Revert Make the new jitter buffer the default jitter buffer.

webrtc/modules/video_coding/frame_buffer2.h

 /*
  *  Copyright (c) 2016 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 10 matching lines @@
 #include "webrtc/base/event.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/video_coding/frame_object.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/modules/video_coding/inter_frame_delay.h"
 #include "webrtc/modules/video_coding/sequence_number_util.h"
 
 namespace webrtc {
 
 class Clock;
-class VCMReceiveStatisticsCallback;
 class VCMJitterEstimator;
 class VCMTiming;
 
 namespace video_coding {
 
 class FrameBuffer {
  public:
   enum ReturnReason { kFrameFound, kTimeout, kStopped };
 
   FrameBuffer(Clock* clock,
               VCMJitterEstimator* jitter_estimator,
-              VCMTiming* timing,
-              VCMReceiveStatisticsCallback* stats_proxy);
+              VCMTiming* timing);
 
   virtual ~FrameBuffer();
 
   // Insert a frame into the frame buffer. Returns the picture id
   // of the last continuous frame or -1 if there is no continuous frame.
   int InsertFrame(std::unique_ptr<FrameObject> frame);
 
   // Get the next frame for decoding. Will return at latest after
   // |max_wait_time_ms|.
   //  - If a frame is available within |max_wait_time_ms| it will return
@@ skipping 81 matching lines @@
 
   // Update the corresponding FrameInfo of |frame| and all FrameInfos that
   // |frame| references.
   // Return false if |frame| will never be decodable, true otherwise.
   bool UpdateFrameInfoWithIncomingFrame(const FrameObject& frame,
                                         FrameMap::iterator info)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   void UpdateJitterDelay() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
+  void UpdateHistograms() const;
+
   void ClearFramesAndHistory() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   FrameMap frames_ GUARDED_BY(crit_);
 
   rtc::CriticalSection crit_;
   Clock* const clock_;
   rtc::Event new_countinuous_frame_event_;
   VCMJitterEstimator* const jitter_estimator_ GUARDED_BY(crit_);
   VCMTiming* const timing_ GUARDED_BY(crit_);
   VCMInterFrameDelay inter_frame_delay_ GUARDED_BY(crit_);
   uint32_t last_decoded_frame_timestamp_ GUARDED_BY(crit_);
   FrameMap::iterator last_decoded_frame_it_ GUARDED_BY(crit_);
   FrameMap::iterator last_continuous_frame_it_ GUARDED_BY(crit_);
   FrameMap::iterator next_frame_it_ GUARDED_BY(crit_);
   int num_frames_history_ GUARDED_BY(crit_);
   int num_frames_buffered_ GUARDED_BY(crit_);
   bool stopped_ GUARDED_BY(crit_);
   VCMVideoProtection protection_mode_ GUARDED_BY(crit_);
-  VCMReceiveStatisticsCallback* const stats_callback_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameBuffer);
+
+  // For WebRTC.Video.JitterBufferDelayInMs metric.
+  int64_t accumulated_delay_ = 0;
+  int64_t accumulated_delay_samples_ = 0;
+
+  // For WebRTC.Video.KeyFramesReceivedInPermille metric.
+  int64_t num_total_frames_ = 0;
+  int64_t num_key_frames_ = 0;
 };
 
 }  // namespace video_coding
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER2_H_