| Index: webrtc/modules/audio_processing/aec3/render_delay_controller.h
|
| diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller.h b/webrtc/modules/audio_processing/aec3/render_delay_controller.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..b22a5fde16d3503c8976108e96e285bc9cc0ea81
|
| --- /dev/null
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| +++ b/webrtc/modules/audio_processing/aec3/render_delay_controller.h
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| @@ -0,0 +1,40 @@
|
| +/*
|
| + * 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;
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| +
|
| + // 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_
|
|
|