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 908 matching lines @@
       _outputSpeechType(AudioFrame::kNormalSpeech),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.enable_voice_pacing),
       feedback_observer_proxy_(new TransportFeedbackProxy()),
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
       rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
       retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
                                                    kMaxRetransmissionWindowMs)),
-      decoder_factory_(config.acm_config.decoder_factory),
-      // Bitrate smoother can be initialized with arbitrary time constant
-      // (0 used here). The actual time constant will be set in SetBitRate.
-      bitrate_smoother_(0, Clock::GetRealTimeClock()) {
+      decoder_factory_(config.acm_config.decoder_factory) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config(config.acm_config);
   acm_config.id = VoEModuleId(instanceId, channelId);
   acm_config.neteq_config.enable_muted_state = true;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
@@ skipping 383 matching lines @@
     }
   }
 
   return 0;
 }
 
 void Channel::SetBitRate(int bitrate_bps, int64_t 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);
+    if (*encoder) {
+      (*encoder)->OnReceivedUplinkBandwidth(
+          bitrate_bps, rtc::Optional<int64_t>(probing_interval_ms));
+    }
   });
   retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
-
-  // 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.
-  bitrate_smoother_.SetTimeConstantMs(probing_interval_ms * 4);
-  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 1928 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