Revert of Log audio network adapter decisions in event log.

webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc

Index: webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
deleted file mode 100644
index 619a2473d93708cf221846151f9b2ebf61d5a815..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  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.
- */
-
-#include <math.h>
-#include <algorithm>
-
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h"
-
-namespace webrtc {
-
-EventLogWriter::EventLogWriter(RtcEventLog* event_log,
-                               int min_bitrate_change_bps,
-                               float min_bitrate_change_fraction,
-                               float min_packet_loss_change_fraction)
-    : event_log_(event_log),
-      min_bitrate_change_bps_(min_bitrate_change_bps),
-      min_bitrate_change_fraction_(min_bitrate_change_fraction),
-      min_packet_loss_change_fraction_(min_packet_loss_change_fraction) {
-  RTC_DCHECK(event_log_);
-}
-
-EventLogWriter::~EventLogWriter() = default;
-
-void EventLogWriter::MaybeLogEncoderConfig(
-    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
-  if (last_logged_config_.num_channels != config.num_channels)
-    return LogEncoderConfig(config);
-  if (last_logged_config_.enable_dtx != config.enable_dtx)
-    return LogEncoderConfig(config);
-  if (last_logged_config_.enable_fec != config.enable_fec)
-    return LogEncoderConfig(config);
-  if (last_logged_config_.frame_length_ms != config.frame_length_ms)
-    return LogEncoderConfig(config);
-  if ((!last_logged_config_.bitrate_bps && config.bitrate_bps) ||
-      (last_logged_config_.bitrate_bps && config.bitrate_bps &&
-       std::abs(*last_logged_config_.bitrate_bps - *config.bitrate_bps) >=
-           std::min(static_cast<int>(*last_logged_config_.bitrate_bps *
-                                     min_bitrate_change_fraction_),
-                    min_bitrate_change_bps_))) {
-    return LogEncoderConfig(config);
-  }
-  if ((!last_logged_config_.uplink_packet_loss_fraction &&
-       config.uplink_packet_loss_fraction) ||
-      (last_logged_config_.uplink_packet_loss_fraction &&
-       config.uplink_packet_loss_fraction &&
-       fabs(*last_logged_config_.uplink_packet_loss_fraction -
-            *config.uplink_packet_loss_fraction) >=
-           min_packet_loss_change_fraction_ *
-               *last_logged_config_.uplink_packet_loss_fraction)) {
-    return LogEncoderConfig(config);
-  }
-}
-
-void EventLogWriter::LogEncoderConfig(
-    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
-  event_log_->LogAudioNetworkAdaptation(config);
-  last_logged_config_ = config;
-}
-
-}  // namespace webrtc