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1 // Copyright 2007-2010, Thomas Walters
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2 //
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3 // AIM-C: A C++ implementation of the Auditory Image Model
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4 // http://www.acousticscale.org/AIMC
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5 //
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6 // This program is free software: you can redistribute it and/or modify
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7 // it under the terms of the GNU General Public License as published by
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8 // the Free Software Foundation, either version 3 of the License, or
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9 // (at your option) any later version.
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10 //
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11 // This program is distributed in the hope that it will be useful,
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12 // but WITHOUT ANY WARRANTY; without even the implied warranty of
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13 // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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14 // GNU General Public License for more details.
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15 //
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16 // You should have received a copy of the GNU General Public License
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17 // along with this program. If not, see <http://www.gnu.org/licenses/>.
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18
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19 /*!
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20 * \file
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21 * \brief Halfwave rectification, compression and lowpass filtering.
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22 *
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23 * \author Tom Walters <tcw24@cam.ac.uk>
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24 * \date created 2007/03/07
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25 * \version \$Id: ModuleHCL.cc 4 2010-02-03 18:44:58Z tcw $
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26 */
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27
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28 #include <math.h>
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29
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30 #include "Modules/NAP/ModuleHCL.h"
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31
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32 namespace aimc {
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33 ModuleHCL::ModuleHCL(Parameters *parameters) : Module(parameters) {
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34 module_identifier_ = "hcl";
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35 module_type_ = "nap";
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36 module_description_ = "Halfwave rectification, compression "
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37 "and lowpass filtering";
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38 module_version_ = "$Id: ModuleHCL.cc 4 2010-02-03 18:44:58Z tcw $";
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39
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40 do_lowpass_ = parameters_->DefaultBool("nap.do_lowpass", false);
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41 do_log_ = parameters_->DefaultBool("nap.do_log_compression", false);
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42 lowpass_cutoff_ = parameters_->DefaultFloat("nap.lowpass_cutoff", 1200.0);
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43 lowpass_order_ = parameters_->DefaultInt("nap.lowpass_order", 2);
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44 }
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45
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46 ModuleHCL::~ModuleHCL() {
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47 }
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48
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49 bool ModuleHCL::InitializeInternal(const SignalBank &input) {
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50 time_constant_ = 1.0f / (2.0f * M_PI * lowpass_cutoff_);
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51 channel_count_ = input.channel_count();
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52 output_.Initialize(input);
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53 ResetInternal();
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54 return true;
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55 }
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56
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57 void ModuleHCL::ResetInternal() {
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58 xn_ = 0.0f;
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59 yn_ = 0.0f;
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60 yns_.clear();
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61 yns_.resize(channel_count_);
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62 for (int c = 0; c < channel_count_; ++c) {
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63 yns_[c].resize(lowpass_order_, 0.0f);
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64 }
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65 }
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66
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67 /* With do_log, the signal is first scaled up so that values <1.0 become
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68 * negligible. This just rescales the sample values to fill the range of a
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69 * 16-bit signed integer, then we lose the bottom bit of resolution. If the
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70 * signal was sampled at 16-bit resolution, there shouldn't be anything to
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71 * speak of there anyway. If it was sampled using a higher resolution, then
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72 * some data will be discarded.
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73 */
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74 void ModuleHCL::Process(const SignalBank &input) {
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75 output_.set_start_time(input.start_time());
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76 for (int c = 0; c < input.channel_count(); ++c) {
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77 for (int i = 0; i < input.buffer_length(); ++i) {
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78 if (input[c][i] < 0.0f) {
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79 output_.set_sample(c, i, 0.0f);
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80 } else {
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81 float s = input[c][i];
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82 if (do_log_) {
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83 s *= pow(2.0f,15);
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84 if (s < 1.0f) s = 1.0f;
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85 s = 20.0f * log10(s);
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86 }
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87 output_.set_sample(c, i, s);
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88 }
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89 }
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90 if (do_lowpass_) {
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91 float b = exp( -1.0f / (input.sample_rate() * time_constant_));
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92 float gain = 1.0f / (1.0f - b);
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93 for (int j = 0; j < lowpass_order_; j++) {
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94 for (int k = 0; k < output_.buffer_length(); ++k) {
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95 xn_ = output_[c][k];
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96 yn_ = xn_ + b * yns_[c][j];
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97 yns_[c][j] = yn_;
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98 output_.set_sample(c, k, yn_ / gain);
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99 }
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100 }
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101 }
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102 }
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103 PushOutput();
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104 }
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105 } // namespace aimc
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