зеркало из https://github.com/mozilla/pjs.git
1690 строки
64 KiB
C++
1690 строки
64 KiB
C++
/* vim:set ts=2 sw=2 sts=2 et cindent: */
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/* ***** BEGIN LICENSE BLOCK *****
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* Version: ML 1.1/GPL 2.0/LGPL 2.1
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*
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* The contents of this file are subject to the Mozilla Public License Version
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* 1.1 (the "License"); you may not use this file except in compliance with
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* the License. You may obtain a copy of the License at
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* http://www.mozilla.org/MPL/
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*
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* Software distributed under the License is distributed on an "AS IS" basis,
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* WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
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* for the specific language governing rights and limitations under the
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* License.
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*
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* The Original Code is Mozilla code.
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*
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* The Initial Developer of the Original Code is the Mozilla Corporation.
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* Portions created by the Initial Developer are Copyright (C) 2007
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* the Initial Developer. All Rights Reserved.
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*
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* Contributor(s):
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* Chris Double <chris.double@double.co.nz>
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* Chris Pearce <chris@pearce.org.nz>
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*
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* Alternatively, the contents of this file may be used under the terms of
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* either the GNU General Public License Version 2 or later (the "GPL"), or
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* the GNU Lesser General Public License Version 2.1 or later (the "LGPL"),
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* in which case the provisions of the GPL or the LGPL are applicable instead
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* of those above. If you wish to allow use of your version of this file only
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* under the terms of either the GPL or the LGPL, and not to allow others to
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* use your version of this file under the terms of the MPL, indicate your
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* decision by deleting the provisions above and replace them with the notice
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* and other provisions required by the GPL or the LGPL. If you do not delete
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* the provisions above, a recipient may use your version of this file under
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* the terms of any one of the MPL, the GPL or the LGPL.
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*
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* ***** END LICENSE BLOCK ***** */
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#include <limits>
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#include "nsAudioStream.h"
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#include "nsTArray.h"
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#include "nsBuiltinDecoder.h"
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#include "nsBuiltinDecoderReader.h"
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#include "nsBuiltinDecoderStateMachine.h"
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#include "mozilla/mozalloc.h"
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#include "VideoUtils.h"
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#include "nsTimeRanges.h"
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using namespace mozilla;
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using namespace mozilla::layers;
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#ifdef PR_LOGGING
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extern PRLogModuleInfo* gBuiltinDecoderLog;
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#define LOG(type, msg) PR_LOG(gBuiltinDecoderLog, type, msg)
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#else
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#define LOG(type, msg)
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#endif
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// Wait this number of seconds when buffering, then leave and play
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// as best as we can if the required amount of data hasn't been
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// retrieved.
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#define BUFFERING_WAIT 30
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// The amount of data to retrieve during buffering is computed based
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// on the download rate. BUFFERING_MIN_RATE is the minimum download
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// rate to be used in that calculation to help avoid constant buffering
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// attempts at a time when the average download rate has not stabilised.
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#define BUFFERING_MIN_RATE 50000
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#define BUFFERING_RATE(x) ((x)< BUFFERING_MIN_RATE ? BUFFERING_MIN_RATE : (x))
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// If audio queue has less than this many usecs of decoded audio, we won't risk
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// trying to decode the video, we'll skip decoding video up to the next
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// keyframe. We may increase this value for an individual decoder if we
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// encounter video frames which take a long time to decode.
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static const PRUint32 LOW_AUDIO_USECS = 300000;
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// If more than this many usecs of decoded audio is queued, we'll hold off
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// decoding more audio. If we increase the low audio threshold (see
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// LOW_AUDIO_USECS above) we'll also increase this value to ensure it's not
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// less than the low audio threshold.
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const PRInt64 AMPLE_AUDIO_USECS = 1000000;
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// Maximum number of bytes we'll allocate and write at once to the audio
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// hardware when the audio stream contains missing samples and we're
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// writing silence in order to fill the gap. We limit our silence-writes
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// to 32KB in order to avoid allocating an impossibly large chunk of
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// memory if we encounter a large chunk of silence.
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const PRUint32 SILENCE_BYTES_CHUNK = 32 * 1024;
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// If we have fewer than LOW_VIDEO_FRAMES decoded frames, and
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// we're not "pumping video", we'll skip the video up to the next keyframe
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// which is at or after the current playback position.
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static const PRUint32 LOW_VIDEO_FRAMES = 1;
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// If we've got more than AMPLE_VIDEO_FRAMES decoded video frames waiting in
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// the video queue, we will not decode any more video frames until some have
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// been consumed by the play state machine thread.
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static const PRUint32 AMPLE_VIDEO_FRAMES = 10;
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// Arbitrary "frame duration" when playing only audio.
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static const int AUDIO_DURATION_USECS = 40000;
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// If we increase our "low audio threshold" (see LOW_AUDIO_USECS above), we
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// use this as a factor in all our calculations. Increasing this will cause
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// us to be more likely to increase our low audio threshold, and to
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// increase it by more.
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static const int THRESHOLD_FACTOR = 2;
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// If we have less than this much undecoded data available, we'll consider
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// ourselves to be running low on undecoded data. We determine how much
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// undecoded data we have remaining using the reader's GetBuffered()
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// implementation.
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static const PRInt64 LOW_DATA_THRESHOLD_USECS = 5000000;
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// LOW_DATA_THRESHOLD_USECS needs to be greater than AMPLE_AUDIO_USECS, otherwise
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// the skip-to-keyframe logic can activate when we're running low on data.
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PR_STATIC_ASSERT(LOW_DATA_THRESHOLD_USECS > AMPLE_AUDIO_USECS);
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// Amount of excess usecs of data to add in to the "should we buffer" calculation.
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static const PRUint32 EXHAUSTED_DATA_MARGIN_USECS = 60000;
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// If we enter buffering within QUICK_BUFFER_THRESHOLD_USECS seconds of starting
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// decoding, we'll enter "quick buffering" mode, which exits a lot sooner than
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// normal buffering mode. This exists so that if the decode-ahead exhausts the
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// downloaded data while decode/playback is just starting up (for example
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// after a seek while the media is still playing, or when playing a media
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// as soon as it's load started), we won't necessarily stop for 30s and wait
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// for buffering. We may actually be able to playback in this case, so exit
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// buffering early and try to play. If it turns out we can't play, we'll fall
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// back to buffering normally.
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static const PRUint32 QUICK_BUFFER_THRESHOLD_USECS = 2000000;
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// If we're quick buffering, we'll remain in buffering mode while we have less than
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// QUICK_BUFFERING_LOW_DATA_USECS of decoded data available.
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static const PRUint32 QUICK_BUFFERING_LOW_DATA_USECS = 1000000;
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// If QUICK_BUFFERING_LOW_DATA_USECS is > AMPLE_AUDIO_USECS, we won't exit
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// quick buffering in a timely fashion, as the decode pauses when it
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// reaches AMPLE_AUDIO_USECS decoded data, and thus we'll never reach
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// QUICK_BUFFERING_LOW_DATA_USECS.
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PR_STATIC_ASSERT(QUICK_BUFFERING_LOW_DATA_USECS <= AMPLE_AUDIO_USECS);
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static TimeDuration UsecsToDuration(PRInt64 aUsecs) {
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return TimeDuration::FromMilliseconds(static_cast<double>(aUsecs) / USECS_PER_MS);
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}
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static PRInt64 DurationToUsecs(TimeDuration aDuration) {
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return static_cast<PRInt64>(aDuration.ToSeconds() * USECS_PER_S);
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}
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class nsAudioMetadataEventRunner : public nsRunnable
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{
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private:
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nsCOMPtr<nsBuiltinDecoder> mDecoder;
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public:
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nsAudioMetadataEventRunner(nsBuiltinDecoder* aDecoder, PRUint32 aChannels,
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PRUint32 aRate) :
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mDecoder(aDecoder),
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mChannels(aChannels),
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mRate(aRate)
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{
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}
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NS_IMETHOD Run()
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{
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mDecoder->MetadataLoaded(mChannels, mRate);
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return NS_OK;
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}
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const PRUint32 mChannels;
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const PRUint32 mRate;
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};
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nsBuiltinDecoderStateMachine::nsBuiltinDecoderStateMachine(nsBuiltinDecoder* aDecoder,
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nsBuiltinDecoderReader* aReader) :
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mDecoder(aDecoder),
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mState(DECODER_STATE_DECODING_METADATA),
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mAudioReentrantMonitor("media.audiostream"),
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mCbCrSize(0),
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mPlayDuration(0),
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mStartTime(-1),
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mEndTime(-1),
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mSeekTime(0),
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mReader(aReader),
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mCurrentFrameTime(0),
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mAudioStartTime(-1),
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mAudioEndTime(-1),
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mVideoFrameEndTime(-1),
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mVolume(1.0),
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mSeekable(PR_TRUE),
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mPositionChangeQueued(PR_FALSE),
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mAudioCompleted(PR_FALSE),
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mGotDurationFromMetaData(PR_FALSE),
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mStopDecodeThreads(PR_TRUE),
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mQuickBuffering(PR_FALSE),
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mEventManager(aDecoder)
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{
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MOZ_COUNT_CTOR(nsBuiltinDecoderStateMachine);
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}
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nsBuiltinDecoderStateMachine::~nsBuiltinDecoderStateMachine()
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{
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MOZ_COUNT_DTOR(nsBuiltinDecoderStateMachine);
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}
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PRBool nsBuiltinDecoderStateMachine::HasFutureAudio() const {
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mDecoder->GetReentrantMonitor().AssertCurrentThreadIn();
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NS_ASSERTION(HasAudio(), "Should only call HasFutureAudio() when we have audio");
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// We've got audio ready to play if:
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// 1. We've not completed playback of audio, and
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// 2. we either have more than the threshold of decoded audio available, or
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// we've completely decoded all audio (but not finished playing it yet
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// as per 1).
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return !mAudioCompleted &&
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(AudioDecodedUsecs() > LOW_AUDIO_USECS || mReader->mAudioQueue.IsFinished());
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}
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PRBool nsBuiltinDecoderStateMachine::HaveNextFrameData() const {
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mDecoder->GetReentrantMonitor().AssertCurrentThreadIn();
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return (!HasAudio() || HasFutureAudio()) &&
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(!HasVideo() || mReader->mVideoQueue.GetSize() > 0);
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}
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PRInt64 nsBuiltinDecoderStateMachine::GetDecodedAudioDuration() {
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NS_ASSERTION(OnDecodeThread(), "Should be on decode thread.");
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mDecoder->GetReentrantMonitor().AssertCurrentThreadIn();
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PRInt64 audioDecoded = mReader->mAudioQueue.Duration();
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if (mAudioEndTime != -1) {
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audioDecoded += mAudioEndTime - GetMediaTime();
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}
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return audioDecoded;
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}
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void nsBuiltinDecoderStateMachine::DecodeLoop()
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{
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NS_ASSERTION(OnDecodeThread(), "Should be on decode thread.");
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// We want to "pump" the decode until we've got a few frames/samples decoded
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// before we consider whether decode is falling behind.
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PRBool audioPump = PR_TRUE;
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PRBool videoPump = PR_TRUE;
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// If the video decode is falling behind the audio, we'll start dropping the
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// inter-frames up until the next keyframe which is at or before the current
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// playback position. skipToNextKeyframe is PR_TRUE if we're currently
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// skipping up to the next keyframe.
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PRBool skipToNextKeyframe = PR_FALSE;
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// Once we've decoded more than videoPumpThreshold video frames, we'll
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// no longer be considered to be "pumping video".
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const unsigned videoPumpThreshold = AMPLE_VIDEO_FRAMES / 2;
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// After the audio decode fills with more than audioPumpThreshold usecs
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// of decoded audio, we'll start to check whether the audio or video decode
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// is falling behind.
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const unsigned audioPumpThreshold = LOW_AUDIO_USECS * 2;
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// Our local low audio threshold. We may increase this if we're slow to
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// decode video frames, in order to reduce the chance of audio underruns.
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PRInt64 lowAudioThreshold = LOW_AUDIO_USECS;
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// Our local ample audio threshold. If we increase lowAudioThreshold, we'll
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// also increase this too appropriately (we don't want lowAudioThreshold to
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// be greater than ampleAudioThreshold, else we'd stop decoding!).
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PRInt64 ampleAudioThreshold = AMPLE_AUDIO_USECS;
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MediaQueue<VideoData>& videoQueue = mReader->mVideoQueue;
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MediaQueue<SoundData>& audioQueue = mReader->mAudioQueue;
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ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor());
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PRBool videoPlaying = HasVideo();
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PRBool audioPlaying = HasAudio();
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// Main decode loop.
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while (mState != DECODER_STATE_SHUTDOWN &&
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!mStopDecodeThreads &&
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(videoPlaying || audioPlaying))
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{
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// We don't want to consider skipping to the next keyframe if we've
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// only just started up the decode loop, so wait until we've decoded
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// some frames before enabling the keyframe skip logic on video.
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if (videoPump &&
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static_cast<PRUint32>(videoQueue.GetSize()) >= videoPumpThreshold)
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{
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videoPump = PR_FALSE;
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}
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// We don't want to consider skipping to the next keyframe if we've
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// only just started up the decode loop, so wait until we've decoded
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// some audio data before enabling the keyframe skip logic on audio.
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if (audioPump && GetDecodedAudioDuration() >= audioPumpThreshold) {
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audioPump = PR_FALSE;
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}
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// We'll skip the video decode to the nearest keyframe if we're low on
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// audio, or if we're low on video, provided we're not running low on
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// data to decode. If we're running low on downloaded data to decode,
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// we won't start keyframe skipping, as we'll be pausing playback to buffer
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// soon anyway and we'll want to be able to display frames immediately
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// after buffering finishes.
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if (mState == DECODER_STATE_DECODING &&
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!skipToNextKeyframe &&
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videoPlaying &&
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((!audioPump && audioPlaying && GetDecodedAudioDuration() < lowAudioThreshold) ||
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(!videoPump &&
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videoPlaying &&
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static_cast<PRUint32>(videoQueue.GetSize()) < LOW_VIDEO_FRAMES)) &&
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!HasLowUndecodedData())
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{
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skipToNextKeyframe = PR_TRUE;
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LOG(PR_LOG_DEBUG, ("%p Skipping video decode to the next keyframe", mDecoder));
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}
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// Video decode.
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if (videoPlaying &&
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static_cast<PRUint32>(videoQueue.GetSize()) < AMPLE_VIDEO_FRAMES)
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{
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// Time the video decode, so that if it's slow, we can increase our low
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// audio threshold to reduce the chance of an audio underrun while we're
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// waiting for a video decode to complete.
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TimeDuration decodeTime;
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{
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PRInt64 currentTime = GetMediaTime();
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ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor());
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TimeStamp start = TimeStamp::Now();
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videoPlaying = mReader->DecodeVideoFrame(skipToNextKeyframe, currentTime);
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decodeTime = TimeStamp::Now() - start;
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}
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if (THRESHOLD_FACTOR * DurationToUsecs(decodeTime) > lowAudioThreshold &&
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!HasLowUndecodedData())
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{
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lowAudioThreshold =
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NS_MIN(THRESHOLD_FACTOR * DurationToUsecs(decodeTime), AMPLE_AUDIO_USECS);
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ampleAudioThreshold = NS_MAX(THRESHOLD_FACTOR * lowAudioThreshold,
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ampleAudioThreshold);
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LOG(PR_LOG_DEBUG,
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("Slow video decode, set lowAudioThreshold=%lld ampleAudioThreshold=%lld",
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lowAudioThreshold, ampleAudioThreshold));
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}
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}
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// Audio decode.
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if (audioPlaying &&
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(GetDecodedAudioDuration() < ampleAudioThreshold || audioQueue.GetSize() == 0))
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{
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ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor());
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audioPlaying = mReader->DecodeAudioData();
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}
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// Notify to ensure that the AudioLoop() is not waiting, in case it was
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// waiting for more audio to be decoded.
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mDecoder->GetReentrantMonitor().NotifyAll();
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if (!IsPlaying()) {
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// Update the ready state, so that the play DOM events fire. We only
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// need to do this if we're not playing; if we're playing the playback
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// code will do an update whenever it advances a frame.
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UpdateReadyState();
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}
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if (mState != DECODER_STATE_SHUTDOWN &&
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!mStopDecodeThreads &&
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(!audioPlaying || (GetDecodedAudioDuration() >= ampleAudioThreshold &&
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audioQueue.GetSize() > 0))
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&&
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(!videoPlaying ||
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static_cast<PRUint32>(videoQueue.GetSize()) >= AMPLE_VIDEO_FRAMES))
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{
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// All active bitstreams' decode is well ahead of the playback
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// position, we may as well wait for the playback to catch up. Note the
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// audio push thread acquires and notifies the decoder monitor every time
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// it pops SoundData off the audio queue. So if the audio push thread pops
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// the last SoundData off the audio queue right after that queue reported
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// it was non-empty here, we'll receive a notification on the decoder
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// monitor which will wake us up shortly after we sleep, thus preventing
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// both the decode and audio push threads waiting at the same time.
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// See bug 620326.
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mon.Wait();
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}
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} // End decode loop.
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if (!mStopDecodeThreads &&
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mState != DECODER_STATE_SHUTDOWN &&
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mState != DECODER_STATE_SEEKING)
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{
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mState = DECODER_STATE_COMPLETED;
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mDecoder->GetReentrantMonitor().NotifyAll();
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}
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LOG(PR_LOG_DEBUG, ("%p Shutting down DecodeLoop this=%p", mDecoder, this));
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}
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PRBool nsBuiltinDecoderStateMachine::IsPlaying()
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{
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mDecoder->GetReentrantMonitor().AssertCurrentThreadIn();
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return !mPlayStartTime.IsNull();
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}
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void nsBuiltinDecoderStateMachine::AudioLoop()
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{
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NS_ASSERTION(OnAudioThread(), "Should be on audio thread.");
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LOG(PR_LOG_DEBUG, ("%p Begun audio thread/loop", mDecoder));
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PRInt64 audioDuration = 0;
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PRInt64 audioStartTime = -1;
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PRUint32 channels, rate;
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double volume = -1;
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PRBool setVolume;
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PRInt32 minWriteSamples = -1;
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PRInt64 samplesAtLastSleep = 0;
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{
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ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor());
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mAudioCompleted = PR_FALSE;
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audioStartTime = mAudioStartTime;
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channels = mInfo.mAudioChannels;
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rate = mInfo.mAudioRate;
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NS_ASSERTION(audioStartTime != -1, "Should have audio start time by now");
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}
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while (1) {
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// Wait while we're not playing, and we're not shutting down, or we're
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// playing and we've got no audio to play.
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{
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ReentrantMonitorAutoEnter mon(mDecoder->GetReentrantMonitor());
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NS_ASSERTION(mState != DECODER_STATE_DECODING_METADATA,
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"Should have meta data before audio started playing.");
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while (mState != DECODER_STATE_SHUTDOWN &&
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!mStopDecodeThreads &&
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(!IsPlaying() ||
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mState == DECODER_STATE_BUFFERING ||
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(mReader->mAudioQueue.GetSize() == 0 &&
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!mReader->mAudioQueue.AtEndOfStream())))
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{
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samplesAtLastSleep = audioDuration;
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mon.Wait();
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}
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// If we're shutting down, break out and exit the audio thread.
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if (mState == DECODER_STATE_SHUTDOWN ||
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mStopDecodeThreads ||
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mReader->mAudioQueue.AtEndOfStream())
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{
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break;
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}
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// We only want to go to the expense of taking the audio monitor and
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// changing the volume if it's the first time we've entered the loop
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// (as we must sync the volume in case it's changed since the
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// nsAudioStream was created) or if the volume has changed.
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setVolume = volume != mVolume;
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volume = mVolume;
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}
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if (setVolume || minWriteSamples == -1) {
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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<PRInt64>(PR_UINT32_MAX), missingSamples);
|
|
audioDuration += PlaySilence(static_cast<PRUint32>(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<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());
|
|
{
|
|
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<nsBuiltinDecoderStateMachine*>(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<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->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<double>(mCurrentFrameTime) / static_cast<double>(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<double>(USECS_PER_S);
|
|
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->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<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::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<PRUint32>(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<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;
|
|
}
|
|
|
|
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<nsIRunnable> 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<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;
|
|
{
|
|
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<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;
|
|
{
|
|
ReentrantMonitorAutoExit exitMon(mDecoder->GetReentrantMonitor());
|
|
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, ("%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<nsIRunnable> 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<double>(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<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->GetReentrantMonitor().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->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<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()
|
|
? (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<PRInt64>(NS_round((end - now).ToSeconds() * 1000));
|
|
if (ms == 0 || ms > PR_UINT32_MAX) {
|
|
break;
|
|
}
|
|
mDecoder->GetReentrantMonitor().Wait(PR_MillisecondsToInterval(static_cast<PRUint32>(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<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->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<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_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;
|
|
}
|