webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
Gustaf Ullberg ee84d39fce AEC3: Downmix multichannel signals before delay estimation
Multichannel signals are downmixed to mono before decimation and
delay estimation. This is useful when not all channels play
audio content. The feature can be toggled in the AEC3 configuration.

Bug: webrtc:10913
Change-Id: I7d40edf7732bb51fec69e7f3ca063d821c5069c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151762
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29126}
2019-09-10 08:16:07 +00:00

77 lines
2.7 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 MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
#include <stddef.h>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/aec3/clockdrift_detector.h"
#include "modules/audio_processing/aec3/decimator.h"
#include "modules/audio_processing/aec3/delay_estimate.h"
#include "modules/audio_processing/aec3/matched_filter.h"
#include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class ApmDataDumper;
struct DownsampledRenderBuffer;
struct EchoCanceller3Config;
// Estimates the delay of the echo path.
class EchoPathDelayEstimator {
public:
EchoPathDelayEstimator(ApmDataDumper* data_dumper,
const EchoCanceller3Config& config);
~EchoPathDelayEstimator();
// Resets the estimation. If the delay confidence is reset, the reset behavior
// is as if the call is restarted.
void Reset(bool reset_delay_confidence);
// Produce a delay estimate if such is avaliable.
absl::optional<DelayEstimate> EstimateDelay(
const DownsampledRenderBuffer& render_buffer,
const std::vector<std::vector<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_);
}
// Returns the level of detected clockdrift.
ClockdriftDetector::Level Clockdrift() const {
return clockdrift_detector_.ClockdriftLevel();
}
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_;
absl::optional<DelayEstimate> old_aggregated_lag_;
size_t consistent_estimate_counter_ = 0;
ClockdriftDetector clockdrift_detector_;
bool downmix_;
// Internal reset method with more granularity.
void Reset(bool reset_lag_aggregator, bool reset_delay_confidence);
RTC_DISALLOW_COPY_AND_ASSIGN(EchoPathDelayEstimator);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_