by the time this runnable executes. So instead // we just trigger UpdateReadyStateForData; when it runs, it // will check the current state and decide whether to tell // the element we're buffering or not. UpdateReadyState(); mState = DECODER_STATE_BUFFERING; LOG(PR_LOG_DEBUG, ("%p Changed state from DECODING to BUFFERING, decoded for %.3lfs", mDecoder, decodeDuration.ToSeconds())); nsMediaDecoder::Statistics stats = mDecoder->GetStatistics(); LOG(PR_LOG_DEBUG, ("%p Playback rate: %.1lfKB/s%s download rate: %.1lfKB/s%s", mDecoder, stats.mPlaybackRate/1024, stats.mPlaybackRateReliable ? "" : " (unreliable)", stats.mDownloadRate/1024, stats.mDownloadRateReliable ? "" : " (unreliable)")); } nsresult nsBuiltinDecoderStateMachine::GetBuffered(nsTimeRanges* aBuffered) { nsMediaStream* stream = mDecoder->GetCurrentStream(); NS_ENSURE_TRUE(stream, NS_ERROR_FAILURE); stream->Pin(); nsresult res = mReader->GetBuffered(aBuffered, mStartTime); stream->Unpin(); return res; }