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/*
* 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
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