view src/Modules/BMM/ModuleGammatone.cc @ 611:0fbaf443ec82

Carfac C++ revision 3, indluding more style improvements. The output structs are now classes again, and have separate storage methods for each output structure along with flags in the Run and RunSegment methods to allow for only storing NAPs if desired.

// 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 <[e-mail]>
 *  \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