Support for remote audio into tracks

talk/media/base/mediachannel.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 13 matching lines @@
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #ifndef TALK_MEDIA_BASE_MEDIACHANNEL_H_
 #define TALK_MEDIA_BASE_MEDIACHANNEL_H_
 
 #include <string>
 #include <vector>
 
+#include "talk/media/base/audiorenderer.h"

[perkj_webrtc, 2015/12/13 19:26:48]

nit , no need since it is still used as a pointer.  

[tommi, 2015/12/13 19:46:33]

Done.

 #include "talk/media/base/codec.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/streamparams.h"
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/window.h"
 // TODO(juberti): re-evaluate this include
 #include "talk/session/media/audiomonitor.h"
 
 namespace rtc {
 class Buffer;
 class RateLimiter;
 class Timing;
 }
 
 namespace cricket {
 
-class AudioRenderer;
 struct RtpHeader;
 class ScreencastId;
 struct VideoFormat;
 class VideoCapturer;
 class VideoRenderer;
 
 const int kMinRtpHeaderExtensionId = 1;
 const int kMaxRtpHeaderExtensionId = 255;
 const int kScreencastDefaultFps = 5;
 
@@ skipping 954 matching lines @@
   virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0;
   // Returns if the telephone-event has been negotiated.
   virtual bool CanInsertDtmf() = 0;
   // Send a DTMF |event|. The DTMF out-of-band signal will be used.
   // The |ssrc| should be either 0 or a valid send stream ssrc.
   // The valid value for the |event| are 0 to 15 which corresponding to
   // DTMF event 0-9, *, #, A-D.
   virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
   // Gets quality stats for the channel.
   virtual bool GetStats(VoiceMediaInfo* info) = 0;
+
+  virtual void SetRawAudioSink(uint32_t ssrc,
+                               rtc::scoped_ptr<webrtc::AudioSink> sink) = 0;
 };
 
 struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> {
 };
 
 struct VideoRecvParameters : RtpParameters<VideoCodec> {
 };
 
 class VideoMediaChannel : public MediaChannel {
  public:
@@ skipping 167 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_BASE_MEDIACHANNEL_H_