| /* |
| * 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 "modules/audio_coding/audio_network_adaptor/event_log_writer.h" |
| |
| #include <math.h> |
| |
| #include <algorithm> |
| #include <cstdlib> |
| #include <memory> |
| #include <utility> |
| |
| #include "absl/types/optional.h" |
| #include "api/rtc_event_log/rtc_event.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" |
| #include "rtc_base/checks.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 AudioEncoderRuntimeConfig& 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 AudioEncoderRuntimeConfig& config) { |
| auto config_copy = std::make_unique<AudioEncoderRuntimeConfig>(config); |
| event_log_->Log( |
| std::make_unique<RtcEventAudioNetworkAdaptation>(std::move(config_copy))); |
| last_logged_config_ = config; |
| } |
| |
| } // namespace webrtc |