1 /* 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ 12 #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ 13 14 #include "api/array_view.h" 15 #include "api/optional.h" 16 #include "modules/audio_processing/aec3/downsampled_render_buffer.h" 17 #include "modules/audio_processing/aec3/render_delay_buffer.h" 18 #include "modules/audio_processing/include/audio_processing.h" 19 #include "modules/audio_processing/logging/apm_data_dumper.h" 20 21 namespace webrtc { 22 23 // Class for aligning the render and capture signal using a RenderDelayBuffer. 24 class RenderDelayController { 25 public: 26 static RenderDelayController* Create(const EchoCanceller3Config& config, 27 int sample_rate_hz); 28 virtual ~RenderDelayController() = default; 29 30 // Resets the delay controller. 31 virtual void Reset() = 0; 32 33 // Receives the externally used delay. 34 virtual void SetDelay(size_t render_delay) = 0; 35 36 // Aligns the render buffer content with the capture signal. 37 virtual size_t GetDelay(const DownsampledRenderBuffer& render_buffer, 38 rtc::ArrayView<const float> capture) = 0; 39 40 // Returns an approximate value for the headroom in the buffer alignment. 41 virtual rtc::Optional<size_t> AlignmentHeadroomSamples() const = 0; 42 }; 43 } // namespace webrtc 44 45 #endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ 46