Implemented the GetSources() in native code.

webrtc/media/engine/webrtcvoiceengine.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.
  */
 
@@ skipping 192 matching lines @@
   void OnReadyToSend(bool ready) override;
   void OnTransportOverheadChanged(int transport_overhead_per_packet) override;
   bool GetStats(VoiceMediaInfo* info) override;
 
   // SSRC=0 will set the audio sink on the latest unsignaled stream, future or
   // current. Only one stream at a time will use the sink.
   void SetRawAudioSink(
       uint32_t ssrc,
       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
+  const std::vector<webrtc::RtpContributingSource*>& GetContributingSources(
+      uint32_t ssrc) override;
+
   // implements Transport interface
   bool SendRtp(const uint8_t* data,
                size_t len,
                const webrtc::PacketOptions& options) override {
     rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
     rtc::PacketOptions rtc_options;
     rtc_options.packet_id = options.packet_id;
     return VoiceMediaChannel::SendPacket(&packet, rtc_options);
   }
 
@@ skipping 65 matching lines @@
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_