| 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 d499233a4b2950cbe14f129beee56c53ed68e5bd..99ada98099fa3163d58fbf46ed0185fe838b8f39 100644
|
| --- a/content/renderer/media/webrtc_local_audio_track.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_track.cc
|
| @@ -42,13 +42,14 @@ void WebRtcLocalAudioTrack::Capture(const int16* audio_data,
|
| base::TimeDelta delay,
|
| int volume,
|
| bool key_pressed,
|
| - bool need_audio_processing) {
|
| + bool need_audio_processing,
|
| + bool force_report_nonzero_energy) {
|
| 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());
|
| + audio_parameters_.frames_per_buffer(), force_report_nonzero_energy);
|
| adapter_->SetSignalLevel(signal_level);
|
|
|
| scoped_refptr<WebRtcAudioCapturer> capturer;
|
|
|