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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
46 lines
1.8 KiB
C++
46 lines
1.8 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#include "webrtc/api/array_view.h"
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#include "webrtc/api/optional.h"
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#include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/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(
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const AudioProcessing::Config::EchoCanceller3& config,
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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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// Receives the externally used delay.
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virtual void SetDelay(size_t render_delay) = 0;
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// Aligns the render buffer content with the capture signal.
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virtual size_t GetDelay(const DownsampledRenderBuffer& render_buffer,
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rtc::ArrayView<const float> capture) = 0;
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// Returns an approximate value for the headroom in the buffer alignment.
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virtual rtc::Optional<size_t> AlignmentHeadroomSamples() const = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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