blob: e840178cb6f1fd00e9050ada96addda594f2a4c2 [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.
#include "webrtc/api/array_view.h"
#include "webrtc/api/optional.h"
#include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {
// Class for aligning the render and capture signal using a RenderDelayBuffer.
class RenderDelayController {
static RenderDelayController* Create(
const AudioProcessing::Config::EchoCanceller3& config,
int sample_rate_hz);
virtual ~RenderDelayController() = default;
// Resets the delay controller.
virtual void Reset() = 0;
// Receives the externally used delay.
virtual void SetDelay(size_t render_delay) = 0;
// Aligns the render buffer content with the capture signal.
virtual size_t GetDelay(const DownsampledRenderBuffer& render_buffer,
rtc::ArrayView<const float> capture) = 0;
// Returns an approximate value for the headroom in the buffer alignment.
virtual rtc::Optional<size_t> AlignmentHeadroomSamples() const = 0;
} // namespace webrtc