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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
73 lines
2.1 KiB
C++
73 lines
2.1 KiB
C++
/*
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* Copyright (c) 2013 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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#include "modules/audio_processing/transient/wpd_node.h"
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#include <math.h>
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#include <string.h>
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#include "common_audio/fir_filter.h"
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#include "common_audio/fir_filter_factory.h"
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#include "modules/audio_processing/transient/dyadic_decimator.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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WPDNode::WPDNode(size_t length,
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const float* coefficients,
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size_t coefficients_length)
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: // The data buffer has parent data length to be able to contain and
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// filter it.
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data_(new float[2 * length + 1]),
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length_(length),
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filter_(
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CreateFirFilter(coefficients, coefficients_length, 2 * length + 1)) {
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RTC_DCHECK_GT(length, 0);
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RTC_DCHECK(coefficients);
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RTC_DCHECK_GT(coefficients_length, 0);
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memset(data_.get(), 0.f, (2 * length + 1) * sizeof(data_[0]));
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}
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WPDNode::~WPDNode() {}
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int WPDNode::Update(const float* parent_data, size_t parent_data_length) {
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if (!parent_data || (parent_data_length / 2) != length_) {
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return -1;
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}
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// Filter data.
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filter_->Filter(parent_data, parent_data_length, data_.get());
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// Decimate data.
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const bool kOddSequence = true;
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size_t output_samples = DyadicDecimate(data_.get(), parent_data_length,
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kOddSequence, data_.get(), length_);
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if (output_samples != length_) {
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return -1;
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}
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// Get abs to all values.
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for (size_t i = 0; i < length_; ++i) {
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data_[i] = fabs(data_[i]);
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}
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return 0;
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}
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int WPDNode::set_data(const float* new_data, size_t length) {
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if (!new_data || length != length_) {
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return -1;
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}
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memcpy(data_.get(), new_data, length * sizeof(data_[0]));
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return 0;
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}
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} // namespace webrtc
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