webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
Byoungchan Lee 604fd2f1ab Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}
2022-01-24 11:50:20 +00:00

56 lines
1.8 KiB
C++

/*
* Copyright (c) 2017 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_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
#include <stddef.h>
#include "absl/types/optional.h"
#include "modules/audio_processing/aec3/clockdrift_detector.h"
namespace webrtc {
// Handles the reporting of metrics for the render delay controller.
class RenderDelayControllerMetrics {
public:
RenderDelayControllerMetrics();
RenderDelayControllerMetrics(const RenderDelayControllerMetrics&) = delete;
RenderDelayControllerMetrics& operator=(const RenderDelayControllerMetrics&) =
delete;
// Updates the metric with new data.
void Update(absl::optional<size_t> delay_samples,
size_t buffer_delay_blocks,
absl::optional<int> skew_shift_blocks,
ClockdriftDetector::Level clockdrift);
// Returns true if the metrics have just been reported, otherwise false.
bool MetricsReported() { return metrics_reported_; }
private:
// Resets the metrics.
void ResetMetrics();
size_t delay_blocks_ = 0;
int reliable_delay_estimate_counter_ = 0;
int delay_change_counter_ = 0;
int call_counter_ = 0;
int skew_report_timer_ = 0;
int initial_call_counter_ = 0;
bool metrics_reported_ = false;
bool initial_update = true;
int skew_shift_count_ = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_