Implemented the GetSources() in native code.

webrtc/api/rtpreceiverinterface.h

Index: webrtc/api/rtpreceiverinterface.h
diff --git a/webrtc/api/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h
index 8607d935a232be6364327ad0af4d966280ec65c4..25738ee72e821bf62f8584a26256a7b83a613d94 100644
--- a/webrtc/api/rtpreceiverinterface.h
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -14,7 +14,9 @@
 #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"
@@ -25,6 +27,30 @@
 
 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,
@@ -61,6 +87,13 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
   // 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() {}
 };
@@ -76,7 +109,8 @@ BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
   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