| Index: content/renderer/media/webrtc_audio_capturer_sink_owner.cc
|
| diff --git a/content/renderer/media/webrtc_audio_capturer_sink_owner.cc b/content/renderer/media/webrtc_audio_capturer_sink_owner.cc
|
| index cbff31bd13784a9b1a3cb27951bf602e3d30a83c..ee9ed66f00a7c709a5e459839afd6c61d99bcd58 100644
|
| --- a/content/renderer/media/webrtc_audio_capturer_sink_owner.cc
|
| +++ b/content/renderer/media/webrtc_audio_capturer_sink_owner.cc
|
| @@ -11,29 +11,19 @@ WebRtcAudioCapturerSinkOwner::WebRtcAudioCapturerSinkOwner(
|
| : delegate_(sink) {
|
| }
|
|
|
| -int WebRtcAudioCapturerSinkOwner::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,
|
| - bool key_pressed) {
|
| +void WebRtcAudioCapturerSinkOwner::CaptureData(const std::vector<int>& channels,
|
| + const int16* audio_data,
|
| + int sample_rate,
|
| + int number_of_channels,
|
| + int number_of_frames) {
|
| base::AutoLock lock(lock_);
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| if (delegate_) {
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| - return delegate_->CaptureData(channels,
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| - audio_data,
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| - sample_rate,
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| - number_of_channels,
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| - number_of_frames,
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| - audio_delay_milliseconds,
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| - current_volume,
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| - need_audio_processing,
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| - key_pressed);
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| + delegate_->CaptureData(channels,
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| + audio_data,
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| + sample_rate,
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| + number_of_channels,
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| + number_of_frames);
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| }
|
| -
|
| - return 0;
|
| }
|
|
|
| void WebRtcAudioCapturerSinkOwner::SetCaptureFormat(
|
|
|