mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 23:01:21 +01:00
a76ef9d0b4
The faster AEC3 alignment introduced recently may in cases cause the alignment (and the AEC3) to repeatedly reset. This CL avoids these resets by handling buffer issues (which are triggering the resets) separately during the initial coarse alignment phase. Change-Id: Idf5e2ffda2591906da8060d03ec8ca73cdaedf53 Bug: webrtc:8798,chromium:805815 Reviewed-on: https://webrtc-review.googlesource.com/43480 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21758}
42 lines
1.6 KiB
C++
42 lines
1.6 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;
|
|
|
|
// 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_
|