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3646f973c2
Avoid including audio_processing.h from within AEC3. Bug: webrtc:8844 Change-Id: I02c475c2fb84e2c24eac86baac3c7edaa08bebc0 Reviewed-on: https://webrtc-review.googlesource.com/53065 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22029}
45 lines
1.7 KiB
C++
45 lines
1.7 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#include "api/array_view.h"
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#include "api/audio/echo_canceller3_config.h"
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#include "api/optional.h"
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#include "modules/audio_processing/aec3/delay_estimate.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/render_delay_buffer.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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namespace webrtc {
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// Class for aligning the render and capture signal using a RenderDelayBuffer.
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class RenderDelayController {
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public:
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static RenderDelayController* Create(const EchoCanceller3Config& config,
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int non_causal_offset,
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int sample_rate_hz);
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virtual ~RenderDelayController() = default;
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// Resets the delay controller.
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virtual void Reset() = 0;
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// Logs a render call.
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virtual void LogRenderCall() = 0;
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// Aligns the render buffer content with the capture signal.
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virtual rtc::Optional<DelayEstimate> GetDelay(
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const DownsampledRenderBuffer& render_buffer,
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rtc::ArrayView<const float> capture) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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