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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_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/delay_estimate.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/render_delay_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {
// Class for aligning the render and capture signal using a RenderDelayBuffer.
class RenderDelayController {
public:
static RenderDelayController* Create(const EchoCanceller3Config& config,
int non_causal_offset,
int sample_rate_hz);
virtual ~RenderDelayController() = default;
// Resets the delay controller.
virtual void Reset() = 0;
// Logs a render call.
virtual void LogRenderCall() = 0;
// Aligns the render buffer content with the capture signal.
virtual absl::optional<DelayEstimate> GetDelay(
const DownsampledRenderBuffer& render_buffer,
size_t render_delay_buffer_delay,
const absl::optional<int>& echo_remover_delay,
rtc::ArrayView<const float> capture) = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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