Implemented the GetSources() in native code.

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

 /*
  *  Copyright (c) 2013 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_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 
 #include <memory>
+#include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
+// The maximum size of the ring buffer of the RtpContributingSource objects.
+static const size_t kContributingSourcesBufferSize = 500;

[hbos, 2017/03/30 09:51:54]

What is the reason for choosing 500? Is this arbitrary?

[Zhi Huang, 2017/03/31 06:44:04]

I'm not sure what is right size of this. It is expected to be larger than the
packet_per_second * 10 seconds. Then the question is what is the typical packet
rate?

+
 class RtpReceiverImpl : public RtpReceiver {
  public:
   // Callbacks passed in here may not be NULL (use Null Object callbacks if you
   // want callbacks to do nothing). This class takes ownership of the media
   // receiver but nothing else.
   RtpReceiverImpl(Clock* clock,
                   RtpFeedback* incoming_messages_callback,
                   RTPPayloadRegistry* rtp_payload_registry,
                   RTPReceiverStrategy* rtp_media_receiver);
 
@@ skipping 15 matching lines @@
   bool LastReceivedTimeMs(int64_t* receive_time_ms) const override;
 
   uint32_t SSRC() const override;
 
   int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override;
 
   int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
 
   TelephoneEventHandler* GetTelephoneEventHandler() override;
 
+  const std::vector<RtpContributingSource*>& GetContributingSources() override;
+
  private:
   bool HaveReceivedFrame() const;
 
   void CheckSSRCChanged(const RTPHeader& rtp_header);
   void CheckCSRC(const WebRtcRTPHeader& rtp_header);
   int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
                               const int8_t first_payload_byte,
                               bool* is_red,
                               PayloadUnion* payload);
 
+  void UpdateContributingSource();
+
   Clock* clock_;
   RTPPayloadRegistry* rtp_payload_registry_;
   std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
 
   RtpFeedback* cb_rtp_feedback_;
 
   rtc::CriticalSection critical_section_rtp_receiver_;
   int64_t last_receive_time_;
   size_t last_received_payload_length_;
 
   // SSRCs.
   uint32_t ssrc_;
   uint8_t num_csrcs_;
   uint32_t current_remote_csrc_[kRtpCsrcSize];
 
   uint32_t last_received_timestamp_;
   int64_t last_received_frame_time_ms_;
   uint16_t last_received_sequence_number_;
+
+  // Contributing Sources.
+  std::vector<RtpContributingSource*> contributing_sources_;
+  RtpContributingSource
+      contributing_sources_buffer_[kContributingSourcesBufferSize];
+  // The contribuing source that uses the |ssrc_|.
+  std::unique_ptr<RtpContributingSource> ssrc_source_;
+
+  size_t current_buffer_index_;
+  size_t current_buffer_size_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_