Update smoothed bitrate.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 1327 matching lines @@
   // We give smoothed bitrate allocation to audio network adaptor as
   // the uplink bandwidth.
   // The probing spikes should not affect the bitrate smoother more than 25%.
   // To simplify the calculations we use a step response as input signal.
   // The step response of an exponential filter is
   // u(t) = 1 - e^(-t / time_constant).
   // In order to limit the affect of a BWE spike within 25% of its value before
   // the next probing, we would choose a time constant that fulfills
   // 1 - e^(-probing_interval_ms / time_constant) < 0.25
   // Then 4 * probing_interval_ms is a good choice.
+  rtc::CritScope lock(&bitrate_smoother_lock_);
   bitrate_smoother_.SetTimeConstantMs(probing_interval_ms * 4);

[ossu, 2016/12/20 13:58:09]

Not strictly part of this CL, but I'm interested: what happens when/if the time
constant changes? Will the smoothing be temporarily wrong, or will old values be
invalidated or will everything be fine? I'm mainly wondering why
SetTimeConstantMs is a separate call? It seems to be you'd want it as part of
AddSample, i.e. this is the value and it's at a point X ms after the previous
sample.

[michaelt, 2016/12/20 15:13:03]

The same filter is used in a different place where we don't adapt the time
constant. That is why we set it in a different function.

As far that i can see. Everything should be fine on a time constant change. The
filter will just adapt faster or slower to a value change.

   bitrate_smoother_.AddSample(bitrate_bps);
-  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
-    if (*encoder) {
-      (*encoder)->OnReceivedUplinkBandwidth(
-          static_cast<int>(*bitrate_smoother_.GetAverage()));
-    }
-  });
 }
 
 void Channel::OnIncomingFractionLoss(int fraction_lost) {
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
     if (*encoder)
       (*encoder)->OnReceivedUplinkPacketLossFraction(fraction_lost / 255.0f);
   });
 }
 
 int32_t Channel::SetVADStatus(bool enableVAD,
@@ skipping 1376 matching lines @@
   // If pacing is enabled we always store packets.
   if (!pacing_enabled_)
     _rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
   rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets);
   if (enable)
     audio_coding_->EnableNack(maxNumberOfPackets);
   else
     audio_coding_->DisableNack();
 }
 
+void Channel::AdaptCodec() {
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    rtc::CritScope lock(&bitrate_smoother_lock_);
+    if (*encoder) {
+      (*encoder)->OnReceivedUplinkBandwidth(
+          static_cast<int>(*bitrate_smoother_.GetAverage()));
+    }
+  });
+}
+
 // Called when we are missing one or more packets.
 int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
   return _rtpRtcpModule->SendNACK(sequence_numbers, length);
 }
 
 uint32_t Channel::Demultiplex(const AudioFrame& audioFrame) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Demultiplex()");
   _audioFrame.CopyFrom(audioFrame);
   _audioFrame.id_ = _channelId;
@@ skipping 517 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc