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. |