commit bd7d8f7e2b824a887aa12236cb6185d446d7da61
author deadbeef <deadbeef@webrtc.org> Sat Dec 19 00:58:44 2015
committer Commit bot <commit-bot@chromium.org> Sat Dec 19 00:58:51 2015
tree c9576bbe1d9274c9c7c9afde21d81d041363cf34
parent 92594a30ce02aed75f8a2a9f21e5b8c5c4e5f199

Adding a MediaStream parameter to createSender.

This will allow an app to create senders with the same stream id,
without SDP munging.

Review URL: https://codereview.webrtc.org/1538673002

Cr-Commit-Position: refs/heads/master@{#11092}

talk/app/webrtc/peerconnection.cc

diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 85e03f9..617eb15 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -813,7 +813,8 @@
 }
 
 rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
-    const std::string& kind) {
+    const std::string& kind,
+    const std::string& stream_id) {
   TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
   RtpSenderInterface* new_sender;
   if (kind == MediaStreamTrackInterface::kAudioKind) {
@@ -824,6 +825,9 @@
     LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
     return rtc::scoped_refptr<RtpSenderInterface>();
   }
+  if (!stream_id.empty()) {
+    new_sender->set_stream_id(stream_id);
+  }
   senders_.push_back(new_sender);
   return RtpSenderProxy::Create(signaling_thread(), new_sender);
 }