| Index: talk/media/webrtc/webrtcvoiceengine.cc
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
|
| index a10c6f8aec12330711daf19e9ad5fd1f3bae4f2b..4d74e915c51441b3a4d403531dc7a87eeaa877c6 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.cc
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.cc
|
| @@ -1330,10 +1330,6 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
|
| RTC_DCHECK(renderer_ == renderer);
|
| return;
|
| }
|
| -
|
| - // TODO(xians): Remove AddChannel() call after Chrome turns on APM
|
| - // in getUserMedia by default.
|
| - renderer->AddChannel(channel_);
|
| renderer->SetSink(this);
|
| renderer_ = renderer;
|
| }
|
| @@ -1343,12 +1339,10 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
|
| // This method is called on the libjingle worker thread.
|
| void Stop() {
|
| rtc::CritScope lock(&lock_);
|
| - if (renderer_ == NULL)
|
| - return;
|
| -
|
| - renderer_->RemoveChannel(channel_);
|
| - renderer_->SetSink(NULL);
|
| - renderer_ = NULL;
|
| + if (renderer_ != NULL) {
|
| + renderer_->SetSink(NULL);
|
| + renderer_ = NULL;
|
| + }
|
| }
|
|
|
| // AudioRenderer::Sink implementation.
|
|
|