blob: 17a75da8d0c9162178e018a1918cc03ca4ae8b9b [file] [log] [blame]
/*
* 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_