webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
Per Åhgren 895ae9a0cd Improving the speed of the delay estimator in AEC3
This CL significantly improves the response time
of the AEC3 delay estimator to audio buffer issues.

The CL adds ensures that the delay estimator
correlators reacts to buffer issues from the
zero state which is much faster than if it has already
achieved a state matching a previous alignment.

The CL has been extensively tested on offline
recordings.

Bug: webrtc:9023, chromium:822245
Change-Id: Ic149b9429e592d4c3535eb8432582f435a1b4745
Reviewed-on: https://webrtc-review.googlesource.com/62081
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22461}
2018-03-15 16:38:07 +00:00

65 lines
2.3 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/audio/echo_canceller3_config.h"
#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 "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. If the soft-reset is specified, only the matched
// filters are reset.
void Reset(bool soft_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::Optional<DelayEstimate> old_aggregated_lag_;
size_t consistent_estimate_counter_ = 0;
RTC_DISALLOW_COPY_AND_ASSIGN(EchoPathDelayEstimator);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_