Support for remote audio into tracks

talk/app/webrtc/webrtcsession.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 12 matching lines @@
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "talk/app/webrtc/webrtcsession.h"
 
 #include <limits.h>
 
 #include <algorithm>
+#include <set>
+#include <utility>
 #include <vector>
-#include <set>
 
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/sctputils.h"
 #include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/videocapturer.h"
 #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"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/call.h"
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/p2p/base/transportchannel.h"
 
@@ skipping 1258 matching lines @@
   if (!voice_channel_) {
     LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
     return;
   }
 
   if (!voice_channel_->SetOutputVolume(ssrc, volume)) {
     ASSERT(false);
   }
 }
 
+void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
+                                    rtc::scoped_ptr<AudioSinkInterface> 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());
 
   if (!video_channel_) {
     // |video_channel_| doesnt't exist. Probably because the remote end doesnt't
     // support video.
     LOG(LS_WARNING) << "Video not used in this call.";
     return false;
   }
@@ skipping 854 matching lines @@
   }
 }
 
 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
                                    const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc