Chromium Code Reviews| Index: talk/app/webrtc/rtpsender.cc |
| diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc |
| index ea10b7b33f9b4e6ef3f92c8e5eab6124b2ac816b..c0d23a0503cc08fb5a7fd8b301f8bd99bf076bd8 100644 |
| --- a/talk/app/webrtc/rtpsender.cc |
| +++ b/talk/app/webrtc/rtpsender.cc |
| @@ -184,7 +184,8 @@ void AudioRtpSender::Stop() { |
| void AudioRtpSender::SetAudioSend() { |
| RTC_DCHECK(!stopped_ && can_send_track()); |
| cricket::AudioOptions options; |
| - if (track_->enabled() && track_->GetSource()) { |
| + if (track_->enabled() && track_->GetSource() && |
| + !track_->GetSource()->remote()) { |
| // TODO(xians): Remove this static_cast since we should be able to connect |
|
perkj_webrtc
2015/12/15 10:15:31
Should we rephrase this todo to say that we should
tommi
2015/12/15 11:00:54
I'd like to keep the TODO untouched for now so tha
|
| // a remote audio track to a peer connection. |
| options = static_cast<LocalAudioSource*>(track_->GetSource())->options(); |