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/* 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.double@double.co.nz>
* Chris Pearce <chris@pearce.org.nz>
*
* 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 <limits>
#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 ms 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_MS = 300;
// If more than this many ms of decoded audio is queued, we'll hold off
// decoding more audio. If we increase the low audio threshold (see
// LOW_AUDIO_MS above) we'll also increase this value to ensure it's not
// less than the low audio threshold.
const PRInt64 AMPLE_AUDIO_MS = 1000;
// 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_MS = 40;
// If we increase our "low audio threshold" (see LOW_AUDIO_MS 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_MS = 5000;
// LOW_DATA_THRESHOLD_MS needs to be greater than AMPLE_AUDIO_MS, otherwise
// the skip-to-keyframe logic can activate when we're running low on data.
PR_STATIC_ASSERT(LOW_DATA_THRESHOLD_MS > AMPLE_AUDIO_MS);
// Amount of excess ms of data to add in to the "should we buffer" calculation.
static const PRUint32 EXHAUSTED_DATA_MARGIN_MS = 60;
// If we enter buffering within QUICK_BUFFER_THRESHOLD_MS 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_MS = 2000;
// If we're quick buffering, we'll remain in buffering mode while we have less than
// QUICK_BUFFERING_LOW_DATA_MS of decoded data available.
static const PRUint32 QUICK_BUFFERING_LOW_DATA_MS = 1000;
// If QUICK_BUFFERING_LOW_DATA_MS is > AMPLE_AUDIO_MS, we won't exit
// quick buffering in a timely fashion, as the decode pauses when it
// reaches AMPLE_AUDIO_MS decoded data, and thus we'll never reach
// QUICK_BUFFERING_LOW_DATA_MS.
PR_STATIC_ASSERT(QUICK_BUFFERING_LOW_DATA_MS <= AMPLE_AUDIO_MS);
static TimeDuration MsToDuration(PRInt64 aMs) {
return TimeDuration::FromMilliseconds(static_cast<double>(aMs));
}
static PRInt64 DurationToMs(TimeDuration aDuration) {
return static_cast<PRInt64>(aDuration.ToSeconds() * 1000);
}
class nsAudioMetadataEventRunner : public nsRunnable
{
private:
nsCOMPtr<nsBuiltinDecoder> mDecoder;
public:
nsAudioMetadataEventRunner(nsBuiltinDecoder* aDecoder, PRUint32 aChannels,
PRUint32 aRate, PRUint32 aFrameBufferLength) :
mDecoder(aDecoder),
mChannels(aChannels),
mRate(aRate),
mFrameBufferLength(aFrameBufferLength)
{
}
NS_IMETHOD Run()
{
mDecoder->MetadataLoaded(mChannels, mRate, mFrameBufferLength);
return NS_OK;
}
const PRUint32 mChannels;
const PRUint32 mRate;
const PRUint32 mFrameBufferLength;
};
nsBuiltinDecoderStateMachine::nsBuiltinDecoderStateMachine(nsBuiltinDecoder* aDecoder,
nsBuiltinDecoderReader* aReader) :
mDecoder(aDecoder),
mState(DECODER_STATE_DECODING_METADATA),
mAudioMonitor("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->GetMonitor().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 &&
(AudioDecodedMs() > LOW_AUDIO_MS || mReader->mAudioQueue.IsFinished());
}
PRBool nsBuiltinDecoderStateMachine::HaveNextFrameData() const {
mDecoder->GetMonitor().AssertCurrentThreadIn();
return (!HasAudio() || HasFutureAudio()) &&
(!HasVideo() || mReader->mVideoQueue.GetSize() > 0);
}
PRInt64 nsBuiltinDecoderStateMachine::GetDecodedAudioDuration() {
NS_ASSERTION(OnDecodeThread(), "Should be on decode thread.");
mDecoder->GetMonitor().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 audioPumpThresholdMs ms
// of decoded audio, we'll start to check whether the audio or video decode
// is falling behind.
const unsigned audioPumpThresholdMs = LOW_AUDIO_MS * 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_MS;
// 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_MS;
MediaQueue<VideoData>& videoQueue = mReader->mVideoQueue;
MediaQueue<SoundData>& audioQueue = mReader->mAudioQueue;
MonitorAutoEnter mon(mDecoder->GetMonitor());
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<PRUint32>(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() >= audioPumpThresholdMs) {
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<PRUint32>(videoQueue.GetSize()) < LOW_VIDEO_FRAMES)) &&
!HasLowUndecodedData())
{
skipToNextKeyframe = PR_TRUE;
LOG(PR_LOG_DEBUG, ("Skipping video decode to the next keyframe"));
}
// Video decode.
if (videoPlaying &&
static_cast<PRUint32>(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();
MonitorAutoExit exitMon(mDecoder->GetMonitor());
TimeStamp start = TimeStamp::Now();
videoPlaying = mReader->DecodeVideoFrame(skipToNextKeyframe, currentTime);
decodeTime = TimeStamp::Now() - start;
}
if (THRESHOLD_FACTOR * DurationToMs(decodeTime) > lowAudioThreshold &&
!HasLowUndecodedData())
{
lowAudioThreshold =
NS_MIN(THRESHOLD_FACTOR * DurationToMs(decodeTime), AMPLE_AUDIO_MS);
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))
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
audioPlaying = mReader->DecodeAudioData();
}
// Notify to ensure that the AudioLoop() is not waiting, in case it was
// waiting for more audio to be decoded.
mDecoder->GetMonitor().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<PRUint32>(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->GetMonitor().NotifyAll();
}
LOG(PR_LOG_DEBUG, ("Shutting down DecodeLoop this=%p", this));
}
PRBool nsBuiltinDecoderStateMachine::IsPlaying()
{
mDecoder->GetMonitor().AssertCurrentThreadIn();
return !mPlayStartTime.IsNull();
}
void nsBuiltinDecoderStateMachine::AudioLoop()
{
NS_ASSERTION(OnAudioThread(), "Should be on audio thread.");
LOG(PR_LOG_DEBUG, ("Begun audio thread/loop"));
PRInt64 audioDuration = 0;
PRInt64 audioStartTime = -1;
PRUint32 channels, rate;
double volume = -1;
PRBool setVolume;
PRInt32 minWriteSamples = -1;
PRInt64 samplesAtLastSleep = 0;
{
MonitorAutoEnter mon(mDecoder->GetMonitor());
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.
{
MonitorAutoEnter mon(mDecoder->GetMonitor());
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) {
MonitorAutoEnter audioMon(mAudioMonitor);
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 (!MsToSamples(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 (!MsToSamples(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<PRInt64>(PR_UINT32_MAX), missingSamples);
audioDuration += PlaySilence(static_cast<PRUint32>(missingSamples),
channels, playedSamples);
} else {
audioDuration += PlayFromAudioQueue(sampleTime, channels);
}
{
MonitorAutoEnter mon(mDecoder->GetMonitor());
PRInt64 playedMs;
if (!SamplesToMs(audioDuration, rate, playedMs)) {
NS_WARNING("Int overflow calculating playedMs");
break;
}
if (!AddOverflow(audioStartTime, playedMs, mAudioEndTime)) {
NS_WARNING("Int overflow calculating audio end time");
break;
}
PRInt64 audioAhead = mAudioEndTime - GetMediaTime();
if (audioAhead > AMPLE_AUDIO_MS &&
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_MS / 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.
MonitorAutoEnter audioMon(mAudioMonitor);
if (mAudioStream) {
PRBool seeking = PR_FALSE;
PRInt64 oldPosition = -1;
{
MonitorAutoExit audioExit(mAudioMonitor);
MonitorAutoEnter mon(mDecoder->GetMonitor());
PRInt64 position = GetMediaTime();
while (oldPosition != position &&
mAudioEndTime - position > 0 &&
mState != DECODER_STATE_SEEKING &&
mState != DECODER_STATE_SHUTDOWN)
{
const PRInt64 DRAIN_BLOCK_MS = 100;
Wait(NS_MIN(mAudioEndTime - position, DRAIN_BLOCK_MS));
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));
}
{
MonitorAutoEnter mon(mDecoder->GetMonitor());
mAudioCompleted = PR_TRUE;
UpdateReadyState();
// Kick the decode and state machine threads; they may be sleeping waiting
// for this to finish.
mDecoder->GetMonitor().NotifyAll();
}
LOG(PR_LOG_DEBUG, ("Audio stream finished playing, audio thread exit"));
}
PRUint32 nsBuiltinDecoderStateMachine::PlaySilence(PRUint32 aSamples,
PRUint32 aChannels,
PRUint64 aSampleOffset)
{
MonitorAutoEnter audioMon(mAudioMonitor);
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<SoundDataValue> 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<SoundData> sound(mReader->mAudioQueue.PopFront());
{
MonitorAutoEnter mon(mDecoder->GetMonitor());
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->GetMonitor().NotifyAll();
}
PRInt64 offset = -1;
PRUint32 samples = 0;
{
MonitorAutoEnter audioMon(mAudioMonitor);
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<nsBuiltinDecoderStateMachine*>(aCloneDonor)->mReader;
}
return mReader->Init(cloneReader);
}
void nsBuiltinDecoderStateMachine::StopPlayback(eStopMode aMode)
{
NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread),
"Should be on state machine thread.");
mDecoder->GetMonitor().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 += TimeStamp::Now() - mPlayStartTime;
mPlayStartTime = TimeStamp();
}
if (HasAudio()) {
MonitorAutoExit exitMon(mDecoder->GetMonitor());
MonitorAutoEnter audioMon(mAudioMonitor);
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->GetMonitor().AssertCurrentThreadIn();
LOG(PR_LOG_DEBUG, ("%p StartPlayback", mDecoder));
mDecoder->mPlaybackStatistics.Start(TimeStamp::Now());
if (HasAudio()) {
PRInt32 rate = mInfo.mAudioRate;
PRInt32 channels = mInfo.mAudioChannels;
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
MonitorAutoEnter audioMon(mAudioMonitor);
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->GetMonitor().NotifyAll();
}
void nsBuiltinDecoderStateMachine::UpdatePlaybackPositionInternal(PRInt64 aTime)
{
NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread),
"Should be on state machine thread.");
mDecoder->GetMonitor().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<nsIRunnable> 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<nsIRunnable> 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->GetMonitor().AssertCurrentThreadIn();
mPositionChangeQueued = PR_FALSE;
}
nsHTMLMediaElement::NextFrameStatus nsBuiltinDecoderStateMachine::GetNextFrameStatus()
{
MonitorAutoEnter mon(mDecoder->GetMonitor());
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.");
MonitorAutoEnter mon(mDecoder->GetMonitor());
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<double>(mCurrentFrameTime) / 1000.0;
}
PRInt64 nsBuiltinDecoderStateMachine::GetDuration()
{
mDecoder->GetMonitor().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->GetMonitor().AssertCurrentThreadIn();
if (mStartTime != -1) {
mEndTime = mStartTime + aDuration;
} else {
mStartTime = 0;
mEndTime = aDuration;
}
}
void nsBuiltinDecoderStateMachine::SetSeekable(PRBool aSeekable)
{
NS_ASSERTION(NS_IsMainThread(), "Should be on main thread.");
mDecoder->GetMonitor().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.
MonitorAutoEnter mon(mDecoder->GetMonitor());
// 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->GetMonitor().NotifyAll();
}
void nsBuiltinDecoderStateMachine::StartDecoding()
{
NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread),
"Should be on state machine thread.");
MonitorAutoEnter mon(mDecoder->GetMonitor());
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.
MonitorAutoEnter mon(mDecoder->GetMonitor());
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->GetMonitor().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.");
MonitorAutoEnter mon(mDecoder->GetMonitor());
// 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 * 1000.0;
if (t > PR_INT64_MAX) {
// Prevent integer overflow.
return;
}
mSeekTime = static_cast<PRInt64>(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->GetMonitor().AssertCurrentThreadIn();
mStopDecodeThreads = PR_TRUE;
mDecoder->GetMonitor().NotifyAll();
if (mDecodeThread) {
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
mDecodeThread->Shutdown();
}
mDecodeThread = nsnull;
}
if (mAudioThread) {
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
mAudioThread->Shutdown();
}
mAudioThread = nsnull;
}
}
nsresult
nsBuiltinDecoderStateMachine::StartDecodeThreads()
{
NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread),
"Should be on state machine thread.");
mDecoder->GetMonitor().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<nsIRunnable> 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<nsIRunnable> event =
NS_NewRunnableMethod(this, &nsBuiltinDecoderStateMachine::AudioLoop);
mAudioThread->Dispatch(event, NS_DISPATCH_NORMAL);
}
return NS_OK;
}
PRInt64 nsBuiltinDecoderStateMachine::AudioDecodedMs() const
{
NS_ASSERTION(HasAudio(),
"Should only call AudioDecodedMs() 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 aAudioMs) const
{
mDecoder->GetMonitor().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() &&
AudioDecodedMs() < aAudioMs)
||
(!HasAudio() &&
HasVideo() &&
!mReader->mVideoQueue.IsFinished() &&
static_cast<PRUint32>(mReader->mVideoQueue.GetSize()) < LOW_VIDEO_FRAMES));
}
PRBool nsBuiltinDecoderStateMachine::HasLowUndecodedData() const
{
return GetUndecodedData() < LOW_DATA_THRESHOLD_MS;
}
PRInt64 nsBuiltinDecoderStateMachine::GetUndecodedData() const
{
mDecoder->GetMonitor().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<nsIDOMTimeRanges*>(&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<PRInt64>((end - currentTime) * 1000);
}
}
return 0;
}
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) {
MonitorAutoEnter mon(mDecoder->GetMonitor());
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;
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
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 %lldms to %lldms (duration %lldms) 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.
PRUint32 frameBufferLength = mInfo.mAudioChannels * FRAMEBUFFER_LENGTH_PER_CHANNEL;
nsCOMPtr<nsIRunnable> metadataLoadedEvent =
new nsAudioMetadataEventRunner(mDecoder, mInfo.mAudioChannels,
mInfo.mAudioRate, frameBufferLength);
NS_DispatchToMainThread(metadataLoadedEvent, NS_DISPATCH_NORMAL);
if (HasAudio()) {
mEventManager.Init(mInfo.mAudioChannels, mInfo.mAudioRate);
mDecoder->RequestFrameBufferLength(frameBufferLength);
}
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
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
nsCOMPtr<nsIRunnable> 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;
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
// 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 = MsToDuration(startTime - mStartTime);
if (HasVideo()) {
nsAutoPtr<VideoData> 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;
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
RenderVideoFrame(video, TimeStamp::Now(), display, aspect);
}
mReader->mVideoQueue.PopFront();
nsCOMPtr<nsIRunnable> 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, ("Seek completed, mCurrentFrameTime=%lld\n", 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<nsIRunnable> stopEvent;
if (GetMediaTime() == mEndTime) {
LOG(PR_LOG_DEBUG, ("%p Changed state from SEEKING (to %lldms) 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 %lldms) to DECODING",
mDecoder, seekTime));
stopEvent = NS_NewRunnableMethod(mDecoder, &nsBuiltinDecoder::SeekingStopped);
StartDecoding();
}
mDecoder->GetMonitor().NotifyAll();
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
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->GetMonitor().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_MS)
: (GetUndecodedData() < BUFFERING_WAIT * 1000)) &&
!stream->IsDataCachedToEndOfStream(mDecoder->mDecoderPosition) &&
!stream->IsSuspended())
{
LOG(PR_LOG_DEBUG,
("Buffering: %.3lfs/%ds, timeout in %.3lfs %s",
GetUndecodedData() / 1000.0,
BUFFERING_WAIT,
BUFFERING_WAIT - elapsed.ToSeconds(),
(mQuickBuffering ? "(quick exit)" : "")));
Wait(1000);
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->GetMonitor().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, ("Shutting down the state machine thread"));
StopDecodeThreads();
if (mDecoder->GetState() == nsBuiltinDecoder::PLAY_STATE_PLAYING) {
PRInt64 videoTime = HasVideo() ? mVideoFrameEndTime : 0;
PRInt64 clockTime = NS_MAX(mEndTime, NS_MAX(videoTime, GetAudioClock()));
UpdatePlaybackPosition(clockTime);
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
nsCOMPtr<nsIRunnable> 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<nsIRunnable> 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->GetMonitor().AssertNotCurrentThreadIn();
if (aData->mDuplicate) {
return;
}
nsRefPtr<Image> 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->GetMonitor().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_MS);
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 = DurationToMs(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 = MsToDuration(clock_time - mStartTime);
mPlayStartTime = TimeStamp::Now();
} else {
// Sound is disabled on this system. Sync to the system clock.
clock_time = DurationToMs(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_MS;
NS_ASSERTION(clock_time >= mStartTime, "Should have positive clock time.");
nsAutoPtr<VideoData> 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()
? DurationToMs(TimeStamp::Now() - mPlayStartTime + mPlayDuration)
: DurationToMs(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_MS) &&
!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->GetMonitor().NotifyAll();
}
if (currentFrame) {
// Decode one frame and display it.
TimeStamp presTime = mPlayStartTime - mPlayDuration +
MsToDuration(currentFrame->mTime - mStartTime);
NS_ASSERTION(currentFrame->mTime >= mStartTime, "Should have positive frame time");
{
nsIntSize display = mInfo.mDisplay;
float aspect = mInfo.mPixelAspectRatio;
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
// 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 = DurationToMs(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->GetMonitor().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->GetMonitor().NotifyAll();
}
if (mState == DECODER_STATE_DECODING ||
mState == DECODER_STATE_COMPLETED) {
mDecoder->GetMonitor().Wait();
}
}
}
void nsBuiltinDecoderStateMachine::Wait(PRInt64 aMs) {
mDecoder->GetMonitor().AssertCurrentThreadIn();
TimeStamp end = TimeStamp::Now() + MsToDuration(aMs);
TimeStamp now;
while ((now = TimeStamp::Now()) < end &&
mState != DECODER_STATE_SHUTDOWN &&
mState != DECODER_STATE_SEEKING)
{
PRInt64 ms = static_cast<PRInt64>(NS_round((end - now).ToSeconds() * 1000));
if (ms == 0 || ms > PR_UINT32_MAX) {
break;
}
NS_ASSERTION(ms <= aMs && ms > 0,
"nsBuiltinDecoderStateMachine::Wait interval very wrong!");
mDecoder->GetMonitor().Wait(PR_MillisecondsToInterval(static_cast<PRUint32>(ms)));
}
}
VideoData* nsBuiltinDecoderStateMachine::FindStartTime()
{
NS_ASSERTION(IsCurrentThread(mDecoder->mStateMachineThread), "Should be on state machine thread.");
mDecoder->GetMonitor().AssertCurrentThreadIn();
PRInt64 startTime = 0;
mStartTime = 0;
VideoData* v = nsnull;
PRInt64 dataOffset = mInfo.mDataOffset;
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
v = mReader->FindStartTime(dataOffset, 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 %lldms", mDecoder, mStartTime));
return v;
}
void nsBuiltinDecoderStateMachine::FindEndTime()
{
NS_ASSERTION(OnStateMachineThread(), "Should be on state machine thread.");
mDecoder->GetMonitor().AssertCurrentThreadIn();
nsMediaStream* stream = mDecoder->GetCurrentStream();
// Seek to the end of file to find the length and duration.
PRInt64 length = stream->GetLength();
NS_ASSERTION(length > 0, "Must have a content length to get end time");
mEndTime = 0;
PRInt64 endTime = 0;
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
endTime = mReader->FindEndTime(length);
}
if (endTime != -1) {
mEndTime = endTime;
}
LOG(PR_LOG_DEBUG, ("%p Media end time is %lldms", mDecoder, mEndTime));
}
void nsBuiltinDecoderStateMachine::UpdateReadyState() {
mDecoder->GetMonitor().AssertCurrentThreadIn();
nsCOMPtr<nsIRunnable> 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->GetMonitor().AssertCurrentThreadIn();
LOG(PR_LOG_DEBUG, ("Loading Media Headers"));
nsresult res;
nsVideoInfo info;
{
MonitorAutoExit exitMon(mDecoder->GetMonitor());
res = mReader->ReadMetadata(&info);
}
mInfo = info;
if (NS_FAILED(res) || (!info.mHasVideo && !info.mHasAudio)) {
mState = DECODER_STATE_SHUTDOWN;
nsCOMPtr<nsIRunnable> 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_MS);
}
void nsBuiltinDecoderStateMachine::StartBuffering()
{
mDecoder->GetMonitor().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 < TimeDuration::FromMilliseconds(QUICK_BUFFER_THRESHOLD_MS);
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, ("Changed state from DECODING to BUFFERING, decoded for %.3lfs",
decodeDuration.ToSeconds()));
nsMediaDecoder::Statistics stats = mDecoder->GetStatistics();
LOG(PR_LOG_DEBUG, ("Playback rate: %.1lfKB/s%s download rate: %.1lfKB/s%s",
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;
}
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