webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
Per Åhgren a76ef9d0b4 Robustify the faster alignment in AEC3 to avoid resets
The faster AEC3 alignment introduced recently may in
cases cause the alignment (and the AEC3) to repeatedly
reset. This CL avoids these resets by handling buffer
issues (which are triggering the resets) separately
during the initial coarse alignment phase.



Change-Id: Idf5e2ffda2591906da8060d03ec8ca73cdaedf53
Bug: webrtc:8798,chromium:805815
Reviewed-on: https://webrtc-review.googlesource.com/43480
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21758}
2018-01-25 09:57:31 +00:00

62 lines
2.1 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 MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
#include <vector>
#include "api/optional.h"
#include "modules/audio_processing/aec3/decimator.h"
#include "modules/audio_processing/aec3/delay_estimate.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/matched_filter.h"
#include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
class ApmDataDumper;
// Estimates the delay of the echo path.
class EchoPathDelayEstimator {
public:
EchoPathDelayEstimator(ApmDataDumper* data_dumper,
const EchoCanceller3Config& config);
~EchoPathDelayEstimator();
// Resets the estimation.
void Reset();
// Produce a delay estimate if such is avaliable.
rtc::Optional<DelayEstimate> EstimateDelay(
const DownsampledRenderBuffer& render_buffer,
rtc::ArrayView<const float> capture);
// Log delay estimator properties.
void LogDelayEstimationProperties(int sample_rate_hz, size_t shift) const {
matched_filter_.LogFilterProperties(sample_rate_hz, shift,
down_sampling_factor_);
}
private:
ApmDataDumper* const data_dumper_;
const size_t down_sampling_factor_;
const size_t sub_block_size_;
Decimator capture_decimator_;
MatchedFilter matched_filter_;
MatchedFilterLagAggregator matched_filter_lag_aggregator_;
RTC_DISALLOW_COPY_AND_ASSIGN(EchoPathDelayEstimator);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_