Revert of Log audio network adapter decisions in event log.

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

 /*
  *  Copyright (c) 2016 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 61 matching lines @@
     case rtclog::Event::BWE_PACKET_DELAY_EVENT:
       return ParsedRtcEventLog::EventType::BWE_PACKET_DELAY_EVENT;
     case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
     case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT;
     case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
     case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
-    case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
-      return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
   }
   RTC_NOTREACHED();
   return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
 }
 
 std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
   uint64_t varint = 0;
   for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
     // The most significant bit of each byte is 0 if it is the last byte in
     // the varint and 1 otherwise. Thus, we take the 7 least significant bits
@@ skipping 355 matching lines @@
   RTC_CHECK(loss_event.has_fraction_loss());
   if (fraction_loss != nullptr) {
     *fraction_loss = loss_event.fraction_loss();
   }
   RTC_CHECK(loss_event.has_total_packets());
   if (total_packets != nullptr) {
     *total_packets = loss_event.total_packets();
   }
 }
 
-void ParsedRtcEventLog::GetAudioNetworkAdaptation(
-    size_t index,
-    AudioNetworkAdaptor::EncoderRuntimeConfig* config) const {
-  RTC_CHECK_LT(index, GetNumberOfEvents());
-  const rtclog::Event& event = events_[index];
-  RTC_CHECK(event.has_type());
-  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
-  RTC_CHECK(event.has_audio_network_adaptation());
-  const rtclog::AudioNetworkAdaptation& ana_event =
-      event.audio_network_adaptation();
-  if (ana_event.has_bitrate_bps())
-    config->bitrate_bps = rtc::Optional<int>(ana_event.bitrate_bps());
-  if (ana_event.has_enable_fec())
-    config->enable_fec = rtc::Optional<bool>(ana_event.enable_fec());
-  if (ana_event.has_enable_dtx())
-    config->enable_dtx = rtc::Optional<bool>(ana_event.enable_dtx());
-  if (ana_event.has_frame_length_ms())
-    config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms());
-  if (ana_event.has_num_channels())
-    config->num_channels = rtc::Optional<size_t>(ana_event.num_channels());
-  if (ana_event.has_uplink_packet_loss_fraction())
-    config->uplink_packet_loss_fraction =
-        rtc::Optional<float>(ana_event.uplink_packet_loss_fraction());
-}
-
 }  // namespace webrtc