Pass event log to ANA.

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.
  */
 
@@ skipping 163 matching lines @@
     const Config& config,
     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)),
       audio_network_adaptor_creator_(
           audio_network_adaptor_creator
               ? std::move(audio_network_adaptor_creator)
-              : [this](const std::string& config_string, const Clock* clock) {
+              : [this](const std::string& config_string,
+                       RtcEventLog* event_log,
+                       const Clock* clock) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
-                                                           clock);
+                                                           event_log, 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) {}
@@ skipping 64 matching lines @@
 }
 
 void AudioEncoderOpus::SetMaxPlaybackRate(int frequency_hz) {
   auto conf = config_;
   conf.max_playback_rate_hz = frequency_hz;
   RTC_CHECK(RecreateEncoderInstance(conf));
 }
 
 bool AudioEncoderOpus::EnableAudioNetworkAdaptor(
     const std::string& config_string,
+    RtcEventLog* event_log,
     const Clock* clock) {
-  audio_network_adaptor_ = audio_network_adaptor_creator_(config_string, clock);
+  audio_network_adaptor_ =
+      audio_network_adaptor_creator_(config_string, event_log, clock);
   return audio_network_adaptor_.get() != nullptr;
 }
 
 void AudioEncoderOpus::DisableAudioNetworkAdaptor() {
   audio_network_adaptor_.reset(nullptr);
 }
 
 void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
   if (!audio_network_adaptor_) {
@@ skipping 236 matching lines @@
     SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
   if (config.enable_dtx)
     SetDtx(*config.enable_dtx);
   if (config.num_channels)
     SetNumChannelsToEncode(*config.num_channels);
 }
 
 std::unique_ptr<AudioNetworkAdaptor>
 AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator(
     const std::string& config_string,
+    RtcEventLog* event_log,
     const Clock* clock) const {
   AudioNetworkAdaptorImpl::Config config;
   config.clock = clock;
+  config.event_log = event_log;
   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