Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
This commit is contained in:
@ -0,0 +1,50 @@
|
||||
/*
|
||||
* 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 MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
|
||||
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
|
||||
|
||||
#include "absl/types/optional.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/audio/echo_canceller3_config.h"
|
||||
#include "modules/audio_processing/aec3/delay_estimate.h"
|
||||
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
|
||||
#include "modules/audio_processing/aec3/render_delay_buffer.h"
|
||||
#include "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(const EchoCanceller3Config& config,
|
||||
int sample_rate_hz,
|
||||
size_t num_capture_channels);
|
||||
virtual ~RenderDelayController() = default;
|
||||
|
||||
// Resets the delay controller. If the delay confidence is reset, the reset
|
||||
// behavior is as if the call is restarted.
|
||||
virtual void Reset(bool reset_delay_confidence) = 0;
|
||||
|
||||
// Logs a render call.
|
||||
virtual void LogRenderCall() = 0;
|
||||
|
||||
// Aligns the render buffer content with the capture signal.
|
||||
virtual absl::optional<DelayEstimate> GetDelay(
|
||||
const DownsampledRenderBuffer& render_buffer,
|
||||
size_t render_delay_buffer_delay,
|
||||
const std::vector<std::vector<float>>& capture) = 0;
|
||||
|
||||
// Returns true if clockdrift has been detected.
|
||||
virtual bool HasClockdrift() const = 0;
|
||||
};
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
|
Reference in New Issue
Block a user