Revert of Implemented the GetSources() in native code.

webrtc/pc/channel.h

 /*
  *  Copyright 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_PC_CHANNEL_H_
 #define WEBRTC_PC_CHANNEL_H_
 
 #include <map>
 #include <memory>
 #include <set>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/call/audio_sink.h"
-#include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/base/asyncinvoker.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/media/base/mediachannel.h"
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/base/videosinkinterface.h"
@@ skipping 452 matching lines @@
   webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const;
   bool SetRtpSendParameters(uint32_t ssrc,
                             const webrtc::RtpParameters& parameters);
   webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const;
   bool SetRtpReceiveParameters(uint32_t ssrc,
                                const webrtc::RtpParameters& parameters);
 
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
 
-  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const;
-
   // Monitoring functions
   sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
       SignalConnectionMonitor;
 
   void StartMediaMonitor(int cms);
   void StopMediaMonitor();
   sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
 
   void StartAudioMonitor(int cms);
   void StopAudioMonitor();
@@ skipping 21 matching lines @@
   const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
   bool SetLocalContent_w(const MediaContentDescription* content,
                          ContentAction action,
                          std::string* error_desc) override;
   bool SetRemoteContent_w(const MediaContentDescription* content,
                           ContentAction action,
                           std::string* error_desc) override;
   void HandleEarlyMediaTimeout();
   bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
   bool SetOutputVolume_w(uint32_t ssrc, double volume);
+  bool GetStats_w(VoiceMediaInfo* stats);
 
   void OnMessage(rtc::Message* pmsg) override;
   void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
   void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) override;
   void OnMediaMonitorUpdate(VoiceMediaChannel* media_channel,
                             const VoiceMediaInfo& info);
   void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
 
@@ skipping 199 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_