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 "absl/types/optional.h" 15 #include "api/array_view.h" 16 #include "api/audio/echo_canceller3_config.h" 17 #include "modules/audio_processing/aec3/delay_estimate.h" 18 #include "modules/audio_processing/aec3/downsampled_render_buffer.h" 19 #include "modules/audio_processing/aec3/render_delay_buffer.h" 20 #include "modules/audio_processing/logging/apm_data_dumper.h" 21 22 namespace webrtc { 23 24 // Class for aligning the render and capture signal using a RenderDelayBuffer. 25 class RenderDelayController { 26 public: 27 static RenderDelayController* Create(const EchoCanceller3Config& config, 28 int sample_rate_hz, 29 size_t num_capture_channels); 30 virtual ~RenderDelayController() = default; 31 32 // Resets the delay controller. If the delay confidence is reset, the reset 33 // behavior is as if the call is restarted. 34 virtual void Reset(bool reset_delay_confidence) = 0; 35 36 // Logs a render call. 37 virtual void LogRenderCall() = 0; 38 39 // Aligns the render buffer content with the capture signal. 40 virtual absl::optional<DelayEstimate> GetDelay( 41 const DownsampledRenderBuffer& render_buffer, 42 size_t render_delay_buffer_delay, 43 const std::vector<std::vector<float>>& capture) = 0; 44 45 // Returns true if clockdrift has been detected. 46 virtual bool HasClockdrift() const = 0; 47 }; 48 } // namespace webrtc 49 50 #endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ 51