Implemented the GetSources() in native code.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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 10 matching lines @@
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/audio_level.h"
 #include "webrtc/voice_engine/file_player.h"
 #include "webrtc/voice_engine/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/shared_data.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace rtc {
 class TimestampWrapAroundHandler;
 }
 
 namespace webrtc {
 
 class AudioDeviceModule;
 class FileWrapper;
 class PacketRouter;
 class ProcessThread;
 class RateLimiter;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
-class RtpReceiver;
 class RTPReceiverAudio;
 class RtpPacketReceived;
 class RtpRtcp;
 class RtpTransportControllerSendInterface;
 class TelephoneEventHandler;
 class VoERTPObserver;
 class VoiceEngineObserver;
 
 struct CallStatistics;
 struct ReportBlock;
@@ skipping 326 matching lines @@
   void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
 
   // The existence of this function alongside OnUplinkPacketLossRate is
   // a compromise. We want the encoder to be agnostic of the PLR source, but
   // we also don't want it to receive conflicting information from TWCC and
   // from RTCP-XR.
   void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
 
   void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate);
 
+  std::vector<RtpSource> GetSources() const {
+    return rtp_receiver_->GetSources();
+  }
+
  private:
   class ProcessAndEncodeAudioTask;
 
   void OnUplinkPacketLossRate(float packet_loss_rate);
   bool InputMute() const;
   bool OnRtpPacketWithHeader(const uint8_t* received_packet,
                              size_t length,
                              RTPHeader *header);
   bool ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
@@ skipping 124 matching lines @@
 
   const bool use_twcc_plr_for_ana_;
 
   rtc::TaskQueue* encoder_queue_ = nullptr;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_