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/*
* Copyright (c) 2012 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/neteq/delay_manager.h"
#include <stdio.h>
#include <stdlib.h>
#include <algorithm>
#include <memory>
#include <numeric>
#include <string>
#include "modules/include/module_common_types_public.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
constexpr int kMinBaseMinimumDelayMs = 0;
constexpr int kMaxBaseMinimumDelayMs = 10000;
constexpr int kStartDelayMs = 80;
std::unique_ptr<ReorderOptimizer> MaybeCreateReorderOptimizer(
const DelayManager::Config& config) {
if (!config.use_reorder_optimizer) {
return nullptr;
}
return std::make_unique<ReorderOptimizer>(
(1 << 15) * config.reorder_forget_factor, config.ms_per_loss_percent,
config.start_forget_weight);
}
} // namespace
DelayManager::Config::Config() {
Parser()->Parse(webrtc::field_trial::FindFullName(
"WebRTC-Audio-NetEqDelayManagerConfig"));
MaybeUpdateFromLegacyFieldTrial();
}
void DelayManager::Config::Log() {
RTC_LOG(LS_INFO) << "Delay manager config:"
" quantile="
<< quantile << " forget_factor=" << forget_factor
<< " start_forget_weight=" << start_forget_weight.value_or(0)
<< " resample_interval_ms="
<< resample_interval_ms.value_or(0)
<< " max_history_ms=" << max_history_ms
<< " use_reorder_optimizer=" << use_reorder_optimizer
<< " reorder_forget_factor=" << reorder_forget_factor
<< " ms_per_loss_percent=" << ms_per_loss_percent;
}
std::unique_ptr<StructParametersParser> DelayManager::Config::Parser() {
return StructParametersParser::Create( //
"quantile", &quantile, //
"forget_factor", &forget_factor, //
"start_forget_weight", &start_forget_weight, //
"resample_interval_ms", &resample_interval_ms, //
"max_history_ms", &max_history_ms, //
"use_reorder_optimizer", &use_reorder_optimizer, //
"reorder_forget_factor", &reorder_forget_factor, //
"ms_per_loss_percent", &ms_per_loss_percent);
}
// TODO(jakobi): remove legacy field trial.
void DelayManager::Config::MaybeUpdateFromLegacyFieldTrial() {
constexpr char kDelayHistogramFieldTrial[] =
"WebRTC-Audio-NetEqDelayHistogram";
if (!webrtc::field_trial::IsEnabled(kDelayHistogramFieldTrial)) {
return;
}
const auto field_trial_string =
webrtc::field_trial::FindFullName(kDelayHistogramFieldTrial);
double percentile = -1.0;
double forget_factor = -1.0;
double start_forget_weight = -1.0;
if (sscanf(field_trial_string.c_str(), "Enabled-%lf-%lf-%lf", &percentile,
&forget_factor, &start_forget_weight) >= 2 &&
percentile >= 0.0 && percentile <= 100.0 && forget_factor >= 0.0 &&
forget_factor <= 1.0) {
this->quantile = percentile / 100;
this->forget_factor = forget_factor;
this->start_forget_weight = start_forget_weight >= 1
? absl::make_optional(start_forget_weight)
: absl::nullopt;
}
}
DelayManager::DelayManager(const Config& config, const TickTimer* tick_timer)
: max_packets_in_buffer_(config.max_packets_in_buffer),
underrun_optimizer_(tick_timer,
(1 << 30) * config.quantile,
(1 << 15) * config.forget_factor,
config.start_forget_weight,
config.resample_interval_ms),
reorder_optimizer_(MaybeCreateReorderOptimizer(config)),
relative_arrival_delay_tracker_(tick_timer, config.max_history_ms),
base_minimum_delay_ms_(config.base_minimum_delay_ms),
effective_minimum_delay_ms_(config.base_minimum_delay_ms),
minimum_delay_ms_(0),
maximum_delay_ms_(0),
target_level_ms_(kStartDelayMs) {
RTC_DCHECK_GE(base_minimum_delay_ms_, 0);
Reset();
}
DelayManager::~DelayManager() {}
absl::optional<int> DelayManager::Update(uint32_t timestamp,
int sample_rate_hz,
bool reset) {
if (reset) {
relative_arrival_delay_tracker_.Reset();
}
absl::optional<int> relative_delay =
relative_arrival_delay_tracker_.Update(timestamp, sample_rate_hz);
if (!relative_delay) {
return absl::nullopt;
}
bool reordered =
relative_arrival_delay_tracker_.newest_timestamp() != timestamp;
if (!reorder_optimizer_ || !reordered) {
underrun_optimizer_.Update(*relative_delay);
}
target_level_ms_ =
underrun_optimizer_.GetOptimalDelayMs().value_or(kStartDelayMs);
if (reorder_optimizer_) {
reorder_optimizer_->Update(*relative_delay, reordered, target_level_ms_);
target_level_ms_ = std::max(
target_level_ms_, reorder_optimizer_->GetOptimalDelayMs().value_or(0));
}
target_level_ms_ = std::max(target_level_ms_, effective_minimum_delay_ms_);
if (maximum_delay_ms_ > 0) {
target_level_ms_ = std::min(target_level_ms_, maximum_delay_ms_);
}
if (packet_len_ms_ > 0) {
// Target level should be at least one packet.
target_level_ms_ = std::max(target_level_ms_, packet_len_ms_);
// Limit to 75% of maximum buffer size.
target_level_ms_ = std::min(
target_level_ms_, 3 * max_packets_in_buffer_ * packet_len_ms_ / 4);
}
return relative_delay;
}
int DelayManager::SetPacketAudioLength(int length_ms) {
if (length_ms <= 0) {
RTC_LOG_F(LS_ERROR) << "length_ms = " << length_ms;
return -1;
}
packet_len_ms_ = length_ms;
return 0;
}
void DelayManager::Reset() {
packet_len_ms_ = 0;
underrun_optimizer_.Reset();
relative_arrival_delay_tracker_.Reset();
target_level_ms_ = kStartDelayMs;
if (reorder_optimizer_) {
reorder_optimizer_->Reset();
}
}
int DelayManager::TargetDelayMs() const {
return target_level_ms_;
}
bool DelayManager::IsValidMinimumDelay(int delay_ms) const {
return 0 <= delay_ms && delay_ms <= MinimumDelayUpperBound();
}
bool DelayManager::IsValidBaseMinimumDelay(int delay_ms) const {
return kMinBaseMinimumDelayMs <= delay_ms &&
delay_ms <= kMaxBaseMinimumDelayMs;
}
bool DelayManager::SetMinimumDelay(int delay_ms) {
if (!IsValidMinimumDelay(delay_ms)) {
return false;
}
minimum_delay_ms_ = delay_ms;
UpdateEffectiveMinimumDelay();
return true;
}
bool DelayManager::SetMaximumDelay(int delay_ms) {
// If `delay_ms` is zero then it unsets the maximum delay and target level is
// unconstrained by maximum delay.
if (delay_ms != 0 &&
(delay_ms < minimum_delay_ms_ || delay_ms < packet_len_ms_)) {
// Maximum delay shouldn't be less than minimum delay or less than a packet.
return false;
}
maximum_delay_ms_ = delay_ms;
UpdateEffectiveMinimumDelay();
return true;
}
bool DelayManager::SetBaseMinimumDelay(int delay_ms) {
if (!IsValidBaseMinimumDelay(delay_ms)) {
return false;
}
base_minimum_delay_ms_ = delay_ms;
UpdateEffectiveMinimumDelay();
return true;
}
int DelayManager::GetBaseMinimumDelay() const {
return base_minimum_delay_ms_;
}
void DelayManager::UpdateEffectiveMinimumDelay() {
// Clamp `base_minimum_delay_ms_` into the range which can be effectively
// used.
const int base_minimum_delay_ms =
rtc::SafeClamp(base_minimum_delay_ms_, 0, MinimumDelayUpperBound());
effective_minimum_delay_ms_ =
std::max(minimum_delay_ms_, base_minimum_delay_ms);
}
int DelayManager::MinimumDelayUpperBound() const {
// Choose the lowest possible bound discarding 0 cases which mean the value
// is not set and unconstrained.
int q75 = max_packets_in_buffer_ * packet_len_ms_ * 3 / 4;
q75 = q75 > 0 ? q75 : kMaxBaseMinimumDelayMs;
const int maximum_delay_ms =
maximum_delay_ms_ > 0 ? maximum_delay_ms_ : kMaxBaseMinimumDelayMs;
return std::min(maximum_delay_ms, q75);
}
} // namespace webrtc