Smooth BWE and pass it to Audio Network Adaptor.

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 30 matching lines @@
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace voe {
 
 namespace {
 
 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
 constexpr int64_t kMinRetransmissionWindowMs = 30;
+constexpr int64_t kDefaultBitrateSmoothingTimeConstantMs = 20000;
 
 }  // namespace
 
 const int kTelephoneEventAttenuationdB = 10;
 
 class RtcEventLogProxy final : public webrtc::RtcEventLog {
  public:
   RtcEventLogProxy() : event_log_(nullptr) {}
 
   bool StartLogging(const std::string& file_name,
@@ skipping 827 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) {
+      decoder_factory_(config.acm_config.decoder_factory),
+      bitrate_bps_smoothed_(kDefaultBitrateSmoothingTimeConstantMs,
+                            Clock::GetRealTimeClock()) {
   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 393 matching lines @@
 }
 
 void Channel::SetBitRate(int bitrate_bps) {
   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);
+  bitrate_bps_smoothed_.AddSample(bitrate_bps);

[minyue-webrtc, 2016/11/15 14:36:03]

will it obtain time constant here? if so, does the default time constant matter
at all?

[michaelt, 2016/11/16 09:50:24]

Yes. 
Not really but we have to set it anyway.

+  OnUplinkBandwidthUpdated(
+      static_cast<int>(*bitrate_bps_smoothed_.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 1904 matching lines @@
   int64_t avg_rtt = 0;
   int64_t max_rtt = 0;
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
+void Channel::OnUplinkBandwidthUpdated(int bitrate_bps) {

[minyue-webrtc, 2016/11/15 14:36:03]

I don't think we need to add this function. put the call to encoder in
SetBitRate

[michaelt, 2016/11/16 09:50:24]

Ok

+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (*encoder) {
+      (*encoder)->OnReceivedUplinkBandwidth(bitrate_bps);
+    }
+  });
+}
+
 }  // namespace voe
 }  // namespace webrtc