Implemented the GetSources() in native code.

webrtc/api/rtpreceiverinterface.h

 /*
  *  Copyright 2015 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.
  */
 
 // This file contains interfaces for RtpReceivers
 // http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
 
 #ifndef WEBRTC_API_RTPRECEIVERINTERFACE_H_
 #define WEBRTC_API_RTPRECEIVERINTERFACE_H_
 
 #include <string>
+#include <vector>
 
 #include "webrtc/api/mediatypes.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/proxy.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
+enum class RtpSourceType {
+  SSRC,
+  CSRC,
+};
+
+class RtpSource {
+ public:
+  RtpSource() = delete;
+  RtpSource(int64_t timestamp_ms, uint32_t source_id, RtpSourceType source_type)
+      : timestamp_ms_(timestamp_ms),
+        source_id_(source_id),
+        source_type_(source_type) {}
+
+  int64_t timestamp_ms() const { return timestamp_ms_; }
+  void update_timestamp_ms(int64_t timestamp_ms) {
+    RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
+    timestamp_ms_ = timestamp_ms;
+  }
+
+  // The identifier of the source can be the CSRC or the SSRC.
+  uint32_t source_id() const { return source_id_; }
+
+  // The source can be either a contributing source or a synchronization source.
+  RtpSourceType source_type() const { return source_type_; }
+
+  // This isn't implemented yet and will always return an empty Optional.
+  // TODO(zhihuang): Implement this to return real audio level.
+  rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); }
+
+ private:
+  int64_t timestamp_ms_;
+  uint32_t source_id_;
+  RtpSourceType source_type_;
+};
+
 class RtpReceiverObserverInterface {
  public:
   // Note: Currently if there are multiple RtpReceivers of the same media type,
   // they will all call OnFirstPacketReceived at once.
   //
   // In the future, it's likely that an RtpReceiver will only call
   // OnFirstPacketReceived when a packet is received specifically for its
   // SSRC/mid.
   virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
 
@@ skipping 16 matching lines @@
   // but this API also applies them to receivers, similar to ORTC:
   // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
   virtual RtpParameters GetParameters() const = 0;
   // Currently, doesn't support changing any parameters, but may in the future.
   virtual bool SetParameters(const RtpParameters& parameters) = 0;
 
   // Does not take ownership of observer.
   // Must call SetObserver(nullptr) before the observer is destroyed.
   virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
 
+  // TODO(zhihuang): Remove the default implementation once the subclasses
+  // implement this. Currently, the only relevant subclass is the
+  // content::FakeRtpReceiver in Chromium.
+  virtual std::vector<RtpSource> GetSources() const {
+    return std::vector<RtpSource>();
+  }
+
  protected:
   virtual ~RtpReceiverInterface() {}
 };
 
 // Define proxy for RtpReceiverInterface.
 // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
 // are called on is an implementation detail.
 BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
   PROXY_SIGNALING_THREAD_DESTRUCTOR()
   PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
   PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
   PROXY_CONSTMETHOD0(std::string, id)
   PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
   PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
   PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
-END_PROXY_MAP()
+  PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
+  END_PROXY_MAP()
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_RTPRECEIVERINTERFACE_H_