Remove unused transport code.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49389004
Cr-Commit-Position: refs/heads/master@{#8719}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8719 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index 1d0a887..6256b35 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -135,17 +135,6 @@
srtp_filter_.set_signal_silent_time(silent_time);
}
- void set_content_name(const std::string& content_name) {
- ASSERT(signaling_thread()->IsCurrent());
- ASSERT(!writable_);
- if (session_->state() != BaseSession::STATE_INIT) {
- LOG(LS_ERROR) << "Content name for a channel can be changed only "
- << "when BaseSession is in STATE_INIT state.";
- return;
- }
- content_name_ = content_name;
- }
-
template <class T>
void RegisterSendSink(T* sink,
void (T::*OnPacket)(const void*, size_t, bool),
@@ -374,7 +363,7 @@
std::vector<StreamParams> local_streams_;
std::vector<StreamParams> remote_streams_;
- std::string content_name_;
+ const std::string content_name_;
bool rtcp_;
TransportChannel* transport_channel_;
TransportChannel* rtcp_transport_channel_;
