c8fa692ec4
I used a command like this to update the paths: perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"` The only manual edit is to add an include of webrtc/rtc_base/checks.h in webrtc/modules/audio_device/android/opensles_common.h, which likely was needed due to changed include paths due to 'git cl format'. BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2969653002 Cr-Commit-Position: refs/heads/master@{#18871}
44 lines
1.7 KiB
C++
44 lines
1.7 KiB
C++
/*
|
|
* 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/modules/audio_processing/aec3/downsampled_render_buffer.h"
|
|
#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
|
|
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
|
|
#include "webrtc/rtc_base/array_view.h"
|
|
#include "webrtc/rtc_base/optional.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Class for aligning the render and capture signal using a RenderDelayBuffer.
|
|
class RenderDelayController {
|
|
public:
|
|
static RenderDelayController* Create(int sample_rate_hz);
|
|
virtual ~RenderDelayController() = default;
|
|
|
|
// Resets the delay controller.
|
|
virtual void Reset() = 0;
|
|
|
|
// Receives the externally used delay.
|
|
virtual void SetDelay(size_t render_delay) = 0;
|
|
|
|
// Aligns the render buffer content with the capture signal.
|
|
virtual size_t GetDelay(const DownsampledRenderBuffer& render_buffer,
|
|
rtc::ArrayView<const float> capture) = 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_
|