Support for remote audio into tracks

talk/app/webrtc/webrtcsession.cc

Index: talk/app/webrtc/webrtcsession.cc
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index ec0456f7cb9f1f8743dabc4794f181353369f439..e7303dac505d76c4b458ac93d071229646f06a06 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -30,8 +30,9 @@
 #include <limits.h>
 
 #include <algorithm>
-#include <vector>
 #include <set>
+#include <utility>
+#include <vector>
 
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
@@ -44,6 +45,7 @@
 #include "talk/session/media/channel.h"
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/mediasession.h"
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/helpers.h"
@@ -1322,6 +1324,15 @@ void WebRtcSession::SetAudioPlayoutVolume(uint32_t ssrc, double volume) {
   }
 }
 
+void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
+                                    rtc::scoped_ptr<AudioSink> sink) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_)
+    return;
+
+  voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
+}
+
 bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,
                                      cricket::VideoCapturer* camera) {
   ASSERT(signaling_thread()->IsCurrent());