Change reporting of timing frames conditions in GetStats on receive side

webrtc/video/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 54 matching lines @@
   bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   void SetSync(Syncable* audio_syncable);
 
   // Implements webrtc::VideoReceiveStream.
   void Start() override;
   void Stop() override;
 
   webrtc::VideoReceiveStream::Stats GetStats() const override;
 
-  rtc::Optional<TimingFrameInfo> GetAndResetTimingFrameInfo() override;
-
   // 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.
   void EnableEncodedFrameRecording(rtc::PlatformFile file,
                                    size_t byte_limit) override;
 
   void AddSecondarySink(RtpPacketSinkInterface* sink) override;
   void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override;
@@ skipping 62 matching lines @@
   // a decoding error) we require a keyframe to restart the stream.
   bool keyframe_required_ = true;
 
   // If we have successfully decoded any frame.
   bool frame_decoded_ = false;
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_