Revert of 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 fd233abe317609d265ca187f54be4781a1a47b75..8607d935a232be6364327ad0af4d966280ec65c4 100644
--- a/webrtc/api/rtpreceiverinterface.h
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -15,7 +15,6 @@
 #define WEBRTC_API_RTPRECEIVERINTERFACE_H_
 
 #include <string>
-#include <vector>
 
 #include "webrtc/api/mediatypes.h"
 #include "webrtc/api/mediastreaminterface.h"
@@ -25,41 +24,6 @@
 #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:
@@ -97,13 +61,6 @@
   // 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() {}
 };
@@ -119,8 +76,7 @@
   PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
   PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
   PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
-  PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
-  END_PROXY_MAP()
+END_PROXY_MAP()
 
 }  // namespace webrtc