Build WebRTC with data channel only.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 
 #include <algorithm>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/rtpparameters.h"
+#include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/window.h"
@@ skipping 973 matching lines @@
   // 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,
       std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
+
+  virtual std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const = 0;
 };
 
 // TODO(deadbeef): Rename to VideoSenderParameters, since they're intended to
 // encapsulate all the parameters needed for a video RtpSender.
 struct VideoSendParameters : RtpSendParameters<VideoCodec> {
   // Use conference mode? This flag comes from the remote
   // description's SDP line 'a=x-google-flag:conference', copied over
   // by VideoChannel::SetRemoteContent_w, and ultimately used by
   // conference mode screencast logic in
   // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig.
@@ skipping 181 matching lines @@
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_