| Index: content/renderer/media/webrtc/processed_local_audio_source.h
|
| diff --git a/content/renderer/media/webrtc/processed_local_audio_source.h b/content/renderer/media/webrtc/processed_local_audio_source.h
|
| index 3cbfe40371ee44e9654b59d602509b29bd14b325..5d2e38862bdf74547a7049065a462968e4415cae 100644
|
| --- a/content/renderer/media/webrtc/processed_local_audio_source.h
|
| +++ b/content/renderer/media/webrtc/processed_local_audio_source.h
|
| @@ -66,7 +66,6 @@ class CONTENT_EXPORT ProcessedLocalAudioSource final
|
|
|
| // The following accessors are not valid until after the source is started
|
| // (when the first track is connected).
|
| - webrtc::AudioSourceInterface* rtc_source() const { return rtc_source_.get(); }
|
| const scoped_refptr<MediaStreamAudioProcessor>& audio_processor() const {
|
| return audio_processor_;
|
| }
|
| @@ -127,10 +126,6 @@ class CONTENT_EXPORT ProcessedLocalAudioSource final
|
| // Lock used to ensure thread-safe access to |source_| by SetVolume().
|
| mutable base::Lock source_lock_;
|
|
|
| - // Holder for WebRTC audio pipeline objects. Created in
|
| - // EnsureSourceIsStarted().
|
| - scoped_refptr<webrtc::AudioSourceInterface> rtc_source_;
|
| -
|
| // Stores latest microphone volume received in a CaptureData() callback.
|
| // Range is [0, 255].
|
| base::subtle::Atomic32 volume_;
|
|
|