Support for remote audio into tracks

talk/session/media/channel.h

 /*
  * libjingle
  * Copyright 2004 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 20 matching lines @@
 #include <string>
 #include <vector>
 #include <map>
 #include <set>
 #include <utility>
 
 #include "talk/media/base/mediachannel.h"
 #include "talk/media/base/mediaengine.h"
 #include "talk/media/base/streamparams.h"
 #include "talk/media/base/videocapturer.h"
-#include "webrtc/p2p/base/transportcontroller.h"
-#include "webrtc/p2p/client/socketmonitor.h"
 #include "talk/session/media/audiomonitor.h"
 #include "talk/session/media/bundlefilter.h"
 #include "talk/session/media/mediamonitor.h"
 #include "talk/session/media/mediasession.h"
 #include "talk/session/media/rtcpmuxfilter.h"
 #include "talk/session/media/srtpfilter.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/window.h"
+#include "webrtc/p2p/base/transportcontroller.h"
+#include "webrtc/p2p/client/socketmonitor.h"
+
+namespace webrtc {
+class AudioSinkInterface;
+}  // namespace webrtc
 
 namespace cricket {
 
 struct CryptoParams;
 class MediaContentDescription;
 struct ViewRequest;
 
 enum SinkType {
   SINK_PRE_CRYPTO,  // Sink packets before encryption or after decryption.
   SINK_POST_CRYPTO  // Sink packets after encryption or before decryption.
@@ skipping 296 matching lines @@
 
   // Returns if the telephone-event has been negotiated.
   bool CanInsertDtmf();
   // Send and/or play a DTMF |event| according to the |flags|.
   // The DTMF out-of-band signal will be used on sending.
   // The |ssrc| should be either 0 or a valid send stream ssrc.
   // The valid value for the |event| are 0 which corresponding to DTMF
   // event 0-9, *, #, A-D.
   bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
   bool SetOutputVolume(uint32_t ssrc, double volume);
+  void SetRawAudioSink(uint32_t ssrc,
+                       rtc::scoped_ptr<webrtc::AudioSinkInterface> sink);
+
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
 
   // Monitoring functions
   sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
       SignalConnectionMonitor;
 
   void StartMediaMonitor(int cms);
   void StopMediaMonitor();
   sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
@@ skipping 256 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_MEDIA_CHANNEL_H_