| /* |
| * Copyright (c) 2015 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. |
| */ |
| |
| #include "webrtc/common_audio/sparse_fir_filter.h" |
| |
| #include "webrtc/rtc_base/checks.h" |
| |
| namespace webrtc { |
| |
| SparseFIRFilter::SparseFIRFilter(const float* nonzero_coeffs, |
| size_t num_nonzero_coeffs, |
| size_t sparsity, |
| size_t offset) |
| : sparsity_(sparsity), |
| offset_(offset), |
| nonzero_coeffs_(nonzero_coeffs, nonzero_coeffs + num_nonzero_coeffs), |
| state_(sparsity_ * (num_nonzero_coeffs - 1) + offset_, 0.f) { |
| RTC_CHECK_GE(num_nonzero_coeffs, 1); |
| RTC_CHECK_GE(sparsity, 1); |
| } |
| |
| SparseFIRFilter::~SparseFIRFilter() = default; |
| |
| void SparseFIRFilter::Filter(const float* in, size_t length, float* out) { |
| // Convolves the input signal |in| with the filter kernel |nonzero_coeffs_| |
| // taking into account the previous state. |
| for (size_t i = 0; i < length; ++i) { |
| out[i] = 0.f; |
| size_t j; |
| for (j = 0; i >= j * sparsity_ + offset_ && |
| j < nonzero_coeffs_.size(); ++j) { |
| out[i] += in[i - j * sparsity_ - offset_] * nonzero_coeffs_[j]; |
| } |
| for (; j < nonzero_coeffs_.size(); ++j) { |
| out[i] += state_[i + (nonzero_coeffs_.size() - j - 1) * sparsity_] * |
| nonzero_coeffs_[j]; |
| } |
| } |
| |
| // Update current state. |
| if (state_.size() > 0u) { |
| if (length >= state_.size()) { |
| std::memcpy(&state_[0], |
| &in[length - state_.size()], |
| state_.size() * sizeof(*in)); |
| } else { |
| std::memmove(&state_[0], |
| &state_[length], |
| (state_.size() - length) * sizeof(state_[0])); |
| std::memcpy(&state_[state_.size() - length], in, length * sizeof(*in)); |
| } |
| } |
| } |
| |
| } // namespace webrtc |