Index: content/renderer/media/webrtc_audio_capturer.cc |
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc |
index 561965ebd803b603639085b9a0f0e2304b8a6777..4fe754c18349786e2203f5192ee360c3e40e992f 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.cc |
+++ b/content/renderer/media/webrtc_audio_capturer.cc |
@@ -297,7 +297,8 @@ void WebRtcAudioCapturer::SetCapturerSource( |
channel_layout, 0, sample_rate, |
16, buffer_size, effects); |
scoped_refptr<MediaStreamAudioProcessor> new_audio_processor( |
- new MediaStreamAudioProcessor(params, constraints, effects)); |
+ new MediaStreamAudioProcessor(params, constraints, effects, |
+ audio_device_)); |
{ |
base::AutoLock auto_lock(lock_); |
audio_processor_ = new_audio_processor; |
@@ -556,27 +557,6 @@ void WebRtcAudioCapturer::GetAudioProcessingParams( |
*key_pressed = key_pressed_; |
} |
-void WebRtcAudioCapturer::FeedRenderDataToAudioProcessor( |
- const int16* render_audio, |
- int sample_rate, |
- int number_of_channels, |
- int number_of_frames, |
- base::TimeDelta render_delay) { |
- scoped_refptr<MediaStreamAudioProcessor> audio_processor; |
- { |
- base::AutoLock auto_lock(lock_); |
- if (!running_) |
- return; |
- |
- audio_processor = audio_processor_; |
- } |
- |
- audio_processor->PushRenderData(render_audio, sample_rate, |
- number_of_channels, |
- number_of_frames, |
- render_delay); |
-} |
- |
void WebRtcAudioCapturer::SetCapturerSourceForTesting( |
const scoped_refptr<media::AudioCapturerSource>& source, |
media::AudioParameters params) { |