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libtgvoip/webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer.h

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/*
* 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 MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#include <stddef.h>
#include <vector>
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/render_buffer.h"
namespace webrtc {
// Class for buffering the incoming render blocks such that these may be
// extracted with a specified delay.
class RenderDelayBuffer {
public:
enum class BufferingEvent {
kNone,
kRenderUnderrun,
kRenderOverrun,
kApiCallSkew
};
static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
size_t num_bands);
static RenderDelayBuffer* Create2(const EchoCanceller3Config& config,
size_t num_bands);
virtual ~RenderDelayBuffer() = default;
// Resets the buffer alignment.
virtual void Reset() = 0;
// Inserts a block into the buffer.
virtual BufferingEvent Insert(
const std::vector<std::vector<float>>& block) = 0;
// Updates the buffers one step based on the specified buffer delay. Returns
// an enum indicating whether there was a special event that occurred.
virtual BufferingEvent PrepareCaptureProcessing() = 0;
// Sets the buffer delay and returns a bool indicating whether the delay
// changed.
virtual bool SetDelay(size_t delay) = 0;
// Gets the buffer delay.
virtual size_t Delay() const = 0;
// Gets the buffer delay.
virtual size_t MaxDelay() const = 0;
// Returns the render buffer for the echo remover.
virtual RenderBuffer* GetRenderBuffer() = 0;
// Returns the downsampled render buffer.
virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
// Returns whether the current delay is noncausal.
virtual bool CausalDelay(size_t delay) const = 0;
// Returns the maximum non calusal offset that can occur in the delay buffer.
static int DelayEstimatorOffset(const EchoCanceller3Config& config);
// Provides an optional external estimate of the audio buffer delay.
virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_