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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc | 207 |
1 files changed, 207 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc new file mode 100644 index 0000000000..bf3a0f18a1 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc @@ -0,0 +1,207 @@ +/* + * 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/experiments/struct_parameters_parser.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() { + StructParametersParser::Create( // + "quantile", &quantile, // + "forget_factor", &forget_factor, // + "start_forget_weight", &start_forget_weight, // + "resample_interval_ms", &resample_interval_ms, // + "use_reorder_optimizer", &use_reorder_optimizer, // + "reorder_forget_factor", &reorder_forget_factor, // + "ms_per_loss_percent", &ms_per_loss_percent) + ->Parse(webrtc::field_trial::FindFullName( + "WebRTC-Audio-NetEqDelayManagerConfig")); +} + +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) + << " use_reorder_optimizer=" << use_reorder_optimizer + << " reorder_forget_factor=" << reorder_forget_factor + << " ms_per_loss_percent=" << ms_per_loss_percent; +} + +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)), + 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() {} + +void DelayManager::Update(int arrival_delay_ms, bool reordered) { + if (!reorder_optimizer_ || !reordered) { + underrun_optimizer_.Update(arrival_delay_ms); + } + target_level_ms_ = + underrun_optimizer_.GetOptimalDelayMs().value_or(kStartDelayMs); + if (reorder_optimizer_) { + reorder_optimizer_->Update(arrival_delay_ms, reordered, target_level_ms_); + target_level_ms_ = std::max( + target_level_ms_, reorder_optimizer_->GetOptimalDelayMs().value_or(0)); + } + unlimited_target_level_ms_ = target_level_ms_; + 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) { + // Limit to 75% of maximum buffer size. + target_level_ms_ = std::min( + target_level_ms_, 3 * max_packets_in_buffer_ * packet_len_ms_ / 4); + } +} + +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(); + target_level_ms_ = kStartDelayMs; + if (reorder_optimizer_) { + reorder_optimizer_->Reset(); + } +} + +int DelayManager::TargetDelayMs() const { + return target_level_ms_; +} + +int DelayManager::UnlimitedTargetLevelMs() const { + return unlimited_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_) { + // 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 |