Smooth BWE and pass it to Audio Network Adaptor.

BUG=webrtc:6443, webrtc:6303

Review-Url: https://codereview.webrtc.org/2503713003
Cr-Commit-Position: refs/heads/master@{#15257}

webrtc/common_audio/mocks/mock_smoothing_filter.h

@@ -20,6 +20,7 @@ class MockSmoothingFilter : public SmoothingFilter {
  public:
   MOCK_METHOD1(AddSample, void(float));
   MOCK_CONST_METHOD0(GetAverage, rtc::Optional<float>());
+  MOCK_METHOD1(SetTimeConstantMs, void(int));
 };
 
 }  // namespace webrtc

webrtc/common_audio/smoothing_filter.cc

@@ -59,4 +59,9 @@ rtc::Optional<float> SmoothingFilterImpl::GetAverage() const {
                                                   : rtc::Optional<float>(value);
 }
 
+void SmoothingFilterImpl::SetTimeConstantMs(int time_constant_ms) {
+  time_constant_ms_ = time_constant_ms;
+  filter_.UpdateBase(exp(1.0f / time_constant_ms_));
+}
+
 }  // namespace webrtc

webrtc/common_audio/smoothing_filter.h

@@ -23,6 +23,7 @@ class SmoothingFilter {
   virtual ~SmoothingFilter() = default;
   virtual void AddSample(float sample) = 0;
   virtual rtc::Optional<float> GetAverage() const = 0;
+  virtual void SetTimeConstantMs(int time_constant_ms) = 0;
 };
 
 // SmoothingFilterImpl applies an exponential filter
@@ -35,9 +36,10 @@ class SmoothingFilterImpl final : public SmoothingFilter {
 
   void AddSample(float sample) override;
   rtc::Optional<float> GetAverage() const override;
+  void SetTimeConstantMs(int time_constant_ms) override;
 
  private:
-  const int time_constant_ms_;
+  int time_constant_ms_;
   const Clock* const clock_;
 
   bool first_sample_received_;

webrtc/voice_engine/channel.cc

@@ -894,7 +894,10 @@ 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 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()) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config(config.acm_config);
@@ -1306,6 +1309,20 @@ 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.
+  // 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_.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) {

webrtc/voice_engine/channel.h

@@ -17,6 +17,7 @@
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/common_audio/smoothing_filter.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
@@ -543,6 +544,8 @@ class Channel
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+
+  SmoothingFilterImpl bitrate_smoother_;
 };
 
 }  // namespace voe