Update smoothed bitrate.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 #include <algorithm>
 #include <iterator>
 
 #include "webrtc/base/analytics/exp_filter.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 namespace {
 
 constexpr int kSampleRateHz = 48000;
 constexpr int kMinBitrateBps = 500;
 constexpr int kMaxBitrateBps = 512000;
 constexpr int kSupportedFrameLengths[] = {20, 60};
@@ skipping 130 matching lines @@
     // Within the hysteresis window; make no change.
     return rtc::Optional<int>();
   }
   return bitrate_bps <= complexity_threshold_bps
              ? rtc::Optional<int>(low_rate_complexity)
              : rtc::Optional<int>(complexity);
 }
 
 AudioEncoderOpus::AudioEncoderOpus(
     const Config& config,
-    AudioNetworkAdaptorCreator&& audio_network_adaptor_creator)
+    AudioNetworkAdaptorCreator&& audio_network_adaptor_creator,
+    std::unique_ptr<SmoothingFilter> bitrate_smoother)
     : packet_loss_rate_(0.0),
       inst_(nullptr),
       packet_loss_fraction_smoother_(new PacketLossFractionSmoother(
-          config.clock ? config.clock : Clock::GetRealTimeClock())),
+          config.clock)),
       audio_network_adaptor_creator_(
           audio_network_adaptor_creator
               ? std::move(audio_network_adaptor_creator)
               : [this](const std::string& config_string, const Clock* clock) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
                                                            clock);
-                }) {
+              }),
+      bitrate_smoother_(bitrate_smoother
+          ? std::move(bitrate_smoother) : std::unique_ptr<SmoothingFilter>(
+              // We choose 5sec as initial time constant due to empirical data.
+              new SmoothingFilterImpl(5000, config.clock))) {
   RTC_CHECK(RecreateEncoderInstance(config));
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst)
     : AudioEncoderOpus(CreateConfig(codec_inst), nullptr) {}
 
 AudioEncoderOpus::~AudioEncoderOpus() {
   RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
 }
 
@@ skipping 68 matching lines @@
     const std::string& config_string,
     const Clock* clock) {
   audio_network_adaptor_ = audio_network_adaptor_creator_(config_string, clock);
   return audio_network_adaptor_.get() != nullptr;
 }
 
 void AudioEncoderOpus::DisableAudioNetworkAdaptor() {
   audio_network_adaptor_.reset(nullptr);
 }
 
-void AudioEncoderOpus::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {
-  if (!audio_network_adaptor_)
-    return;
-  audio_network_adaptor_->SetUplinkBandwidth(uplink_bandwidth_bps);
-  ApplyAudioNetworkAdaptor();
-}
-
 void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
   if (!audio_network_adaptor_) {
     packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction);
     float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage();
     return SetProjectedPacketLossRate(average_fraction_loss);
   }
   audio_network_adaptor_->SetUplinkPacketLossFraction(
       uplink_packet_loss_fraction);
   ApplyAudioNetworkAdaptor();
 }
 
-void AudioEncoderOpus::OnReceivedTargetAudioBitrate(
-    int target_audio_bitrate_bps) {
+void AudioEncoderOpus::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    rtc::Optional<int64_t> probing_interval_ms) {
   if (audio_network_adaptor_) {
     audio_network_adaptor_->SetTargetAudioBitrate(target_audio_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.
+    if (probing_interval_ms)
+      bitrate_smoother_->SetTimeConstantMs(*probing_interval_ms * 4);
+    bitrate_smoother_->AddSample(target_audio_bitrate_bps);
+
     ApplyAudioNetworkAdaptor();
   } else if (webrtc::field_trial::FindFullName(
                  "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
     if (!overhead_bytes_per_packet_) {
       LOG(LS_INFO)
           << "AudioEncoderOpus: Overhead unknown, target audio bitrate "
           << target_audio_bitrate_bps << " bps is ignored.";
       return;
     }
     const int overhead_bps = static_cast<int>(
@@ skipping 39 matching lines @@
                         frame_length_ms <= max_frame_length_ms;
                });
   RTC_DCHECK(std::is_sorted(config_.supported_frame_lengths_ms.begin(),
                             config_.supported_frame_lengths_ms.end()));
 }
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
+  MaybeUpdateUplinkBandwidth();
 
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
   if (input_buffer_.size() <
       (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
   RTC_CHECK_EQ(input_buffer_.size(),
@@ skipping 147 matching lines @@
     const Clock* clock) const {
   AudioNetworkAdaptorImpl::Config config;
   config.clock = clock;
   return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
       config, ControllerManagerImpl::Create(
                   config_string, NumChannels(), supported_frame_lengths_ms(),
                   num_channels_to_encode_, next_frame_length_ms_,
                   GetTargetBitrate(), config_.fec_enabled, GetDtx(), clock)));
 }
 
+void AudioEncoderOpus::MaybeUpdateUplinkBandwidth() {
+  if (audio_network_adaptor_) {
+    int64_t now_ms = rtc::TimeMillis();
+    if (!bitrate_smoother_last_update_time_ ||
+        now_ms - *bitrate_smoother_last_update_time_ >=
+            config_.uplink_bandwidth_update_interval_ms) {
+      rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
+      if (smoothed_bitrate)
+        audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
+      bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
+    }
+  }
+}
+
 }  // namespace webrtc