| Index: content/renderer/media/webrtc_local_audio_renderer.cc
|
| diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
|
| index 11c125f63748078254a0fbdba0b9c89c06d31ecb..af65d8d685e03045db35ed9f61b169affdaefe99 100644
|
| --- a/content/renderer/media/webrtc_local_audio_renderer.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_renderer.cc
|
| @@ -52,13 +52,14 @@ void WebRtcLocalAudioRenderer::OnRenderError() {
|
|
|
| // content::WebRtcAudioCapturerSink implementation
|
| int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels,
|
| - 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) {
|
| + 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) {
|
| TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
|
| base::AutoLock auto_lock(thread_lock_);
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|
|
|
|