Smooth BWE and pass it to Audio Network Adaptor.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 3d195eb141aaf47af41f7d9bcdfbf5fcfb7a67b9..5712332aa618873b45c19bf46d80bbf883afd287 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -48,6 +48,7 @@ namespace {
 
 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
 constexpr int64_t kMinRetransmissionWindowMs = 30;
+constexpr int64_t kDefaultBitrateSmoothingTimeConstantMs = 20000;
 
 }  // namespace
 
@@ -895,7 +896,9 @@ Channel::Channel(int32_t channelId,
       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);
@@ -1309,6 +1312,16 @@ void Channel::SetBitRate(int bitrate_bps) {
       (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps);
   });
   retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
+
+  // We give smoothed bitrate allocation to audio network adaptor as
+  // the uplink bandwidth.
+  bitrate_bps_smoothed_.AddSample(bitrate_bps);
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (*encoder) {
+      (*encoder)->OnReceivedUplinkBandwidth(
+          static_cast<int>(*bitrate_bps_smoothed_.GetAverage()));
+    }
+  });
 }
 
 void Channel::OnIncomingFractionLoss(int fraction_lost) {