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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/audio_network_adaptor_impl.h"
#include <stdint.h>
#include <utility>
#include <vector>
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
#include "rtc_base/checks.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
constexpr int kEventLogMinBitrateChangeBps = 5000;
constexpr float kEventLogMinBitrateChangeFraction = 0.25;
constexpr float kEventLogMinPacketLossChangeFraction = 0.5;
} // namespace
AudioNetworkAdaptorImpl::Config::Config() : event_log(nullptr) {}
AudioNetworkAdaptorImpl::Config::~Config() = default;
AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
const Config& config,
std::unique_ptr<ControllerManager> controller_manager,
std::unique_ptr<DebugDumpWriter> debug_dump_writer)
: config_(config),
controller_manager_(std::move(controller_manager)),
debug_dump_writer_(std::move(debug_dump_writer)),
event_log_writer_(
config.event_log
? new EventLogWriter(config.event_log,
kEventLogMinBitrateChangeBps,
kEventLogMinBitrateChangeFraction,
kEventLogMinPacketLossChangeFraction)
: nullptr) {
RTC_DCHECK(controller_manager_);
}
AudioNetworkAdaptorImpl::~AudioNetworkAdaptorImpl() = default;
void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) {
last_metrics_.uplink_bandwidth_bps = uplink_bandwidth_bps;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetUplinkPacketLossFraction(
float uplink_packet_loss_fraction) {
last_metrics_.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
network_metrics.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) {
last_metrics_.rtt_ms = rtt_ms;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
network_metrics.rtt_ms = rtt_ms;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetTargetAudioBitrate(
int target_audio_bitrate_bps) {
last_metrics_.target_audio_bitrate_bps = target_audio_bitrate_bps;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
network_metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
UpdateNetworkMetrics(network_metrics);
}
void AudioNetworkAdaptorImpl::SetOverhead(size_t overhead_bytes_per_packet) {
last_metrics_.overhead_bytes_per_packet = overhead_bytes_per_packet;
DumpNetworkMetrics();
Controller::NetworkMetrics network_metrics;
network_metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
UpdateNetworkMetrics(network_metrics);
}
AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
AudioEncoderRuntimeConfig config;
for (auto& controller :
controller_manager_->GetSortedControllers(last_metrics_))
controller->MakeDecision(&config);
// Update ANA stats.
auto increment_opt = [](absl::optional<uint32_t>& a) {
a = a.value_or(0) + 1;
};
if (prev_config_) {
if (config.bitrate_bps != prev_config_->bitrate_bps) {
increment_opt(stats_.bitrate_action_counter);
}
if (config.enable_dtx != prev_config_->enable_dtx) {
increment_opt(stats_.dtx_action_counter);
}
if (config.enable_fec != prev_config_->enable_fec) {
increment_opt(stats_.fec_action_counter);
}
if (config.frame_length_ms && prev_config_->frame_length_ms) {
if (*config.frame_length_ms > *prev_config_->frame_length_ms) {
increment_opt(stats_.frame_length_increase_counter);
} else if (*config.frame_length_ms < *prev_config_->frame_length_ms) {
increment_opt(stats_.frame_length_decrease_counter);
}
}
if (config.num_channels != prev_config_->num_channels) {
increment_opt(stats_.channel_action_counter);
}
if (config.uplink_packet_loss_fraction) {
stats_.uplink_packet_loss_fraction = *config.uplink_packet_loss_fraction;
}
}
prev_config_ = config;
if (debug_dump_writer_)
debug_dump_writer_->DumpEncoderRuntimeConfig(config, rtc::TimeMillis());
if (event_log_writer_)
event_log_writer_->MaybeLogEncoderConfig(config);
return config;
}
void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) {
debug_dump_writer_ = DebugDumpWriter::Create(file_handle);
}
void AudioNetworkAdaptorImpl::StopDebugDump() {
debug_dump_writer_.reset(nullptr);
}
ANAStats AudioNetworkAdaptorImpl::GetStats() const {
return stats_;
}
void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
if (debug_dump_writer_)
debug_dump_writer_->DumpNetworkMetrics(last_metrics_, rtc::TimeMillis());
}
void AudioNetworkAdaptorImpl::UpdateNetworkMetrics(
const Controller::NetworkMetrics& network_metrics) {
for (auto& controller : controller_manager_->GetControllers())
controller->UpdateNetworkMetrics(network_metrics);
}
} // namespace webrtc
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