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This is done by adding a reorder optimizer that estimates the probability of receiving reordered packets. The optimal delay is decided by balancing the cost of increasing the delay against the probability of missing a reordered packet, resulting in a loss. This balance is decided using the `ms_per_loss_percent` parameter. The usage and parameters can be controlled via field trial. Bug: webrtc:10178 Change-Id: Ic484df0412af35610e74b3a6070f2bac7a926a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231541 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34954}
127 lines
4.4 KiB
C++
127 lines
4.4 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
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#define MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
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#include <string.h> // Provide access to size_t.
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#include <deque>
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#include <memory>
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#include "absl/types/optional.h"
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#include "api/neteq/tick_timer.h"
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#include "modules/audio_coding/neteq/histogram.h"
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#include "modules/audio_coding/neteq/relative_arrival_delay_tracker.h"
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#include "modules/audio_coding/neteq/reorder_optimizer.h"
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#include "modules/audio_coding/neteq/underrun_optimizer.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/experiments/struct_parameters_parser.h"
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namespace webrtc {
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class DelayManager {
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public:
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struct Config {
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Config();
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void Log();
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// Options that can be configured via field trial.
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double quantile = 0.97;
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double forget_factor = 0.9993;
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absl::optional<double> start_forget_weight = 2;
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absl::optional<int> resample_interval_ms;
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int max_history_ms = 2000;
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bool use_reorder_optimizer = true;
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double reorder_forget_factor = 0.9993;
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int ms_per_loss_percent = 20;
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// Options that are externally populated.
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int max_packets_in_buffer = 200;
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int base_minimum_delay_ms = 0;
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private:
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std::unique_ptr<StructParametersParser> Parser();
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// TODO(jakobi): remove legacy field trial.
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void MaybeUpdateFromLegacyFieldTrial();
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};
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DelayManager(const Config& config, const TickTimer* tick_timer);
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virtual ~DelayManager();
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// Updates the delay manager with a new incoming packet, with `timestamp` from
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// the RTP header. This updates the statistics and a new target buffer level
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// is calculated. Returns the relative delay if it can be calculated. If
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// `reset` is true, restarts the relative arrival delay calculation from this
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// packet.
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virtual absl::optional<int> Update(uint32_t timestamp,
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int sample_rate_hz,
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bool reset = false);
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// Resets all state.
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virtual void Reset();
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// Gets the target buffer level in milliseconds.
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virtual int TargetDelayMs() const;
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// Notifies the DelayManager of how much audio data is carried in each packet.
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virtual int SetPacketAudioLength(int length_ms);
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// Accessors and mutators.
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// Assuming `delay` is in valid range.
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virtual bool SetMinimumDelay(int delay_ms);
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virtual bool SetMaximumDelay(int delay_ms);
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virtual bool SetBaseMinimumDelay(int delay_ms);
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virtual int GetBaseMinimumDelay() const;
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// These accessors are only intended for testing purposes.
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int effective_minimum_delay_ms_for_test() const {
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return effective_minimum_delay_ms_;
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}
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private:
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// Provides value which minimum delay can't exceed based on current buffer
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// size and given `maximum_delay_ms_`. Lower bound is a constant 0.
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int MinimumDelayUpperBound() const;
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// Updates `effective_minimum_delay_ms_` delay based on current
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// `minimum_delay_ms_`, `base_minimum_delay_ms_` and `maximum_delay_ms_`
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// and buffer size.
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void UpdateEffectiveMinimumDelay();
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// Makes sure that `delay_ms` is less than maximum delay, if any maximum
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// is set. Also, if possible check `delay_ms` to be less than 75% of
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// `max_packets_in_buffer_`.
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bool IsValidMinimumDelay(int delay_ms) const;
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bool IsValidBaseMinimumDelay(int delay_ms) const;
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// TODO(jakobi): set maximum buffer delay instead of number of packets.
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const int max_packets_in_buffer_;
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UnderrunOptimizer underrun_optimizer_;
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std::unique_ptr<ReorderOptimizer> reorder_optimizer_;
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RelativeArrivalDelayTracker relative_arrival_delay_tracker_;
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int base_minimum_delay_ms_;
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int effective_minimum_delay_ms_; // Used as lower bound for target delay.
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int minimum_delay_ms_; // Externally set minimum delay.
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int maximum_delay_ms_; // Externally set maximum allowed delay.
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int packet_len_ms_ = 0;
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int target_level_ms_; // Currently preferred buffer level.
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RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
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