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 <memory>
 #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 {
 
+struct RtpContributingSource {

[the sun, 2017/04/04 21:09:59]

I'd suggest instead:

class RtpContributingSource {
 public:
  RtpContributingSource() = delete;
  RtpContributingSource(int64_t timestamp, uint32_t source)
      : timestamp_(timestamp), source_(source) {}

  int64_t timestamp() const { return timestamp_; }
  void update_timestamp(int64_t timestamp) {
    RTC_DCHECK_LE(timestamp_, timestamp);
    timestamp_ = timestamp;
  }

  uint32_t source() const { return source_; }

  // 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_;
  uint32_t source_;
};

1. We shouldn't lug around the extra weight of two optionals we're not using.
2. Adding the ctor arguments allows us to use emplace_back() with the
containers, and enforces the invariant of the class - you *can't* construct it
without giving it a timestamp and source.

[Zhi Huang, 2017/04/05 04:16:04]

The working group had some discussions and achieved an agreement on what the api
should look like. I've made some changes based on your suggestion.

+ public:
+  RtpContributingSource() : timestamp_(0), ssrc_(0) {}
+
+  int64_t timestamp() { return timestamp_; }
+  void set_timestamp(int64_t timestamp) { timestamp_ = timestamp; }
+
+  uint32_t source() { return csrc_ ? *csrc_ : ssrc_; }
+
+  uint32_t ssrc() { return ssrc_; }
+  void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; }
+  void set_csrc(uint32_t csrc) { csrc_ = rtc::Optional<uint32_t>(csrc); }
+
+  // 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() { return audio_level_; }
+
+ private:
+  int64_t timestamp_;
+  uint32_t ssrc_;
+  rtc::Optional<uint32_t> csrc_;
+  rtc::Optional<int8_t> audio_level_;
+};
+
 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<RtpContributingSource> GetContributingSources() {
+    return std::vector<RtpContributingSource>();
+  }
+
  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_METHOD0(std::vector<RtpContributingSource>, GetContributingSources);
+  END_PROXY_MAP()
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_RTPRECEIVERINTERFACE_H_