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/*
* Copyright (c) 2021 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_UNDERRUN_OPTIMIZER_H_
#define MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
#include <memory>
#include "absl/types/optional.h"
#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/histogram.h"
namespace webrtc {
// Estimates probability of buffer underrun due to late packet arrival.
// The optimal delay is decided such that the probability of underrun is lower
// than 1 - `histogram_quantile`.
class UnderrunOptimizer {
public:
UnderrunOptimizer(const TickTimer* tick_timer,
int histogram_quantile,
int forget_factor,
absl::optional<int> start_forget_weight,
absl::optional<int> resample_interval_ms);
void Update(int relative_delay_ms);
absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
void Reset();
private:
const TickTimer* tick_timer_;
Histogram histogram_;
const int histogram_quantile_; // In Q30.
const absl::optional<int> resample_interval_ms_;
std::unique_ptr<TickTimer::Stopwatch> resample_stopwatch_;
int max_delay_in_interval_ms_ = 0;
absl::optional<int> optimal_delay_ms_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
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