| /* |
| * 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. |
| */ |
| |
| #ifndef MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_ |
| #define MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_ |
| |
| #include <string.h> // Provide access to size_t. |
| |
| #include <deque> |
| #include <memory> |
| |
| #include "absl/types/optional.h" |
| #include "api/neteq/tick_timer.h" |
| #include "modules/audio_coding/neteq/histogram.h" |
| #include "modules/audio_coding/neteq/relative_arrival_delay_tracker.h" |
| #include "modules/audio_coding/neteq/reorder_optimizer.h" |
| #include "modules/audio_coding/neteq/underrun_optimizer.h" |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/experiments/struct_parameters_parser.h" |
| |
| namespace webrtc { |
| |
| class DelayManager { |
| public: |
| struct Config { |
| Config(); |
| void Log(); |
| |
| // Options that can be configured via field trial. |
| double quantile = 0.97; |
| double forget_factor = 0.9993; |
| absl::optional<double> start_forget_weight = 2; |
| absl::optional<int> resample_interval_ms; |
| int max_history_ms = 2000; |
| |
| bool use_reorder_optimizer = true; |
| double reorder_forget_factor = 0.9993; |
| int ms_per_loss_percent = 20; |
| |
| // Options that are externally populated. |
| int max_packets_in_buffer = 200; |
| int base_minimum_delay_ms = 0; |
| |
| private: |
| std::unique_ptr<StructParametersParser> Parser(); |
| |
| // TODO(jakobi): remove legacy field trial. |
| void MaybeUpdateFromLegacyFieldTrial(); |
| }; |
| |
| DelayManager(const Config& config, const TickTimer* tick_timer); |
| |
| virtual ~DelayManager(); |
| |
| // Updates the delay manager with a new incoming packet, with `timestamp` from |
| // the RTP header. This updates the statistics and a new target buffer level |
| // is calculated. Returns the relative delay if it can be calculated. If |
| // `reset` is true, restarts the relative arrival delay calculation from this |
| // packet. |
| virtual absl::optional<int> Update(uint32_t timestamp, |
| int sample_rate_hz, |
| bool reset = false); |
| |
| // Resets all state. |
| virtual void Reset(); |
| |
| // Gets the target buffer level in milliseconds. |
| virtual int TargetDelayMs() const; |
| |
| // Notifies the DelayManager of how much audio data is carried in each packet. |
| virtual int SetPacketAudioLength(int length_ms); |
| |
| // Accessors and mutators. |
| // Assuming `delay` is in valid range. |
| virtual bool SetMinimumDelay(int delay_ms); |
| virtual bool SetMaximumDelay(int delay_ms); |
| virtual bool SetBaseMinimumDelay(int delay_ms); |
| virtual int GetBaseMinimumDelay() const; |
| |
| // These accessors are only intended for testing purposes. |
| int effective_minimum_delay_ms_for_test() const { |
| return effective_minimum_delay_ms_; |
| } |
| |
| private: |
| // Provides value which minimum delay can't exceed based on current buffer |
| // size and given `maximum_delay_ms_`. Lower bound is a constant 0. |
| int MinimumDelayUpperBound() const; |
| |
| // Updates `effective_minimum_delay_ms_` delay based on current |
| // `minimum_delay_ms_`, `base_minimum_delay_ms_` and `maximum_delay_ms_` |
| // and buffer size. |
| void UpdateEffectiveMinimumDelay(); |
| |
| // Makes sure that `delay_ms` is less than maximum delay, if any maximum |
| // is set. Also, if possible check `delay_ms` to be less than 75% of |
| // `max_packets_in_buffer_`. |
| bool IsValidMinimumDelay(int delay_ms) const; |
| |
| bool IsValidBaseMinimumDelay(int delay_ms) const; |
| |
| // TODO(jakobi): set maximum buffer delay instead of number of packets. |
| const int max_packets_in_buffer_; |
| UnderrunOptimizer underrun_optimizer_; |
| std::unique_ptr<ReorderOptimizer> reorder_optimizer_; |
| RelativeArrivalDelayTracker relative_arrival_delay_tracker_; |
| |
| int base_minimum_delay_ms_; |
| int effective_minimum_delay_ms_; // Used as lower bound for target delay. |
| int minimum_delay_ms_; // Externally set minimum delay. |
| int maximum_delay_ms_; // Externally set maximum allowed delay. |
| |
| int packet_len_ms_ = 0; |
| int target_level_ms_; // Currently preferred buffer level. |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager); |
| }; |
| |
| } // namespace webrtc |
| #endif // MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_ |