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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_MATCHED_FILTER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
#include <stddef.h>
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "rtc_base/system/arch.h"
namespace webrtc {
class ApmDataDumper;
struct DownsampledRenderBuffer;
namespace aec3 {
#if defined(WEBRTC_HAS_NEON)
// Filter core for the matched filter that is optimized for NEON.
void MatchedFilterCore_NEON(size_t x_start_index,
float x2_sum_threshold,
float smoothing,
rtc::ArrayView<const float> x,
rtc::ArrayView<const float> y,
rtc::ArrayView<float> h,
bool* filters_updated,
float* error_sum);
#endif
#if defined(WEBRTC_ARCH_X86_FAMILY)
// Filter core for the matched filter that is optimized for SSE2.
void MatchedFilterCore_SSE2(size_t x_start_index,
float x2_sum_threshold,
float smoothing,
rtc::ArrayView<const float> x,
rtc::ArrayView<const float> y,
rtc::ArrayView<float> h,
bool* filters_updated,
float* error_sum);
// Filter core for the matched filter that is optimized for AVX2.
void MatchedFilterCore_AVX2(size_t x_start_index,
float x2_sum_threshold,
float smoothing,
rtc::ArrayView<const float> x,
rtc::ArrayView<const float> y,
rtc::ArrayView<float> h,
bool* filters_updated,
float* error_sum);
#endif
// Filter core for the matched filter.
void MatchedFilterCore(size_t x_start_index,
float x2_sum_threshold,
float smoothing,
rtc::ArrayView<const float> x,
rtc::ArrayView<const float> y,
rtc::ArrayView<float> h,
bool* filters_updated,
float* error_sum);
} // namespace aec3
// Produces recursively updated cross-correlation estimates for several signal
// shifts where the intra-shift spacing is uniform.
class MatchedFilter {
public:
// Stores properties for the lag estimate corresponding to a particular signal
// shift.
struct LagEstimate {
LagEstimate() = default;
LagEstimate(float accuracy, bool reliable, size_t lag, bool updated)
: accuracy(accuracy), reliable(reliable), lag(lag), updated(updated) {}
float accuracy = 0.f;
bool reliable = false;
size_t lag = 0;
bool updated = false;
};
MatchedFilter(ApmDataDumper* data_dumper,
Aec3Optimization optimization,
size_t sub_block_size,
size_t window_size_sub_blocks,
int num_matched_filters,
size_t alignment_shift_sub_blocks,
float excitation_limit,
float smoothing,
float matching_filter_threshold);
MatchedFilter() = delete;
MatchedFilter(const MatchedFilter&) = delete;
MatchedFilter& operator=(const MatchedFilter&) = delete;
~MatchedFilter();
// Updates the correlation with the values in the capture buffer.
void Update(const DownsampledRenderBuffer& render_buffer,
rtc::ArrayView<const float> capture);
// Resets the matched filter.
void Reset();
// Returns the current lag estimates.
rtc::ArrayView<const MatchedFilter::LagEstimate> GetLagEstimates() const {
return lag_estimates_;
}
// Returns the maximum filter lag.
size_t GetMaxFilterLag() const {
return filters_.size() * filter_intra_lag_shift_ + filters_[0].size();
}
// Log matched filter properties.
void LogFilterProperties(int sample_rate_hz,
size_t shift,
size_t downsampling_factor) const;
private:
ApmDataDumper* const data_dumper_;
const Aec3Optimization optimization_;
const size_t sub_block_size_;
const size_t filter_intra_lag_shift_;
std::vector<std::vector<float>> filters_;
std::vector<LagEstimate> lag_estimates_;
std::vector<size_t> filters_offsets_;
const float excitation_limit_;
const float smoothing_;
const float matching_filter_threshold_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
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