| 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 b10c6782d40dab712fd893b3418aa71288a2bb63..0b06c11c2907dcb38bdb7f4032f8846450fa1580 100644
|
| --- a/content/renderer/media/webrtc_local_audio_track.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_track.cc
|
| @@ -5,6 +5,7 @@
|
| #include "content/renderer/media/webrtc_local_audio_track.h"
|
|
|
| #include "content/public/renderer/media_stream_audio_sink.h"
|
| +#include "content/renderer/media/media_stream_audio_level_calculator.h"
|
| #include "content/renderer/media/media_stream_audio_sink_owner.h"
|
| #include "content/renderer/media/media_stream_audio_track_sink.h"
|
| #include "content/renderer/media/peer_connection_audio_sink_owner.h"
|
| @@ -46,6 +47,13 @@ void WebRtcLocalAudioTrack::Capture(const int16* audio_data,
|
| bool key_pressed,
|
| bool need_audio_processing) {
|
| DCHECK(capture_thread_checker_.CalledOnValidThread());
|
| +
|
| + // Calculate the signal level regardless if the track is disabled or enabled.
|
| + int signal_level = level_calculator_->Calculate(
|
| + audio_data, audio_parameters_.channels(),
|
| + audio_parameters_.frames_per_buffer());
|
| + adapter_->SetSignalLevel(signal_level);
|
| +
|
| scoped_refptr<WebRtcAudioCapturer> capturer;
|
| SinkList::ItemList sinks;
|
| SinkList::ItemList sinks_to_notify_format;
|
| @@ -98,6 +106,7 @@ void WebRtcLocalAudioTrack::OnSetFormat(
|
| DCHECK(capture_thread_checker_.CalledOnValidThread());
|
|
|
| audio_parameters_ = params;
|
| + level_calculator_.reset(new MediaStreamAudioLevelCalculator());
|
|
|
| base::AutoLock auto_lock(lock_);
|
| // Remember to notify all sinks of the new format.
|
|
|