This CL centralizes the render buffering in AEC3 so that all render buffers are updated and synchronized/aligned with the render alignment buffer. Bug: webrtc:8597, chromium:790905 Change-Id: I8a94e5c1f27316b6100b420eec9652ea31c1a91d Reviewed-on: https://webrtc-review.googlesource.com/25680 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20989}
76 lines
2.4 KiB
C++
76 lines
2.4 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 MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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#include <stddef.h>
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#include <array>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/fft_data.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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// Class for buffering the incoming render blocks such that these may be
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// extracted with a specified delay.
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class RenderDelayBuffer {
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public:
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enum class BufferingEvent {
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kNone,
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kRenderUnderrun,
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kRenderOverrun,
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kApiCallSkew,
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kRenderDataLost
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};
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static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
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size_t num_bands);
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virtual ~RenderDelayBuffer() = default;
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// Resets the buffer alignment.
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virtual void Reset() = 0;
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// Inserts a block into the buffer.
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virtual BufferingEvent Insert(
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const std::vector<std::vector<float>>& block) = 0;
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// Updates the buffers one step based on the specified buffer delay. Returns
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// an enum indicating whether there was a special event that occurred.
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virtual BufferingEvent PrepareCaptureCall() = 0;
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// Sets the buffer delay.
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virtual void SetDelay(size_t delay) = 0;
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// Gets the buffer delay.
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virtual size_t Delay() const = 0;
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// Gets the buffer delay.
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virtual size_t MaxDelay() const = 0;
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// Gets the observed jitter in the render and capture call sequence.
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virtual size_t MaxApiJitter() const = 0;
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// Returns the render buffer for the echo remover.
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virtual const RenderBuffer& GetRenderBuffer() const = 0;
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// Returns the downsampled render buffer.
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virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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