Implemented the GetSources() in native code.

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 4b5524877c77206c58eb8db145ba0f88202b75c9..e4ec8f5d049d472a11db110d5de5128fdd4e4c1b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -12,6 +12,7 @@
 #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"
@@ -21,6 +22,9 @@
 
 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
@@ -56,6 +60,8 @@ class RtpReceiverImpl : public RtpReceiver {
 
   TelephoneEventHandler* GetTelephoneEventHandler() override;
 
+  const std::vector<RtpContributingSource*>& GetContributingSources() override;
+
  private:
   bool HaveReceivedFrame() const;
 
@@ -66,6 +72,8 @@ class RtpReceiverImpl : public RtpReceiver {
                               bool* is_red,
                               PayloadUnion* payload);
 
+  void UpdateContributingSource();
+
   Clock* clock_;
   RTPPayloadRegistry* rtp_payload_registry_;
   std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
@@ -84,6 +92,16 @@ class RtpReceiverImpl : public RtpReceiver {
   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_