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..43965d325cd96daa5a365c1abef01927c7341fe4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -11,7 +11,10 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 
+#include <list>
 #include <memory>
+#include <unordered_map>
+#include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
@@ -56,6 +59,16 @@ class RtpReceiverImpl : public RtpReceiver {
 
   TelephoneEventHandler* GetTelephoneEventHandler() override;
 
+  std::vector<RtpSource> GetSources() const override;
+
+  const std::vector<RtpSource>& ssrc_sources_for_testing() const {
+    return ssrc_sources_;
+  }
+
+  const std::list<RtpSource>& csrc_sources_for_testing() const {
+    return csrc_sources_;
+  }
+
  private:
   bool HaveReceivedFrame() const;
 
@@ -66,6 +79,9 @@ class RtpReceiverImpl : public RtpReceiver {
                               bool* is_red,
                               PayloadUnion* payload);
 
+  void UpdateSources();
+  void RemoveOutdatedSources(int64_t now_ms);
+
   Clock* clock_;
   RTPPayloadRegistry* rtp_payload_registry_;
   std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
@@ -84,6 +100,12 @@ class RtpReceiverImpl : public RtpReceiver {
   uint32_t last_received_timestamp_;
   int64_t last_received_frame_time_ms_;
   uint16_t last_received_sequence_number_;
+
+  std::unordered_map<uint32_t, std::list<RtpSource>::iterator>
+      iterator_by_csrc_;
+  // The RtpSource objects are sorted chronologically.
+  std::list<RtpSource> csrc_sources_;
+  std::vector<RtpSource> ssrc_sources_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_