mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
74158ff761
Split out `RelativeArrivalDelayTracker` and `DelayOptimizer` logic. This is in preparation for adding another `DelayOptimizer` specialized in handling reordered packets. Bug: webrtc:10178 Change-Id: Id3c1746d91980b171fa524f9b2b71cf11fc75f64 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231224 Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34938}
50 lines
1.6 KiB
C++
50 lines
1.6 KiB
C++
/*
|
|
* 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_
|