Mercurial > hg > aimc
view src/Modules/BMM/ModuleGammatone.cc @ 121:3cdaa81c3aca
- Massive refactoring to make module tree stuff work. In theory we now support configuration files again. The graphics stuff is untested as yet.
author | tomwalters |
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date | Mon, 18 Oct 2010 04:42:28 +0000 |
parents | c5f5e9569863 |
children | 9fcf55c040fe |
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// Copyright 2009-2010, Thomas Walters // // AIM-C: A C++ implementation of the Auditory Image Model // http://www.acousticscale.org/AIMC // // Licensed under the Apache License, Version 2.0 (the "License"); // you may not use this file except in compliance with the License. // You may obtain a copy of the License at // // http://www.apache.org/licenses/LICENSE-2.0 // // Unless required by applicable law or agreed to in writing, software // distributed under the License is distributed on an "AS IS" BASIS, // WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. // See the License for the specific language governing permissions and // limitations under the License. /*! \file * \brief Slaney's gammatone filterbank * * \author Thomas Walters <tom@acousticscale.org> * \date created 2009/11/13 * \version \$Id$ */ #include <cmath> #include <complex> #include "Support/ERBTools.h" #include "Modules/BMM/ModuleGammatone.h" namespace aimc { using std::vector; using std::complex; ModuleGammatone::ModuleGammatone(Parameters *params) : Module(params) { module_identifier_ = "gt"; module_type_ = "bmm"; module_description_ = "Gammatone filterbank (Slaney's IIR gammatone)"; module_version_ = "$Id$"; num_channels_ = parameters_->DefaultInt("gtfb.channel_count", 200); min_frequency_ = parameters_->DefaultFloat("gtfb.min_frequency", 86.0f); max_frequency_ = parameters_->DefaultFloat("gtfb.max_frequency", 16000.0f); } ModuleGammatone::~ModuleGammatone() { } void ModuleGammatone::ResetInternal() { state_1_.resize(num_channels_); state_2_.resize(num_channels_); state_3_.resize(num_channels_); state_4_.resize(num_channels_); for (int i = 0; i < num_channels_; ++i) { state_1_[i].clear(); state_1_[i].resize(3, 0.0f); state_2_[i].clear(); state_2_[i].resize(3, 0.0f); state_3_[i].clear(); state_3_[i].resize(3, 0.0f); state_4_[i].clear(); state_4_[i].resize(3, 0.0f); } } bool ModuleGammatone::InitializeInternal(const SignalBank& input) { // Calculate number of channels, and centre frequencies float erb_max = ERBTools::Freq2ERB(max_frequency_); float erb_min = ERBTools::Freq2ERB(min_frequency_); float delta_erb = (erb_max - erb_min) / (num_channels_ - 1); centre_frequencies_.resize(num_channels_); float erb_current = erb_min; output_.Initialize(num_channels_, input.buffer_length(), input.sample_rate()); for (int i = 0; i < num_channels_; ++i) { centre_frequencies_[i] = ERBTools::ERB2Freq(erb_current); erb_current += delta_erb; output_.set_centre_frequency(i, centre_frequencies_[i]); } a_.resize(num_channels_); b1_.resize(num_channels_); b2_.resize(num_channels_); b3_.resize(num_channels_); b4_.resize(num_channels_); state_1_.resize(num_channels_); state_2_.resize(num_channels_); state_3_.resize(num_channels_); state_4_.resize(num_channels_); for (int ch = 0; ch < num_channels_; ++ch) { double cf = centre_frequencies_[ch]; double erb = ERBTools::Freq2ERBw(cf); // LOG_INFO("%e", erb); // Sample interval double dt = 1.0f / input.sample_rate(); // Bandwidth parameter double b = 1.019f * 2.0f * M_PI * erb; // The following expressions are derived in Apple TR #35, "An // Efficient Implementation of the Patterson-Holdsworth Cochlear // Filter Bank" and used in Malcolm Slaney's auditory toolbox, where he // defines this alternaltive four stage cascade of second-order filters. // Calculate the gain: double cpt = cf * M_PI * dt; complex<double> exponent(0.0, 2.0 * cpt); complex<double> ec = exp(2.0 * exponent); complex<double> two_cf_pi_t(2.0 * cpt, 0.0); complex<double> two_pow(pow(2.0, (3.0 / 2.0)), 0.0); complex<double> p1 = -2.0 * ec * dt; complex<double> p2 = 2.0 * exp(-(b * dt) + exponent) * dt; complex<double> b_dt(b * dt, 0.0); double gain = abs( (p1 + p2 * (cos(two_cf_pi_t) - sqrt(3.0 - two_pow) * sin(two_cf_pi_t))) * (p1 + p2 * (cos(two_cf_pi_t) + sqrt(3.0 - two_pow) * sin(two_cf_pi_t))) * (p1 + p2 * (cos(two_cf_pi_t) - sqrt(3.0 + two_pow) * sin(two_cf_pi_t))) * (p1 + p2 * (cos(two_cf_pi_t) + sqrt(3.0 + two_pow) * sin(two_cf_pi_t))) / pow((-2.0 / exp(2.0 * b_dt) - 2.0 * ec + 2.0 * (1.0 + ec) / exp(b_dt)), 4)); // LOG_INFO("%e", gain); // The filter coefficients themselves: const int coeff_count = 3; a_[ch].resize(coeff_count, 0.0f); b1_[ch].resize(coeff_count, 0.0f); b2_[ch].resize(coeff_count, 0.0f); b3_[ch].resize(coeff_count, 0.0f); b4_[ch].resize(coeff_count, 0.0f); state_1_[ch].resize(coeff_count, 0.0f); state_2_[ch].resize(coeff_count, 0.0f); state_3_[ch].resize(coeff_count, 0.0f); state_4_[ch].resize(coeff_count, 0.0f); double B0 = dt; double B2 = 0.0f; double B11 = -(2.0f * dt * cos(2.0f * cf * M_PI * dt) / exp(b * dt) + 2.0f * sqrt(3 + pow(2.0f, 1.5f)) * dt * sin(2.0f * cf * M_PI * dt) / exp(b * dt)) / 2.0f; double B12 = -(2.0f * dt * cos(2.0f * cf * M_PI * dt) / exp(b * dt) - 2.0f * sqrt(3 + pow(2.0f, 1.5f)) * dt * sin(2.0f * cf * M_PI * dt) / exp(b * dt)) / 2.0f; double B13 = -(2.0f * dt * cos(2.0f * cf * M_PI * dt) / exp(b * dt) + 2.0f * sqrt(3 - pow(2.0f, 1.5f)) * dt * sin(2.0f * cf * M_PI * dt) / exp(b * dt)) / 2.0f; double B14 = -(2.0f * dt * cos(2.0f * cf * M_PI * dt) / exp(b * dt) - 2.0f * sqrt(3 - pow(2.0f, 1.5f)) * dt * sin(2.0f * cf * M_PI * dt) / exp(b * dt)) / 2.0f; a_[ch][0] = 1.0f; a_[ch][1] = -2.0f * cos(2.0f * cf * M_PI * dt) / exp(b * dt); a_[ch][2] = exp(-2.0f * b * dt); b1_[ch][0] = B0 / gain; b1_[ch][1] = B11 / gain; b1_[ch][2] = B2 / gain; b2_[ch][0] = B0; b2_[ch][1] = B12; b2_[ch][2] = B2; b3_[ch][0] = B0; b3_[ch][1] = B13; b3_[ch][2] = B2; b4_[ch][0] = B0; b4_[ch][1] = B14; b4_[ch][2] = B2; } return true; } void ModuleGammatone::Process(const SignalBank &input) { output_.set_start_time(input.start_time()); int audio_channel = 0; vector<vector<double> >::iterator b1 = b1_.begin(); vector<vector<double> >::iterator b2 = b2_.begin(); vector<vector<double> >::iterator b3 = b3_.begin(); vector<vector<double> >::iterator b4 = b4_.begin(); vector<vector<double> >::iterator a = a_.begin(); vector<vector<double> >::iterator s1 = state_1_.begin(); vector<vector<double> >::iterator s2 = state_2_.begin(); vector<vector<double> >::iterator s3 = state_3_.begin(); vector<vector<double> >::iterator s4 = state_4_.begin(); // Temporary storage between filter stages vector<double> out(input.buffer_length()); for (int ch = 0; ch < num_channels_; ++ch, ++b1, ++b2, ++b3, ++b4, ++a, ++s1, ++s2, ++s3, ++s4) { for (int i = 0; i < input.buffer_length(); ++i) { // Direct-form-II IIR filter double in = input.sample(audio_channel, i); out[i] = (*b1)[0] * in + (*s1)[0]; for (unsigned int stage = 1; stage < s1->size(); ++stage) (*s1)[stage - 1] = (*b1)[stage] * in - (*a)[stage] * out[i] + (*s1)[stage]; } for (int i = 0; i < input.buffer_length(); ++i) { // Direct-form-II IIR filter double in = out[i]; out[i] = (*b2)[0] * in + (*s2)[0]; for (unsigned int stage = 1; stage < s2->size(); ++stage) (*s2)[stage - 1] = (*b2)[stage] * in - (*a)[stage] * out[i] + (*s2)[stage]; } for (int i = 0; i < input.buffer_length(); ++i) { // Direct-form-II IIR filter double in = out[i]; out[i] = (*b3)[0] * in + (*s3)[0]; for (unsigned int stage = 1; stage < s3->size(); ++stage) (*s3)[stage - 1] = (*b3)[stage] * in - (*a)[stage] * out[i] + (*s3)[stage]; } for (int i = 0; i < input.buffer_length(); ++i) { // Direct-form-II IIR filter double in = out[i]; out[i] = (*b4)[0] * in + (*s4)[0]; for (unsigned int stage = 1; stage < s4->size(); ++stage) (*s4)[stage - 1] = (*b4)[stage] * in - (*a)[stage] * out[i] + (*s4)[stage]; output_.set_sample(ch, i, out[i]); } } PushOutput(); } } // namespace aimc