/* * 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); static RenderDelayController* Create2(const EchoCanceller3Config& config, int sample_rate_hz); virtual ~RenderDelayController() = default; // Resets the delay controller. If the delay confidence is reset, the reset // behavior is as if the call is restarted. virtual void Reset(bool reset_delay_confidence) = 0; // Logs a render call. virtual void LogRenderCall() = 0; // Aligns the render buffer content with the capture signal. virtual absl::optional GetDelay( const DownsampledRenderBuffer& render_buffer, size_t render_delay_buffer_delay, const absl::optional& echo_remover_delay, rtc::ArrayView capture) = 0; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_