Index: media/filters/audio_renderer_impl.cc |
diff --git a/media/filters/audio_renderer_impl.cc b/media/filters/audio_renderer_impl.cc |
index 81fb574e8bfdd50a99bd064a05937431931e4689..164864db5f31f406f8d87a32c0c4016698fe584e 100644 |
--- a/media/filters/audio_renderer_impl.cc |
+++ b/media/filters/audio_renderer_impl.cc |
@@ -148,6 +148,7 @@ void AudioRendererImpl::SetMediaTime(base::TimeDelta time) { |
DCHECK_EQ(state_, kFlushed); |
start_timestamp_ = time; |
+ ended_timestamp_ = kInfiniteDuration(); |
audio_clock_.reset(new AudioClock(time, audio_parameters_.sample_rate())); |
} |
@@ -547,7 +548,6 @@ int AudioRendererImpl::Render(AudioBus* audio_bus, |
const int delay_frames = static_cast<int>(playback_delay.InSecondsF() * |
audio_parameters_.sample_rate()); |
int frames_written = 0; |
- base::Closure time_cb; |
{ |
base::AutoLock auto_lock(lock_); |
@@ -587,46 +587,59 @@ int AudioRendererImpl::Render(AudioBus* audio_bus, |
frames_written = |
algorithm_->FillBuffer(audio_bus, requested_frames, playback_rate_); |
} |
- audio_clock_->WroteAudio( |
- frames_written, requested_frames, delay_frames, playback_rate_); |
+ // Per the TimeSource API the media time should always increase even after |
+ // we've rendered all known audio data. Doing so simplifies scenarios where |
+ // we have other sources of media data that need to be scheduled after audio |
+ // data has ended. |
+ // |
+ // That being said, we don't want to advance time when underflowed as we |
+ // know more decoded frames will eventually arrive. If we did, we would |
+ // throw things out of sync when said decoded frames arrive. |
+ int frames_after_end_of_stream = 0; |
if (frames_written == 0) { |
- if (received_end_of_stream_ && !rendered_end_of_stream_ && |
- !audio_clock_->audio_data_buffered()) { |
- rendered_end_of_stream_ = true; |
- task_runner_->PostTask(FROM_HERE, ended_cb_); |
- } else if (!received_end_of_stream_ && state_ == kPlaying) { |
- if (buffering_state_ != BUFFERING_HAVE_NOTHING) { |
- algorithm_->IncreaseQueueCapacity(); |
- SetBufferingState_Locked(BUFFERING_HAVE_NOTHING); |
- } |
+ if (received_end_of_stream_) { |
+ if (ended_timestamp_ == kInfiniteDuration()) |
+ ended_timestamp_ = audio_clock_->back_timestamp(); |
+ frames_after_end_of_stream = requested_frames; |
+ } else if (state_ == kPlaying && |
+ buffering_state_ != BUFFERING_HAVE_NOTHING) { |
+ algorithm_->IncreaseQueueCapacity(); |
+ SetBufferingState_Locked(BUFFERING_HAVE_NOTHING); |
} |
} |
+ audio_clock_->WroteAudio(frames_written + frames_after_end_of_stream, |
+ requested_frames, |
+ delay_frames, |
+ playback_rate_); |
+ |
if (CanRead_Locked()) { |
task_runner_->PostTask(FROM_HERE, |
base::Bind(&AudioRendererImpl::AttemptRead, |
weak_factory_.GetWeakPtr())); |
} |
- // Firing |ended_cb_| means we no longer need to run |time_cb_|. |
- if (!rendered_end_of_stream_ && |
- last_timestamp_update_ != audio_clock_->current_media_timestamp()) { |
+ if (last_timestamp_update_ != audio_clock_->front_timestamp()) { |
// Since |max_time| uses linear interpolation, only provide an upper bound |
// that is for audio data at the same playback rate. Failing to do so can |
// make time jump backwards when the linear interpolated time advances |
// past buffered regions of audio at different rates. |
- last_timestamp_update_ = audio_clock_->current_media_timestamp(); |
+ last_timestamp_update_ = audio_clock_->front_timestamp(); |
base::TimeDelta max_time = |
last_timestamp_update_ + |
audio_clock_->contiguous_audio_data_buffered_at_same_rate(); |
- time_cb = base::Bind(time_cb_, last_timestamp_update_, max_time); |
+ task_runner_->PostTask( |
+ FROM_HERE, base::Bind(time_cb_, last_timestamp_update_, max_time)); |
+ |
+ if (last_timestamp_update_ >= ended_timestamp_ && |
+ !rendered_end_of_stream_) { |
+ rendered_end_of_stream_ = true; |
+ task_runner_->PostTask(FROM_HERE, ended_cb_); |
+ } |
} |
} |
- if (!time_cb.is_null()) |
- task_runner_->PostTask(FROM_HERE, time_cb); |
- |
DCHECK_LE(frames_written, requested_frames); |
return frames_written; |
} |