Removing not used files.

Added a wy to not use the Reverb-Anitalias-Filter and tried the one mentioned in https://github.com/PaulStoffregen/Audio/pull/357/files.
pull/32/head
Holger Wirtz 4 years ago
parent 69fbd6a4c0
commit 0c27d1bb60
  1. 9
      MicroDexed.ino
  2. 16
      config.h
  3. 144
      effect_dynamics.cpp
  4. 194
      effect_dynamics.h

@ -61,9 +61,11 @@ AudioMixer4 reverb_mixer_r;
AudioMixer4 reverb_mixer_l; AudioMixer4 reverb_mixer_l;
AudioEffectFreeverb freeverb_r; AudioEffectFreeverb freeverb_r;
AudioEffectFreeverb freeverb_l; AudioEffectFreeverb freeverb_l;
#if defined(REVERB_ANTIALIAS_FRQ)
AudioFilterBiquad freeverb_antialias_r; AudioFilterBiquad freeverb_antialias_r;
AudioFilterBiquad freeverb_antialias_l; AudioFilterBiquad freeverb_antialias_l;
#endif #endif
#endif
AudioMixer4 master_mixer_r; AudioMixer4 master_mixer_r;
AudioMixer4 master_mixer_l; AudioMixer4 master_mixer_l;
AudioAmplifier volume_r; AudioAmplifier volume_r;
@ -76,10 +78,15 @@ AudioAnalyzePeak master_peak_l;
#if defined(USE_FX) #if defined(USE_FX)
AudioConnection patchCord0(reverb_mixer_r, freeverb_r); AudioConnection patchCord0(reverb_mixer_r, freeverb_r);
AudioConnection patchCord1(reverb_mixer_l, freeverb_l); AudioConnection patchCord1(reverb_mixer_l, freeverb_l);
#if defined(REVERB_ANTIALIAS_FRQ)
AudioConnection patchCord2(freeverb_r, freeverb_antialias_r); AudioConnection patchCord2(freeverb_r, freeverb_antialias_r);
AudioConnection patchCord3(freeverb_l, freeverb_antialias_l); AudioConnection patchCord3(freeverb_l, freeverb_antialias_l);
AudioConnection patchCord4(freeverb_antialias_r, 0, master_mixer_r, 3); AudioConnection patchCord4(freeverb_antialias_r, 0, master_mixer_r, 3);
AudioConnection patchCord5(freeverb_antialias_l, 0, master_mixer_l, 3); AudioConnection patchCord5(freeverb_antialias_l, 0, master_mixer_l, 3);
#else
AudioConnection patchCord2(freeverb_r, 0, master_mixer_r, 3);
AudioConnection patchCord3(freeverb_l, 0, master_mixer_l, 3);
#endif
#endif #endif
AudioConnection patchCord6(master_mixer_r, volume_r); AudioConnection patchCord6(master_mixer_r, volume_r);
AudioConnection patchCord7(master_mixer_l, volume_l); AudioConnection patchCord7(master_mixer_l, volume_l);
@ -1466,7 +1473,7 @@ void initial_values_from_eeprom(bool init)
} }
check_configuration(); check_configuration();
for (uint8_t instance_id = 0; instance_id < NUM_DEXED; instance_id++) for (uint8_t instance_id = 0; instance_id < NUM_DEXED; instance_id++)
{ {
set_voiceconfig_params(instance_id); set_voiceconfig_params(instance_id);

@ -56,7 +56,7 @@
// sed -i.orig 's/^#define USB_MIDI_SYSEX_MAX 290/#define USB_MIDI_SYSEX_MAX 4104/' /usr/local/arduino-teensy/hardware/teensy/avr/cores/teensy3/usb_midi.h // sed -i.orig 's/^#define USB_MIDI_SYSEX_MAX 290/#define USB_MIDI_SYSEX_MAX 4104/' /usr/local/arduino-teensy/hardware/teensy/avr/cores/teensy3/usb_midi.h
//#define USB_MIDI_SYSEX_MAX 4104 //#define USB_MIDI_SYSEX_MAX 4104
#define VERSION "1.0.3" #define VERSION "1.0.4"
//************************************************************************************************* //*************************************************************************************************
//* DEVICE SETTINGS //* DEVICE SETTINGS
@ -74,14 +74,13 @@
//************************************************************************************************* //*************************************************************************************************
// If nothing is defined Teensy internal DAC is used as audio output device! // If nothing is defined Teensy internal DAC is used as audio output device!
// Left and right channel audio signal is presented on pins A21 and A22. // Left and right channel audio signal is presented on pins A21 and A22.
#define TEENSY_AUDIO_BOARD
#define AUDIO_DEVICE_USB #define AUDIO_DEVICE_USB
#define SGTL5000_HEADPHONE_VOLUME 0.8 #define TEENSY_AUDIO_BOARD
//#define PT8211_AUDIO
//#define TGA_AUDIO_BOARD
//#define TEENSY_DAC //#define TEENSY_DAC
//#define TEENSY_DAC_SYMMETRIC //#define TEENSY_DAC_SYMMETRIC
//#define I2S_AUDIO_ONLY //#define I2S_AUDIO_ONLY
//#define TGA_AUDIO_BOARD
//#define PT8211_AUDIO
//************************************************************************************************* //*************************************************************************************************
//* MIDI SOFTWARE SETTINGS //* MIDI SOFTWARE SETTINGS
@ -93,7 +92,7 @@
//************************************************************************************************* //*************************************************************************************************
//* DEBUG OUTPUT SETTINGS //* DEBUG OUTPUT SETTINGS
//************************************************************************************************* //*************************************************************************************************
//#define DEBUG 1 #define DEBUG 1
#define SERIAL_SPEED 230400 #define SERIAL_SPEED 230400
#define SHOW_XRUN 1 #define SHOW_XRUN 1
#define SHOW_CPU_LOAD_MSEC 5000 #define SHOW_CPU_LOAD_MSEC 5000
@ -113,10 +112,13 @@
#define MOD_FILTER_OUTPUT MOD_BUTTERWORTH_FILTER_OUTPUT // MOD_LINKWITZ_RILEY_FILTER_OUTPUT MOD_BUTTERWORTH_FILTER_OUTPUT MOD_NO_FILTER_OUTPUT #define MOD_FILTER_OUTPUT MOD_BUTTERWORTH_FILTER_OUTPUT // MOD_LINKWITZ_RILEY_FILTER_OUTPUT MOD_BUTTERWORTH_FILTER_OUTPUT MOD_NO_FILTER_OUTPUT
#define MOD_FILTER_CUTOFF_HZ 2000 #define MOD_FILTER_CUTOFF_HZ 2000
// REVERB parameters // REVERB parameters
#define REVERB_ANTIALIAS_FRQ 7500 //#define REVERB_ANTIALIAS_FRQ 7500
// SGTL5000 // SGTL5000
#ifdef TEENSY_AUDIO_BOARD
#define SGTL5000_AUDIO_ENHANCE 1 #define SGTL5000_AUDIO_ENHANCE 1
#define SGTL5000_AUDIO_THRU 1 #define SGTL5000_AUDIO_THRU 1
#define SGTL5000_HEADPHONE_VOLUME 0.8
#endif
//************************************************************************************************* //*************************************************************************************************
//* AUDIO SOFTWARE SETTINGS //* AUDIO SOFTWARE SETTINGS
//************************************************************************************************* //*************************************************************************************************

@ -1,144 +0,0 @@
/* Audio Library for Teensy 3.X
* Dynamics Processor (Gate, Compressor & Limiter)
* Copyright (c) 2017, Marc Paquette (marc@dacsystemes.com)
* Based on analyse_rms & mixer objects by Paul Stoffregen
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development funding notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "effect_dynamics.h"
#include "fast_log.h"
#include "utility/dspinst.h"
#include "utility/sqrt_integer.h"
static float analyse_rms(int16_t *data) {
uint32_t *p = (uint32_t *)data;
const uint32_t *end = p + AUDIO_BLOCK_SAMPLES / 2;
int64_t sum = 0;
do {
uint32_t n1 = *p++;
uint32_t n2 = *p++;
uint32_t n3 = *p++;
uint32_t n4 = *p++;
sum = multiply_accumulate_16tx16t_add_16bx16b(sum, n1, n1);
sum = multiply_accumulate_16tx16t_add_16bx16b(sum, n2, n2);
sum = multiply_accumulate_16tx16t_add_16bx16b(sum, n3, n3);
sum = multiply_accumulate_16tx16t_add_16bx16b(sum, n4, n4);
} while (p < end);
int32_t meansq = sum / AUDIO_BLOCK_SAMPLES;
return sqrt_uint32(meansq) / 32767.0f;
}
static void applyGain(int16_t *data, int32_t mult1, int32_t mult2) {
uint32_t *p = (uint32_t *)data;
const uint32_t *end = p + AUDIO_BLOCK_SAMPLES / 2;
int32_t inc = (mult2 - mult1) / (AUDIO_BLOCK_SAMPLES / 2);
do {
uint32_t tmp32 = *p; // read 2 samples from *data
int32_t val1 = signed_multiply_32x16b(mult1, tmp32);
mult1 += inc;
int32_t val2 = signed_multiply_32x16t(mult1, tmp32);
mult1 += inc;
val1 = signed_saturate_rshift(val1, 16, 0);
val2 = signed_saturate_rshift(val2, 16, 0);
*p++ = pack_16b_16b(val2, val1);
} while (p < end);
}
void AudioEffectDynamics::update(void) {
audio_block_t *block;
block = receiveWritable(0);
if (!block) return;
if (!gateEnabled && !compEnabled && !limiterEnabled) {
//Transmit & release
transmit(block);
release(block);
return;
}
//Analyze received block
float rms = analyse_rms(block->data);
//Compute block RMS level in Db
float inputdb = MIN_DB;
if (rms > 0) inputdb = unitToDb(rms);
//Gate
if (gateEnabled) {
if (inputdb >= gateThresholdOpen) gatedb = (aGateAttack * gatedb) + (aOneMinusGateAttack * MAX_DB);
else if (inputdb < gateThresholdClose) gatedb = (aGateRelease * gatedb) + (aOneMinusGateRelease * MIN_DB);
}
else gatedb = MAX_DB;
//Compressor
if (compEnabled) {
float attdb = MAX_DB; //Below knee
if (inputdb >= aLowKnee) {
if(inputdb <= aHighKnee) {
//Knee transition
float knee = inputdb - aLowKnee;
attdb = aKneeRatio * knee * knee * aTwoKneeWidth;
}
else {
//Above knee
attdb = compThreshold + ((inputdb - compThreshold) * compRatio) - inputdb;
}
}
if (attdb <= compdb) compdb = (aCompAttack * compdb) + (aOneMinusCompAttack * attdb);
else compdb = (aCompRelease * compdb) + (aOneMinusCompRelease * attdb);
}
else compdb = MAX_DB;
//Brickwall Limiter
if (limiterEnabled) {
float outdb = inputdb + compdb + makeupdb;
if (outdb >= limitThreshold) limitdb = (aLimitAttack * limitdb) + (aOneMinusLimitAttack * (limitThreshold - outdb));
else limitdb *= aLimitRelease;
}
else limitdb = MAX_DB;
//Compute linear gain
float totalGain = gatedb + compdb + makeupdb + limitdb;
int32_t mult = dbToUnit(totalGain) * 65536.0f;
//Apply gain to block
applyGain(block->data, last_mult, mult);
last_mult = mult;
//Transmit & release
transmit(block);
release(block);
}

@ -1,194 +0,0 @@
/* Audio Library for Teensy 3.X
* Dynamics Processor (Gate, Compressor & Limiter)
* Copyright (c) 2018, Marc Paquette (marc@dacsystemes.com)
* Based on analyse_rms, effect_envelope & mixer objects by Paul Stoffregen
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development funding notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#ifndef effect_dynamics_h_
#define effect_dynamics_h_
#include "Arduino.h"
#include "AudioStream.h"
#define MIN_DB -110.0f
#define MAX_DB 0.0f
#define MIN_T 0.03f //Roughly 1 block
#define MAX_T 4.00f
#define RATIO_OFF 1.0f
#define RATIO_INFINITY 60.0f
class AudioEffectDynamics : public AudioStream
{
public:
AudioEffectDynamics(void) : AudioStream(1, inputQueueArray) {
gate();
compression();
limit();
autoMakeupGain();
gatedb = MIN_DB;
compdb = MIN_DB;
limitdb = MIN_DB;
}
//Sets the gate parameters.
//threshold is in dbFS
//attack & release are in seconds
void gate(float threshold = -50.0f, float attack = MIN_T, float release = 0.3f, float hysterisis = 6.0f) {
gateEnabled = threshold > MIN_DB;
gateThresholdOpen = constrain(threshold, MIN_DB, MAX_DB);
gateThresholdClose = gateThresholdOpen - constrain(hysterisis, 0.0f, 6.0f);
float gateAttackTime = constrain(attack, MIN_T, MAX_T);
float gateReleaseTime = constrain(release, MIN_T, MAX_T);
aGateAttack = timeToAlpha(gateAttackTime);
aOneMinusGateAttack = 1.0f - aGateAttack;
aGateRelease = timeToAlpha(gateReleaseTime);
aOneMinusGateRelease = 1.0f - aGateRelease;
}
//Sets the compression parameters.
//threshold & kneeWidth are in db(FS)
//attack and release are in seconds
//ratio is expressed as x:1 i.e. 1 for no compression, 60 for brickwall limiting
//Set kneeWidth to 0 for hard knee
void compression(float threshold = -40.0f, float attack = MIN_T, float release = 0.5f, float ratio = 35.0f, float kneeWidth = 6.0f) {
compEnabled = threshold < MAX_DB;
compThreshold = constrain(threshold, MIN_DB, MAX_DB);
float compAttackTime = constrain(attack, MIN_T, MAX_T);
float compReleaseTime = constrain(release, MIN_T, MAX_T);
compRatio = 1.0f / constrain(abs(ratio), RATIO_OFF, RATIO_INFINITY);
float compKneeWidth = constrain(abs(kneeWidth), 0.0f, 32.0f);
computeMakeupGain();
aCompAttack = timeToAlpha(compAttackTime);
aOneMinusCompAttack = 1.0f - aCompAttack;
aCompRelease = timeToAlpha(compReleaseTime);
aOneMinusCompRelease = 1.0f - aCompRelease;
aHalfKneeWidth = compKneeWidth / 2.0f;
aTwoKneeWidth = 1.0f / (compKneeWidth * 2.0f);
aKneeRatio = compRatio - 1.0f;
aLowKnee = compThreshold - aHalfKneeWidth;
aHighKnee = compThreshold + aHalfKneeWidth;
}
//Sets the hard limiter parameters
//threshold is in dbFS
//attack & release are in seconds
void limit(float threshold = -3.0f, float attack = MIN_T, float release = MIN_T) {
limiterEnabled = threshold < MAX_DB;
limitThreshold = constrain(threshold, MIN_DB, MAX_DB);
float limitAttackTime = constrain(attack, MIN_T, MAX_T);
float limitReleaseTime = constrain(release, MIN_T, MAX_T);
computeMakeupGain();
aLimitAttack = timeToAlpha(limitAttackTime);
aOneMinusLimitAttack = 1.0f - aLimitAttack;
aLimitRelease = timeToAlpha(limitReleaseTime);
}
//Enables automatic makeup gain setting
//headroom is in dbFS
void autoMakeupGain(float headroom = 6.0f) {
mgAutoEnabled = true;
mgHeadroom = constrain(headroom, 0.0f, 60.0f);
computeMakeupGain();
}
//Sets a fixed makeup gain value.
//gain is in dbFS
void makeupGain(float gain = 0.0f) {
mgAutoEnabled = false;
makeupdb = constrain(gain, -12.0f, 24.0f);
}
private:
audio_block_t *inputQueueArray[1];
bool gateEnabled = false;
float gateThresholdOpen;
float gateThresholdClose;
float gatedb;
bool compEnabled = false;
float compThreshold;
float compRatio;
float compdb;
bool limiterEnabled = false;
float limitThreshold;
float limitdb;
bool mgAutoEnabled;
float mgHeadroom;
float makeupdb;
float aGateAttack;
float aOneMinusGateAttack;
float aGateRelease;
float aOneMinusGateRelease;
float aHalfKneeWidth;
float aTwoKneeWidth;
float aKneeRatio;
float aLowKnee;
float aHighKnee;
float aCompAttack;
float aOneMinusCompAttack;
float aCompRelease;
float aOneMinusCompRelease;
float aLimitAttack;
float aOneMinusLimitAttack;
float aLimitRelease;
int32_t last_mult;
void computeMakeupGain() {
if (mgAutoEnabled) {
makeupdb = -compThreshold + (compThreshold * compRatio) + limitThreshold - mgHeadroom;
}
}
//Computes smoothing time constants for a 10% to 90% change
float timeToAlpha(float time) {
return expf(-0.9542f / (((float)AUDIO_SAMPLE_RATE_EXACT / (float)AUDIO_BLOCK_SAMPLES) * time));
}
virtual void update(void);
};
#endif
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