Update talk to 58037405.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/5579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5267 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
index e0aff89..8579db2 100644
--- a/talk/session/media/channel.cc
+++ b/talk/session/media/channel.cc
@@ -587,11 +587,16 @@
int BaseChannel::SetOption(SocketType type, talk_base::Socket::Option opt,
int value) {
+ TransportChannel* channel = NULL;
switch (type) {
- case ST_RTP: return transport_channel_->SetOption(opt, value);
- case ST_RTCP: return rtcp_transport_channel_->SetOption(opt, value);
- default: return -1;
+ case ST_RTP:
+ channel = transport_channel_;
+ break;
+ case ST_RTCP:
+ channel = rtcp_transport_channel_;
+ break;
}
+ return channel ? channel->SetOption(opt, value) : -1;
}
void BaseChannel::OnWritableState(TransportChannel* channel) {
@@ -2152,7 +2157,9 @@
// Tweak our video processing settings, if needed.
VideoOptions video_options;
media_channel()->GetOptions(&video_options);
- video_options.conference_mode.Set(video->conference_mode());
+ if (video->conference_mode()) {
+ video_options.conference_mode.Set(true);
+ }
video_options.buffered_mode_latency.Set(video->buffered_mode_latency());
if (!media_channel()->SetOptions(video_options)) {