webrtc/modules/audio_processing/aec3/render_delay_controller.h
Per Åhgren 5c532d3774 Robustification of the echo suppression behavior during headset usage.
This CL robustifies the echo removal behavior when headsets are used.
In particular it:
-Introduces a secondary, more refined alignment when no alignment can
be found using the delay estimator.
-Changes decision logic for when to use the linear filter output.
-Changes the decision logic for when to be transparent.
-Changes the way that the transparent mode works.
-Makes the nonlinear mode less aggressive.
-Removes the detector for non-audible echoes.
-Makes the attenuation when there are signals with strong narrowband
characteristics more mild in scenarios with low render.

Furthermore the CL:
-Removes the input of external echo leakage information.


Bug: webrtc:9047,chromium:824111,webrtc:8314,webrtc:8671,webrtc:5201,webrtc:5919
Change-Id: Ied1fe0c0a35d3c31b47606ed2db319a73644d406
Reviewed-on: https://webrtc-review.googlesource.com/60866
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22548}
2018-03-22 00:23:23 +00:00

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1.8 KiB
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/*
* 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 "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "api/optional.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 non_causal_offset,
int sample_rate_hz);
virtual ~RenderDelayController() = default;
// Resets the delay controller.
virtual void Reset() = 0;
// Logs a render call.
virtual void LogRenderCall() = 0;
// Aligns the render buffer content with the capture signal.
virtual rtc::Optional<DelayEstimate> GetDelay(
const DownsampledRenderBuffer& render_buffer,
size_t render_delay_buffer_delay,
const rtc::Optional<int>& echo_remover_delay,
rtc::ArrayView<const float> capture) = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_