Added new preset filters

pull/1/head
Edmund Blackaddr 7 years ago
parent 354cd93bab
commit df6c4c5e38
  1. 7
      examples/Delay/AnalogDelayDemo/AnalogDelayDemo.ino
  2. 25
      src/AudioEffectAnalogDelay.h
  3. 29
      src/effects/AudioEffectAnalogDelay.cpp
  4. 83
      src/effects/AudioEffectAnalogDelayFilters.h

@ -94,6 +94,13 @@ void setup() {
analogDelay.mix(0.5f); analogDelay.mix(0.5f);
analogDelay.feedback(0.0f); analogDelay.feedback(0.0f);
//////////////////////////////////
// AnalogDelay filter selection //
// Uncomment to tryout the 3 different built-in filters.
//analogDelay.setFilter(AudioEffectAnalogDelay::Filter::DM3); // The default filter. Naturally bright echo (highs stay, lows fade away)
//analogDelay.setFilter(AudioEffectAnalogDelay::Filter::WARM); // A warm filter with a smooth frequency rolloff above 2Khz
//analogDelay.setFilter(AudioEffectAnalogDelay::Filter::DARK); // A very dark filter, with a sharp rolloff above 1Khz
// Setup 2-stages of LPF, cutoff 4500 Hz, Q-factor 0.7071 (a 'normal' Q-factor) // Setup 2-stages of LPF, cutoff 4500 Hz, Q-factor 0.7071 (a 'normal' Q-factor)
cabFilter.setLowpass(0, 4500, .7071); cabFilter.setLowpass(0, 4500, .7071);
cabFilter.setLowpass(1, 4500, .7071); cabFilter.setLowpass(1, 4500, .7071);

@ -30,10 +30,6 @@
namespace BAGuitar { namespace BAGuitar {
/// The number of stages in the analog-response Biquad filter
constexpr unsigned MAX_NUM_FILTER_STAGES = 4;
constexpr unsigned NUM_COEFFS_PER_STAGE = 5;
/**************************************************************************//** /**************************************************************************//**
* AudioEffectAnalogDelay models BBD based analog delays. It provides controls * AudioEffectAnalogDelay models BBD based analog delays. It provides controls
* for delay, feedback (or regen), mix and output level. All parameters can be * for delay, feedback (or regen), mix and output level. All parameters can be
@ -54,6 +50,12 @@ public:
NUM_CONTROLS ///< this can be used as an alias for the number of MIDI controls NUM_CONTROLS ///< this can be used as an alias for the number of MIDI controls
}; };
enum class Filter {
DM3 = 0,
WARM,
DARK
};
// *** CONSTRUCTORS *** // *** CONSTRUCTORS ***
AudioEffectAnalogDelay() = delete; AudioEffectAnalogDelay() = delete;
@ -133,9 +135,17 @@ public:
/// @param value the CC value from 0 to 127 /// @param value the CC value from 0 to 127
void processMidi(int channel, int midiCC, int value); void processMidi(int channel, int midiCC, int value);
// ** FILTER COEFFICIENTS **
/// Set the filter coefficients to one of the presets. See AudioEffectAnalogDelay::Filter
/// for options.
/// @details See AudioEffectAnalogDelayFIlters.h for more details.
/// @param filter the preset filter. E.g. AudioEffectAnalogDelay::Filter::WARM
void setFilter(Filter filter);
/// Override the default coefficients with your own. The number of filters stages affects how /// Override the default coefficients with your own. The number of filters stages affects how
/// much CPU is consumed. /// much CPU is consumed.
/// @details The effect uses the CMSIS-DSP library for biquads which requires coefficents /// @details The effect uses the CMSIS-DSP library for biquads which requires coefficents.
/// be in q31 format, which means they are 32-bit signed integers representing -1.0 to slightly /// be in q31 format, which means they are 32-bit signed integers representing -1.0 to slightly
/// less than +1.0. The coeffShift parameter effectively multiplies the coefficients by 2^shift. <br> /// less than +1.0. The coeffShift parameter effectively multiplies the coefficients by 2^shift. <br>
/// Example: If you really want +1.5, must instead use +0.75 * 2^1, thus 0.75 in q31 format is /// Example: If you really want +1.5, must instead use +0.75 * 2^1, thus 0.75 in q31 format is
@ -171,11 +181,6 @@ private:
// Coefficients // Coefficients
void m_constructFilter(void); void m_constructFilter(void);
// int m_numStages;
// int m_coeffShift;
// int m_coeffs[MAX_NUM_FILTER_STAGES*NUM_COEFFS_PER_STAGE] = {};
//size_t m_callCount = 0;
}; };
} }

@ -5,6 +5,7 @@
* Author: slascos * Author: slascos
*/ */
#include <new> #include <new>
#include "AudioEffectAnalogDelayFilters.h"
#include "AudioEffectAnalogDelay.h" #include "AudioEffectAnalogDelay.h"
namespace BAGuitar { namespace BAGuitar {
@ -12,16 +13,6 @@ namespace BAGuitar {
constexpr int MIDI_CHANNEL = 0; constexpr int MIDI_CHANNEL = 0;
constexpr int MIDI_CONTROL = 1; constexpr int MIDI_CONTROL = 1;
// BOSS DM-3 Filters
constexpr unsigned DM3_COEFF_SHIFT = 2;
constexpr int32_t DM3[5*MAX_NUM_FILTER_STAGES] = {
536870912, 988616936, 455608573, 834606945, -482959709,
536870912, 1031466345, 498793368, 965834205, -467402235,
536870912, 1105821939, 573646688, 928470657, -448083489,
2339, 5093, 2776, 302068995, 4412722
};
AudioEffectAnalogDelay::AudioEffectAnalogDelay(float maxDelayMs) AudioEffectAnalogDelay::AudioEffectAnalogDelay(float maxDelayMs)
: AudioStream(1, m_inputQueueArray) : AudioStream(1, m_inputQueueArray)
{ {
@ -58,7 +49,7 @@ AudioEffectAnalogDelay::~AudioEffectAnalogDelay()
void AudioEffectAnalogDelay::m_constructFilter(void) void AudioEffectAnalogDelay::m_constructFilter(void)
{ {
// Use DM3 coefficients by default // Use DM3 coefficients by default
m_iir = new IirBiQuadFilterHQ(MAX_NUM_FILTER_STAGES, reinterpret_cast<const int32_t *>(&DM3), DM3_COEFF_SHIFT); m_iir = new IirBiQuadFilterHQ(DM3_NUM_STAGES, reinterpret_cast<const int32_t *>(&DM3), DM3_COEFF_SHIFT);
} }
void AudioEffectAnalogDelay::setFilterCoeffs(int numStages, const int32_t *coeffs, int coeffShift) void AudioEffectAnalogDelay::setFilterCoeffs(int numStages, const int32_t *coeffs, int coeffShift)
@ -66,6 +57,22 @@ void AudioEffectAnalogDelay::setFilterCoeffs(int numStages, const int32_t *coeff
m_iir->changeFilterCoeffs(numStages, coeffs, coeffShift); m_iir->changeFilterCoeffs(numStages, coeffs, coeffShift);
} }
void AudioEffectAnalogDelay::setFilter(Filter filter)
{
switch(filter) {
case Filter::WARM :
m_iir->changeFilterCoeffs(WARM_NUM_STAGES, reinterpret_cast<const int32_t *>(&WARM), WARM_COEFF_SHIFT);
break;
case Filter::DARK :
m_iir->changeFilterCoeffs(DARK_NUM_STAGES, reinterpret_cast<const int32_t *>(&DARK), DARK_COEFF_SHIFT);
break;
case Filter::DM3 :
default:
m_iir->changeFilterCoeffs(DM3_NUM_STAGES, reinterpret_cast<const int32_t *>(&DM3), DM3_COEFF_SHIFT);
break;
}
}
void AudioEffectAnalogDelay::update(void) void AudioEffectAnalogDelay::update(void)
{ {
audio_block_t *inputAudioBlock = receiveReadOnly(); // get the next block of input samples audio_block_t *inputAudioBlock = receiveReadOnly(); // get the next block of input samples

@ -0,0 +1,83 @@
/**************************************************************************//**
* @file
* @author Steve Lascos
* @company Blackaddr Audio
*
* This file constains precomputed co-efficients for the AudioEffectAnalogDelay
* class.
*
* @copyright This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.*
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*****************************************************************************/
#include <cstdint>
namespace BAGuitar {
// The number of stages in the analog-response Biquad filter
constexpr unsigned MAX_NUM_FILTER_STAGES = 4;
constexpr unsigned NUM_COEFFS_PER_STAGE = 5;
// Matlab/Octave can be helpful to design a filter. Once you have the IIR filter (bz,az) coefficients
// in the z-domain, they can be converted to second-order-sections. AudioEffectAnalogDelay is designed
// to accept up to a maximum of an 8th order filter, broken into four, 2nd order stages.
//
// Second order sections can be created with:
// [sos] = tf2sos(bz,az);
// The results coefficents must be converted the Q31 format required by the ARM CMSIS-DSP library. This means
// all coefficients must lie between -1.0 and +0.9999. If your (bz,az) coefficients exceed this, you must divide
// them down by a power of 2. For example, if your largest magnitude coefficient is -3.5, you must divide by
// 2^shift where 4=2^2 and thus shift = 2. You must then mutliply by 2^31 to get a 32-bit signed integer value
// that represents the required Q31 coefficient.
// BOSS DM-3 Filters
// b(z) = 1.0e-03 * (0.0032 0.0257 0.0900 0.1800 0.2250 0.1800 0.0900 0.0257 0.0032)
// a(z) = 1.0000 -5.7677 14.6935 -21.3811 19.1491 -10.5202 3.2584 -0.4244 -0.0067
constexpr unsigned DM3_NUM_STAGES = 4;
constexpr unsigned DM3_COEFF_SHIFT = 2;
constexpr int32_t DM3[5*MAX_NUM_FILTER_STAGES] = {
536870912, 988616936, 455608573, 834606945, -482959709,
536870912, 1031466345, 498793368, 965834205, -467402235,
536870912, 1105821939, 573646688, 928470657, -448083489,
2339, 5093, 2776, 302068995, 4412722
};
// Blackaddr WARM Filter
// Butterworth, 8th order, cutoff = 2000 Hz
// Matlab/Octave command: [bz, az] = butter(8, 2000/44100/2);
// b(z) = 1.0e-05 * (0.0086 0.0689 0.2411 0.4821 0.6027 0.4821 0.2411 0.0689 0.0086_
// a(z) = 1.0000 -6.5399 18.8246 -31.1340 32.3473 -21.6114 9.0643 -2.1815 0.2306
constexpr unsigned WARM_NUM_STAGES = 4;
constexpr unsigned WARM_COEFF_SHIFT = 2;
constexpr int32_t WARM[5*MAX_NUM_FILTER_STAGES] = {
536870912,1060309346,523602393,976869875,-481046241,
536870912,1073413910,536711084,891250612,-391829326,
536870912,1087173998,550475248,835222426,-333446881,
46,92,46,807741349,-304811072
};
// Blackaddr DARK Filter
// Chebychev Type II, 8th order, stopband = 60db, cutoff = 1000 Hz
// Matlab command: [bz, az] = cheby2(8, 60, 1000/44100/2);
// b(z) = 0.0009 -0.0066 0.0219 -0.0423 0.0522 -0.0423 0.0219 -0.0066 0.0009
// a(z) = 1.0000 -7.4618 24.3762 -45.5356 53.1991 -39.8032 18.6245 -4.9829 0.5836
constexpr unsigned DARK_NUM_STAGES = 4;
constexpr unsigned DARK_COEFF_SHIFT = 1;
constexpr int32_t DARK[5*MAX_NUM_FILTER_STAGES] = {
1073741824,-2124867808,1073741824,2107780229,-1043948409,
1073741824,-2116080466,1073741824,2042553796,-979786242,
1073741824,-2077777790,1073741824,1964779896,-904264933,
957356,-1462833,957356,1896884898,-838694612
};
};
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