mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
4712776bf4
This CL resets the AEC3 realignment functionality when a significant and persistent skew in the number of render and capture API calls is detected. Bug: chromium:811658,webrtc:8879 Change-Id: Ib5c727b38f427da2a7d25eac7c939a17bdaabe74 Reviewed-on: https://webrtc-review.googlesource.com/52260 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21997}
45 lines
1.7 KiB
C++
45 lines
1.7 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_H_
|
|
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
|
|
|
|
#include "api/array_view.h"
|
|
#include "api/optional.h"
|
|
#include "modules/audio_processing/aec3/delay_estimate.h"
|
|
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
|
|
#include "modules/audio_processing/aec3/render_delay_buffer.h"
|
|
#include "modules/audio_processing/include/audio_processing.h"
|
|
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Class for aligning the render and capture signal using a RenderDelayBuffer.
|
|
class RenderDelayController {
|
|
public:
|
|
static RenderDelayController* Create(const EchoCanceller3Config& config,
|
|
int non_causal_offset,
|
|
int sample_rate_hz);
|
|
virtual ~RenderDelayController() = default;
|
|
|
|
// Resets the delay controller.
|
|
virtual void Reset() = 0;
|
|
|
|
// Logs a render call.
|
|
virtual void LogRenderCall() = 0;
|
|
|
|
// Aligns the render buffer content with the capture signal.
|
|
virtual rtc::Optional<DelayEstimate> GetDelay(
|
|
const DownsampledRenderBuffer& render_buffer,
|
|
rtc::ArrayView<const float> capture) = 0;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
|