Edward Lemur c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00

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2.0 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#include <stddef.h>
#include <array>
#include <vector>
#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
#include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "webrtc/modules/audio_processing/aec3/fft_data.h"
#include "webrtc/modules/audio_processing/aec3/render_buffer.h"
#include "webrtc/rtc_base/array_view.h"
namespace webrtc {
// Class for buffering the incoming render blocks such that these may be
// extracted with a specified delay.
class RenderDelayBuffer {
public:
static RenderDelayBuffer* Create(size_t num_bands);
virtual ~RenderDelayBuffer() = default;
// Resets the buffer data.
virtual void Reset() = 0;
// Inserts a block into the buffer and returns true if the insert is
// successful.
virtual bool Insert(const std::vector<std::vector<float>>& block) = 0;
// Updates the buffers one step based on the specified buffer delay. Returns
// true if there was no overrun, otherwise returns false.
virtual bool UpdateBuffers() = 0;
// Sets the buffer delay.
virtual void SetDelay(size_t delay) = 0;
// Gets the buffer delay.
virtual size_t Delay() const = 0;
// Returns the render buffer for the echo remover.
virtual const RenderBuffer& GetRenderBuffer() const = 0;
// Returns the downsampled render buffer.
virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_