Change reporting of timing frames conditions in GetStats on receive side

webrtc/call/video_receive_stream.h

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 76 matching lines @@
     int width = 0;
     int height = 0;
 
     int sync_offset_ms = std::numeric_limits<int>::max();
 
     uint32_t ssrc = 0;
     std::string c_name;
     StreamDataCounters rtp_stats;
     RtcpPacketTypeCounter rtcp_packet_type_counts;
     RtcpStatistics rtcp_stats;
+
+    // Timing frame info: all important timestamps for a full lifetime of a
+    // single 'timing frame'.
+    rtc::Optional<webrtc::TimingFrameInfo> timing_frame_info;
   };
 
   struct Config {
    private:
     // Access to the copy constructor is private to force use of the Copy()
     // method for those exceptional cases where we do use it.
     Config(const Config&);
 
    public:
     Config() = delete;
@@ skipping 107 matching lines @@
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
 
   // TODO(pbos): Add info on currently-received codec to Stats.
   virtual Stats GetStats() const = 0;
 
-  virtual rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() = 0;
-
   // Takes ownership of the file, is responsible for closing it later.
   // Calling this method will close and finalize any current log.
   // Giving rtc::kInvalidPlatformFileValue disables logging.
   // If a frame to be written would make the log too large the write fails and
   // the log is closed and finalized. A |byte_limit| of 0 means no limit.
   virtual void EnableEncodedFrameRecording(rtc::PlatformFile file,
                                            size_t byte_limit) = 0;
   inline void DisableEncodedFrameRecording() {
     EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0);
   }
 
   // RtpDemuxer only forwards a given RTP packet to one sink. However, some
   // sinks, such as FlexFEC, might wish to be informed of all of the packets
   // a given sink receives (or any set of sinks). They may do so by registering
   // themselves as secondary sinks.
   virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0;
   virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0;
 
  protected:
   virtual ~VideoReceiveStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_