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Revert "Revert "Audio system overhaul (#11820)" due to freezing issues"

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This reverts commit 996a19ee7b.
This commit is contained in:
Drashna Jael're
2021-12-07 09:22:22 -08:00
parent acec40a11a
commit 43002bdf77
22 changed files with 2000 additions and 805 deletions
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21
common_features.mk
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@@ -58,12 +58,31 @@ ifeq ($(strip $(COMMAND_ENABLE)), yes)
OPT_DEFS += -DCOMMAND_ENABLE OPT_DEFS += -DCOMMAND_ENABLE
endif endif
AUDIO_ENABLE ?= no
ifeq ($(strip $(AUDIO_ENABLE)), yes) ifeq ($(strip $(AUDIO_ENABLE)), yes)
ifeq ($(PLATFORM),CHIBIOS)
AUDIO_DRIVER ?= dac_basic
ifeq ($(strip $(AUDIO_DRIVER)), dac_basic)
OPT_DEFS += -DAUDIO_DRIVER_DAC
else ifeq ($(strip $(AUDIO_DRIVER)), dac_additive)
OPT_DEFS += -DAUDIO_DRIVER_DAC
## stm32f2 and above have a usable DAC unit, f1 do not, and need to use pwm instead
else ifeq ($(strip $(AUDIO_DRIVER)), pwm_software)
OPT_DEFS += -DAUDIO_DRIVER_PWM
else ifeq ($(strip $(AUDIO_DRIVER)), pwm_hardware)
OPT_DEFS += -DAUDIO_DRIVER_PWM
endif
else
# fallback for all other platforms is pwm
AUDIO_DRIVER ?= pwm_hardware
OPT_DEFS += -DAUDIO_DRIVER_PWM
endif
OPT_DEFS += -DAUDIO_ENABLE OPT_DEFS += -DAUDIO_ENABLE
MUSIC_ENABLE = yes MUSIC_ENABLE = yes
SRC += $(QUANTUM_DIR)/process_keycode/process_audio.c SRC += $(QUANTUM_DIR)/process_keycode/process_audio.c
SRC += $(QUANTUM_DIR)/process_keycode/process_clicky.c SRC += $(QUANTUM_DIR)/process_keycode/process_clicky.c
SRC += $(QUANTUM_DIR)/audio/audio_$(PLATFORM_KEY).c SRC += $(QUANTUM_DIR)/audio/audio.c ## common audio code, hardware agnostic
SRC += $(QUANTUM_DIR)/audio/driver_$(PLATFORM_KEY)_$(strip $(AUDIO_DRIVER)).c
SRC += $(QUANTUM_DIR)/audio/voices.c SRC += $(QUANTUM_DIR)/audio/voices.c
SRC += $(QUANTUM_DIR)/audio/luts.c SRC += $(QUANTUM_DIR)/audio/luts.c
endif endif

2
keyboards/planck/config.h
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@@ -40,7 +40,7 @@ along with this program. If not, see <http://www.gnu.org/licenses/>.
#define QMK_SPEAKER C6 #define QMK_SPEAKER C6
#define AUDIO_VOICES #define AUDIO_VOICES
#define C6_AUDIO #define AUDIO_PIN C6
#define BACKLIGHT_PIN B7 #define BACKLIGHT_PIN B7

5
keyboards/planck/ez/config.h
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@@ -58,7 +58,10 @@
#define MUSIC_MAP #define MUSIC_MAP
#undef AUDIO_VOICES #undef AUDIO_VOICES
#undef C6_AUDIO #undef AUDIO_PIN
#define AUDIO_PIN A5
#define AUDIO_PIN_ALT A4
#define AUDIO_PIN_ALT_AS_NEGATIVE
/* Debounce reduces chatter (unintended double-presses) - set 0 if debouncing is not needed */ /* Debounce reduces chatter (unintended double-presses) - set 0 if debouncing is not needed */
// #define DEBOUNCE 6 // #define DEBOUNCE 6

295
quantum/audio/audio.h
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@@ -1,4 +1,5 @@
/* Copyright 2016 Jack Humbert /* Copyright 2016-2020 Jack Humbert
* Copyright 2020 JohSchneider
* *
* This program is free software: you can redistribute it and/or modify * 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 * it under the terms of the GNU General Public License as published by
@@ -13,28 +14,30 @@
* You should have received a copy of the GNU General Public License * You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>. * along with this program. If not, see <http://www.gnu.org/licenses/>.
*/ */
#pragma once #pragma once
#include <stdint.h> #include <stdint.h>
#include <stdbool.h> #include <stdbool.h>
#if defined(__AVR__)
# include <avr/io.h>
#endif
#include "wait.h"
#include "musical_notes.h" #include "musical_notes.h"
#include "song_list.h" #include "song_list.h"
#include "voices.h" #include "voices.h"
#include "quantum.h" #include "quantum.h"
#include <math.h> #include <math.h>
// Largely untested PWM audio mode (doesn't sound as good) #if defined(__AVR__)
// #define PWM_AUDIO # include <avr/io.h>
# if defined(AUDIO_DRIVER_PWM)
# include "driver_avr_pwm.h"
# endif
#endif
// #define VIBRATO_ENABLE #if defined(PROTOCOL_CHIBIOS)
# if defined(AUDIO_DRIVER_PWM)
// Enable vibrato strength/amplitude - slows down ISR too much # include "driver_chibios_pwm.h"
// #define VIBRATO_STRENGTH_ENABLE # elif defined(AUDIO_DRIVER_DAC)
# include "driver_chibios_dac.h"
# endif
#endif
typedef union { typedef union {
uint8_t raw; uint8_t raw;
@@ -45,62 +48,238 @@ typedef union {
}; };
} audio_config_t; } audio_config_t;
bool is_audio_on(void); // AVR/LUFA has a MIN, arm/chibios does not
void audio_toggle(void); #ifndef MIN
void audio_on(void); # define MIN(a, b) (((a) < (b)) ? (a) : (b))
void audio_off(void);
// Vibrato rate functions
#ifdef VIBRATO_ENABLE
void set_vibrato_rate(float rate);
void increase_vibrato_rate(float change);
void decrease_vibrato_rate(float change);
# ifdef VIBRATO_STRENGTH_ENABLE
void set_vibrato_strength(float strength);
void increase_vibrato_strength(float change);
void decrease_vibrato_strength(float change);
#endif #endif
#endif /*
* a 'musical note' is represented by pitch and duration; a 'musical tone' adds intensity and timbre
* https://en.wikipedia.org/wiki/Musical_tone
* "A musical tone is characterized by its duration, pitch, intensity (or loudness), and timbre (or quality)"
*/
typedef struct {
uint16_t time_started; // timestamp the tone/note was started, system time runs with 1ms resolution -> 16bit timer overflows every ~64 seconds, long enough under normal circumstances; but might be too soon for long-duration notes when the note_tempo is set to a very low value
float pitch; // aka frequency, in Hz
uint16_t duration; // in ms, converted from the musical_notes.h unit which has 64parts to a beat, factoring in the current tempo in beats-per-minute
// float intensity; // aka volume [0,1] TODO: not used at the moment; pwm drivers can't handle it
// uint8_t timbre; // range: [0,100] TODO: this currently kept track of globally, should we do this per tone instead?
} musical_tone_t;
// Polyphony functions // public interface
void set_polyphony_rate(float rate);
void enable_polyphony(void);
void disable_polyphony(void);
void increase_polyphony_rate(float change);
void decrease_polyphony_rate(float change);
void set_timbre(float timbre);
void set_tempo(uint8_t tempo);
void increase_tempo(uint8_t tempo_change);
void decrease_tempo(uint8_t tempo_change);
/**
* @brief one-time initialization called by quantum/quantum.c
* @details usually done lazy, when some tones are to be played
*
* @post audio system (and hardware) initialized and ready to play tones
*/
void audio_init(void); void audio_init(void);
void audio_startup(void); void audio_startup(void);
#ifdef PWM_AUDIO /**
void play_sample(uint8_t* s, uint16_t l, bool r); * @brief en-/disable audio output, save this choice to the eeprom
#endif */
void play_note(float freq, int vol); void audio_toggle(void);
void stop_note(float freq); /**
void stop_all_notes(void); * @brief enable audio output, save this choice to the eeprom
void play_notes(float (*np)[][2], uint16_t n_count, bool n_repeat); */
void audio_on(void);
/**
* @brief disable audio output, save this choice to the eeprom
*/
void audio_off(void);
/**
* @brief query the if audio output is enabled
*/
bool audio_is_on(void);
#define SCALE \ /**
(int8_t[]) { 0 + (12 * 0), 2 + (12 * 0), 4 + (12 * 0), 5 + (12 * 0), 7 + (12 * 0), 9 + (12 * 0), 11 + (12 * 0), 0 + (12 * 1), 2 + (12 * 1), 4 + (12 * 1), 5 + (12 * 1), 7 + (12 * 1), 9 + (12 * 1), 11 + (12 * 1), 0 + (12 * 2), 2 + (12 * 2), 4 + (12 * 2), 5 + (12 * 2), 7 + (12 * 2), 9 + (12 * 2), 11 + (12 * 2), 0 + (12 * 3), 2 + (12 * 3), 4 + (12 * 3), 5 + (12 * 3), 7 + (12 * 3), 9 + (12 * 3), 11 + (12 * 3), 0 + (12 * 4), 2 + (12 * 4), 4 + (12 * 4), 5 + (12 * 4), 7 + (12 * 4), 9 + (12 * 4), 11 + (12 * 4), } * @brief start playback of a tone with the given frequency and duration
*
* @details starts the playback of a given note, which is automatically stopped
* at the the end of its duration = fire&forget
*
* @param[in] pitch frequency of the tone be played
* @param[in] duration in milliseconds, use 'audio_duration_to_ms' to convert
* from the musical_notes.h unit to ms
*/
void audio_play_note(float pitch, uint16_t duration);
// TODO: audio_play_note(float pitch, uint16_t duration, float intensity, float timbre);
// audio_play_note_with_instrument ifdef AUDIO_ENABLE_VOICES
// These macros are used to allow play_notes to play an array of indeterminate /**
* @brief start playback of a tone with the given frequency
*
* @details the 'frequency' is put on-top the internal stack of active tones,
* as a new tone with indefinite duration. this tone is played by
* the hardware until a call to 'audio_stop_tone'.
* should a tone with that frequency already be active, its entry
* is put on the top of said internal stack - so no duplicate
* entries are kept.
* 'hardware_start' is called upon the first note.
*
* @param[in] pitch frequency of the tone be played
*/
void audio_play_tone(float pitch);
/**
* @brief stop a given tone/frequency
*
* @details removes a tone matching the given frequency from the internal
* playback stack
* the hardware is stopped in case this was the last/only frequency
* being played.
*
* @param[in] pitch tone/frequency to be stopped
*/
void audio_stop_tone(float pitch);
/**
* @brief play a melody
*
* @details starts playback of a melody passed in from a SONG definition - an
* array of {pitch, duration} float-tuples
*
* @param[in] np note-pointer to the SONG array
* @param[in] n_count number of MUSICAL_NOTES of the SONG
* @param[in] n_repeat false for onetime, true for looped playback
*/
void audio_play_melody(float (*np)[][2], uint16_t n_count, bool n_repeat);
/**
* @brief play a short tone of a specific frequency to emulate a 'click'
*
* @details constructs a two-note melody (one pause plus a note) and plays it through
* audio_play_melody. very short durations might not quite work due to
* hardware limitations (DAC: added pulses from zero-crossing feature;...)
*
* @param[in] delay in milliseconds, length for the pause before the pulses, can be zero
* @param[in] pitch
* @param[in] duration in milliseconds, length of the 'click'
*/
void audio_play_click(uint16_t delay, float pitch, uint16_t duration);
/**
* @brief stops all playback
*
* @details stops playback of both a melody as well as single tones, resetting
* the internal state
*/
void audio_stop_all(void);
/**
* @brief query if one/multiple tones are playing
*/
bool audio_is_playing_note(void);
/**
* @brief query if a melody/SONG is playing
*/
bool audio_is_playing_melody(void);
// These macros are used to allow audio_play_melody to play an array of indeterminate
// length. This works around the limitation of C's sizeof operation on pointers. // length. This works around the limitation of C's sizeof operation on pointers.
// The global float array for the song must be used here. // The global float array for the song must be used here.
#define NOTE_ARRAY_SIZE(x) ((int16_t)(sizeof(x) / (sizeof(x[0])))) #define NOTE_ARRAY_SIZE(x) ((int16_t)(sizeof(x) / (sizeof(x[0]))))
#define PLAY_SONG(note_array) play_notes(&note_array, NOTE_ARRAY_SIZE((note_array)), false)
#define PLAY_LOOP(note_array) play_notes(&note_array, NOTE_ARRAY_SIZE((note_array)), true)
bool is_playing_notes(void); /**
* @brief convenience macro, to play a melody/SONG once
*/
#define PLAY_SONG(note_array) audio_play_melody(&note_array, NOTE_ARRAY_SIZE((note_array)), false)
// TODO: a 'song' is a melody plus singing/vocals -> PLAY_MELODY
/**
* @brief convenience macro, to play a melody/SONG in a loop, until stopped by 'audio_stop_all'
*/
#define PLAY_LOOP(note_array) audio_play_melody(&note_array, NOTE_ARRAY_SIZE((note_array)), true)
// Tone-Multiplexing functions
// this feature only makes sense for hardware setups which can't do proper
// audio-wave synthesis = have no DAC and need to use PWM for tone generation
#ifdef AUDIO_ENABLE_TONE_MULTIPLEXING
# ifndef AUDIO_TONE_MULTIPLEXING_RATE_DEFAULT
# define AUDIO_TONE_MULTIPLEXING_RATE_DEFAULT 0
// 0=off, good starting value is 4; the lower the value the higher the cpu-load
# endif
void audio_set_tone_multiplexing_rate(uint16_t rate);
void audio_enable_tone_multiplexing(void);
void audio_disable_tone_multiplexing(void);
void audio_increase_tone_multiplexing_rate(uint16_t change);
void audio_decrease_tone_multiplexing_rate(uint16_t change);
#endif
// Tempo functions
void audio_set_tempo(uint8_t tempo);
void audio_increase_tempo(uint8_t tempo_change);
void audio_decrease_tempo(uint8_t tempo_change);
// conversion macros, from 64parts-to-a-beat to milliseconds and back
uint16_t audio_duration_to_ms(uint16_t duration_bpm);
uint16_t audio_ms_to_duration(uint16_t duration_ms);
void audio_startup(void);
// hardware interface
// implementation in the driver_avr/arm_* respective parts
void audio_driver_initialize(void);
void audio_driver_start(void);
void audio_driver_stop(void);
/**
* @brief get the number of currently active tones
* @return number, 0=none active
*/
uint8_t audio_get_number_of_active_tones(void);
/**
* @brief access to the raw/unprocessed frequency for a specific tone
* @details each active tone has a frequency associated with it, which
* the internal state keeps track of, and is usually influenced
* by various effects
* @param[in] tone_index, ranging from 0 to number_of_active_tones-1, with the
* first being the most recent and each increment yielding the next
* older one
* @return a positive frequency, in Hz; or zero if the tone is a pause
*/
float audio_get_frequency(uint8_t tone_index);
/**
* @brief calculate and return the frequency for the requested tone
* @details effects like glissando, vibrato, ... are post-processed onto the
* each active tones 'base'-frequency; this function returns the
* post-processed result.
* @param[in] tone_index, ranging from 0 to number_of_active_tones-1, with the
* first being the most recent and each increment yielding the next
* older one
* @return a positive frequency, in Hz; or zero if the tone is a pause
*/
float audio_get_processed_frequency(uint8_t tone_index);
/**
* @brief update audio internal state: currently playing and active tones,...
* @details This function is intended to be called by the audio-hardware
* specific implementation on a somewhat regular basis while a SONG
* or notes (pitch+duration) are playing to 'advance' the internal
* state (current playing notes, position in the melody, ...)
*
* @return true if something changed in the currently active tones, which the
* hardware might need to react to
*/
bool audio_update_state(void);
// legacy and back-warts compatibility stuff
#define is_audio_on() audio_is_on()
#define is_playing_notes() audio_is_playing_melody()
#define is_playing_note() audio_is_playing_note()
#define stop_all_notes() audio_stop_all()
#define stop_note(f) audio_stop_tone(f)
#define play_note(f, v) audio_play_tone(f)
#define set_timbre(t) voice_set_timbre(t)
#define set_tempo(t) audio_set_tempo(t)
#define increase_tempo(t) audio_increase_tempo(t)
#define decrease_tempo(t) audio_decrease_tempo(t)
// vibrato functions are not used in any keyboards

4
quantum/audio/audio_chibios.c
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@@ -88,19 +88,23 @@ static void gpt_cb8(GPTDriver *gptp);
gptStart(&GPTD6, &gpt6cfg1); \ gptStart(&GPTD6, &gpt6cfg1); \
gptStartContinuous(&GPTD6, 2U); \ gptStartContinuous(&GPTD6, 2U); \
palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG) palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG)
#define START_CHANNEL_2() \ #define START_CHANNEL_2() \
gptStart(&GPTD7, &gpt7cfg1); \ gptStart(&GPTD7, &gpt7cfg1); \
gptStartContinuous(&GPTD7, 2U); \ gptStartContinuous(&GPTD7, 2U); \
palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG) palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG)
#define STOP_CHANNEL_1() \ #define STOP_CHANNEL_1() \
gptStopTimer(&GPTD6); \ gptStopTimer(&GPTD6); \
palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL); \ palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL); \
palSetPad(GPIOA, 4) palSetPad(GPIOA, 4)
#define STOP_CHANNEL_2() \ #define STOP_CHANNEL_2() \
gptStopTimer(&GPTD7); \ gptStopTimer(&GPTD7); \
palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL); \ palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL); \
palSetPad(GPIOA, 5) palSetPad(GPIOA, 5)
#define RESTART_CHANNEL_1() \ #define RESTART_CHANNEL_1() \
STOP_CHANNEL_1(); \ STOP_CHANNEL_1(); \
START_CHANNEL_1() START_CHANNEL_1()

606
quantum/audio/audio_pwm.c
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@@ -1,606 +0,0 @@
/* Copyright 2016 Jack Humbert
*
* 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 2 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 <stdio.h>
#include <string.h>
//#include <math.h>
#include <avr/pgmspace.h>
#include <avr/interrupt.h>
#include <avr/io.h>
#include "print.h"
#include "audio.h"
#include "keymap.h"
#include "eeconfig.h"
#define PI 3.14159265
#define CPU_PRESCALER 8
#ifndef STARTUP_SONG
# define STARTUP_SONG SONG(STARTUP_SOUND)
#endif
float startup_song[][2] = STARTUP_SONG;
// Timer Abstractions
// TIMSK3 - Timer/Counter #3 Interrupt Mask Register
// Turn on/off 3A interputs, stopping/enabling the ISR calls
#define ENABLE_AUDIO_COUNTER_3_ISR TIMSK3 |= _BV(OCIE3A)
#define DISABLE_AUDIO_COUNTER_3_ISR TIMSK3 &= ~_BV(OCIE3A)
// TCCR3A: Timer/Counter #3 Control Register
// Compare Output Mode (COM3An) = 0b00 = Normal port operation, OC3A disconnected from PC6
#define ENABLE_AUDIO_COUNTER_3_OUTPUT TCCR3A |= _BV(COM3A1);
#define DISABLE_AUDIO_COUNTER_3_OUTPUT TCCR3A &= ~(_BV(COM3A1) | _BV(COM3A0));
#define NOTE_PERIOD ICR3
#define NOTE_DUTY_CYCLE OCR3A
#ifdef PWM_AUDIO
# include "wave.h"
# define SAMPLE_DIVIDER 39
# define SAMPLE_RATE (2000000.0 / SAMPLE_DIVIDER / 2048)
// Resistor value of 1/ (2 * PI * 10nF * (2000000 hertz / SAMPLE_DIVIDER / 10)) for 10nF cap
float places[8] = {0, 0, 0, 0, 0, 0, 0, 0};
uint16_t place_int = 0;
bool repeat = true;
#endif
void delay_us(int count) {
while (count--) {
_delay_us(1);
}
}
int voices = 0;
int voice_place = 0;
float frequency = 0;
int volume = 0;
long position = 0;
float frequencies[8] = {0, 0, 0, 0, 0, 0, 0, 0};
int volumes[8] = {0, 0, 0, 0, 0, 0, 0, 0};
bool sliding = false;
float place = 0;
uint8_t* sample;
uint16_t sample_length = 0;
// float freq = 0;
bool playing_notes = false;
bool playing_note = false;
float note_frequency = 0;
float note_length = 0;
uint8_t note_tempo = TEMPO_DEFAULT;
float note_timbre = TIMBRE_DEFAULT;
uint16_t note_position = 0;
float (*notes_pointer)[][2];
uint16_t notes_count;
bool notes_repeat;
float notes_rest;
bool note_resting = false;
uint16_t current_note = 0;
uint8_t rest_counter = 0;
#ifdef VIBRATO_ENABLE
float vibrato_counter = 0;
float vibrato_strength = .5;
float vibrato_rate = 0.125;
#endif
float polyphony_rate = 0;
static bool audio_initialized = false;
audio_config_t audio_config;
uint16_t envelope_index = 0;
void audio_init() {
// Check EEPROM
if (!eeconfig_is_enabled()) {
eeconfig_init();
}
audio_config.raw = eeconfig_read_audio();
#ifdef PWM_AUDIO
PLLFRQ = _BV(PDIV2);
PLLCSR = _BV(PLLE);
while (!(PLLCSR & _BV(PLOCK)))
;
PLLFRQ |= _BV(PLLTM0); /* PCK 48MHz */
/* Init a fast PWM on Timer4 */
TCCR4A = _BV(COM4A0) | _BV(PWM4A); /* Clear OC4A on Compare Match */
TCCR4B = _BV(CS40); /* No prescaling => f = PCK/256 = 187500Hz */
OCR4A = 0;
/* Enable the OC4A output */
DDRC |= _BV(PORTC6);
DISABLE_AUDIO_COUNTER_3_ISR; // Turn off 3A interputs
TCCR3A = 0x0; // Options not needed
TCCR3B = _BV(CS31) | _BV(CS30) | _BV(WGM32); // 64th prescaling and CTC
OCR3A = SAMPLE_DIVIDER - 1; // Correct count/compare, related to sample playback
#else
// Set port PC6 (OC3A and /OC4A) as output
DDRC |= _BV(PORTC6);
DISABLE_AUDIO_COUNTER_3_ISR;
// TCCR3A / TCCR3B: Timer/Counter #3 Control Registers
// Compare Output Mode (COM3An) = 0b00 = Normal port operation, OC3A disconnected from PC6
// Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14 (Period = ICR3, Duty Cycle = OCR3A)
// Clock Select (CS3n) = 0b010 = Clock / 8
TCCR3A = (0 << COM3A1) | (0 << COM3A0) | (1 << WGM31) | (0 << WGM30);
TCCR3B = (1 << WGM33) | (1 << WGM32) | (0 << CS32) | (1 << CS31) | (0 << CS30);
#endif
audio_initialized = true;
}
void audio_startup() {
if (audio_config.enable) {
PLAY_SONG(startup_song);
}
}
void stop_all_notes() {
if (!audio_initialized) {
audio_init();
}
voices = 0;
#ifdef PWM_AUDIO
DISABLE_AUDIO_COUNTER_3_ISR;
#else
DISABLE_AUDIO_COUNTER_3_ISR;
DISABLE_AUDIO_COUNTER_3_OUTPUT;
#endif
playing_notes = false;
playing_note = false;
frequency = 0;
volume = 0;
for (uint8_t i = 0; i < 8; i++) {
frequencies[i] = 0;
volumes[i] = 0;
}
}
void stop_note(float freq) {
if (playing_note) {
if (!audio_initialized) {
audio_init();
}
#ifdef PWM_AUDIO
freq = freq / SAMPLE_RATE;
#endif
for (int i = 7; i >= 0; i--) {
if (frequencies[i] == freq) {
frequencies[i] = 0;
volumes[i] = 0;
for (int j = i; (j < 7); j++) {
frequencies[j] = frequencies[j + 1];
frequencies[j + 1] = 0;
volumes[j] = volumes[j + 1];
volumes[j + 1] = 0;
}
break;
}
}
voices--;
if (voices < 0) voices = 0;
if (voice_place >= voices) {
voice_place = 0;
}
if (voices == 0) {
#ifdef PWM_AUDIO
DISABLE_AUDIO_COUNTER_3_ISR;
#else
DISABLE_AUDIO_COUNTER_3_ISR;
DISABLE_AUDIO_COUNTER_3_OUTPUT;
#endif
frequency = 0;
volume = 0;
playing_note = false;
}
}
}
#ifdef VIBRATO_ENABLE
float mod(float a, int b) {
float r = fmod(a, b);
return r < 0 ? r + b : r;
}
float vibrato(float average_freq) {
# ifdef VIBRATO_STRENGTH_ENABLE
float vibrated_freq = average_freq * pow(vibrato_lut[(int)vibrato_counter], vibrato_strength);
# else
float vibrated_freq = average_freq * vibrato_lut[(int)vibrato_counter];
# endif
vibrato_counter = mod((vibrato_counter + vibrato_rate * (1.0 + 440.0 / average_freq)), VIBRATO_LUT_LENGTH);
return vibrated_freq;
}
#endif
ISR(TIMER3_COMPA_vect) {
if (playing_note) {
#ifdef PWM_AUDIO
if (voices == 1) {
// SINE
OCR4A = pgm_read_byte(&sinewave[(uint16_t)place]) >> 2;
// SQUARE
// if (((int)place) >= 1024){
// OCR4A = 0xFF >> 2;
// } else {
// OCR4A = 0x00;
// }
// SAWTOOTH
// OCR4A = (int)place / 4;
// TRIANGLE
// if (((int)place) >= 1024) {
// OCR4A = (int)place / 2;
// } else {
// OCR4A = 2048 - (int)place / 2;
// }
place += frequency;
if (place >= SINE_LENGTH) place -= SINE_LENGTH;
} else {
int sum = 0;
for (int i = 0; i < voices; i++) {
// SINE
sum += pgm_read_byte(&sinewave[(uint16_t)places[i]]) >> 2;
// SQUARE
// if (((int)places[i]) >= 1024){
// sum += 0xFF >> 2;
// } else {
// sum += 0x00;
// }
places[i] += frequencies[i];
if (places[i] >= SINE_LENGTH) places[i] -= SINE_LENGTH;
}
OCR4A = sum;
}
#else
if (voices > 0) {
float freq;
if (polyphony_rate > 0) {
if (voices > 1) {
voice_place %= voices;
if (place++ > (frequencies[voice_place] / polyphony_rate / CPU_PRESCALER)) {
voice_place = (voice_place + 1) % voices;
place = 0.0;
}
}
# ifdef VIBRATO_ENABLE
if (vibrato_strength > 0) {
freq = vibrato(frequencies[voice_place]);
} else {
# else
{
# endif
freq = frequencies[voice_place];
}
} else {
if (frequency != 0 && frequency < frequencies[voices - 1] && frequency < frequencies[voices - 1] * pow(2, -440 / frequencies[voices - 1] / 12 / 2)) {
frequency = frequency * pow(2, 440 / frequency / 12 / 2);
} else if (frequency != 0 && frequency > frequencies[voices - 1] && frequency > frequencies[voices - 1] * pow(2, 440 / frequencies[voices - 1] / 12 / 2)) {
frequency = frequency * pow(2, -440 / frequency / 12 / 2);
} else {
frequency = frequencies[voices - 1];
}
# ifdef VIBRATO_ENABLE
if (vibrato_strength > 0) {
freq = vibrato(frequency);
} else {
# else
{
# endif
freq = frequency;
}
}
if (envelope_index < 65535) {
envelope_index++;
}
freq = voice_envelope(freq);
if (freq < 30.517578125) freq = 30.52;
NOTE_PERIOD = (int)(((double)F_CPU) / (freq * CPU_PRESCALER)); // Set max to the period
NOTE_DUTY_CYCLE = (int)((((double)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre); // Set compare to half the period
}
#endif
}
// SAMPLE
// OCR4A = pgm_read_byte(&sample[(uint16_t)place_int]);
// place_int++;
// if (place_int >= sample_length)
// if (repeat)
// place_int -= sample_length;
// else
// DISABLE_AUDIO_COUNTER_3_ISR;
if (playing_notes) {
#ifdef PWM_AUDIO
OCR4A = pgm_read_byte(&sinewave[(uint16_t)place]) >> 0;
place += note_frequency;
if (place >= SINE_LENGTH) place -= SINE_LENGTH;
#else
if (note_frequency > 0) {
float freq;
# ifdef VIBRATO_ENABLE
if (vibrato_strength > 0) {
freq = vibrato(note_frequency);
} else {
# else
{
# endif
freq = note_frequency;
}
if (envelope_index < 65535) {
envelope_index++;
}
freq = voice_envelope(freq);
NOTE_PERIOD = (int)(((double)F_CPU) / (freq * CPU_PRESCALER)); // Set max to the period
NOTE_DUTY_CYCLE = (int)((((double)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre); // Set compare to half the period
} else {
NOTE_PERIOD = 0;
NOTE_DUTY_CYCLE = 0;
}
#endif
note_position++;
bool end_of_note = false;
if (NOTE_PERIOD > 0)
end_of_note = (note_position >= (note_length / NOTE_PERIOD * 0xFFFF));
else
end_of_note = (note_position >= (note_length * 0x7FF));
if (end_of_note) {
current_note++;
if (current_note >= notes_count) {
if (notes_repeat) {
current_note = 0;
} else {
#ifdef PWM_AUDIO
DISABLE_AUDIO_COUNTER_3_ISR;
#else
DISABLE_AUDIO_COUNTER_3_ISR;
DISABLE_AUDIO_COUNTER_3_OUTPUT;
#endif
playing_notes = false;
return;
}
}
if (!note_resting && (notes_rest > 0)) {
note_resting = true;
note_frequency = 0;
note_length = notes_rest;
current_note--;
} else {
note_resting = false;
#ifdef PWM_AUDIO
note_frequency = (*notes_pointer)[current_note][0] / SAMPLE_RATE;
note_length = (*notes_pointer)[current_note][1] * (((float)note_tempo) / 100);
#else
envelope_index = 0;
note_frequency = (*notes_pointer)[current_note][0];
note_length = ((*notes_pointer)[current_note][1] / 4) * (((float)note_tempo) / 100);
#endif
}
note_position = 0;
}
}
if (!audio_config.enable) {
playing_notes = false;
playing_note = false;
}
}
void play_note(float freq, int vol) {
if (!audio_initialized) {
audio_init();
}
if (audio_config.enable && voices < 8) {
DISABLE_AUDIO_COUNTER_3_ISR;
// Cancel notes if notes are playing
if (playing_notes) stop_all_notes();
playing_note = true;
envelope_index = 0;
#ifdef PWM_AUDIO
freq = freq / SAMPLE_RATE;
#endif
if (freq > 0) {
frequencies[voices] = freq;
volumes[voices] = vol;
voices++;
}
#ifdef PWM_AUDIO
ENABLE_AUDIO_COUNTER_3_ISR;
#else
ENABLE_AUDIO_COUNTER_3_ISR;
ENABLE_AUDIO_COUNTER_3_OUTPUT;
#endif
}
}
void play_notes(float (*np)[][2], uint16_t n_count, bool n_repeat, float n_rest) {
if (!audio_initialized) {
audio_init();
}
if (audio_config.enable) {
DISABLE_AUDIO_COUNTER_3_ISR;
// Cancel note if a note is playing
if (playing_note) stop_all_notes();
playing_notes = true;
notes_pointer = np;
notes_count = n_count;
notes_repeat = n_repeat;
notes_rest = n_rest;
place = 0;
current_note = 0;
#ifdef PWM_AUDIO
note_frequency = (*notes_pointer)[current_note][0] / SAMPLE_RATE;
note_length = (*notes_pointer)[current_note][1] * (((float)note_tempo) / 100);
#else
note_frequency = (*notes_pointer)[current_note][0];
note_length = ((*notes_pointer)[current_note][1] / 4) * (((float)note_tempo) / 100);
#endif
note_position = 0;
#ifdef PWM_AUDIO
ENABLE_AUDIO_COUNTER_3_ISR;
#else
ENABLE_AUDIO_COUNTER_3_ISR;
ENABLE_AUDIO_COUNTER_3_OUTPUT;
#endif
}
}
#ifdef PWM_AUDIO
void play_sample(uint8_t* s, uint16_t l, bool r) {
if (!audio_initialized) {
audio_init();
}
if (audio_config.enable) {
DISABLE_AUDIO_COUNTER_3_ISR;
stop_all_notes();
place_int = 0;
sample = s;
sample_length = l;
repeat = r;
ENABLE_AUDIO_COUNTER_3_ISR;
}
}
#endif
void audio_toggle(void) {
audio_config.enable ^= 1;
eeconfig_update_audio(audio_config.raw);
}
void audio_on(void) {
audio_config.enable = 1;
eeconfig_update_audio(audio_config.raw);
}
void audio_off(void) {
audio_config.enable = 0;
eeconfig_update_audio(audio_config.raw);
}
#ifdef VIBRATO_ENABLE
// Vibrato rate functions
void set_vibrato_rate(float rate) { vibrato_rate = rate; }
void increase_vibrato_rate(float change) { vibrato_rate *= change; }
void decrease_vibrato_rate(float change) { vibrato_rate /= change; }
# ifdef VIBRATO_STRENGTH_ENABLE
void set_vibrato_strength(float strength) { vibrato_strength = strength; }
void increase_vibrato_strength(float change) { vibrato_strength *= change; }
void decrease_vibrato_strength(float change) { vibrato_strength /= change; }
# endif /* VIBRATO_STRENGTH_ENABLE */
#endif /* VIBRATO_ENABLE */
// Polyphony functions
void set_polyphony_rate(float rate) { polyphony_rate = rate; }
void enable_polyphony() { polyphony_rate = 5; }
void disable_polyphony() { polyphony_rate = 0; }
void increase_polyphony_rate(float change) { polyphony_rate *= change; }
void decrease_polyphony_rate(float change) { polyphony_rate /= change; }
// Timbre function
void set_timbre(float timbre) { note_timbre = timbre; }
// Tempo functions
void set_tempo(uint8_t tempo) { note_tempo = tempo; }
void decrease_tempo(uint8_t tempo_change) { note_tempo += tempo_change; }
void increase_tempo(uint8_t tempo_change) {
if (note_tempo - tempo_change < 10) {
note_tempo = 10;
} else {
note_tempo -= tempo_change;
}
}
//------------------------------------------------------------------------------
// Override these functions in your keymap file to play different tunes on
// startup and bootloader jump
__attribute__((weak)) void play_startup_tone() {}
__attribute__((weak)) void play_goodbye_tone() {}
//------------------------------------------------------------------------------

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/* Copyright 2020 Jack Humbert
* Copyright 2020 JohSchneider
*
* 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 2 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/>.
*/
#pragma once

332
quantum/audio/driver_avr_pwm_hardware.c Normal file
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/* Copyright 2016 Jack Humbert
* Copyright 2020 JohSchneider
*
* 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 2 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/>.
*/
#if defined(__AVR__)
# include <avr/pgmspace.h>
# include <avr/interrupt.h>
# include <avr/io.h>
#endif
#include "audio.h"
extern bool playing_note;
extern bool playing_melody;
extern uint8_t note_timbre;
#define CPU_PRESCALER 8
/*
Audio Driver: PWM
drive up to two speakers through the AVR PWM hardware-peripheral, using timer1 and/or timer3 on Atmega32U4.
the primary channel_1 can be connected to either pin PC4 PC5 or PC6 (the later being used by most AVR based keyboards) with a PMW signal generated by timer3
and an optional secondary channel_2 on either pin PB5, PB6 or PB7, with a PWM signal from timer1
alternatively, the PWM pins on PORTB can be used as only/primary speaker
*/
#if defined(AUDIO_PIN) && (AUDIO_PIN != C4) && (AUDIO_PIN != C5) && (AUDIO_PIN != C6) && (AUDIO_PIN != B5) && (AUDIO_PIN != B6) && (AUDIO_PIN != B7) && (AUDIO_PIN != D5)
# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under the AVR settings for available options."
#endif
#if (AUDIO_PIN == C4) || (AUDIO_PIN == C5) || (AUDIO_PIN == C6)
# define AUDIO1_PIN_SET
# define AUDIO1_TIMSKx TIMSK3
# define AUDIO1_TCCRxA TCCR3A
# define AUDIO1_TCCRxB TCCR3B
# define AUDIO1_ICRx ICR3
# define AUDIO1_WGMx0 WGM30
# define AUDIO1_WGMx1 WGM31
# define AUDIO1_WGMx2 WGM32
# define AUDIO1_WGMx3 WGM33
# define AUDIO1_CSx0 CS30
# define AUDIO1_CSx1 CS31
# define AUDIO1_CSx2 CS32
# if (AUDIO_PIN == C6)
# define AUDIO1_COMxy0 COM3A0
# define AUDIO1_COMxy1 COM3A1
# define AUDIO1_OCIExy OCIE3A
# define AUDIO1_OCRxy OCR3A
# define AUDIO1_PIN C6
# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPA_vect
# elif (AUDIO_PIN == C5)
# define AUDIO1_COMxy0 COM3B0
# define AUDIO1_COMxy1 COM3B1
# define AUDIO1_OCIExy OCIE3B
# define AUDIO1_OCRxy OCR3B
# define AUDIO1_PIN C5
# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPB_vect
# elif (AUDIO_PIN == C4)
# define AUDIO1_COMxy0 COM3C0
# define AUDIO1_COMxy1 COM3C1
# define AUDIO1_OCIExy OCIE3C
# define AUDIO1_OCRxy OCR3C
# define AUDIO1_PIN C4
# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPC_vect
# endif
#endif
#if defined(AUDIO_PIN) && defined(AUDIO_PIN_ALT) && (AUDIO_PIN == AUDIO_PIN_ALT)
# error "Audio feature: AUDIO_PIN and AUDIO_PIN_ALT on the same pin makes no sense."
#endif
#if ((AUDIO_PIN == B5) && ((AUDIO_PIN_ALT == B6) || (AUDIO_PIN_ALT == B7))) || ((AUDIO_PIN == B6) && ((AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B7))) || ((AUDIO_PIN == B7) && ((AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B6)))
# error "Audio feature: PORTB as AUDIO_PIN and AUDIO_PIN_ALT at the same time is not supported."
#endif
#if defined(AUDIO_PIN_ALT) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7)
# error "Audio feature: the pin selected as AUDIO_PIN_ALT is not supported."
#endif
#if (AUDIO_PIN == B5) || (AUDIO_PIN == B6) || (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B6) || (AUDIO_PIN_ALT == B7) || (AUDIO_PIN == D5)
# define AUDIO2_PIN_SET
# define AUDIO2_TIMSKx TIMSK1
# define AUDIO2_TCCRxA TCCR1A
# define AUDIO2_TCCRxB TCCR1B
# define AUDIO2_ICRx ICR1
# define AUDIO2_WGMx0 WGM10
# define AUDIO2_WGMx1 WGM11
# define AUDIO2_WGMx2 WGM12
# define AUDIO2_WGMx3 WGM13
# define AUDIO2_CSx0 CS10
# define AUDIO2_CSx1 CS11
# define AUDIO2_CSx2 CS12
# if (AUDIO_PIN == B5) || (AUDIO_PIN_ALT == B5)
# define AUDIO2_COMxy0 COM1A0
# define AUDIO2_COMxy1 COM1A1
# define AUDIO2_OCIExy OCIE1A
# define AUDIO2_OCRxy OCR1A
# define AUDIO2_PIN B5
# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect
# elif (AUDIO_PIN == B6) || (AUDIO_PIN_ALT == B6)
# define AUDIO2_COMxy0 COM1B0
# define AUDIO2_COMxy1 COM1B1
# define AUDIO2_OCIExy OCIE1B
# define AUDIO2_OCRxy OCR1B
# define AUDIO2_PIN B6
# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPB_vect
# elif (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B7)
# define AUDIO2_COMxy0 COM1C0
# define AUDIO2_COMxy1 COM1C1
# define AUDIO2_OCIExy OCIE1C
# define AUDIO2_OCRxy OCR1C
# define AUDIO2_PIN B7
# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPC_vect
# elif (AUDIO_PIN == D5) && defined(__AVR_ATmega32A__)
# pragma message "Audio support for ATmega32A is experimental and can cause crashes."
# undef AUDIO2_TIMSKx
# define AUDIO2_TIMSKx TIMSK
# define AUDIO2_COMxy0 COM1A0
# define AUDIO2_COMxy1 COM1A1
# define AUDIO2_OCIExy OCIE1A
# define AUDIO2_OCRxy OCR1A
# define AUDIO2_PIN D5
# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect
# endif
#endif
// C6 seems to be the assumed default by many existing keyboard - but sill warn the user
#if !defined(AUDIO1_PIN_SET) && !defined(AUDIO2_PIN_SET)
# pragma message "Audio feature enabled, but no suitable pin selected - see docs/feature_audio under the AVR settings for available options. Don't expect to hear anything... :-)"
// TODO: make this an error - go through the breaking-change-process and change all keyboards to the new define
#endif
// -----------------------------------------------------------------------------
#ifdef AUDIO1_PIN_SET
static float channel_1_frequency = 0.0f;
void channel_1_set_frequency(float freq) {
if (freq == 0.0f) // a pause/rest is a valid "note" with freq=0
{
// disable the output, but keep the pwm-ISR going (with the previous
// frequency) so the audio-state keeps getting updated
// Note: setting the duty-cycle 0 is not possible on non-inverting PWM mode - see the AVR data-sheet
AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0));
return;
} else {
AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); // enable output, PWM mode
}
channel_1_frequency = freq;
// set pwm period
AUDIO1_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER));
// and duty cycle
AUDIO1_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100);
}
void channel_1_start(void) {
// enable timer-counter ISR
AUDIO1_TIMSKx |= _BV(AUDIO1_OCIExy);
// enable timer-counter output
AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1);
}
void channel_1_stop(void) {
// disable timer-counter ISR
AUDIO1_TIMSKx &= ~_BV(AUDIO1_OCIExy);
// disable timer-counter output
AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0));
}
#endif
#ifdef AUDIO2_PIN_SET
static float channel_2_frequency = 0.0f;
void channel_2_set_frequency(float freq) {
if (freq == 0.0f) {
AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0));
return;
} else {
AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1);
}
channel_2_frequency = freq;
AUDIO2_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER));
AUDIO2_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100);
}
float channel_2_get_frequency(void) { return channel_2_frequency; }
void channel_2_start(void) {
AUDIO2_TIMSKx |= _BV(AUDIO2_OCIExy);
AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1);
}
void channel_2_stop(void) {
AUDIO2_TIMSKx &= ~_BV(AUDIO2_OCIExy);
AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0));
}
#endif
void audio_driver_initialize() {
#ifdef AUDIO1_PIN_SET
channel_1_stop();
setPinOutput(AUDIO1_PIN);
#endif
#ifdef AUDIO2_PIN_SET
channel_2_stop();
setPinOutput(AUDIO2_PIN);
#endif
// TCCR3A / TCCR3B: Timer/Counter #3 Control Registers TCCR3A/TCCR3B, TCCR1A/TCCR1B
// Compare Output Mode (COM3An and COM1An) = 0b00 = Normal port operation
// OC3A -- PC6
// OC3B -- PC5
// OC3C -- PC4
// OC1A -- PB5
// OC1B -- PB6
// OC1C -- PB7
// Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14. Period = ICR3, Duty Cycle OCR3A)
// OCR3A - PC6
// OCR3B - PC5
// OCR3C - PC4
// OCR1A - PB5
// OCR1B - PB6
// OCR1C - PB7
// Clock Select (CS3n) = 0b010 = Clock / 8
#ifdef AUDIO1_PIN_SET
// initialize timer-counter
AUDIO1_TCCRxA = (0 << AUDIO1_COMxy1) | (0 << AUDIO1_COMxy0) | (1 << AUDIO1_WGMx1) | (0 << AUDIO1_WGMx0);
AUDIO1_TCCRxB = (1 << AUDIO1_WGMx3) | (1 << AUDIO1_WGMx2) | (0 << AUDIO1_CSx2) | (1 << AUDIO1_CSx1) | (0 << AUDIO1_CSx0);
#endif
#ifdef AUDIO2_PIN_SET
AUDIO2_TCCRxA = (0 << AUDIO2_COMxy1) | (0 << AUDIO2_COMxy0) | (1 << AUDIO2_WGMx1) | (0 << AUDIO2_WGMx0);
AUDIO2_TCCRxB = (1 << AUDIO2_WGMx3) | (1 << AUDIO2_WGMx2) | (0 << AUDIO2_CSx2) | (1 << AUDIO2_CSx1) | (0 << AUDIO2_CSx0);
#endif
}
void audio_driver_stop() {
#ifdef AUDIO1_PIN_SET
channel_1_stop();
#endif
#ifdef AUDIO2_PIN_SET
channel_2_stop();
#endif
}
void audio_driver_start(void) {
#ifdef AUDIO1_PIN_SET
channel_1_start();
if (playing_note) {
channel_1_set_frequency(audio_get_processed_frequency(0));
}
#endif
#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET)
channel_2_start();
if (playing_note) {
channel_2_set_frequency(audio_get_processed_frequency(0));
}
#endif
}
static volatile uint32_t isr_counter = 0;
#ifdef AUDIO1_PIN_SET
ISR(AUDIO1_TIMERx_COMPy_vect) {
isr_counter++;
if (isr_counter < channel_1_frequency / (CPU_PRESCALER * 8)) return;
isr_counter = 0;
bool state_changed = audio_update_state();
if (!playing_note && !playing_melody) {
channel_1_stop();
# ifdef AUDIO2_PIN_SET
channel_2_stop();
# endif
return;
}
if (state_changed) {
channel_1_set_frequency(audio_get_processed_frequency(0));
# ifdef AUDIO2_PIN_SET
if (audio_get_number_of_active_tones() > 1) {
channel_2_set_frequency(audio_get_processed_frequency(1));
} else {
channel_2_stop();
}
# endif
}
}
#endif
#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET)
ISR(AUDIO2_TIMERx_COMPy_vect) {
isr_counter++;
if (isr_counter < channel_2_frequency / (CPU_PRESCALER * 8)) return;
isr_counter = 0;
bool state_changed = audio_update_state();
if (!playing_note && !playing_melody) {
channel_2_stop();
return;
}
if (state_changed) {
channel_2_set_frequency(audio_get_processed_frequency(0));
}
}
#endif

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/* Copyright 2019 Jack Humbert
* Copyright 2020 JohSchneider
*
* 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 2 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/>.
*/
#pragma once
#ifndef A4
# define A4 PAL_LINE(GPIOA, 4)
#endif
#ifndef A5
# define A5 PAL_LINE(GPIOA, 5)
#endif
/**
* Size of the dac_buffer arrays. All must be the same size.
*/
#define AUDIO_DAC_BUFFER_SIZE 256U
/**
* Highest value allowed sample value.
* since the DAC is limited to 12 bit, the absolute max is 0xfff = 4095U;
* lower values adjust the peak-voltage aka volume down.
* adjusting this value has only an effect on a sample-buffer whose values are
* are NOT pregenerated - see square-wave
*/
#ifndef AUDIO_DAC_SAMPLE_MAX
# define AUDIO_DAC_SAMPLE_MAX 4095U
#endif
#if !defined(AUDIO_DAC_SAMPLE_RATE) && !defined(AUDIO_MAX_SIMULTANEOUS_TONES) && !defined(AUDIO_DAC_QUALITY_VERY_LOW) && !defined(AUDIO_DAC_QUALITY_LOW) && !defined(AUDIO_DAC_QUALITY_HIGH) && !defined(AUDIO_DAC_QUALITY_VERY_HIGH)
# define AUDIO_DAC_QUALITY_SANE_MINIMUM
#endif
/**
* These presets allow you to quickly switch between quality settings for
* the DAC. The sample rate and maximum number of simultaneous tones roughly
* has an inverse relationship - slightly higher sample rates may be possible.
*
* NOTE: a high sample-rate results in a higher cpu-load, which might lead to
* (audible) discontinuities and/or starve other processes of cpu-time
* (like RGB-led back-lighting, ...)
*/
#ifdef AUDIO_DAC_QUALITY_VERY_LOW
# define AUDIO_DAC_SAMPLE_RATE 11025U
# define AUDIO_MAX_SIMULTANEOUS_TONES 8
#endif
#ifdef AUDIO_DAC_QUALITY_LOW
# define AUDIO_DAC_SAMPLE_RATE 22050U
# define AUDIO_MAX_SIMULTANEOUS_TONES 4
#endif
#ifdef AUDIO_DAC_QUALITY_HIGH
# define AUDIO_DAC_SAMPLE_RATE 44100U
# define AUDIO_MAX_SIMULTANEOUS_TONES 2
#endif
#ifdef AUDIO_DAC_QUALITY_VERY_HIGH
# define AUDIO_DAC_SAMPLE_RATE 88200U
# define AUDIO_MAX_SIMULTANEOUS_TONES 1
#endif
#ifdef AUDIO_DAC_QUALITY_SANE_MINIMUM
/* a sane-minimum config: with a trade-off between cpu-load and tone-range
*
* the (currently) highest defined note is NOTE_B8 with 7902Hz; if we now
* aim for an even even multiple of the buffer-size, we end up with:
* ( roundUptoPow2(highest note / AUDIO_DAC_BUFFER_SIZE) * nyquist-rate * AUDIO_DAC_BUFFER_SIZE)
* 7902/256 = 30.867 * 2 * 256 ~= 16384
* which works out (but the 'scope shows some sampling artifacts with lower harmonics :-P)
*/
# define AUDIO_DAC_SAMPLE_RATE 16384U
# define AUDIO_MAX_SIMULTANEOUS_TONES 8
#endif
/**
* Effective bit-rate of the DAC. 44.1khz is the standard for most audio - any
* lower will sacrifice perceptible audio quality. Any higher will limit the
* number of simultaneous tones. In most situations, a tenth (1/10) of the
* sample rate is where notes become unbearable.
*/
#ifndef AUDIO_DAC_SAMPLE_RATE
# define AUDIO_DAC_SAMPLE_RATE 44100U
#endif
/**
* The number of tones that can be played simultaneously. If too high a value
* is used here, the keyboard will freeze and glitch-out when that many tones
* are being played.
*/
#ifndef AUDIO_MAX_SIMULTANEOUS_TONES
# define AUDIO_MAX_SIMULTANEOUS_TONES 2
#endif
/**
* The default value of the DAC when not playing anything. Certain hardware
* setups may require a high (AUDIO_DAC_SAMPLE_MAX) or low (0) value here.
* Since multiple added sine waves tend to oscillate around the midpoint,
* and possibly never/rarely reach either 0 of MAX, 1/2 MAX can be a
* reasonable default value.
*/
#ifndef AUDIO_DAC_OFF_VALUE
# define AUDIO_DAC_OFF_VALUE AUDIO_DAC_SAMPLE_MAX / 2
#endif
#if AUDIO_DAC_OFF_VALUE > AUDIO_DAC_SAMPLE_MAX
# error "AUDIO_DAC: OFF_VALUE may not be larger than SAMPLE_MAX"
#endif
/**
*user overridable sample generation/processing
*/
uint16_t dac_value_generate(void);

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/* Copyright 2016-2019 Jack Humbert
* Copyright 2020 JohSchneider
*
* 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 2 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 "audio.h"
#include <ch.h>
#include <hal.h>
/*
Audio Driver: DAC
which utilizes the dac unit many STM32 are equipped with, to output a modulated waveform from samples stored in the dac_buffer_* array who are passed to the hardware through DMA
it is also possible to have a custom sample-LUT by implementing/overriding 'dac_value_generate'
this driver allows for multiple simultaneous tones to be played through one single channel by doing additive wave-synthesis
*/
#if !defined(AUDIO_PIN)
# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC additive)' for available options."
#endif
#if defined(AUDIO_PIN_ALT) && !defined(AUDIO_PIN_ALT_AS_NEGATIVE)
# pragma message "Audio feature: AUDIO_PIN_ALT set, but not AUDIO_PIN_ALT_AS_NEGATIVE - pin will be left unused; audio might still work though."
#endif
#if !defined(AUDIO_PIN_ALT)
// no ALT pin defined is valid, but the c-ifs below need some value set
# define AUDIO_PIN_ALT PAL_NOLINE
#endif
#if !defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
# define AUDIO_DAC_SAMPLE_WAVEFORM_SINE
#endif
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SINE
/* one full sine wave over [0,2*pi], but shifted up one amplitude and left pi/4; for the samples to start at 0
*/
static const dacsample_t dac_buffer_sine[AUDIO_DAC_BUFFER_SIZE] = {
// 256 values, max 4095
0x0, 0x1, 0x2, 0x6, 0xa, 0xf, 0x16, 0x1e, 0x27, 0x32, 0x3d, 0x4a, 0x58, 0x67, 0x78, 0x89, 0x9c, 0xb0, 0xc5, 0xdb, 0xf2, 0x10a, 0x123, 0x13e, 0x159, 0x175, 0x193, 0x1b1, 0x1d1, 0x1f1, 0x212, 0x235, 0x258, 0x27c, 0x2a0, 0x2c6, 0x2ed, 0x314, 0x33c, 0x365, 0x38e, 0x3b8, 0x3e3, 0x40e, 0x43a, 0x467, 0x494, 0x4c2, 0x4f0, 0x51f, 0x54e, 0x57d, 0x5ad, 0x5dd, 0x60e, 0x63f, 0x670, 0x6a1, 0x6d3, 0x705, 0x737, 0x769, 0x79b, 0x7cd, 0x800, 0x832, 0x864, 0x896, 0x8c8, 0x8fa, 0x92c, 0x95e, 0x98f, 0x9c0, 0x9f1, 0xa22, 0xa52, 0xa82, 0xab1, 0xae0, 0xb0f, 0xb3d, 0xb6b, 0xb98, 0xbc5, 0xbf1, 0xc1c, 0xc47, 0xc71, 0xc9a, 0xcc3, 0xceb, 0xd12, 0xd39, 0xd5f, 0xd83, 0xda7, 0xdca, 0xded, 0xe0e, 0xe2e, 0xe4e, 0xe6c, 0xe8a, 0xea6, 0xec1, 0xedc, 0xef5, 0xf0d, 0xf24, 0xf3a, 0xf4f, 0xf63, 0xf76, 0xf87, 0xf98, 0xfa7, 0xfb5, 0xfc2, 0xfcd, 0xfd8, 0xfe1, 0xfe9, 0xff0, 0xff5, 0xff9, 0xffd, 0xffe,
0xfff, 0xffe, 0xffd, 0xff9, 0xff5, 0xff0, 0xfe9, 0xfe1, 0xfd8, 0xfcd, 0xfc2, 0xfb5, 0xfa7, 0xf98, 0xf87, 0xf76, 0xf63, 0xf4f, 0xf3a, 0xf24, 0xf0d, 0xef5, 0xedc, 0xec1, 0xea6, 0xe8a, 0xe6c, 0xe4e, 0xe2e, 0xe0e, 0xded, 0xdca, 0xda7, 0xd83, 0xd5f, 0xd39, 0xd12, 0xceb, 0xcc3, 0xc9a, 0xc71, 0xc47, 0xc1c, 0xbf1, 0xbc5, 0xb98, 0xb6b, 0xb3d, 0xb0f, 0xae0, 0xab1, 0xa82, 0xa52, 0xa22, 0x9f1, 0x9c0, 0x98f, 0x95e, 0x92c, 0x8fa, 0x8c8, 0x896, 0x864, 0x832, 0x800, 0x7cd, 0x79b, 0x769, 0x737, 0x705, 0x6d3, 0x6a1, 0x670, 0x63f, 0x60e, 0x5dd, 0x5ad, 0x57d, 0x54e, 0x51f, 0x4f0, 0x4c2, 0x494, 0x467, 0x43a, 0x40e, 0x3e3, 0x3b8, 0x38e, 0x365, 0x33c, 0x314, 0x2ed, 0x2c6, 0x2a0, 0x27c, 0x258, 0x235, 0x212, 0x1f1, 0x1d1, 0x1b1, 0x193, 0x175, 0x159, 0x13e, 0x123, 0x10a, 0xf2, 0xdb, 0xc5, 0xb0, 0x9c, 0x89, 0x78, 0x67, 0x58, 0x4a, 0x3d, 0x32, 0x27, 0x1e, 0x16, 0xf, 0xa, 0x6, 0x2, 0x1};
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SINE
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
static const dacsample_t dac_buffer_triangle[AUDIO_DAC_BUFFER_SIZE] = {
// 256 values, max 4095
0x0, 0x20, 0x40, 0x60, 0x80, 0xa0, 0xc0, 0xe0, 0x100, 0x120, 0x140, 0x160, 0x180, 0x1a0, 0x1c0, 0x1e0, 0x200, 0x220, 0x240, 0x260, 0x280, 0x2a0, 0x2c0, 0x2e0, 0x300, 0x320, 0x340, 0x360, 0x380, 0x3a0, 0x3c0, 0x3e0, 0x400, 0x420, 0x440, 0x460, 0x480, 0x4a0, 0x4c0, 0x4e0, 0x500, 0x520, 0x540, 0x560, 0x580, 0x5a0, 0x5c0, 0x5e0, 0x600, 0x620, 0x640, 0x660, 0x680, 0x6a0, 0x6c0, 0x6e0, 0x700, 0x720, 0x740, 0x760, 0x780, 0x7a0, 0x7c0, 0x7e0, 0x800, 0x81f, 0x83f, 0x85f, 0x87f, 0x89f, 0x8bf, 0x8df, 0x8ff, 0x91f, 0x93f, 0x95f, 0x97f, 0x99f, 0x9bf, 0x9df, 0x9ff, 0xa1f, 0xa3f, 0xa5f, 0xa7f, 0xa9f, 0xabf, 0xadf, 0xaff, 0xb1f, 0xb3f, 0xb5f, 0xb7f, 0xb9f, 0xbbf, 0xbdf, 0xbff, 0xc1f, 0xc3f, 0xc5f, 0xc7f, 0xc9f, 0xcbf, 0xcdf, 0xcff, 0xd1f, 0xd3f, 0xd5f, 0xd7f, 0xd9f, 0xdbf, 0xddf, 0xdff, 0xe1f, 0xe3f, 0xe5f, 0xe7f, 0xe9f, 0xebf, 0xedf, 0xeff, 0xf1f, 0xf3f, 0xf5f, 0xf7f, 0xf9f, 0xfbf, 0xfdf,
0xfff, 0xfdf, 0xfbf, 0xf9f, 0xf7f, 0xf5f, 0xf3f, 0xf1f, 0xeff, 0xedf, 0xebf, 0xe9f, 0xe7f, 0xe5f, 0xe3f, 0xe1f, 0xdff, 0xddf, 0xdbf, 0xd9f, 0xd7f, 0xd5f, 0xd3f, 0xd1f, 0xcff, 0xcdf, 0xcbf, 0xc9f, 0xc7f, 0xc5f, 0xc3f, 0xc1f, 0xbff, 0xbdf, 0xbbf, 0xb9f, 0xb7f, 0xb5f, 0xb3f, 0xb1f, 0xaff, 0xadf, 0xabf, 0xa9f, 0xa7f, 0xa5f, 0xa3f, 0xa1f, 0x9ff, 0x9df, 0x9bf, 0x99f, 0x97f, 0x95f, 0x93f, 0x91f, 0x8ff, 0x8df, 0x8bf, 0x89f, 0x87f, 0x85f, 0x83f, 0x81f, 0x800, 0x7e0, 0x7c0, 0x7a0, 0x780, 0x760, 0x740, 0x720, 0x700, 0x6e0, 0x6c0, 0x6a0, 0x680, 0x660, 0x640, 0x620, 0x600, 0x5e0, 0x5c0, 0x5a0, 0x580, 0x560, 0x540, 0x520, 0x500, 0x4e0, 0x4c0, 0x4a0, 0x480, 0x460, 0x440, 0x420, 0x400, 0x3e0, 0x3c0, 0x3a0, 0x380, 0x360, 0x340, 0x320, 0x300, 0x2e0, 0x2c0, 0x2a0, 0x280, 0x260, 0x240, 0x220, 0x200, 0x1e0, 0x1c0, 0x1a0, 0x180, 0x160, 0x140, 0x120, 0x100, 0xe0, 0xc0, 0xa0, 0x80, 0x60, 0x40, 0x20};
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
static const dacsample_t dac_buffer_square[AUDIO_DAC_BUFFER_SIZE] = {
[0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = 0, // first and
[AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX, // second half
};
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
/*
// four steps: 0, 1/3, 2/3 and 1
static const dacsample_t dac_buffer_staircase[AUDIO_DAC_BUFFER_SIZE] = {
[0 ... AUDIO_DAC_BUFFER_SIZE/3 -1 ] = 0,
[AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE / 2 -1 ] = AUDIO_DAC_SAMPLE_MAX / 3,
[AUDIO_DAC_BUFFER_SIZE / 2 ... 3 * AUDIO_DAC_BUFFER_SIZE / 4 -1 ] = 2 * AUDIO_DAC_SAMPLE_MAX / 3,
[3 * AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE -1 ] = AUDIO_DAC_SAMPLE_MAX,
}
*/
#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
static const dacsample_t dac_buffer_trapezoid[AUDIO_DAC_BUFFER_SIZE] = {0x0, 0x1f, 0x7f, 0xdf, 0x13f, 0x19f, 0x1ff, 0x25f, 0x2bf, 0x31f, 0x37f, 0x3df, 0x43f, 0x49f, 0x4ff, 0x55f, 0x5bf, 0x61f, 0x67f, 0x6df, 0x73f, 0x79f, 0x7ff, 0x85f, 0x8bf, 0x91f, 0x97f, 0x9df, 0xa3f, 0xa9f, 0xaff, 0xb5f, 0xbbf, 0xc1f, 0xc7f, 0xcdf, 0xd3f, 0xd9f, 0xdff, 0xe5f, 0xebf, 0xf1f, 0xf7f, 0xfdf, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff,
0xfff, 0xfdf, 0xf7f, 0xf1f, 0xebf, 0xe5f, 0xdff, 0xd9f, 0xd3f, 0xcdf, 0xc7f, 0xc1f, 0xbbf, 0xb5f, 0xaff, 0xa9f, 0xa3f, 0x9df, 0x97f, 0x91f, 0x8bf, 0x85f, 0x7ff, 0x79f, 0x73f, 0x6df, 0x67f, 0x61f, 0x5bf, 0x55f, 0x4ff, 0x49f, 0x43f, 0x3df, 0x37f, 0x31f, 0x2bf, 0x25f, 0x1ff, 0x19f, 0x13f, 0xdf, 0x7f, 0x1f, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0};
#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
static dacsample_t dac_buffer_empty[AUDIO_DAC_BUFFER_SIZE] = {AUDIO_DAC_OFF_VALUE};
/* keep track of the sample position for for each frequency */
static float dac_if[AUDIO_MAX_SIMULTANEOUS_TONES] = {0.0};
static float active_tones_snapshot[AUDIO_MAX_SIMULTANEOUS_TONES] = {0, 0};
static uint8_t active_tones_snapshot_length = 0;
typedef enum {
OUTPUT_SHOULD_START,
OUTPUT_RUN_NORMALLY,
// path 1: wait for zero, then change/update active tones
OUTPUT_TONES_CHANGED,
OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE,
// path 2: hardware should stop, wait for zero then turn output off = stop the timer
OUTPUT_SHOULD_STOP,
OUTPUT_REACHED_ZERO_BEFORE_OFF,
OUTPUT_OFF,
OUTPUT_OFF_1,
OUTPUT_OFF_2, // trailing off: giving the DAC two more conversion cycles until the AUDIO_DAC_OFF_VALUE reaches the output, then turn the timer off, which leaves the output at that level
number_of_output_states
} output_states_t;
output_states_t state = OUTPUT_OFF_2;
/**
* Generation of the waveform being passed to the callback. Declared weak so users
* can override it with their own wave-forms/noises.
*/
__attribute__((weak)) uint16_t dac_value_generate(void) {
// DAC is running/asking for values but snapshot length is zero -> must be playing a pause
if (active_tones_snapshot_length == 0) {
return AUDIO_DAC_OFF_VALUE;
}
/* doing additive wave synthesis over all currently playing tones = adding up
* sine-wave-samples for each frequency, scaled by the number of active tones
*/
uint16_t value = 0;
float frequency = 0.0f;
for (uint8_t i = 0; i < active_tones_snapshot_length; i++) {
/* Note: a user implementation does not have to rely on the active_tones_snapshot, but
* could directly query the active frequencies through audio_get_processed_frequency */
frequency = active_tones_snapshot[i];
dac_if[i] = dac_if[i] + ((frequency * AUDIO_DAC_BUFFER_SIZE) / AUDIO_DAC_SAMPLE_RATE) * 2 / 3;
/*Note: the 2/3 are necessary to get the correct frequencies on the
* DAC output (as measured with an oscilloscope), since the gpt
* timer runs with 3*AUDIO_DAC_SAMPLE_RATE; and the DAC callback
* is called twice per conversion.*/
dac_if[i] = fmod(dac_if[i], AUDIO_DAC_BUFFER_SIZE);
// Wavetable generation/lookup
uint16_t dac_i = (uint16_t)dac_if[i];
#if defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE)
value += dac_buffer_sine[dac_i] / active_tones_snapshot_length;
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE)
value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length;
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
value += dac_buffer_trapezoid[dac_i] / active_tones_snapshot_length;
#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE)
value += dac_buffer_square[dac_i] / active_tones_snapshot_length;
#endif
/*
// SINE
value += dac_buffer_sine[dac_i] / active_tones_snapshot_length / 3;
// TRIANGLE
value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length / 3;
// SQUARE
value += dac_buffer_square[dac_i] / active_tones_snapshot_length / 3;
//NOTE: combination of these three wave-forms is more exemplary - and doesn't sound particularly good :-P
*/
// STAIRS (mostly usefully as test-pattern)
// value_avg = dac_buffer_staircase[dac_i] / active_tones_snapshot_length;
}
return value;
}
/**
* DAC streaming callback. Does all of the main computing for playing songs.
*
* Note: chibios calls this CB twice: during the 'half buffer event', and the 'full buffer event'.
*/
static void dac_end(DACDriver *dacp) {
dacsample_t *sample_p = (dacp)->samples;
// work on the other half of the buffer
if (dacIsBufferComplete(dacp)) {
sample_p += AUDIO_DAC_BUFFER_SIZE / 2; // 'half_index'
}
for (uint8_t s = 0; s < AUDIO_DAC_BUFFER_SIZE / 2; s++) {
if (OUTPUT_OFF <= state) {
sample_p[s] = AUDIO_DAC_OFF_VALUE;
continue;
} else {
sample_p[s] = dac_value_generate();
}
/* zero crossing (or approach, whereas zero == DAC_OFF_VALUE, which can be configured to anything from 0 to DAC_SAMPLE_MAX)
* ============================*=*========================== AUDIO_DAC_SAMPLE_MAX
* * *
* * *
* ---------------------------------------------------------
* * * } AUDIO_DAC_SAMPLE_MAX/100
* --------------------------------------------------------- AUDIO_DAC_OFF_VALUE
* * * } AUDIO_DAC_SAMPLE_MAX/100
* ---------------------------------------------------------
* *
* * *
* * *
* =====*=*================================================= 0x0
*/
if (((sample_p[s] + (AUDIO_DAC_SAMPLE_MAX / 100)) > AUDIO_DAC_OFF_VALUE) && // value approaches from below
(sample_p[s] < (AUDIO_DAC_OFF_VALUE + (AUDIO_DAC_SAMPLE_MAX / 100))) // or above
) {
if ((OUTPUT_SHOULD_START == state) && (active_tones_snapshot_length > 0)) {
state = OUTPUT_RUN_NORMALLY;
} else if (OUTPUT_TONES_CHANGED == state) {
state = OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE;
} else if (OUTPUT_SHOULD_STOP == state) {
state = OUTPUT_REACHED_ZERO_BEFORE_OFF;
}
}
// still 'ramping up', reset the output to OFF_VALUE until the generated values reach that value, to do a smooth handover
if (OUTPUT_SHOULD_START == state) {
sample_p[s] = AUDIO_DAC_OFF_VALUE;
}
if ((OUTPUT_SHOULD_START == state) || (OUTPUT_REACHED_ZERO_BEFORE_OFF == state) || (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state)) {
uint8_t active_tones = MIN(AUDIO_MAX_SIMULTANEOUS_TONES, audio_get_number_of_active_tones());
active_tones_snapshot_length = 0;
// update the snapshot - once, and only on occasion that something changed;
// -> saves cpu cycles (?)
for (uint8_t i = 0; i < active_tones; i++) {
float freq = audio_get_processed_frequency(i);
if (freq > 0) { // disregard 'rest' notes, with valid frequency 0.0f; which would only lower the resulting waveform volume during the additive synthesis step
active_tones_snapshot[active_tones_snapshot_length++] = freq;
}
}
if ((0 == active_tones_snapshot_length) && (OUTPUT_REACHED_ZERO_BEFORE_OFF == state)) {
state = OUTPUT_OFF;
}
if (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state) {
state = OUTPUT_RUN_NORMALLY;
}
}
}
// update audio internal state (note position, current_note, ...)
if (audio_update_state()) {
if (OUTPUT_SHOULD_STOP != state) {
state = OUTPUT_TONES_CHANGED;
}
}
if (OUTPUT_OFF <= state) {
if (OUTPUT_OFF_2 == state) {
// stopping timer6 = stopping the DAC at whatever value it is currently pushing to the output = AUDIO_DAC_OFF_VALUE
gptStopTimer(&GPTD6);
} else {
state++;
}
}
}
static void dac_error(DACDriver *dacp, dacerror_t err) {
(void)dacp;
(void)err;
chSysHalt("DAC failure. halp");
}
static const GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE * 3,
.callback = NULL,
.cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
.dier = 0U};
static const DACConfig dac_conf = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
/**
* @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
* on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
* to be a third of what we expect.
*
* Here are all the values for DAC_TRG (TSEL in the ref manual)
* TIM15_TRGO 0b011
* TIM2_TRGO 0b100
* TIM3_TRGO 0b001
* TIM6_TRGO 0b000
* TIM7_TRGO 0b010
* EXTI9 0b110
* SWTRIG 0b111
*/
static const DACConversionGroup dac_conv_cfg = {.num_channels = 1U, .end_cb = dac_end, .error_cb = dac_error, .trigger = DAC_TRG(0b000)};
void audio_driver_initialize() {
if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
palSetLineMode(A4, PAL_MODE_INPUT_ANALOG);
dacStart(&DACD1, &dac_conf);
}
if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
palSetLineMode(A5, PAL_MODE_INPUT_ANALOG);
dacStart(&DACD2, &dac_conf);
}
/* enable the output buffer, to directly drive external loads with no additional circuitry
*
* see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
* Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
* Note: enabling the output buffer imparts an additional dc-offset of a couple mV
*
* this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
* (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
*/
DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
if (AUDIO_PIN == A4) {
dacStartConversion(&DACD1, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
} else if (AUDIO_PIN == A5) {
dacStartConversion(&DACD2, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
}
// no inverted/out-of-phase waveform (yet?), only pulling AUDIO_PIN_ALT to AUDIO_DAC_OFF_VALUE
#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
if (AUDIO_PIN_ALT == A4) {
dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
} else if (AUDIO_PIN_ALT == A5) {
dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
}
#endif
gptStart(&GPTD6, &gpt6cfg1);
}
void audio_driver_stop(void) { state = OUTPUT_SHOULD_STOP; }
void audio_driver_start(void) {
gptStartContinuous(&GPTD6, 2U);
for (uint8_t i = 0; i < AUDIO_MAX_SIMULTANEOUS_TONES; i++) {
dac_if[i] = 0.0f;
active_tones_snapshot[i] = 0.0f;
}
active_tones_snapshot_length = 0;
state = OUTPUT_SHOULD_START;
}

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/* Copyright 2016-2020 Jack Humbert
* Copyright 2020 JohSchneider
*
* 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 2 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 "audio.h"
#include "ch.h"
#include "hal.h"
/*
Audio Driver: DAC
which utilizes both channels of the DAC unit many STM32 are equipped with to output a modulated square-wave, from precomputed samples stored in a buffer, which is passed to the hardware through DMA
this driver can either be used to drive to separate speakers, wired to A4+Gnd and A5+Gnd, which allows two tones to be played simultaneously
OR
one speaker wired to A4+A5 with the AUDIO_PIN_ALT_AS_NEGATIVE define set - see docs/feature_audio
*/
#if !defined(AUDIO_PIN)
# pragma message "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC basic)' for available options."
// TODO: make this an 'error' instead; go through a breaking change, and add AUDIO_PIN A5 to all keyboards currently using AUDIO on STM32 based boards? - for now: set the define here
# define AUDIO_PIN A5
#endif
// check configuration for ONE speaker, connected to both DAC pins
#if defined(AUDIO_PIN_ALT_AS_NEGATIVE) && !defined(AUDIO_PIN_ALT)
# error "Audio feature: AUDIO_PIN_ALT_AS_NEGATIVE set, but no pin configured as AUDIO_PIN_ALT"
#endif
#ifndef AUDIO_PIN_ALT
// no ALT pin defined is valid, but the c-ifs below need some value set
# define AUDIO_PIN_ALT -1
#endif
#if !defined(AUDIO_STATE_TIMER)
# define AUDIO_STATE_TIMER GPTD8
#endif
// square-wave
static const dacsample_t dac_buffer_1[AUDIO_DAC_BUFFER_SIZE] = {
// First half is max, second half is 0
[0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = AUDIO_DAC_SAMPLE_MAX,
[AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = 0,
};
// square-wave
static const dacsample_t dac_buffer_2[AUDIO_DAC_BUFFER_SIZE] = {
// opposite of dac_buffer above
[0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = 0,
[AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX,
};
GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
.callback = NULL,
.cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
.dier = 0U};
GPTConfig gpt7cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
.callback = NULL,
.cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
.dier = 0U};
static void gpt_audio_state_cb(GPTDriver *gptp);
GPTConfig gptStateUpdateCfg = {.frequency = 10,
.callback = gpt_audio_state_cb,
.cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
.dier = 0U};
static const DACConfig dac_conf_ch1 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
static const DACConfig dac_conf_ch2 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
/**
* @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
* on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
* to be a third of what we expect.
*
* Here are all the values for DAC_TRG (TSEL in the ref manual)
* TIM15_TRGO 0b011
* TIM2_TRGO 0b100
* TIM3_TRGO 0b001
* TIM6_TRGO 0b000
* TIM7_TRGO 0b010
* EXTI9 0b110
* SWTRIG 0b111
*/
static const DACConversionGroup dac_conv_grp_ch1 = {.num_channels = 1U, .trigger = DAC_TRG(0b000)};
static const DACConversionGroup dac_conv_grp_ch2 = {.num_channels = 1U, .trigger = DAC_TRG(0b010)};
void channel_1_start(void) {
gptStart(&GPTD6, &gpt6cfg1);
gptStartContinuous(&GPTD6, 2U);
palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
}
void channel_1_stop(void) {
gptStopTimer(&GPTD6);
palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL);
palSetPad(GPIOA, 4);
}
static float channel_1_frequency = 0.0f;
void channel_1_set_frequency(float freq) {
channel_1_frequency = freq;
channel_1_stop();
if (freq <= 0.0) // a pause/rest has freq=0
return;
gpt6cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
channel_1_start();
}
float channel_1_get_frequency(void) { return channel_1_frequency; }
void channel_2_start(void) {
gptStart(&GPTD7, &gpt7cfg1);
gptStartContinuous(&GPTD7, 2U);
palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
}
void channel_2_stop(void) {
gptStopTimer(&GPTD7);
palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL);
palSetPad(GPIOA, 5);
}
static float channel_2_frequency = 0.0f;
void channel_2_set_frequency(float freq) {
channel_2_frequency = freq;
channel_2_stop();
if (freq <= 0.0) // a pause/rest has freq=0
return;
gpt7cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
channel_2_start();
}
float channel_2_get_frequency(void) { return channel_2_frequency; }
static void gpt_audio_state_cb(GPTDriver *gptp) {
if (audio_update_state()) {
#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
// one piezo/speaker connected to both audio pins, the generated square-waves are inverted
channel_1_set_frequency(audio_get_processed_frequency(0));
channel_2_set_frequency(audio_get_processed_frequency(0));
#else // two separate audio outputs/speakers
// primary speaker on A4, optional secondary on A5
if (AUDIO_PIN == A4) {
channel_1_set_frequency(audio_get_processed_frequency(0));
if (AUDIO_PIN_ALT == A5) {
if (audio_get_number_of_active_tones() > 1) {
channel_2_set_frequency(audio_get_processed_frequency(1));
} else {
channel_2_stop();
}
}
}
// primary speaker on A5, optional secondary on A4
if (AUDIO_PIN == A5) {
channel_2_set_frequency(audio_get_processed_frequency(0));
if (AUDIO_PIN_ALT == A4) {
if (audio_get_number_of_active_tones() > 1) {
channel_1_set_frequency(audio_get_processed_frequency(1));
} else {
channel_1_stop();
}
}
}
#endif
}
}
void audio_driver_initialize() {
if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
dacStart(&DACD1, &dac_conf_ch1);
// initial setup of the dac-triggering timer is still required, even
// though it gets reconfigured and restarted later on
gptStart(&GPTD6, &gpt6cfg1);
}
if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
dacStart(&DACD2, &dac_conf_ch2);
gptStart(&GPTD7, &gpt7cfg1);
}
/* enable the output buffer, to directly drive external loads with no additional circuitry
*
* see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
* Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
* Note: enabling the output buffer imparts an additional dc-offset of a couple mV
*
* this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
* (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
*/
DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
// start state-updater
gptStart(&AUDIO_STATE_TIMER, &gptStateUpdateCfg);
}
void audio_driver_stop(void) {
if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
gptStopTimer(&GPTD6);
// stop the ongoing conversion and put the output in a known state
dacStopConversion(&DACD1);
dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
}
if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
gptStopTimer(&GPTD7);
dacStopConversion(&DACD2);
dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
}
gptStopTimer(&AUDIO_STATE_TIMER);
}
void audio_driver_start(void) {
if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
dacStartConversion(&DACD1, &dac_conv_grp_ch1, (dacsample_t *)dac_buffer_1, AUDIO_DAC_BUFFER_SIZE);
}
if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
dacStartConversion(&DACD2, &dac_conv_grp_ch2, (dacsample_t *)dac_buffer_2, AUDIO_DAC_BUFFER_SIZE);
}
gptStartContinuous(&AUDIO_STATE_TIMER, 2U);
}

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/* Copyright 2020 Jack Humbert
* Copyright 2020 JohSchneider
*
* 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 2 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/>.
*/
#pragma once
#if !defined(AUDIO_PWM_DRIVER)
// NOTE: Timer2 seems to be used otherwise in QMK, otherwise we could default to A5 (= TIM2_CH1, with PWMD2 and alternate-function(1))
# define AUDIO_PWM_DRIVER PWMD1
#endif
#if !defined(AUDIO_PWM_CHANNEL)
// NOTE: sticking to the STM data-sheet numbering: TIMxCH1 to TIMxCH4
// default: STM32F303CC PA8+TIM1_CH1 -> 1
# define AUDIO_PWM_CHANNEL 1
#endif
#if !defined(AUDIO_PWM_PAL_MODE)
// pin-alternate function: see the data-sheet for which pin needs what AF to connect to TIMx_CHy
// default: STM32F303CC PA8+TIM1_CH1 -> 6
# define AUDIO_PWM_PAL_MODE 6
#endif
#if !defined(AUDIO_STATE_TIMER)
// timer used to trigger updates in the audio-system, configured/enabled in chibios mcuconf.
// Tim6 is the default for "larger" STMs, smaller ones might not have this one (enabled) and need to switch to a different one (e.g.: STM32F103 has only Tim1-Tim4)
# define AUDIO_STATE_TIMER GPTD6
#endif

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quantum/audio/driver_chibios_pwm_hardware.c Normal file
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/* Copyright 2020 Jack Humbert
* Copyright 2020 JohSchneider
*
* 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 2 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/>.
*/
/*
Audio Driver: PWM
the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
this driver uses the chibios-PWM system to produce a square-wave on specific output pins that are connected to the PWM hardware.
The hardware directly toggles the pin via its alternate function. see your MCUs data-sheet for which pin can be driven by what timer - looking for TIMx_CHy and the corresponding alternate function.
*/
#include "audio.h"
#include "ch.h"
#include "hal.h"
#if !defined(AUDIO_PIN)
# error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
#endif
extern bool playing_note;
extern bool playing_melody;
extern uint8_t note_timbre;
static PWMConfig pwmCFG = {
.frequency = 100000, /* PWM clock frequency */
// CHIBIOS-BUG? can't set the initial period to <2, or the pwm (hard or software) takes ~130ms with .frequency=500000 for a pwmChangePeriod to take effect; with no output=silence in the meantime
.period = 2, /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
.callback = NULL, /* no callback, the hardware directly toggles the pin */
.channels =
{
#if AUDIO_PWM_CHANNEL == 4
{PWM_OUTPUT_DISABLED, NULL}, /* channel 0 -> TIMx_CH1 */
{PWM_OUTPUT_DISABLED, NULL}, /* channel 1 -> TIMx_CH2 */
{PWM_OUTPUT_DISABLED, NULL}, /* channel 2 -> TIMx_CH3 */
{PWM_OUTPUT_ACTIVE_HIGH, NULL} /* channel 3 -> TIMx_CH4 */
#elif AUDIO_PWM_CHANNEL == 3
{PWM_OUTPUT_DISABLED, NULL},
{PWM_OUTPUT_DISABLED, NULL},
{PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH3 */
{PWM_OUTPUT_DISABLED, NULL}
#elif AUDIO_PWM_CHANNEL == 2
{PWM_OUTPUT_DISABLED, NULL},
{PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH2 */
{PWM_OUTPUT_DISABLED, NULL},
{PWM_OUTPUT_DISABLED, NULL}
#else /*fallback to CH1 */
{PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH1 */
{PWM_OUTPUT_DISABLED, NULL},
{PWM_OUTPUT_DISABLED, NULL},
{PWM_OUTPUT_DISABLED, NULL}
#endif
},
};
static float channel_1_frequency = 0.0f;
void channel_1_set_frequency(float freq) {
channel_1_frequency = freq;
if (freq <= 0.0) // a pause/rest has freq=0
return;
pwmcnt_t period = (pwmCFG.frequency / freq);
pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
// adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
}
float channel_1_get_frequency(void) { return channel_1_frequency; }
void channel_1_start(void) {
pwmStop(&AUDIO_PWM_DRIVER);
pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
}
void channel_1_stop(void) { pwmStop(&AUDIO_PWM_DRIVER); }
static void gpt_callback(GPTDriver *gptp);
GPTConfig gptCFG = {
/* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
the tempo (which might vary!) is in bpm (beats per minute)
therefore: if the timer ticks away at .frequency = (60*64)Hz,
and the .interval counts from 64 downwards - audio_update_state is
called just often enough to not miss any notes
*/
.frequency = 60 * 64,
.callback = gpt_callback,
};
void audio_driver_initialize(void) {
pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
// connect the AUDIO_PIN to the PWM hardware
#if defined(USE_GPIOV1) // STM32F103C8
palSetLineMode(AUDIO_PIN, PAL_MODE_STM32_ALTERNATE_PUSHPULL);
#else // GPIOv2 (or GPIOv3 for f4xx, which is the same/compatible at this command)
palSetLineMode(AUDIO_PIN, PAL_STM32_MODE_ALTERNATE | PAL_STM32_ALTERNATE(AUDIO_PWM_PAL_MODE));
#endif
gptStart(&AUDIO_STATE_TIMER, &gptCFG);
}
void audio_driver_start(void) {
channel_1_stop();
channel_1_start();
if (playing_note || playing_melody) {
gptStartContinuous(&AUDIO_STATE_TIMER, 64);
}
}
void audio_driver_stop(void) {
channel_1_stop();
gptStopTimer(&AUDIO_STATE_TIMER);
}
/* a regular timer task, that checks the note to be currently played
* and updates the pwm to output that frequency
*/
static void gpt_callback(GPTDriver *gptp) {
float freq; // TODO: freq_alt
if (audio_update_state()) {
freq = audio_get_processed_frequency(0); // freq_alt would be index=1
channel_1_set_frequency(freq);
}
}

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/* Copyright 2020 Jack Humbert
* Copyright 2020 JohSchneider
*
* 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 2 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/>.
*/
/*
Audio Driver: PWM
the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
this driver uses the chibios-PWM system to produce a square-wave on any given output pin in software
- a pwm callback is used to set/clear the configured pin.
*/
#include "audio.h"
#include "ch.h"
#include "hal.h"
#if !defined(AUDIO_PIN)
# error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
#endif
extern bool playing_note;
extern bool playing_melody;
extern uint8_t note_timbre;
static void pwm_audio_period_callback(PWMDriver *pwmp);
static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp);
static PWMConfig pwmCFG = {
.frequency = 100000, /* PWM clock frequency */
// CHIBIOS-BUG? can't set the initial period to <2, or the pwm (hard or software) takes ~130ms with .frequency=500000 for a pwmChangePeriod to take effect; with no output=silence in the meantime
.period = 2, /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
.callback = pwm_audio_period_callback,
.channels =
{
// software-PWM just needs another callback on any channel
{PWM_OUTPUT_ACTIVE_HIGH, pwm_audio_channel_interrupt_callback}, /* channel 0 -> TIMx_CH1 */
{PWM_OUTPUT_DISABLED, NULL}, /* channel 1 -> TIMx_CH2 */
{PWM_OUTPUT_DISABLED, NULL}, /* channel 2 -> TIMx_CH3 */
{PWM_OUTPUT_DISABLED, NULL} /* channel 3 -> TIMx_CH4 */
},
};
static float channel_1_frequency = 0.0f;
void channel_1_set_frequency(float freq) {
channel_1_frequency = freq;
if (freq <= 0.0) // a pause/rest has freq=0
return;
pwmcnt_t period = (pwmCFG.frequency / freq);
pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
// adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
}
float channel_1_get_frequency(void) { return channel_1_frequency; }
void channel_1_start(void) {
pwmStop(&AUDIO_PWM_DRIVER);
pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER);
pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
}
void channel_1_stop(void) {
pwmStop(&AUDIO_PWM_DRIVER);
palClearLine(AUDIO_PIN); // leave the line low, after last note was played
#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
palClearLine(AUDIO_PIN_ALT); // leave the line low, after last note was played
#endif
}
// generate a PWM signal on any pin, not necessarily the one connected to the timer
static void pwm_audio_period_callback(PWMDriver *pwmp) {
(void)pwmp;
palClearLine(AUDIO_PIN);
#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
palSetLine(AUDIO_PIN_ALT);
#endif
}
static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp) {
(void)pwmp;
if (channel_1_frequency > 0) {
palSetLine(AUDIO_PIN); // generate a PWM signal on any pin, not necessarily the one connected to the timer
#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
palClearLine(AUDIO_PIN_ALT);
#endif
}
}
static void gpt_callback(GPTDriver *gptp);
GPTConfig gptCFG = {
/* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
the tempo (which might vary!) is in bpm (beats per minute)
therefore: if the timer ticks away at .frequency = (60*64)Hz,
and the .interval counts from 64 downwards - audio_update_state is
called just often enough to not miss anything
*/
.frequency = 60 * 64,
.callback = gpt_callback,
};
void audio_driver_initialize(void) {
pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
palSetLineMode(AUDIO_PIN, PAL_MODE_OUTPUT_PUSHPULL);
palClearLine(AUDIO_PIN);
#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
palSetLineMode(AUDIO_PIN_ALT, PAL_MODE_OUTPUT_PUSHPULL);
palClearLine(AUDIO_PIN_ALT);
#endif
pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER); // enable pwm callbacks
pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
gptStart(&AUDIO_STATE_TIMER, &gptCFG);
}
void audio_driver_start(void) {
channel_1_stop();
channel_1_start();
if (playing_note || playing_melody) {
gptStartContinuous(&AUDIO_STATE_TIMER, 64);
}
}
void audio_driver_stop(void) {
channel_1_stop();
gptStopTimer(&AUDIO_STATE_TIMER);
}
/* a regular timer task, that checks the note to be currently played
* and updates the pwm to output that frequency
*/
static void gpt_callback(GPTDriver *gptp) {
float freq; // TODO: freq_alt
if (audio_update_state()) {
freq = audio_get_processed_frequency(0); // freq_alt would be index=1
channel_1_set_frequency(freq);
}
}

29
quantum/audio/musical_notes.h
View File

@@ -1,4 +1,5 @@
/* Copyright 2016 Jack Humbert /* Copyright 2016 Jack Humbert
* Copyright 2020 JohSchneider
* *
* This program is free software: you can redistribute it and/or modify * 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 * it under the terms of the GNU General Public License as published by
@@ -13,12 +14,11 @@
* You should have received a copy of the GNU General Public License * You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>. * along with this program. If not, see <http://www.gnu.org/licenses/>.
*/ */
#pragma once #pragma once
// Tempo Placeholder
#ifndef TEMPO_DEFAULT #ifndef TEMPO_DEFAULT
# define TEMPO_DEFAULT 100 # define TEMPO_DEFAULT 120
// in beats-per-minute
#endif #endif
#define SONG(notes...) \ #define SONG(notes...) \
@@ -27,12 +27,14 @@
// Note Types // Note Types
#define MUSICAL_NOTE(note, duration) \ #define MUSICAL_NOTE(note, duration) \
{ (NOTE##note), duration } { (NOTE##note), duration }
#define BREVE_NOTE(note) MUSICAL_NOTE(note, 128) #define BREVE_NOTE(note) MUSICAL_NOTE(note, 128)
#define WHOLE_NOTE(note) MUSICAL_NOTE(note, 64) #define WHOLE_NOTE(note) MUSICAL_NOTE(note, 64)
#define HALF_NOTE(note) MUSICAL_NOTE(note, 32) #define HALF_NOTE(note) MUSICAL_NOTE(note, 32)
#define QUARTER_NOTE(note) MUSICAL_NOTE(note, 16) #define QUARTER_NOTE(note) MUSICAL_NOTE(note, 16)
#define EIGHTH_NOTE(note) MUSICAL_NOTE(note, 8) #define EIGHTH_NOTE(note) MUSICAL_NOTE(note, 8)
#define SIXTEENTH_NOTE(note) MUSICAL_NOTE(note, 4) #define SIXTEENTH_NOTE(note) MUSICAL_NOTE(note, 4)
#define THIRTYSECOND_NOTE(note) MUSICAL_NOTE(note, 2)
#define BREVE_DOT_NOTE(note) MUSICAL_NOTE(note, 128 + 64) #define BREVE_DOT_NOTE(note) MUSICAL_NOTE(note, 128 + 64)
#define WHOLE_DOT_NOTE(note) MUSICAL_NOTE(note, 64 + 32) #define WHOLE_DOT_NOTE(note) MUSICAL_NOTE(note, 64 + 32)
@@ -40,6 +42,9 @@
#define QUARTER_DOT_NOTE(note) MUSICAL_NOTE(note, 16 + 8) #define QUARTER_DOT_NOTE(note) MUSICAL_NOTE(note, 16 + 8)
#define EIGHTH_DOT_NOTE(note) MUSICAL_NOTE(note, 8 + 4) #define EIGHTH_DOT_NOTE(note) MUSICAL_NOTE(note, 8 + 4)
#define SIXTEENTH_DOT_NOTE(note) MUSICAL_NOTE(note, 4 + 2) #define SIXTEENTH_DOT_NOTE(note) MUSICAL_NOTE(note, 4 + 2)
#define THIRTYSECOND_DOT_NOTE(note) MUSICAL_NOTE(note, 2 + 1)
// duration of 64 units == one beat == one whole note
// with a tempo of 60bpm this comes to a length of one second
// Note Type Shortcuts // Note Type Shortcuts
#define M__NOTE(note, duration) MUSICAL_NOTE(note, duration) #define M__NOTE(note, duration) MUSICAL_NOTE(note, duration)
@@ -49,31 +54,29 @@
#define Q__NOTE(n) QUARTER_NOTE(n) #define Q__NOTE(n) QUARTER_NOTE(n)
#define E__NOTE(n) EIGHTH_NOTE(n) #define E__NOTE(n) EIGHTH_NOTE(n)
#define S__NOTE(n) SIXTEENTH_NOTE(n) #define S__NOTE(n) SIXTEENTH_NOTE(n)
#define T__NOTE(n) THIRTYSECOND_NOTE(n)
#define BD_NOTE(n) BREVE_DOT_NOTE(n) #define BD_NOTE(n) BREVE_DOT_NOTE(n)
#define WD_NOTE(n) WHOLE_DOT_NOTE(n) #define WD_NOTE(n) WHOLE_DOT_NOTE(n)
#define HD_NOTE(n) HALF_DOT_NOTE(n) #define HD_NOTE(n) HALF_DOT_NOTE(n)
#define QD_NOTE(n) QUARTER_DOT_NOTE(n) #define QD_NOTE(n) QUARTER_DOT_NOTE(n)
#define ED_NOTE(n) EIGHTH_DOT_NOTE(n) #define ED_NOTE(n) EIGHTH_DOT_NOTE(n)
#define SD_NOTE(n) SIXTEENTH_DOT_NOTE(n) #define SD_NOTE(n) SIXTEENTH_DOT_NOTE(n)
#define TD_NOTE(n) THIRTYSECOND_DOT_NOTE(n)
// Note Timbre // Note Timbre
// Changes how the notes sound // Changes how the notes sound
#define TIMBRE_12 0.125f #define TIMBRE_12 12
#define TIMBRE_25 0.250f #define TIMBRE_25 25
#define TIMBRE_50 0.500f #define TIMBRE_50 50
#define TIMBRE_75 0.750f #define TIMBRE_75 75
#ifndef TIMBRE_DEFAULT #ifndef TIMBRE_DEFAULT
# define TIMBRE_DEFAULT TIMBRE_50 # define TIMBRE_DEFAULT TIMBRE_50
#endif #endif
// Notes - # = Octave // Notes - # = Octave
#ifdef __arm__
# define NOTE_REST 1.00f
#else
#define NOTE_REST 0.00f #define NOTE_REST 0.00f
#endif
/* These notes are currently bugged
#define NOTE_C0 16.35f #define NOTE_C0 16.35f
#define NOTE_CS0 17.32f #define NOTE_CS0 17.32f
#define NOTE_D0 18.35f #define NOTE_D0 18.35f
@@ -97,8 +100,6 @@
#define NOTE_GS1 51.91f #define NOTE_GS1 51.91f
#define NOTE_A1 55.00f #define NOTE_A1 55.00f
#define NOTE_AS1 58.27f #define NOTE_AS1 58.27f
*/
#define NOTE_B1 61.74f #define NOTE_B1 61.74f
#define NOTE_C2 65.41f #define NOTE_C2 65.41f
#define NOTE_CS2 69.30f #define NOTE_CS2 69.30f

152
quantum/audio/voices.c
View File

@@ -1,4 +1,5 @@
/* Copyright 2016 Jack Humbert /* Copyright 2016 Jack Humbert
* Copyright 2020 JohSchneider
* *
* This program is free software: you can redistribute it and/or modify * 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 * it under the terms of the GNU General Public License as published by
@@ -17,35 +18,73 @@
#include "audio.h" #include "audio.h"
#include <stdlib.h> #include <stdlib.h>
// these are imported from audio.c uint8_t note_timbre = TIMBRE_DEFAULT;
extern uint16_t envelope_index; bool glissando = false;
extern float note_timbre; bool vibrato = false;
extern float polyphony_rate; float vibrato_strength = 0.5;
extern bool glissando; float vibrato_rate = 0.125;
uint16_t voices_timer = 0;
#ifdef AUDIO_VOICE_DEFAULT
voice_type voice = AUDIO_VOICE_DEFAULT;
#else
voice_type voice = default_voice; voice_type voice = default_voice;
#endif
void set_voice(voice_type v) { voice = v; } void set_voice(voice_type v) { voice = v; }
void voice_iterate() { voice = (voice + 1) % number_of_voices; } void voice_iterate() { voice = (voice + 1) % number_of_voices; }
void voice_deiterate() { voice = (voice - 1 + number_of_voices) % number_of_voices; } void voice_deiterate() { voice = (voice - 1 + number_of_voices) % number_of_voices; }
#ifdef AUDIO_VOICES
float mod(float a, int b) {
float r = fmod(a, b);
return r < 0 ? r + b : r;
}
// Effect: 'vibrate' a given target frequency slightly above/below its initial value
float voice_add_vibrato(float average_freq) {
float vibrato_counter = mod(timer_read() / (100 * vibrato_rate), VIBRATO_LUT_LENGTH);
return average_freq * pow(vibrato_lut[(int)vibrato_counter], vibrato_strength);
}
// Effect: 'slides' the 'frequency' from the starting-point, to the target frequency
float voice_add_glissando(float from_freq, float to_freq) {
if (to_freq != 0 && from_freq < to_freq && from_freq < to_freq * pow(2, -440 / to_freq / 12 / 2)) {
return from_freq * pow(2, 440 / from_freq / 12 / 2);
} else if (to_freq != 0 && from_freq > to_freq && from_freq > to_freq * pow(2, 440 / to_freq / 12 / 2)) {
return from_freq * pow(2, -440 / from_freq / 12 / 2);
} else {
return to_freq;
}
}
#endif
float voice_envelope(float frequency) { float voice_envelope(float frequency) {
// envelope_index ranges from 0 to 0xFFFF, which is preserved at 880.0 Hz // envelope_index ranges from 0 to 0xFFFF, which is preserved at 880.0 Hz
__attribute__((unused)) uint16_t compensated_index = (uint16_t)((float)envelope_index * (880.0 / frequency)); // __attribute__((unused)) uint16_t compensated_index = (uint16_t)((float)envelope_index * (880.0 / frequency));
#ifdef AUDIO_VOICES
uint16_t envelope_index = timer_elapsed(voices_timer); // TODO: multiply in some factor?
uint16_t compensated_index = envelope_index / 100; // TODO: correct factor would be?
#endif
switch (voice) { switch (voice) {
case default_voice: case default_voice:
glissando = false; glissando = false;
note_timbre = TIMBRE_50; // note_timbre = TIMBRE_50; //Note: leave the user the possibility to adjust the timbre with 'audio_set_timbre'
polyphony_rate = 0;
break; break;
#ifdef AUDIO_VOICES #ifdef AUDIO_VOICES
case vibrating:
glissando = false;
vibrato = true;
break;
case something: case something:
glissando = false; glissando = false;
polyphony_rate = 0;
switch (compensated_index) { switch (compensated_index) {
case 0 ... 9: case 0 ... 9:
note_timbre = TIMBRE_12; note_timbre = TIMBRE_12;
@@ -56,24 +95,23 @@ float voice_envelope(float frequency) {
break; break;
case 20 ... 200: case 20 ... 200:
note_timbre = .125 + .125; note_timbre = 12 + 12;
break; break;
default: default:
note_timbre = .125; note_timbre = 12;
break; break;
} }
break; break;
case drums: case drums:
glissando = false; glissando = false;
polyphony_rate = 0;
// switch (compensated_index) { // switch (compensated_index) {
// case 0 ... 10: // case 0 ... 10:
// note_timbre = 0.5; // note_timbre = 50;
// break; // break;
// case 11 ... 20: // case 11 ... 20:
// note_timbre = 0.5 * (21 - compensated_index) / 10; // note_timbre = 50 * (21 - compensated_index) / 10;
// break; // break;
// default: // default:
// note_timbre = 0; // note_timbre = 0;
@@ -87,10 +125,10 @@ float voice_envelope(float frequency) {
frequency = (rand() % (int)(40)) + 60; frequency = (rand() % (int)(40)) + 60;
switch (envelope_index) { switch (envelope_index) {
case 0 ... 10: case 0 ... 10:
note_timbre = 0.5; note_timbre = 50;
break; break;
case 11 ... 20: case 11 ... 20:
note_timbre = 0.5 * (21 - envelope_index) / 10; note_timbre = 50 * (21 - envelope_index) / 10;
break; break;
default: default:
note_timbre = 0; note_timbre = 0;
@@ -102,10 +140,10 @@ float voice_envelope(float frequency) {
frequency = (rand() % (int)(1000)) + 1000; frequency = (rand() % (int)(1000)) + 1000;
switch (envelope_index) { switch (envelope_index) {
case 0 ... 5: case 0 ... 5:
note_timbre = 0.5; note_timbre = 50;
break; break;
case 6 ... 20: case 6 ... 20:
note_timbre = 0.5 * (21 - envelope_index) / 15; note_timbre = 50 * (21 - envelope_index) / 15;
break; break;
default: default:
note_timbre = 0; note_timbre = 0;
@@ -117,10 +155,10 @@ float voice_envelope(float frequency) {
frequency = (rand() % (int)(2000)) + 3000; frequency = (rand() % (int)(2000)) + 3000;
switch (envelope_index) { switch (envelope_index) {
case 0 ... 15: case 0 ... 15:
note_timbre = 0.5; note_timbre = 50;
break; break;
case 16 ... 20: case 16 ... 20:
note_timbre = 0.5 * (21 - envelope_index) / 5; note_timbre = 50 * (21 - envelope_index) / 5;
break; break;
default: default:
note_timbre = 0; note_timbre = 0;
@@ -132,10 +170,10 @@ float voice_envelope(float frequency) {
frequency = (rand() % (int)(2000)) + 3000; frequency = (rand() % (int)(2000)) + 3000;
switch (envelope_index) { switch (envelope_index) {
case 0 ... 35: case 0 ... 35:
note_timbre = 0.5; note_timbre = 50;
break; break;
case 36 ... 50: case 36 ... 50:
note_timbre = 0.5 * (51 - envelope_index) / 15; note_timbre = 50 * (51 - envelope_index) / 15;
break; break;
default: default:
note_timbre = 0; note_timbre = 0;
@@ -145,7 +183,6 @@ float voice_envelope(float frequency) {
break; break;
case butts_fader: case butts_fader:
glissando = true; glissando = true;
polyphony_rate = 0;
switch (compensated_index) { switch (compensated_index) {
case 0 ... 9: case 0 ... 9:
frequency = frequency / 4; frequency = frequency / 4;
@@ -158,7 +195,7 @@ float voice_envelope(float frequency) {
break; break;
case 20 ... 200: case 20 ... 200:
note_timbre = .125 - pow(((float)compensated_index - 20) / (200 - 20), 2) * .125; note_timbre = 12 - (uint8_t)(pow(((float)compensated_index - 20) / (200 - 20), 2) * 12.5);
break; break;
default: default:
@@ -168,7 +205,6 @@ float voice_envelope(float frequency) {
break; break;
// case octave_crunch: // case octave_crunch:
// polyphony_rate = 0;
// switch (compensated_index) { // switch (compensated_index) {
// case 0 ... 9: // case 0 ... 9:
// case 20 ... 24: // case 20 ... 24:
@@ -193,7 +229,6 @@ float voice_envelope(float frequency) {
case duty_osc: case duty_osc:
// This slows the loop down a substantial amount, so higher notes may freeze // This slows the loop down a substantial amount, so higher notes may freeze
glissando = true; glissando = true;
polyphony_rate = 0;
switch (compensated_index) { switch (compensated_index) {
default: default:
# define OCS_SPEED 10 # define OCS_SPEED 10
@@ -201,21 +236,19 @@ float voice_envelope(float frequency) {
// sine wave is slow // sine wave is slow
// note_timbre = (sin((float)compensated_index/10000*OCS_SPEED) * OCS_AMP / 2) + .5; // note_timbre = (sin((float)compensated_index/10000*OCS_SPEED) * OCS_AMP / 2) + .5;
// triangle wave is a bit faster // triangle wave is a bit faster
note_timbre = (float)abs((compensated_index * OCS_SPEED % 3000) - 1500) * (OCS_AMP / 1500) + (1 - OCS_AMP) / 2; note_timbre = (uint8_t)abs((compensated_index * OCS_SPEED % 3000) - 1500) * (OCS_AMP / 1500) + (1 - OCS_AMP) / 2;
break; break;
} }
break; break;
case duty_octave_down: case duty_octave_down:
glissando = true; glissando = true;
polyphony_rate = 0; note_timbre = (uint8_t)(100 * (envelope_index % 2) * .125 + .375 * 2);
note_timbre = (envelope_index % 2) * .125 + .375 * 2; if ((envelope_index % 4) == 0) note_timbre = 50;
if ((envelope_index % 4) == 0) note_timbre = 0.5;
if ((envelope_index % 8) == 0) note_timbre = 0; if ((envelope_index % 8) == 0) note_timbre = 0;
break; break;
case delayed_vibrato: case delayed_vibrato:
glissando = true; glissando = true;
polyphony_rate = 0;
note_timbre = TIMBRE_50; note_timbre = TIMBRE_50;
# define VOICE_VIBRATO_DELAY 150 # define VOICE_VIBRATO_DELAY 150
# define VOICE_VIBRATO_SPEED 50 # define VOICE_VIBRATO_SPEED 50
@@ -223,16 +256,16 @@ float voice_envelope(float frequency) {
case 0 ... VOICE_VIBRATO_DELAY: case 0 ... VOICE_VIBRATO_DELAY:
break; break;
default: default:
frequency = frequency * vibrato_lut[(int)fmod((((float)compensated_index - (VOICE_VIBRATO_DELAY + 1)) / 1000 * VOICE_VIBRATO_SPEED), VIBRATO_LUT_LENGTH)]; frequency = frequency * vibrato_lut[(int)fmod((((float)compensated_index - (VOICE_VIBRATO_DELAY + 1)) / 1000 * VOICE_VIBRATO_SPEED), VIBRATO_LUT_LENGTH)];
break; break;
} }
break; break;
// case delayed_vibrato_octave: // case delayed_vibrato_octave:
// polyphony_rate = 0;
// if ((envelope_index % 2) == 1) { // if ((envelope_index % 2) == 1) {
// note_timbre = 0.55; // note_timbre = 55;
// } else { // } else {
// note_timbre = 0.45; // note_timbre = 45;
// } // }
// #define VOICE_VIBRATO_DELAY 150 // #define VOICE_VIBRATO_DELAY 150
// #define VOICE_VIBRATO_SPEED 50 // #define VOICE_VIBRATO_SPEED 50
@@ -245,35 +278,64 @@ float voice_envelope(float frequency) {
// } // }
// break; // break;
// case duty_fifth_down: // case duty_fifth_down:
// note_timbre = 0.5; // note_timbre = TIMBRE_50;
// if ((envelope_index % 3) == 0) // if ((envelope_index % 3) == 0)
// note_timbre = 0.75; // note_timbre = TIMBRE_75;
// break; // break;
// case duty_fourth_down: // case duty_fourth_down:
// note_timbre = 0.0; // note_timbre = 0;
// if ((envelope_index % 12) == 0) // if ((envelope_index % 12) == 0)
// note_timbre = 0.75; // note_timbre = TIMBRE_75;
// if (((envelope_index % 12) % 4) != 1) // if (((envelope_index % 12) % 4) != 1)
// note_timbre = 0.75; // note_timbre = TIMBRE_75;
// break; // break;
// case duty_third_down: // case duty_third_down:
// note_timbre = 0.5; // note_timbre = TIMBRE_50;
// if ((envelope_index % 5) == 0) // if ((envelope_index % 5) == 0)
// note_timbre = 0.75; // note_timbre = TIMBRE_75;
// break; // break;
// case duty_fifth_third_down: // case duty_fifth_third_down:
// note_timbre = 0.5; // note_timbre = TIMBRE_50;
// if ((envelope_index % 5) == 0) // if ((envelope_index % 5) == 0)
// note_timbre = 0.75; // note_timbre = TIMBRE_75;
// if ((envelope_index % 3) == 0) // if ((envelope_index % 3) == 0)
// note_timbre = 0.25; // note_timbre = TIMBRE_25;
// break; // break;
#endif #endif // AUDIO_VOICES
default: default:
break; break;
} }
#ifdef AUDIO_VOICES
if (vibrato && (vibrato_strength > 0)) {
frequency = voice_add_vibrato(frequency);
}
if (glissando) {
// TODO: where to keep track of the start-frequency?
// frequency = voice_add_glissando(??, frequency);
}
#endif // AUDIO_VOICES
return frequency; return frequency;
} }
// Vibrato functions
void voice_set_vibrato_rate(float rate) { vibrato_rate = rate; }
void voice_increase_vibrato_rate(float change) { vibrato_rate *= change; }
void voice_decrease_vibrato_rate(float change) { vibrato_rate /= change; }
void voice_set_vibrato_strength(float strength) { vibrato_strength = strength; }
void voice_increase_vibrato_strength(float change) { vibrato_strength *= change; }
void voice_decrease_vibrato_strength(float change) { vibrato_strength /= change; }
// Timbre functions
void voice_set_timbre(uint8_t timbre) {
if ((timbre > 0) && (timbre < 100)) {
note_timbre = timbre;
}
}
uint8_t voice_get_timbre(void) { return note_timbre; }

20
quantum/audio/voices.h
View File

@@ -1,4 +1,5 @@
/* Copyright 2016 Jack Humbert /* Copyright 2016 Jack Humbert
* Copyright 2020 JohSchneider
* *
* This program is free software: you can redistribute it and/or modify * 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 * it under the terms of the GNU General Public License as published by
@@ -26,6 +27,7 @@ float voice_envelope(float frequency);
typedef enum { typedef enum {
default_voice, default_voice,
#ifdef AUDIO_VOICES #ifdef AUDIO_VOICES
vibrating,
something, something,
drums, drums,
butts_fader, butts_fader,
@@ -45,3 +47,21 @@ typedef enum {
void set_voice(voice_type v); void set_voice(voice_type v);
void voice_iterate(void); void voice_iterate(void);
void voice_deiterate(void); void voice_deiterate(void);
// Vibrato functions
void voice_set_vibrato_rate(float rate);
void voice_increase_vibrato_rate(float change);
void voice_decrease_vibrato_rate(float change);
void voice_set_vibrato_strength(float strength);
void voice_increase_vibrato_strength(float change);
void voice_decrease_vibrato_strength(float change);
// Timbre functions
/**
* @brief set the global timbre for tones to be played
* @note: only applies to pwm implementations - where it adjusts the duty-cycle
* @note: using any instrument from voices.[ch] other than 'default' may override the set value
* @param[in]: timbre: valid range is (0,100)
*/
void voice_set_timbre(uint8_t timbre);
uint8_t voice_get_timbre(void);

36
quantum/audio/wave.h
View File

@@ -1,36 +0,0 @@
/* Copyright 2016 Jack Humbert
*
* 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 2 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 <avr/io.h>
#include <avr/interrupt.h>
#include <avr/pgmspace.h>
#define SINE_LENGTH 2048
const uint8_t sinewave[] PROGMEM = // 2048 values
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0x60, 0x5f, 0x5f, 0x5f, 0x5e, 0x5e, 0x5d, 0x5d, 0x5d, 0x5c, 0x5c, 0x5c, 0x5b, 0x5b, 0x5a, 0x5a, 0x5a, 0x59, 0x59, 0x59, 0x58, 0x58, 0x58, 0x57, 0x57, 0x56, 0x56, 0x56, 0x55, 0x55, 0x55, 0x54, 0x54, 0x53, 0x53, 0x53, 0x52, 0x52, 0x52, 0x51, 0x51, 0x51, 0x50, 0x50, 0x4f, 0x4f, 0x4f, 0x4e, 0x4e, 0x4e, 0x4d, 0x4d, 0x4d, 0x4c, 0x4c, 0x4b, 0x4b, 0x4b, 0x4a, 0x4a, 0x4a, 0x49, 0x49, 0x49, 0x48, 0x48, 0x48, 0x47, 0x47, 0x47, 0x46, 0x46, 0x45, 0x45, 0x45, 0x44, 0x44, 0x44, 0x43, 0x43, 0x43, 0x42, 0x42, 0x42, 0x41, 0x41, 0x41, 0x40, 0x40, 0x40, 0x3f, 0x3f, 0x3f, 0x3e, 0x3e, 0x3e, 0x3d, 0x3d, 0x3d, 0x3c, 0x3c, 0x3c, 0x3b, 0x3b, 0x3b, 0x3a, 0x3a, 0x3a, 0x39, 0x39, 0x39, 0x38, 0x38, 0x38, 0x37, 0x37, 0x37, 0x36, 0x36, 0x36, 0x35, 0x35, 0x35, 0x34, 0x34, 0x34, 0x34, 0x33, 0x33, 0x33, 0x32, 0x32, 0x32, 0x31, 0x31, 0x31, 0x30, 0x30, 0x30, 0x30, 0x2f, 0x2f, 0x2f, 0x2e, 0x2e, 0x2e, 0x2d, 0x2d, 0x2d, 0x2d, 0x2c, 0x2c, 0x2c, 0x2b, 0x2b, 0x2b, 0x2a, 0x2a,
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0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x6, 0x6, 0x6, 0x6, 0x6, 0x6, 0x6, 0x6, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x8, 0x8, 0x8, 0x8, 0x8, 0x8, 0x8, 0x9, 0x9, 0x9, 0x9, 0x9, 0x9, 0x9, 0xa, 0xa, 0xa, 0xa, 0xa, 0xa, 0xa, 0xb, 0xb, 0xb, 0xb, 0xb, 0xb, 0xc, 0xc, 0xc, 0xc, 0xc, 0xc, 0xd, 0xd, 0xd, 0xd, 0xd, 0xd, 0xe, 0xe, 0xe, 0xe, 0xe, 0xf, 0xf, 0xf, 0xf, 0xf, 0xf, 0x10, 0x10, 0x10, 0x10, 0x10, 0x11, 0x11, 0x11, 0x11, 0x11, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x15, 0x15, 0x15, 0x15, 0x15, 0x16, 0x16, 0x16, 0x16, 0x17, 0x17, 0x17, 0x17, 0x17,
0x18, 0x18, 0x18, 0x18, 0x19, 0x19, 0x19, 0x19, 0x1a, 0x1a, 0x1a, 0x1a, 0x1b, 0x1b, 0x1b, 0x1b, 0x1b, 0x1c, 0x1c, 0x1c, 0x1c, 0x1d, 0x1d, 0x1d, 0x1d, 0x1e, 0x1e, 0x1e, 0x1e, 0x1f, 0x1f, 0x1f, 0x1f, 0x20, 0x20, 0x20, 0x21, 0x21, 0x21, 0x21, 0x22, 0x22, 0x22, 0x22, 0x23, 0x23, 0x23, 0x23, 0x24, 0x24, 0x24, 0x25, 0x25, 0x25, 0x25, 0x26, 0x26, 0x26, 0x26, 0x27, 0x27, 0x27, 0x28, 0x28, 0x28, 0x28, 0x29, 0x29, 0x29, 0x2a, 0x2a, 0x2a, 0x2a, 0x2b, 0x2b, 0x2b, 0x2c, 0x2c, 0x2c, 0x2d, 0x2d, 0x2d, 0x2d, 0x2e, 0x2e, 0x2e, 0x2f, 0x2f, 0x2f, 0x30, 0x30, 0x30, 0x30, 0x31, 0x31, 0x31, 0x32, 0x32, 0x32, 0x33, 0x33, 0x33, 0x34, 0x34, 0x34, 0x34, 0x35, 0x35, 0x35, 0x36, 0x36, 0x36, 0x37, 0x37, 0x37, 0x38, 0x38, 0x38, 0x39, 0x39, 0x39, 0x3a, 0x3a, 0x3a, 0x3b, 0x3b, 0x3b, 0x3c, 0x3c, 0x3c, 0x3d, 0x3d, 0x3d, 0x3e, 0x3e, 0x3e, 0x3f, 0x3f, 0x3f, 0x40, 0x40, 0x40, 0x41, 0x41, 0x41, 0x42, 0x42, 0x42, 0x43, 0x43, 0x43, 0x44, 0x44, 0x44, 0x45, 0x45, 0x45, 0x46,
0x46, 0x47, 0x47, 0x47, 0x48, 0x48, 0x48, 0x49, 0x49, 0x49, 0x4a, 0x4a, 0x4a, 0x4b, 0x4b, 0x4b, 0x4c, 0x4c, 0x4d, 0x4d, 0x4d, 0x4e, 0x4e, 0x4e, 0x4f, 0x4f, 0x4f, 0x50, 0x50, 0x51, 0x51, 0x51, 0x52, 0x52, 0x52, 0x53, 0x53, 0x53, 0x54, 0x54, 0x55, 0x55, 0x55, 0x56, 0x56, 0x56, 0x57, 0x57, 0x58, 0x58, 0x58, 0x59, 0x59, 0x59, 0x5a, 0x5a, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c, 0x5c, 0x5d, 0x5d, 0x5d, 0x5e, 0x5e, 0x5f, 0x5f, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x61, 0x62, 0x62, 0x62, 0x63, 0x63, 0x64, 0x64, 0x64, 0x65, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x67, 0x68, 0x68, 0x69, 0x69, 0x69, 0x6a, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c, 0x6d, 0x6d, 0x6e, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71, 0x71, 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75, 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79, 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f};

4
quantum/backlight/backlight_avr.c
View File

@@ -126,7 +126,7 @@
# define COMxx1 COM1B1 # define COMxx1 COM1B1
# define OCRxx OCR1B # define OCRxx OCR1B
# endif # endif
#elif !defined(B5_AUDIO) && !defined(B6_AUDIO) && !defined(B7_AUDIO) #elif (AUDIO_PIN != B5) && (AUDIO_PIN != B6) && (AUDIO_PIN != B7) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7)
// Timer 1 is not in use by Audio feature, Backlight can use it // Timer 1 is not in use by Audio feature, Backlight can use it
# pragma message "Using hardware timer 1 with software PWM" # pragma message "Using hardware timer 1 with software PWM"
# define HARDWARE_PWM # define HARDWARE_PWM
@@ -145,7 +145,7 @@
# define OCIExA OCIE1A # define OCIExA OCIE1A
# define OCRxx OCR1A # define OCRxx OCR1A
#elif !defined(C6_AUDIO) && !defined(C5_AUDIO) && !defined(C4_AUDIO) #elif (AUDIO_PIN != C4) && (AUDIO_PIN != C5) && (AUDIO_PIN != C6)
# pragma message "Using hardware timer 3 with software PWM" # pragma message "Using hardware timer 3 with software PWM"
// Timer 3 is not in use by Audio feature, Backlight can use it // Timer 3 is not in use by Audio feature, Backlight can use it
# define HARDWARE_PWM # define HARDWARE_PWM

67
util/audio_generate_dac_lut.py Executable file
View File

@@ -0,0 +1,67 @@
#!/usr/bin/env python3
#
# Copyright 2020 JohSchneider
#
# 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 2 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/>.
#
AUDIO_DAC_BUFFER_SIZE=256
AUDIO_DAC_SAMPLE_MAX=4095
def plot(values):
for v in values:
print('0'* int(v * 80/AUDIO_DAC_SAMPLE_MAX))
def to_lut(values):
for v in values:
print(hex(int(v)), end=", ")
from math import sin, tau, pi
samples=[]
def sampleSine():
for s in range(AUDIO_DAC_BUFFER_SIZE):
samples.append((sin((s/AUDIO_DAC_BUFFER_SIZE)*tau - pi/2) + 1 )/2* AUDIO_DAC_SAMPLE_MAX)
def sampleTriangle():
for s in range(AUDIO_DAC_BUFFER_SIZE):
if s < AUDIO_DAC_BUFFER_SIZE/2:
samples.append(s/(AUDIO_DAC_BUFFER_SIZE/2) * AUDIO_DAC_SAMPLE_MAX)
else:
samples.append(AUDIO_DAC_SAMPLE_MAX - (s-AUDIO_DAC_BUFFER_SIZE/2)/(AUDIO_DAC_BUFFER_SIZE/2) * AUDIO_DAC_SAMPLE_MAX)
#compromise between square and triangle wave,
def sampleTrapezoidal():
for i in range(AUDIO_DAC_BUFFER_SIZE):
a=3 #slope/inclination
if (i < AUDIO_DAC_BUFFER_SIZE/2):
s = a * (i * AUDIO_DAC_SAMPLE_MAX/(AUDIO_DAC_BUFFER_SIZE/2)) + (1-a)*AUDIO_DAC_SAMPLE_MAX/2
else:
i = i - AUDIO_DAC_BUFFER_SIZE/2
s = AUDIO_DAC_SAMPLE_MAX - a * (i * AUDIO_DAC_SAMPLE_MAX/(AUDIO_DAC_BUFFER_SIZE/2)) - (1-a)*AUDIO_DAC_SAMPLE_MAX/2
if s < 0:
s=0
if s> AUDIO_DAC_SAMPLE_MAX:
s=AUDIO_DAC_SAMPLE_MAX
samples.append(s)
#sampleSine()
sampleTrapezoidal()
#print(samples)
plot(samples)
to_lut(samples)

39
util/sample_parser.py Executable file
View File

@@ -0,0 +1,39 @@
#!/usr/bin/env python3
#
# Copyright 2019 Jack Humbert
#
# 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 2 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/>.
#
import wave, struct, sys
waveFile = wave.open(sys.argv[1], 'r')
# print(str(waveFile.getparams()))
# sys.exit()
if (waveFile.getsampwidth() != 2):
raise(Exception("This script currently only works with 16bit audio files"))
length = waveFile.getnframes()
out = "#define DAC_SAMPLE_CUSTOM_LENGTH " + str(length) + "\n\n"
out += "static const dacsample_t dac_sample_custom[" + str(length) + "] = {"
for i in range(0,length):
if (i % 8 == 0):
out += "\n "
waveData = waveFile.readframes(1)
data = struct.unpack("<h", waveData)
out += str(int((int(data[0]) + 0x8000) / 16)) + ", "
out = out[:-2]
out += "\n};"
print(out)

40
util/wavetable_parser.py Executable file
View File

@@ -0,0 +1,40 @@
#!/usr/bin/env python3
#
# Copyright 2019 Jack Humbert
#
# 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 2 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/>.
#
import wave, struct, sys
waveFile = wave.open(sys.argv[1], 'r')
length = waveFile.getnframes()
out = "#define DAC_WAVETABLE_CUSTOM_LENGTH " + str(int(length / 256)) + "\n\n"
out += "static const dacsample_t dac_wavetable_custom[" + str(int(length / 256)) + "][256] = {"
for i in range(0,length):
if (i % 8 == 0):
out += "\n "
if (i % 256 == 0):
out = out[:-2]
out += "{\n "
waveData = waveFile.readframes(1)
data = struct.unpack("<h", waveData)
out += str(int((int(data[0]) + 0x8000) / 16)) + ", "
if (i % 256 == 255):
out = out[:-2]
out += "\n },"
out = out[:-1]
out += "\n};"
print(out)