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..fe1b5a9d0bf824dae4f9203f1f325425504ae806 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>

[the sun, 2017/04/05 14:58:29]

Why the need to add this?

[Zhi Huang, 2017/04/06 03:09:49]

Oh, I missed this when removing the unique_ptr.

 #include <string>
+#include <vector>
 
 #include "webrtc/api/mediatypes.h"
 #include "webrtc/api/mediastreaminterface.h"
@@ -25,6 +27,41 @@
 
 namespace webrtc {
 
+enum class RtpSourceType {

[hbos, 2017/04/05 15:20:16]

nit: With enum class you have to qualify the name with "RtpSourceType::" so it
would be enough to call these "SSRC" and "CSRC".

[Zhi Huang, 2017/04/06 03:09:49]

Done.

+  RTP_SSRC_SOURCE,
+  RTP_CSRC_SOURCE,
+};
+
+struct RtpSource {

[hbos, 2017/04/05 11:15:41]

Considering the getters/setters and private members, I'd make this class rather
than struct.

[the sun, 2017/04/05 14:58:29]

+1, make it a class!

Btw, thanks for not adding any more ways to modify the data than
update_timestamp() - this looks great!

[Zhi Huang, 2017/04/06 03:09:49]

Done.

+ public:
+  RtpSource() = delete;
+  RtpSource(int64_t timestamp, uint32_t source_id, RtpSourceType source_type)
+      : timestamp_(timestamp),
+        source_id_(source_id),
+        source_type_(source_type) {}
+
+  int64_t timestamp() const { return timestamp_; }

[danilchap, 2017/04/05 16:24:21]

can you add units, is it timestamp_ms?

[Zhi Huang, 2017/04/06 03:09:49]

Done.

+  void update_timestamp(int64_t timestamp) {
+    RTC_DCHECK_LE(timestamp_, timestamp);
+    timestamp_ = timestamp;
+  }
+
+  // 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_;
+  uint32_t source_id_;
+  RtpSourceType source_type_;
+};
+
 class RtpReceiverObserverInterface {
  public:
   // Note: Currently if there are multiple RtpReceivers of the same media type,
@@ -61,6 +98,13 @@ 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<RtpSource> GetSources() {

[the sun, 2017/04/05 14:58:29]

Make this method const:
virtual std::vector<RtpSource> GetSources() const {
  ...

[Zhi Huang, 2017/04/06 03:09:49]

Done.

+    return std::vector<RtpSource>();
+  }
+
  protected:
   virtual ~RtpReceiverInterface() {}
 };
@@ -76,7 +120,8 @@ 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<RtpSource>, GetSources);
+  END_PROXY_MAP()
 
 }  // namespace webrtc