cf02cf13a7
This CL adds major render pipeline changes to the AEC3 code. The reason for these are that 1) It allows the echo removal unit to receive information about the content in bands beyond band 0, thereby allowing removal of high-frequency echoes 2) It allows more controlled handling of the render buffers, allowing proper buffer behaviour during capture glitches and clock-drift. Unfortunately, the render pipeline caused a lot of related changes in much of the rest of the AEC3 files. Most of these are, however, caused by a change of class name. Another unfortunate effect of this CL, is that a number of unittest cease to compile. I chose to temporarily solve that by removing them from the build using #if/#endif. The reason for that is that those will anyway again need to be changed in the next review, and doing like this avoids them having to be reviewed twice. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2784023002 Cr-Commit-Position: refs/heads/master@{#17547}
44 lines
1.7 KiB
C++
44 lines
1.7 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#include "webrtc/base/array_view.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
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#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
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namespace webrtc {
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// Class for aligning the render and capture signal using a RenderDelayBuffer.
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class RenderDelayController {
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public:
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static RenderDelayController* Create(int sample_rate_hz);
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virtual ~RenderDelayController() = default;
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// Resets the delay controller.
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virtual void Reset() = 0;
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// Receives the externally used delay.
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virtual void SetDelay(size_t render_delay) = 0;
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// Aligns the render buffer content with the capture signal.
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virtual size_t GetDelay(const DownsampledRenderBuffer& render_buffer,
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rtc::ArrayView<const float> capture) = 0;
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// Returns an approximate value for the headroom in the buffer alignment.
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virtual rtc::Optional<size_t> AlignmentHeadroomSamples() const = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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