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..bd2a4a3c5e8576f98778b1e180b68ffcb8a394c5 100644 |
--- a/content/renderer/media/webrtc_audio_renderer.cc |
+++ b/content/renderer/media/webrtc_audio_renderer.cc |
@@ -39,6 +39,13 @@ namespace { |
// between each callback leads to one UMA update each 100ms. |
const int kNumCallbacksBetweenRenderTimeHistograms = 10; |
+// Audio parameters that don't change. |
+const media::AudioParameters::Format kFormat = |
+ media::AudioParameters::AUDIO_PCM_LOW_LATENCY; |
+const media::ChannelLayout kChannelLayout = media::CHANNEL_LAYOUT_STEREO; |
+const int kChannels = 2; |
+const int kBitsPerSample = 16; |
+ |
// This is a simple wrapper class that's handed out to users of a shared |
// WebRtcAudioRenderer instance. This class maintains the per-user 'playing' |
// and 'started' states to avoid problems related to incorrect usage which |
@@ -187,12 +194,7 @@ 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, |
- 16, |
- 0), |
+ sink_params_(kFormat, kChannelLayout, 0, kBitsPerSample, 0), |
output_device_id_(device_id), |
security_origin_(security_origin), |
render_callback_count_(0) { |
@@ -434,6 +436,28 @@ 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. We always |
+ // pull 10 ms of data from the source (see PrepareSink()), so the fifo is only |
+ // required if the number of frames to drop doesn't correspond to 10 ms. |
+ if (frames_skipped > 0) { |
+ const uint32_t source_frames_per_buffer = |
+ static_cast<uint32_t>(sink_params_.sample_rate() / 100); |
+ if (!audio_fifo_ && frames_skipped != source_frames_per_buffer) { |
+ audio_fifo_.reset(new media::AudioPullFifo( |
+ kChannels, source_frames_per_buffer, |
+ 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 will deliver. |
if (audio_fifo_) |
audio_fifo_->Consume(audio_bus, audio_bus->frames()); |
else |
@@ -457,7 +481,9 @@ 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_|, |
@@ -606,55 +632,38 @@ void WebRtcAudioRenderer::PrepareSink() { |
UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected", sample_rate); |
} |
- // Set up audio parameters for the source, i.e., the WebRTC client. |
+ // Calculate the frames per buffer for the source, i.e. the WebRTC client. We |
+ // use 10 ms of data since the WebRTC client only supports multiples of 10 ms |
+ // as buffer size where 10 ms is preferred for lowest possible delay. |
+ const int source_frames_per_buffer = (sample_rate / 100); |
+ DVLOG(1) << "Using WebRTC output buffer size: " << source_frames_per_buffer; |
- // The WebRTC client only supports multiples of 10ms as buffer size where |
- // 10ms is preferred for lowest possible delay. |
- const int frames_per_10ms = (sample_rate / 100); |
- DVLOG(1) << "Using WebRTC output buffer size: " << frames_per_10ms; |
- media::AudioParameters source_params( |
- media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
- new_sink_params.channel_layout(), sample_rate, 16, frames_per_10ms); |
- source_params.set_channels_for_discrete(new_sink_params.channels()); |
- |
- const int frames_per_buffer = GetOptimalBufferSize( |
+ // Setup sink parameters. |
+ const int sink_frames_per_buffer = GetOptimalBufferSize( |
sample_rate, sink_->GetOutputParameters().frames_per_buffer()); |
- |
- new_sink_params.Reset( |
- new_sink_params.format(), new_sink_params.channel_layout(), |
- sample_rate, 16, frames_per_buffer); |
+ new_sink_params.set_sample_rate(sample_rate); |
+ new_sink_params.set_frames_per_buffer(sink_frames_per_buffer); |
// 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()) { |
- 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; |
- } |
+ const bool different_source_sink_frames = |
+ source_frames_per_buffer != new_sink_params.frames_per_buffer(); |
+ if (different_source_sink_frames) { |
+ DVLOG(1) << "Rebuffering from " << source_frames_per_buffer << " to " |
+ << new_sink_params.frames_per_buffer(); |
} |
- |
{ |
base::AutoLock lock(lock_); |
+ if ((!audio_fifo_ && different_source_sink_frames) || |
+ (audio_fifo_ && |
o1ka
2016/01/22 13:15:03
So in certain cases an existing fifo is reset. Doe
Henrik Grunell
2016/01/25 11:56:23
[We talked about this offline, summarizing here.]
|
+ audio_fifo_->SizeInFrames() != source_frames_per_buffer)) { |
+ audio_fifo_.reset(new media::AudioPullFifo( |
+ kChannels, source_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); |