Pass time constant to bwe smoothing filter.

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 1286 matching lines @@
 
   if (_rtpRtcpModule->SetAudioPacketSize(codec.pacsize) != 0) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                  "SetSendCodec() failed to set audio packet size");
     return -1;
   }
 
   return 0;
 }
 
-void Channel::SetBitRate(int bitrate_bps) {
+void Channel::SetBitRate(int bitrate_bps, int probing_interval_ms) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
     if (*encoder)
       (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps);
   });
   retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
 
   // We give smoothed bitrate allocation to audio network adaptor as
   // the uplink bandwidth.
-  // TODO(michaelt) : Remove kDefaultBitrateSmoothingTimeConstantMs as soon as
-  // we pass the probing interval to this function.
-  constexpr int64_t kDefaultBitrateSmoothingTimeConstantMs = 20000;
-  bitrate_smoother_.SetTimeConstantMs(kDefaultBitrateSmoothingTimeConstantMs);
+  bitrate_smoother_.SetTimeConstantMs(probing_interval_ms * 4);

[minyue-webrtc, 2016/11/22 10:14:25]

add some reason of choosing 4

[michaelt, 2016/11/22 16:29:00]

Done.

   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) {
@@ skipping 1924 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