c8fa692ec4
I used a command like this to update the paths: perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"` The only manual edit is to add an include of webrtc/rtc_base/checks.h in webrtc/modules/audio_device/android/opensles_common.h, which likely was needed due to changed include paths due to 'git cl format'. BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2969653002 Cr-Commit-Position: refs/heads/master@{#18871}
60 lines
2.0 KiB
C++
60 lines
2.0 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_BUFFER_H_
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#define WEBRTC_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 "webrtc/modules/audio_processing/aec3/aec3_common.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/fft_data.h"
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#include "webrtc/modules/audio_processing/aec3/render_buffer.h"
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#include "webrtc/rtc_base/array_view.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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static RenderDelayBuffer* Create(size_t num_bands);
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virtual ~RenderDelayBuffer() = default;
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// Resets the buffer data.
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virtual void Reset() = 0;
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// Inserts a block into the buffer and returns true if the insert is
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// successful.
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virtual bool Insert(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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// true if there was no overrun, otherwise returns false.
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virtual bool UpdateBuffers() = 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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// 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 // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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