Bug 1350814 - Replace use of int64_t for microseconds by TimeUnit in AudioSink. r=kikuo

MozReview-Commit-ID: 3diOpJu7g8i

--HG--
extra : rebase_source : 16d7050cd29ea774fce7e76a16ae9100c766b9f8
extra : source : cf6946ba307d04b00e02e353b4724655929eaa69

dom/media/MediaDecoderStateMachine.cpp

@@ -2700,7 +2700,8 @@ MediaDecoderStateMachine::CreateAudioSink()
   auto audioSinkCreator = [self] () {
     MOZ_ASSERT(self->OnTaskQueue());
     AudioSink* audioSink = new AudioSink(
-      self->mTaskQueue, self->mAudioQueue, self->GetMediaTime(),
+      self->mTaskQueue, self->mAudioQueue,
+      TimeUnit::FromMicroseconds(self->GetMediaTime()),
       self->Info().mAudio, self->mAudioChannel);
 
     self->mAudibleListener = audioSink->AudibleEvent().Connect(

dom/media/mediasink/AudioSink.cpp

@@ -33,11 +33,10 @@ static const int32_t LOW_AUDIO_USECS = 300000;
 
 AudioSink::AudioSink(AbstractThread* aThread,
                      MediaQueue<AudioData>& aAudioQueue,
-                     int64_t aStartTime,
+                     TimeUnit aStartTime,
                      const AudioInfo& aInfo,
                      dom::AudioChannel aChannel)
   : mStartTime(aStartTime)
-  , mLastGoodPosition(0)
   , mInfo(aInfo)
   , mChannel(aChannel)
   , mPlaying(true)
@@ -48,7 +47,6 @@ AudioSink::AudioSink(AbstractThread* aThread,
   , mOwnerThread(aThread)
   , mProcessedQueueLength(0)
   , mFramesParsed(0)
-  , mLastEndTime(0)
   , mIsAudioDataAudible(false)
   , mAudioQueue(aAudioQueue)
 {
@@ -103,12 +101,13 @@ AudioSink::Init(const PlaybackParams& aParams)
   return p;
 }
 
-int64_t
+TimeUnit
 AudioSink::GetPosition()
 {
-  int64_t pos;
+  int64_t tmp;
   if (mAudioStream &&
-      (pos = mAudioStream->GetPosition()) >= 0) {
+      (tmp = mAudioStream->GetPosition()) >= 0) {
+    TimeUnit pos = TimeUnit::FromMicroseconds(tmp);
     NS_ASSERTION(pos >= mLastGoodPosition,
                  "AudioStream position shouldn't go backward");
     // Update the last good position when we got a good one.
@@ -221,7 +220,7 @@ AudioSink::InitializeAudioStream(const PlaybackParams& aParams)
   return NS_OK;
 }
 
-int64_t
+TimeUnit
 AudioSink::GetEndTime() const
 {
   int64_t written;
@@ -229,14 +228,14 @@ AudioSink::GetEndTime() const
     MonitorAutoLock mon(mMonitor);
     written = mWritten;
   }
-  CheckedInt64 playedUsecs = FramesToUsecs(written, mOutputRate) + mStartTime;
-  if (!playedUsecs.isValid()) {
+  TimeUnit played = FramesToTimeUnit(written, mOutputRate) + mStartTime;
+  if (!played.IsValid()) {
     NS_WARNING("Int overflow calculating audio end time");
-    return 0;
+    return TimeUnit::Zero();
   }
   // As we may be resampling, rounding errors may occur. Ensure we never get
   // past the original end time.
-  return std::min<int64_t>(mLastEndTime, playedUsecs.value());
+  return std::min(mLastEndTime, played);
 }
 
 UniquePtr<AudioStream::Chunk>
@@ -407,8 +406,8 @@ AudioSink::NotifyAudioNeeded()
     // audio hardware, so we can play across the gap.
     // Calculate the timestamp of the next chunk of audio in numbers of
     // samples.
-    CheckedInt64 sampleTime = UsecsToFrames(data->mTime - mStartTime,
-                                            data->mRate);
+    CheckedInt64 sampleTime = TimeUnitToFrames(
+      TimeUnit::FromMicroseconds(data->mTime) - mStartTime, data->mRate);
     // Calculate the number of frames that have been pushed onto the audio hardware.
     CheckedInt64 missingFrames = sampleTime - mFramesParsed;
 
@@ -450,7 +449,7 @@ AudioSink::NotifyAudioNeeded()
       }
     }
 
-    mLastEndTime = data->GetEndTime();
+    mLastEndTime = TimeUnit::FromMicroseconds(data->GetEndTime());
     mFramesParsed += data->mFrames;
 
     if (mConverter->InputConfig() != mConverter->OutputConfig()) {

dom/media/mediasink/AudioSink.h

@@ -32,7 +32,7 @@ class AudioSink : private AudioStream::DataSource {
 public:
   AudioSink(AbstractThread* aThread,
             MediaQueue<AudioData>& aAudioQueue,
-            int64_t aStartTime,
+            TimeUnit aStartTime,
             const AudioInfo& aInfo,
             dom::AudioChannel aChannel);
 
@@ -46,8 +46,8 @@ public:
    * All public functions are not thread-safe.
    * Called on the task queue of MDSM only.
    */
-  int64_t GetPosition();
-  int64_t GetEndTime() const;
+  TimeUnit GetPosition();
+  TimeUnit GetEndTime() const;
 
   // Check whether we've pushed more frames to the audio hardware than it has
   // played.
@@ -80,15 +80,15 @@ private:
   // The audio stream resource. Used on the task queue of MDSM only.
   RefPtr<AudioStream> mAudioStream;
 
-  // The presentation time of the first audio frame that was played in
-  // microseconds. We can add this to the audio stream position to determine
+  // The presentation time of the first audio frame that was played.
+  // We can add this to the audio stream position to determine
   // the current audio time.
-  const int64_t mStartTime;
+  const TimeUnit mStartTime;
 
   // Keep the last good position returned from the audio stream. Used to ensure
   // position returned by GetPosition() is mono-increasing in spite of audio
   // stream error. Used on the task queue of MDSM only.
-  int64_t mLastGoodPosition;
+  TimeUnit mLastGoodPosition;
 
   const AudioInfo mInfo;
 
@@ -149,7 +149,7 @@ private:
   // at the current input framerate.
   int64_t mFramesParsed;
   Maybe<RefPtr<AudioData>> mLastProcessedPacket;
-  int64_t mLastEndTime;
+  TimeUnit mLastEndTime;
   // Never modifed after construction.
   uint32_t mOutputRate;
   uint32_t mOutputChannels;

dom/media/mediasink/AudioSinkWrapper.cpp

@@ -58,7 +58,7 @@ AudioSinkWrapper::GetEndTime(TrackType aType) const
   AssertOwnerThread();
   MOZ_ASSERT(mIsStarted, "Must be called after playback starts.");
   if (aType == TrackInfo::kAudioTrack && mAudioSink) {
-    return mAudioSink->GetEndTime();
+    return mAudioSink->GetEndTime().ToMicroseconds();
   }
   return 0;
 }
@@ -85,7 +85,7 @@ AudioSinkWrapper::GetPosition(TimeStamp* aTimeStamp) const
 
   if (!mAudioEnded) {
     // Rely on the audio sink to report playback position when it is not ended.
-    pos = mAudioSink->GetPosition();
+    pos = mAudioSink->GetPosition().ToMicroseconds();
   } else if (!mPlayStartTime.IsNull()) {
     // Calculate playback position using system clock if we are still playing.
     pos = GetVideoPosition(t);