| 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");
|
|
|