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This CL adds code for doing signal-dependent downmixing before the delay estimation in the multichannel case. As part of the CL, the unittests of the render delay controller are corrected. However, as that caused some of them to fail, the CL (for now) as well disables the failing test. Bug: webrtc:11153,chromium:1029740, webrtc:11161 Change-Id: I0b765c28fa5e547aabd6dfbd24b626ff9a16346f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161045 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29980}
79 lines
2.8 KiB
C++
79 lines
2.8 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
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#include <stddef.h>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "modules/audio_processing/aec3/alignment_mixer.h"
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#include "modules/audio_processing/aec3/clockdrift_detector.h"
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#include "modules/audio_processing/aec3/decimator.h"
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#include "modules/audio_processing/aec3/delay_estimate.h"
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#include "modules/audio_processing/aec3/matched_filter.h"
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#include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class ApmDataDumper;
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struct DownsampledRenderBuffer;
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struct EchoCanceller3Config;
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// Estimates the delay of the echo path.
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class EchoPathDelayEstimator {
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public:
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EchoPathDelayEstimator(ApmDataDumper* data_dumper,
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const EchoCanceller3Config& config,
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size_t num_capture_channels);
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~EchoPathDelayEstimator();
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// Resets the estimation. If the delay confidence is reset, the reset behavior
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// is as if the call is restarted.
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void Reset(bool reset_delay_confidence);
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// Produce a delay estimate if such is avaliable.
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absl::optional<DelayEstimate> EstimateDelay(
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const DownsampledRenderBuffer& render_buffer,
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const std::vector<std::vector<float>>& capture);
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// Log delay estimator properties.
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void LogDelayEstimationProperties(int sample_rate_hz, size_t shift) const {
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matched_filter_.LogFilterProperties(sample_rate_hz, shift,
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down_sampling_factor_);
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}
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// Returns the level of detected clockdrift.
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ClockdriftDetector::Level Clockdrift() const {
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return clockdrift_detector_.ClockdriftLevel();
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}
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private:
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ApmDataDumper* const data_dumper_;
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const size_t down_sampling_factor_;
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const size_t sub_block_size_;
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AlignmentMixer capture_mixer_;
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Decimator capture_decimator_;
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MatchedFilter matched_filter_;
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MatchedFilterLagAggregator matched_filter_lag_aggregator_;
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absl::optional<DelayEstimate> old_aggregated_lag_;
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size_t consistent_estimate_counter_ = 0;
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ClockdriftDetector clockdrift_detector_;
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// Internal reset method with more granularity.
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void Reset(bool reset_lag_aggregator, bool reset_delay_confidence);
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RTC_DISALLOW_COPY_AND_ASSIGN(EchoPathDelayEstimator);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
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