Index: content/renderer/media/webrtc_local_audio_source_provider.cc |
diff --git a/content/renderer/media/webrtc_local_audio_source_provider.cc b/content/renderer/media/webrtc_local_audio_source_provider.cc |
index f18d8c54be2004f8ed85225779b99527f2044888..bc5f80dfe376a83193421afeb88c4f296aec7a5f 100644 |
--- a/content/renderer/media/webrtc_local_audio_source_provider.cc |
+++ b/content/renderer/media/webrtc_local_audio_source_provider.cc |
@@ -43,10 +43,10 @@ WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() { |
audio_converter_->RemoveInput(this); |
} |
-void WebRtcLocalAudioSourceProvider::SetCaptureFormat( |
+void WebRtcLocalAudioSourceProvider::OnSetFormat( |
const media::AudioParameters& params) { |
// We need detach the thread here because it will be a new capture thread |
- // calling SetCaptureFormat() and CaptureData() if the source is restarted. |
+ // calling OnSetFormat() and OnData() if the source is restarted. |
capture_thread_checker_.DetachFromThread(); |
DCHECK(capture_thread_checker_.CalledOnValidThread()); |
DCHECK(params.IsValid()); |
@@ -68,20 +68,15 @@ void WebRtcLocalAudioSourceProvider::SetCaptureFormat( |
params.frames_per_buffer()); |
} |
-int WebRtcLocalAudioSourceProvider::CaptureData( |
- const std::vector<int>& channels, |
+void WebRtcLocalAudioSourceProvider::OnData( |
const int16* audio_data, |
int sample_rate, |
int number_of_channels, |
- int number_of_frames, |
- int audio_delay_milliseconds, |
- int current_volume, |
- bool need_audio_processing, |
- bool key_pressed) { |
+ int number_of_frames) { |
DCHECK(capture_thread_checker_.CalledOnValidThread()); |
base::AutoLock auto_lock(lock_); |
if (!is_enabled_) |
- return 0; |
+ return; |
DCHECK(fifo_.get()); |
@@ -98,8 +93,6 @@ int WebRtcLocalAudioSourceProvider::CaptureData( |
// WebAudio stops consuming data. |
DLOG(WARNING) << "Local source provicer FIFO is full" << fifo_->frames(); |
} |
- |
- return 0; |
} |
void WebRtcLocalAudioSourceProvider::setClient( |