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