| /* |
| * 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ |
| |
| #include "webrtc/base/array_view.h" |
| #include "webrtc/base/optional.h" |
| #include "webrtc/modules/audio_processing/aec3/render_delay_buffer.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 { |
| public: |
| static RenderDelayController* Create( |
| int sample_rate_hz, |
| const RenderDelayBuffer& render_delay_buffer); |
| virtual ~RenderDelayController() = default; |
| |
| // Aligns the render buffer content with the capture signal. |
| virtual size_t GetDelay(rtc::ArrayView<const float> capture) = 0; |
| |
| // Analyzes the render signal and returns false if there is a buffer overrun. |
| virtual bool AnalyzeRender(rtc::ArrayView<const float> render) = 0; |
| |
| // Returns an approximate value for the headroom in the buffer alignment. |
| virtual rtc::Optional<size_t> AlignmentHeadroomSamples() const = 0; |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ |