| Index: content/renderer/media/webrtc_local_audio_track.cc
|
| diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc
|
| index 37846335e8531270c5c6c66658c400137b82bcf4..c79757613fe192636f2048435f11b35c1414068e 100644
|
| --- a/content/renderer/media/webrtc_local_audio_track.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_track.cc
|
| @@ -156,12 +156,6 @@ void WebRtcLocalAudioTrack::Capture(media::AudioBus* audio_source,
|
| scoped_refptr<ConfiguredBuffer> current_buffer;
|
| {
|
| base::AutoLock auto_lock(lock_);
|
| - // When the track is disabled, we simply return here.
|
| - // TODO(xians): Figure out if we should feed zero to sinks instead, in
|
| - // order to inject VAD data in such case.
|
| - if (!enabled())
|
| - return;
|
| -
|
| capturer = capturer_;
|
| voe_channels = voe_channels_;
|
| current_buffer = buffer_;
|
| @@ -173,13 +167,17 @@ void WebRtcLocalAudioTrack::Capture(media::AudioBus* audio_source,
|
|
|
| // Push the data to the fifo.
|
| current_buffer->Push(audio_source);
|
| - // Only turn off the audio processing when the constrain is set to false as
|
| + // Only turn off the audio processing when the constraint is set to false as
|
| // well as there is no correct delay value.
|
| bool need_audio_processing = need_audio_processing_ ?
|
| need_audio_processing_ : (audio_delay_milliseconds != 0);
|
| int current_volume = volume;
|
| while (current_buffer->Consume()) {
|
| // Feed the data to the sinks.
|
| + // TODO (jiayl): we should not pass the real audio data down if the track is
|
| + // disabled. This is currently done so to feed input to WebRTC typing
|
| + // detection and should be changed when audio processing is moved from
|
| + // WebRTC to the track.
|
| for (SinkList::const_iterator it = sinks.begin(); it != sinks.end(); ++it) {
|
| int new_volume = (*it)->CaptureData(voe_channels,
|
| current_buffer->buffer(),
|
|
|