Revert of Log audio network adapter decisions in event log.

webrtc/logging/rtc_event_log/rtc_event_log.cc

 /*
  *  Copyright (c) 2015 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 59 matching lines @@
                     const uint8_t* header,
                     size_t packet_length) override;
   void LogRtcpPacket(PacketDirection direction,
                      MediaType media_type,
                      const uint8_t* packet,
                      size_t length) override;
   void LogAudioPlayout(uint32_t ssrc) override;
   void LogBwePacketLossEvent(int32_t bitrate,
                              uint8_t fraction_loss,
                              int32_t total_packets) override;
-  void LogAudioNetworkAdaptation(
-      const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override;
 
  private:
   void StoreEvent(std::unique_ptr<rtclog::Event>* event);
 
   // Message queue for passing control messages to the logging thread.
   SwapQueue<RtcEventLogHelperThread::ControlMessage> message_queue_;
 
   // Message queue for passing events to the logging thread.
   SwapQueue<std::unique_ptr<rtclog::Event> > event_queue_;
 
@@ skipping 337 matching lines @@
   std::unique_ptr<rtclog::Event> event(new rtclog::Event());
   event->set_timestamp_us(rtc::TimeMicros());
   event->set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
   auto bwe_event = event->mutable_bwe_packet_loss_event();
   bwe_event->set_bitrate(bitrate);
   bwe_event->set_fraction_loss(fraction_loss);
   bwe_event->set_total_packets(total_packets);
   StoreEvent(&event);
 }
 
-void RtcEventLogImpl::LogAudioNetworkAdaptation(
-    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
-  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
-  event->set_timestamp_us(rtc::TimeMicros());
-  event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
-  auto audio_network_adaptation = event->mutable_audio_network_adaptation();
-  if (config.bitrate_bps)
-    audio_network_adaptation->set_bitrate_bps(*config.bitrate_bps);
-  if (config.frame_length_ms)
-    audio_network_adaptation->set_frame_length_ms(*config.frame_length_ms);
-  if (config.uplink_packet_loss_fraction) {
-    audio_network_adaptation->set_uplink_packet_loss_fraction(
-        *config.uplink_packet_loss_fraction);
-  }
-  if (config.enable_fec)
-    audio_network_adaptation->set_enable_fec(*config.enable_fec);
-  if (config.enable_dtx)
-    audio_network_adaptation->set_enable_dtx(*config.enable_dtx);
-  if (config.num_channels)
-    audio_network_adaptation->set_num_channels(*config.num_channels);
-  StoreEvent(&event);
-}
-
 void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event>* event) {
   if (!event_queue_.Insert(event)) {
     LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event.";
   }
   helper_thread_.SignalNewEvent();
 }
 
 bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
                                    rtclog::EventStream* result) {
   char tmp_buffer[1024];
@@ skipping 28 matching lines @@
 #else
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 #endif  // ENABLE_RTC_EVENT_LOG
 }
 
 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 }
 
 }  // namespace webrtc