Index: talk/media/webrtc/webrtcvideoengine2.cc |
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc |
index bc303cd98a9c4ea835796ac9f1e783b71aa8894c..7754fedfc456bafc9df433018e47250e7b6bc957 100644 |
--- a/talk/media/webrtc/webrtcvideoengine2.cc |
+++ b/talk/media/webrtc/webrtcvideoengine2.cc |
@@ -1095,6 +1095,26 @@ bool WebRtcVideoChannel2::SetSend(bool send) { |
return true; |
} |
+bool WebRtcVideoChannel2::SetVideoSend(uint32 ssrc, bool mute, |
+ const VideoOptions* options) { |
+ LOG(LS_VERBOSE) << "SetVideoSend: " << ssrc << " -> " |
+ << (mute ? "mute" : "unmute"); |
+ DCHECK(ssrc != 0); |
+ rtc::CritScope stream_lock(&stream_crit_); |
+ if (send_streams_.find(ssrc) == send_streams_.end()) { |
+ LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; |
+ return false; |
+ } |
+ |
+ send_streams_[ssrc]->MuteStream(mute); |
+ |
+ if (!mute && options) { |
+ return SetOptions(*options); |
+ } else { |
+ return true; |
+ } |
+} |
+ |
bool WebRtcVideoChannel2::ValidateSendSsrcAvailability( |
const StreamParams& sp) const { |
for (uint32_t ssrc: sp.ssrcs) { |
@@ -1514,20 +1534,6 @@ void WebRtcVideoChannel2::OnReadyToSend(bool ready) { |
call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); |
} |
-bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) { |
- LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> " |
- << (mute ? "mute" : "unmute"); |
- DCHECK(ssrc != 0); |
- rtc::CritScope stream_lock(&stream_crit_); |
- if (send_streams_.find(ssrc) == send_streams_.end()) { |
- LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; |
- return false; |
- } |
- |
- send_streams_[ssrc]->MuteStream(mute); |
- return true; |
-} |
- |
bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions( |
const std::vector<RtpHeaderExtension>& extensions) { |
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions"); |