blob: 277b80de74650d685c446ddcf3137b3727493567 [file] [log] [blame]
/*
* 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_