Index: content/renderer/media/webrtc_audio_renderer.cc |
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc |
index 640d8dc4ad689ba6db96781017231662b532d0be..6689222262dac1be09a3c8330de50cddbee3c557 100644 |
--- a/content/renderer/media/webrtc_audio_renderer.cc |
+++ b/content/renderer/media/webrtc_audio_renderer.cc |
@@ -187,7 +187,6 @@ WebRtcAudioRenderer::WebRtcAudioRenderer( |
play_ref_count_(0), |
start_ref_count_(0), |
audio_delay_milliseconds_(0), |
- fifo_delay_milliseconds_(0), |
sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
media::CHANNEL_LAYOUT_STEREO, |
0, |
@@ -434,6 +433,26 @@ int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus, |
DCHECK_LE(audio_delay_milliseconds, static_cast<uint32_t>(INT_MAX)); |
audio_delay_milliseconds_ = static_cast<int>(audio_delay_milliseconds); |
+ // If there are skipped frames, pull and throw away the same amount. |
+ if (frames_skipped > 0) { |
+ const uint32_t frames_per_10ms = |
+ static_cast<uint32_t>(sink_params_.sample_rate() / 100); |
+ if (!audio_fifo_ && frames_skipped != frames_per_10ms) { |
+ audio_fifo_.reset(new media::AudioPullFifo( |
+ sink_params_.channels(), frames_per_10ms, |
+ base::Bind(&WebRtcAudioRenderer::SourceCallback, |
+ base::Unretained(this)))); |
+ } |
+ |
+ scoped_ptr<media::AudioBus> drop_bus = |
+ media::AudioBus::Create(audio_bus->channels(), frames_skipped); |
+ if (audio_fifo_) |
+ audio_fifo_->Consume(drop_bus.get(), drop_bus->frames()); |
+ else |
+ SourceCallback(0, drop_bus.get()); |
+ } |
+ |
+ // Pull the data we shall deliver. |
tommi (sloooow) - chröme
2016/01/15 18:50:03
nit: s/shall/will
Henrik Grunell
2016/01/18 13:00:54
Done.
|
if (audio_fifo_) |
audio_fifo_->Consume(audio_bus, audio_bus->frames()); |
else |
@@ -457,7 +476,9 @@ void WebRtcAudioRenderer::SourceCallback( |
<< audio_bus->frames() << ")"; |
int output_delay_milliseconds = audio_delay_milliseconds_; |
- output_delay_milliseconds += fifo_delay_milliseconds_; |
+ output_delay_milliseconds += fifo_frame_delay * |
+ base::Time::kMillisecondsPerSecond / |
+ sink_params_.sample_rate(); |
DVLOG(2) << "output_delay_milliseconds: " << output_delay_milliseconds; |
// We need to keep render data for the |source_| regardless of |state_|, |
@@ -627,34 +648,28 @@ void WebRtcAudioRenderer::PrepareSink() { |
// Create a FIFO if re-buffering is required to match the source input with |
// the sink request. The source acts as provider here and the sink as |
// consumer. |
- int new_fifo_delay_milliseconds = 0; |
scoped_ptr<media::AudioPullFifo> new_audio_fifo; |
- if (source_params.frames_per_buffer() != |
- new_sink_params.frames_per_buffer()) { |
+ DCHECK_EQ(source_params.channels(), |
+ media::ChannelLayoutToChannelCount(media::CHANNEL_LAYOUT_STEREO)); |
tommi (sloooow) - chröme
2016/01/15 18:50:03
new assumption or did the previous code assume thi
Henrik Grunell
2016/01/18 13:00:54
Actually, the assumption is that source and sink o
|
+ const bool different_source_sink_frames = |
+ source_params.frames_per_buffer() != new_sink_params.frames_per_buffer(); |
+ if (different_source_sink_frames) { |
DVLOG(1) << "Rebuffering from " << source_params.frames_per_buffer() |
<< " to " << new_sink_params.frames_per_buffer(); |
- new_audio_fifo.reset(new media::AudioPullFifo( |
- source_params.channels(), source_params.frames_per_buffer(), |
- base::Bind(&WebRtcAudioRenderer::SourceCallback, |
- base::Unretained(this)))); |
- |
- if (new_sink_params.frames_per_buffer() > |
- source_params.frames_per_buffer()) { |
- int frame_duration_milliseconds = |
- base::Time::kMillisecondsPerSecond / |
- static_cast<double>(source_params.sample_rate()); |
- new_fifo_delay_milliseconds = (new_sink_params.frames_per_buffer() - |
- source_params.frames_per_buffer()) * |
- frame_duration_milliseconds; |
- } |
} |
{ |
base::AutoLock lock(lock_); |
+ const bool source_frames_changed = |
+ audio_fifo_->SizeInFrames() != source_params.frames_per_buffer(); |
+ if ((!audio_fifo_ && different_source_sink_frames) || |
+ (audio_fifo_ && source_frames_changed)) { |
+ audio_fifo_.reset(new media::AudioPullFifo( |
+ source_params.channels(), source_params.frames_per_buffer(), |
+ base::Bind(&WebRtcAudioRenderer::SourceCallback, |
+ base::Unretained(this)))); |
+ } |
sink_params_ = new_sink_params; |
- fifo_delay_milliseconds_ = new_fifo_delay_milliseconds; |
- if (new_audio_fifo.get()) |
- audio_fifo_ = std::move(new_audio_fifo); |
} |
sink_->Initialize(new_sink_params, this); |