Fix unit for logged bitrates at the end of a call.

webrtc/call/call.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.
  */
 
@@ skipping 351 matching lines @@
   }
 }
 
 void Call::UpdateReceiveHistograms() {
   const int kMinRequiredPeriodicSamples = 5;
   AggregatedStats video_bytes_per_sec =
       received_video_bytes_per_second_counter_.GetStats();
   if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
                                 video_bytes_per_sec.average * 8 / 1000);
-    LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInKbps, "
+    LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBytesPerSec, "

[stefan-webrtc, 2016/11/16 08:05:34]

Why bytes per sec? I think it makes more sense to use bits or kbits?

                  << video_bytes_per_sec.ToString();
   }
   AggregatedStats audio_bytes_per_sec =
       received_audio_bytes_per_second_counter_.GetStats();
   if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
                                 audio_bytes_per_sec.average * 8 / 1000);
-    LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInKbps, "
+    LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInBytesPerSec, "
                  << audio_bytes_per_sec.ToString();
   }
   AggregatedStats rtcp_bytes_per_sec =
       received_rtcp_bytes_per_second_counter_.GetStats();
   if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
                                 rtcp_bytes_per_sec.average * 8);
-    LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBps, "
+    LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBytesPerSec, "
                  << rtcp_bytes_per_sec.ToString();
   }
   AggregatedStats recv_bytes_per_sec =
       received_bytes_per_second_counter_.GetStats();
   if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
                                 recv_bytes_per_sec.average * 8 / 1000);
-    LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInKbps, "
+    LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBytesPerSec, "
                  << recv_bytes_per_sec.ToString();
   }
 }
 
 PacketReceiver* Call::Receiver() {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   return this;
 }
@@ skipping 692 matching lines @@
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
   ReadLockScoped read_lock(*receive_crit_);
   auto it = video_receive_ssrcs_.find(ssrc);
   if (it == video_receive_ssrcs_.end())
     return false;
   return it->second->OnRecoveredPacket(packet, length);
 }
 
 }  // namespace internal
 }  // namespace webrtc