Calculate JitterBufferDelayInMs in the new jitter buffer.

webrtc/video/receive_statistics_proxy.cc

 /*
  *  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.
  */
 
 #include "webrtc/video/receive_statistics_proxy.h"
 
 #include <cmath>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 namespace {
 // Periodic time interval for processing samples for |freq_offset_counter_|.
 const int64_t kFreqOffsetProcessIntervalMs = 40000;
 }  // namespace
 
 ReceiveStatisticsProxy::ReceiveStatisticsProxy(
     const VideoReceiveStream::Config* config,
@@ skipping 57 matching lines @@
   if (qp != -1)
     RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
 
   // TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
   // not per frame. Change decode time to include every frame.
   const int kMinRequiredDecodeSamples = 5;
   int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
   if (decode_ms != -1)
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
 
-  int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
-  if (jb_delay_ms != -1) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
-                               jb_delay_ms);
+  if (field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") !=
+        "Enabled") {
+    int jb_delay_ms =
+        jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
+    if (jb_delay_ms != -1) {
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
+                                 jb_delay_ms);
+    }
   }
   int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (target_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
   }
   int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (current_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
                                current_delay_ms);
   }
@@ skipping 225 matching lines @@
   ++num_samples;
 }
 
 int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return sum / num_samples;
 }
 
 }  // namespace webrtc