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authorDrashna Jael're <drashna@live.com>2021-06-29 15:36:35 -0700
committerDrashna Jael're <drashna@live.com>2021-06-29 15:36:35 -0700
commit996a19ee7ba3308e17fd347afde0b135852835cc (patch)
tree54f8d2c9179ad3318146179722a0924419411e98
parentacf2c323e2927f6007b17ded577cf49fd86fec6c (diff)
Revert "Audio system overhaul (#11820)" due to freezing issues
This reverts commit c80e5f9f8868ccaa8cb990be6f4da3f1011c2b78.
Diffstat
-rw-r--r--common_features.mk21
-rw-r--r--keyboards/planck/config.h2
-rw-r--r--keyboards/planck/ez/config.h5
-rw-r--r--quantum/audio/audio.c539
-rw-r--r--quantum/audio/audio.h281
-rw-r--r--quantum/audio/audio_chibios.c20
-rw-r--r--quantum/audio/audio_pwm.c606
-rw-r--r--quantum/audio/driver_avr_pwm.h17
-rw-r--r--quantum/audio/driver_avr_pwm_hardware.c332
-rw-r--r--quantum/audio/driver_chibios_dac.h126
-rw-r--r--quantum/audio/driver_chibios_dac_additive.c335
-rw-r--r--quantum/audio/driver_chibios_dac_basic.c245
-rw-r--r--quantum/audio/driver_chibios_pwm.h40
-rw-r--r--quantum/audio/driver_chibios_pwm_hardware.c144
-rw-r--r--quantum/audio/driver_chibios_pwm_software.c164
-rw-r--r--quantum/audio/musical_notes.h77
-rw-r--r--quantum/audio/voices.c170
-rw-r--r--quantum/audio/voices.h20
-rw-r--r--quantum/audio/wave.h36
-rw-r--r--quantum/backlight/backlight_avr.c4
-rwxr-xr-xutil/audio_generate_dac_lut.py67
-rwxr-xr-xutil/sample_parser.py39
-rwxr-xr-xutil/wavetable_parser.py40
23 files changed, 798 insertions, 2532 deletions
diff --git a/common_features.mk b/common_features.mk
index 8b51a60fb9..74b8c1046b 100644
--- a/common_features.mk
+++ b/common_features.mk
@@ -43,31 +43,12 @@ ifeq ($(strip $(COMMAND_ENABLE)), yes)
OPT_DEFS += -DCOMMAND_ENABLE
endif
-AUDIO_ENABLE ?= no
ifeq ($(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
MUSIC_ENABLE = yes
SRC += $(QUANTUM_DIR)/process_keycode/process_audio.c
SRC += $(QUANTUM_DIR)/process_keycode/process_clicky.c
- SRC += $(QUANTUM_DIR)/audio/audio.c ## common audio code, hardware agnostic
- SRC += $(QUANTUM_DIR)/audio/driver_$(PLATFORM_KEY)_$(strip $(AUDIO_DRIVER)).c
+ SRC += $(QUANTUM_DIR)/audio/audio_$(PLATFORM_KEY).c
SRC += $(QUANTUM_DIR)/audio/voices.c
SRC += $(QUANTUM_DIR)/audio/luts.c
endif
diff --git a/keyboards/planck/config.h b/keyboards/planck/config.h
index 71111eca21..9ef2b0b0dd 100644
--- a/keyboards/planck/config.h
+++ b/keyboards/planck/config.h
@@ -40,7 +40,7 @@ along with this program. If not, see <http://www.gnu.org/licenses/>.
#define QMK_SPEAKER C6
#define AUDIO_VOICES
-#define AUDIO_PIN C6
+#define C6_AUDIO
#define BACKLIGHT_PIN B7
diff --git a/keyboards/planck/ez/config.h b/keyboards/planck/ez/config.h
index a3713a5d2b..b37b2570ca 100644
--- a/keyboards/planck/ez/config.h
+++ b/keyboards/planck/ez/config.h
@@ -57,10 +57,7 @@
#define MUSIC_MAP
#undef AUDIO_VOICES
-#undef AUDIO_PIN
-#define AUDIO_PIN A5
-#define AUDIO_PIN_ALT A4
-#define AUDIO_PIN_ALT_AS_NEGATIVE
+#undef C6_AUDIO
/* Debounce reduces chatter (unintended double-presses) - set 0 if debouncing is not needed */
// #define DEBOUNCE 6
diff --git a/quantum/audio/audio.c b/quantum/audio/audio.c
deleted file mode 100644
index 46277dd70b..0000000000
--- a/quantum/audio/audio.c
+++ /dev/null
@@ -1,539 +0,0 @@
-/* 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 "eeconfig.h"
-#include "timer.h"
-#include "wait.h"
-
-/* audio system:
- *
- * audio.[ch] takes care of all overall state, tracking the actively playing
- * notes/tones; the notes a SONG consists of;
- * ...
- * = everything audio-related that is platform agnostic
- *
- * driver_[avr|chibios]_[dac|pwm] take care of the lower hardware dependent parts,
- * specific to each platform and the used subsystem/driver to drive
- * the output pins/channels with the calculated frequencies for each
- * active tone
- * as part of this, the driver has to trigger regular state updates by
- * calling 'audio_update_state' through some sort of timer - be it a
- * dedicated one or piggybacking on for example the timer used to
- * generate a pwm signal/clock.
- *
- *
- * A Note on terminology:
- * tone, pitch and frequency are used somewhat interchangeably, in a strict Wikipedia-sense:
- * "(Musical) tone, a sound characterized by its duration, pitch (=frequency),
- * intensity (=volume), and timbre"
- * - intensity/volume is currently not handled at all, although the 'dac_additive' driver could do so
- * - timbre is handled globally (TODO: only used with the pwm drivers at the moment)
- *
- * in musical_note.h a 'note' is the combination of a pitch and a duration
- * these are used to create SONG arrays; during playback their frequencies
- * are handled as single successive tones, while the durations are
- * kept track of in 'audio_update_state'
- *
- * 'voice' as it is used here, equates to a sort of instrument with its own
- * characteristics sound and effects
- * the audio system as-is deals only with (possibly multiple) tones of one
- * instrument/voice at a time (think: chords). since the number of tones that
- * can be reproduced depends on the hardware/driver in use: pwm can only
- * reproduce one tone per output/speaker; DACs can reproduce/mix multiple
- * when doing additive synthesis.
- *
- * 'duration' can either be in the beats-per-minute related unit found in
- * musical_notes.h, OR in ms; keyboards create SONGs with the former, while
- * the internal state of the audio system does its calculations with the later - ms
- */
-
-#ifndef AUDIO_TONE_STACKSIZE
-# define AUDIO_TONE_STACKSIZE 8
-#endif
-uint8_t active_tones = 0; // number of tones pushed onto the stack by audio_play_tone - might be more than the hardware is able to reproduce at any single time
-musical_tone_t tones[AUDIO_TONE_STACKSIZE]; // stack of currently active tones
-
-bool playing_melody = false; // playing a SONG?
-bool playing_note = false; // or (possibly multiple simultaneous) tones
-bool state_changed = false; // global flag, which is set if anything changes with the active_tones
-
-// melody/SONG related state variables
-float (*notes_pointer)[][2]; // SONG, an array of MUSICAL_NOTEs
-uint16_t notes_count; // length of the notes_pointer array
-bool notes_repeat; // PLAY_SONG or PLAY_LOOP?
-uint16_t melody_current_note_duration = 0; // duration of the currently playing note from the active melody, in ms
-uint8_t note_tempo = TEMPO_DEFAULT; // beats-per-minute
-uint16_t current_note = 0; // index into the array at notes_pointer
-bool note_resting = false; // if a short pause was introduced between two notes with the same frequency while playing a melody
-uint16_t last_timestamp = 0;
-
-#ifdef AUDIO_ENABLE_TONE_MULTIPLEXING
-# ifndef AUDIO_MAX_SIMULTANEOUS_TONES
-# define AUDIO_MAX_SIMULTANEOUS_TONES 3
-# endif
-uint16_t tone_multiplexing_rate = AUDIO_TONE_MULTIPLEXING_RATE_DEFAULT;
-uint8_t tone_multiplexing_index_shift = 0; // offset used on active-tone array access
-#endif
-
-// provided and used by voices.c
-extern uint8_t note_timbre;
-extern bool glissando;
-extern bool vibrato;
-extern uint16_t voices_timer;
-
-#ifndef STARTUP_SONG
-# define STARTUP_SONG SONG(STARTUP_SOUND)
-#endif
-#ifndef AUDIO_ON_SONG
-# define AUDIO_ON_SONG SONG(AUDIO_ON_SOUND)
-#endif
-#ifndef AUDIO_OFF_SONG
-# define AUDIO_OFF_SONG SONG(AUDIO_OFF_SOUND)
-#endif
-float startup_song[][2] = STARTUP_SONG;
-float audio_on_song[][2] = AUDIO_ON_SONG;
-float audio_off_song[][2] = AUDIO_OFF_SONG;
-
-static bool audio_initialized = false;
-static bool audio_driver_stopped = true;
-audio_config_t audio_config;
-
-void audio_init() {
- if (audio_initialized) {
- return;
- }
-
- // Check EEPROM
-#ifdef EEPROM_ENABLE
- if (!eeconfig_is_enabled()) {
- eeconfig_init();
- }
- audio_config.raw = eeconfig_read_audio();
-#else // EEPROM settings
- audio_config.enable = true;
-# ifdef AUDIO_CLICKY_ON
- audio_config.clicky_enable = true;
-# endif
-#endif // EEPROM settings
-
- for (uint8_t i = 0; i < AUDIO_TONE_STACKSIZE; i++) {
- tones[i] = (musical_tone_t){.time_started = 0, .pitch = -1.0f, .duration = 0};
- }
-
- if (!audio_initialized) {
- audio_driver_initialize();
- audio_initialized = true;
- }
- stop_all_notes();
-}
-
-void audio_startup(void) {
- if (audio_config.enable) {
- PLAY_SONG(startup_song);
- }
-
- last_timestamp = timer_read();
-}
-
-void audio_toggle(void) {
- if (audio_config.enable) {
- stop_all_notes();
- }
- audio_config.enable ^= 1;
- eeconfig_update_audio(audio_config.raw);
- if (audio_config.enable) {
- audio_on_user();
- }
-}
-
-void audio_on(void) {
- audio_config.enable = 1;
- eeconfig_update_audio(audio_config.raw);
- audio_on_user();
- PLAY_SONG(audio_on_song);
-}
-
-void audio_off(void) {
- PLAY_SONG(audio_off_song);
- wait_ms(100);
- audio_stop_all();
- audio_config.enable = 0;
- eeconfig_update_audio(audio_config.raw);
-}
-
-bool audio_is_on(void) { return (audio_config.enable != 0); }
-
-void audio_stop_all() {
- if (audio_driver_stopped) {
- return;
- }
-
- active_tones = 0;
-
- audio_driver_stop();
-
- playing_melody = false;
- playing_note = false;
-
- melody_current_note_duration = 0;
-
- for (uint8_t i = 0; i < AUDIO_TONE_STACKSIZE; i++) {
- tones[i] = (musical_tone_t){.time_started = 0, .pitch = -1.0f, .duration = 0};
- }
-
- audio_driver_stopped = true;
-}
-
-void audio_stop_tone(float pitch) {
- if (pitch < 0.0f) {
- pitch = -1 * pitch;
- }
-
- if (playing_note) {
- if (!audio_initialized) {
- audio_init();
- }
- bool found = false;
- for (int i = AUDIO_TONE_STACKSIZE - 1; i >= 0; i--) {
- found = (tones[i].pitch == pitch);
- if (found) {
- tones[i] = (musical_tone_t){.time_started = 0, .pitch = -1.0f, .duration = 0};
- for (int j = i; (j < AUDIO_TONE_STACKSIZE - 1); j++) {
- tones[j] = tones[j + 1];
- tones[j + 1] = (musical_tone_t){.time_started = 0, .pitch = -1.0f, .duration = 0};
- }
- break;
- }
- }
- if (!found) {
- return;
- }
-
- state_changed = true;
- active_tones--;
- if (active_tones < 0) active_tones = 0;
-#ifdef AUDIO_ENABLE_TONE_MULTIPLEXING
- if (tone_multiplexing_index_shift >= active_tones) {
- tone_multiplexing_index_shift = 0;
- }
-#endif
- if (active_tones == 0) {
- audio_driver_stop();
- audio_driver_stopped = true;
- playing_note = false;
- }
- }
-}
-
-void audio_play_note(float pitch, uint16_t duration) {
- if (!audio_config.enable) {
- return;
- }
-
- if (!audio_initialized) {
- audio_init();
- }
-
- if (pitch < 0.0f) {
- pitch = -1 * pitch;
- }
-
- // round-robin: shifting out old tones, keeping only unique ones
- // if the new frequency is already amongst the active tones, shift it to the top of the stack
- bool found = false;
- for (int i = active_tones - 1; i >= 0; i--) {
- found = (tones[i].pitch == pitch);
- if (found) {
- for (int j = i; (j < active_tones - 1); j++) {
- tones[j] = tones[j + 1];
- tones[j + 1] = (musical_tone_t){.time_started = timer_read(), .pitch = pitch, .duration = duration};
- }
- return; // since this frequency played already, the hardware was already started
- }
- }
-
- // frequency/tone is actually new, so we put it on the top of the stack
- active_tones++;
- if (active_tones > AUDIO_TONE_STACKSIZE) {
- active_tones = AUDIO_TONE_STACKSIZE;
- // shift out the oldest tone to make room
- for (int i = 0; i < active_tones - 1; i++) {
- tones[i] = tones[i + 1];
- }
- }
- state_changed = true;
- playing_note = true;
- tones[active_tones - 1] = (musical_tone_t){.time_started = timer_read(), .pitch = pitch, .duration = duration};
-
- // TODO: needs to be handled per note/tone -> use its timestamp instead?
- voices_timer = timer_read(); // reset to zero, for the effects added by voices.c
-
- if (audio_driver_stopped) {
- audio_driver_start();
- audio_driver_stopped = false;
- }
-}
-
-void audio_play_tone(float pitch) { audio_play_note(pitch, 0xffff); }
-
-void audio_play_melody(float (*np)[][2], uint16_t n_count, bool n_repeat) {
- if (!audio_config.enable) {
- audio_stop_all();
- return;
- }
-
- if (!audio_initialized) {
- audio_init();
- }
-
- // Cancel note if a note is playing
- if (playing_note) audio_stop_all();
-
- playing_melody = true;
- note_resting = false;
-
- notes_pointer = np;
- notes_count = n_count;
- notes_repeat = n_repeat;
-
- current_note = 0; // note in the melody-array/list at note_pointer
-
- // start first note manually, which also starts the audio_driver
- // all following/remaining notes are played by 'audio_update_state'
- audio_play_note((*notes_pointer)[current_note][0], audio_duration_to_ms((*notes_pointer)[current_note][1]));
- last_timestamp = timer_read();
- melody_current_note_duration = audio_duration_to_ms((*notes_pointer)[current_note][1]);
-}
-
-float click[2][2];
-void audio_play_click(uint16_t delay, float pitch, uint16_t duration) {
- uint16_t duration_tone = audio_ms_to_duration(duration);
- uint16_t duration_delay = audio_ms_to_duration(delay);
-
- if (delay <= 0.0f) {
- click[0][0] = pitch;
- click[0][1] = duration_tone;
- click[1][0] = 0.0f;
- click[1][1] = 0.0f;
- audio_play_melody(&click, 1, false);
- } else {
- // first note is a rest/pause
- click[0][0] = 0.0f;
- click[0][1] = duration_delay;
- // second note is the actual click
- click[1][0] = pitch;
- click[1][1] = duration_tone;
- audio_play_melody(&click, 2, false);
- }
-}
-
-bool audio_is_playing_note(void) { return playing_note; }
-
-bool audio_is_playing_melody(void) { return playing_melody; }
-
-uint8_t audio_get_number_of_active_tones(void) { return active_tones; }
-
-float audio_get_frequency(uint8_t tone_index) {
- if (tone_index >= active_tones) {
- return 0.0f;
- }
- return tones[active_tones - tone_index - 1].pitch;
-}
-
-float audio_get_processed_frequency(uint8_t tone_index) {
- if (tone_index >= active_tones) {
- return 0.0f;
- }
-
- int8_t index = active_tones - tone_index - 1;
- // new tones are stacked on top (= appended at the end), so the most recent/current is MAX-1
-
-#ifdef AUDIO_ENABLE_TONE_MULTIPLEXING
- index = index - tone_multiplexing_index_shift;
- if (index < 0) // wrap around
- index += active_tones;
-#endif
-
- if (tones[index].pitch <= 0.0f) {
- return 0.0f;
- }
-
- return voice_envelope(tones[index].pitch);
-}
-
-bool audio_update_state(void) {
- if (!playing_note && !playing_melody) {
- return false;
- }
-
- bool goto_next_note = false;
- uint16_t current_time = timer_read();
-
- if (playing_melody) {
- goto_next_note = timer_elapsed(last_timestamp) >= melody_current_note_duration;
- if (goto_next_note) {
- uint16_t delta = timer_elapsed(last_timestamp) - melody_current_note_duration;
- last_timestamp = current_time;
- uint16_t previous_note = current_note;
- current_note++;
- voices_timer = timer_read(); // reset to zero, for the effects added by voices.c
-
- if (current_note >= notes_count) {
- if (notes_repeat) {
- current_note = 0;
- } else {
- audio_stop_all();
- return false;
- }
- }
-
- if (!note_resting && (*notes_pointer)[previous_note][0] == (*notes_pointer)[current_note][0]) {
- note_resting = true;
-
- // special handling for successive notes of the same frequency:
- // insert a short pause to separate them audibly
- audio_play_note(0.0f, audio_duration_to_ms(2));
- current_note = previous_note;
- melody_current_note_duration = audio_duration_to_ms(2);
-
- } else {
- note_resting = false;
-
- // TODO: handle glissando here (or remember previous and current tone)
- /* there would need to be a freq(here we are) -> freq(next note)
- * and do slide/glissando in between problem here is to know which
- * frequency on the stack relates to what other? e.g. a melody starts
- * tones in a sequence, and stops expiring one, so the most recently
- * stopped is the starting point for a glissando to the most recently started?
- * how to detect and preserve this relation?
- * and what about user input, chords, ...?
- */
-
- // '- delta': Skip forward in the next note's length if we've over shot
- // the last, so the overall length of the song is the same
- uint16_t duration = audio_duration_to_ms((*notes_pointer)[current_note][1]);
-
- // Skip forward past any completely missed notes
- while (delta > duration && current_note < notes_count - 1) {
- delta -= duration;
- current_note++;
- duration = audio_duration_to_ms((*notes_pointer)[current_note][1]);
- }
-
- if (delta < duration) {
- duration -= delta;
- } else {
- // Only way to get here is if it is the last note and
- // we have completely missed it. Play it for 1ms...
- duration = 1;
- }
-
- audio_play_note((*notes_pointer)[current_note][0], duration);
- melody_current_note_duration = duration;
- }
- }
- }
-
- if (playing_note) {
-#ifdef AUDIO_ENABLE_TONE_MULTIPLEXING
- tone_multiplexing_index_shift = (int)(current_time / tone_multiplexing_rate) % MIN(AUDIO_MAX_SIMULTANEOUS_TONES, active_tones);
- goto_next_note = true;
-#endif
- if (vibrato || glissando) {
- // force update on each cycle, since vibrato shifts the frequency slightly
- goto_next_note = true;
- }
-
- // housekeeping: stop notes that have no playtime left
- for (int i = 0; i < active_tones; i++) {
- if ((tones[i].duration != 0xffff) // indefinitely playing notes, started by 'audio_play_tone'
- && (tones[i].duration != 0) // 'uninitialized'
- ) {
- if (timer_elapsed(tones[i].time_started) >= tones[i].duration) {
- audio_stop_tone(tones[i].pitch); // also sets 'state_changed=true'
- }
- }
- }
- }
-
- // state-changes have a higher priority, always triggering the hardware to update
- if (state_changed) {
- state_changed = false;
- return true;
- }
-
- return goto_next_note;
-}
-
-// Tone-multiplexing functions
-#ifdef AUDIO_ENABLE_TONE_MULTIPLEXING
-void audio_set_tone_multiplexing_rate(uint16_t rate) { tone_multiplexing_rate = rate; }
-void audio_enable_tone_multiplexing(void) { tone_multiplexing_rate = AUDIO_TONE_MULTIPLEXING_RATE_DEFAULT; }
-void audio_disable_tone_multiplexing(void) { tone_multiplexing_rate = 0; }
-void audio_increase_tone_multiplexing_rate(uint16_t change) {
- if ((0xffff - change) > tone_multiplexing_rate) {
- tone_multiplexing_rate += change;
- }
-}
-void audio_decrease_tone_multiplexing_rate(uint16_t change) {
- if (change <= tone_multiplexing_rate) {
- tone_multiplexing_rate -= change;
- }
-}
-#endif
-
-// Tempo functions
-
-void audio_set_tempo(uint8_t tempo) {
- if (tempo < 10) note_tempo = 10;
- // else if (tempo > 250)
- // note_tempo = 250;
- else
- note_tempo = tempo;
-}
-
-void audio_increase_tempo(uint8_t tempo_change) {
- if (tempo_change > 255 - note_tempo)
- note_tempo = 255;
- else
- note_tempo += tempo_change;
-}
-
-void audio_decrease_tempo(uint8_t tempo_change) {
- if (tempo_change >= note_tempo - 10)
- note_tempo = 10;
- else
- note_tempo -= tempo_change;
-}
-
-// TODO in the int-math version are some bugs; songs sometimes abruptly end - maybe an issue with the timer/system-tick wrapping around?
-uint16_t audio_duration_to_ms(uint16_t duration_bpm) {
-#if defined(__AVR__)
- // doing int-math saves us some bytes in the overall firmware size, but the intermediate result is less accurate before being cast to/returned as uint
- return ((uint32_t)duration_bpm * 60 * 1000) / (64 * note_tempo);
- // NOTE: beware of uint16_t overflows when note_tempo is low and/or the duration is long
-#else
- return ((float)duration_bpm * 60) / (64 * note_tempo) * 1000;
-#endif
-}
-uint16_t audio_ms_to_duration(uint16_t duration_ms) {
-#if defined(__AVR__)
- return ((uint32_t)duration_ms * 64 * note_tempo) / 60 / 1000;
-#else
- return ((float)duration_ms * 64 * note_tempo) / 60 / 1000;
-#endif
-}
diff --git a/quantum/audio/audio.h b/quantum/audio/audio.h
index 56b9158a1a..dccf03d5f6 100644
--- a/quantum/audio/audio.h
+++ b/quantum/audio/audio.h
@@ -1,5 +1,4 @@
-/* Copyright 2016-2020 Jack Humbert
- * Copyright 2020 JohSchneider
+/* 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
@@ -14,30 +13,28 @@
* 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
#include <stdint.h>
#include <stdbool.h>
+#if defined(__AVR__)
+# include <avr/io.h>
+#endif
+#include "wait.h"
#include "musical_notes.h"
#include "song_list.h"
#include "voices.h"
#include "quantum.h"
#include <math.h>
-#if defined(__AVR__)
-# include <avr/io.h>
-# if defined(AUDIO_DRIVER_PWM)
-# include "driver_avr_pwm.h"
-# endif
-#endif
+// Largely untested PWM audio mode (doesn't sound as good)
+// #define PWM_AUDIO
-#if defined(PROTOCOL_CHIBIOS)
-# if defined(AUDIO_DRIVER_PWM)
-# include "driver_chibios_pwm.h"
-# elif defined(AUDIO_DRIVER_DAC)
-# include "driver_chibios_dac.h"
-# endif
-#endif
+// #define VIBRATO_ENABLE
+
+// Enable vibrato strength/amplitude - slows down ISR too much
+// #define VIBRATO_STRENGTH_ENABLE
typedef union {
uint8_t raw;
@@ -48,238 +45,62 @@ typedef union {
};
} audio_config_t;
-// AVR/LUFA has a MIN, arm/chibios does not
-#ifndef MIN
-# define MIN(a, b) (((a) < (b)) ? (a) : (b))
-#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;
-
-// public interface
-
-/**
- * @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_startup(void);
-
-/**
- * @brief en-/disable audio output, save this choice to the eeprom
- */
+bool is_audio_on(void);
void audio_toggle(void);
-/**
- * @brief enable audio output, save this choice to the eeprom
- */
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);
-/**
- * @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
+// Vibrato rate functions
-/**
- * @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);
+#ifdef VIBRATO_ENABLE
-/**
- * @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);
+void set_vibrato_rate(float rate);
+void increase_vibrato_rate(float change);
+void decrease_vibrato_rate(float change);
-/**
- * @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);
+# ifdef VIBRATO_STRENGTH_ENABLE
-// 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.
-// The global float array for the song must be used here.
-#define NOTE_ARRAY_SIZE(x) ((int16_t)(sizeof(x) / (sizeof(x[0]))))
-
-/**
- * @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)
+void set_vibrato_strength(float strength);
+void increase_vibrato_strength(float change);
+void decrease_vibrato_strength(float change);
-// 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);
+#endif
-// hardware interface
+// Polyphony functions
-// implementation in the driver_avr/arm_* respective parts
-void audio_driver_initialize(void);
-void audio_driver_start(void);
-void audio_driver_stop(void);
+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);
-/**
- * @brief get the number of currently active tones
- * @return number, 0=none active
- */
-uint8_t audio_get_number_of_active_tones(void);
+void set_timbre(float timbre);
+void set_tempo(uint8_t tempo);
-/**
- * @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);
+void increase_tempo(uint8_t tempo_change);
+void decrease_tempo(uint8_t tempo_change);
-/**
- * @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);
+void audio_init(void);
+void audio_startup(void);
-/**
- * @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);
+#ifdef PWM_AUDIO
+void play_sample(uint8_t* s, uint16_t l, bool r);
+#endif
+void play_note(float freq, int vol);
+void stop_note(float freq);
+void stop_all_notes(void);
+void play_notes(float (*np)[][2], uint16_t n_count, bool n_repeat);
-// legacy and back-warts compatibility stuff
+#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), }
-#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)
+// These macros are used to allow play_notes to play an array of indeterminate
+// length. This works around the limitation of C's sizeof operation on pointers.
+// The global float array for the song must be used here.
+#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)
-#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
+bool is_playing_notes(void);
diff --git a/quantum/audio/audio_chibios.c b/quantum/audio/audio_chibios.c
index 377f93de5d..3640423e91 100644
--- a/quantum/audio/audio_chibios.c
+++ b/quantum/audio/audio_chibios.c
@@ -84,27 +84,23 @@ static void gpt_cb8(GPTDriver *gptp);
# define DAC_SAMPLE_MAX 65535U
#endif
-#define START_CHANNEL_1() \
- gptStart(&GPTD6, &gpt6cfg1); \
+#define START_CHANNEL_1() \
+ gptStart(&GPTD6, &gpt6cfg1); \
gptStartContinuous(&GPTD6, 2U); \
palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG)
-
-#define START_CHANNEL_2() \
- gptStart(&GPTD7, &gpt7cfg1); \
+#define START_CHANNEL_2() \
+ gptStart(&GPTD7, &gpt7cfg1); \
gptStartContinuous(&GPTD7, 2U); \
palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG)
-
-#define STOP_CHANNEL_1() \
- gptStopTimer(&GPTD6); \
+#define STOP_CHANNEL_1() \
+ gptStopTimer(&GPTD6); \
palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL); \
palSetPad(GPIOA, 4)
-
-#define STOP_CHANNEL_2() \
- gptStopTimer(&GPTD7); \
+#define STOP_CHANNEL_2() \
+ gptStopTimer(&GPTD7); \
palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL); \
palSetPad(GPIOA, 5)
-
#define RESTART_CHANNEL_1() \
STOP_CHANNEL_1(); \
START_CHANNEL_1()
diff --git a/quantum/audio/audio_pwm.c b/quantum/audio/audio_pwm.c
new file mode 100644
index 0000000000..d93ac4bb40
--- /dev/null
+++ b/quantum/audio/audio_pwm.c
@@ -0,0 +1,606 @@
+/* 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() {}
+//------------------------------------------------------------------------------
diff --git a/quantum/audio/driver_avr_pwm.h b/quantum/audio/driver_avr_pwm.h
deleted file mode 100644
index d6eb3571da..0000000000
--- a/quantum/audio/driver_avr_pwm.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/* 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
diff --git a/quantum/audio/driver_avr_pwm_hardware.c b/quantum/audio/driver_avr_pwm_hardware.c
deleted file mode 100644
index df03a4558c..0000000000
--- a/quantum/audio/driver_avr_pwm_hardware.c
+++ /dev/null
@@ -1,332 +0,0 @@
-/* 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
diff --git a/quantum/audio/driver_chibios_dac.h b/quantum/audio/driver_chibios_dac.h
deleted file mode 100644
index 07cd622ead..0000000000
--- a/quantum/audio/driver_chibios_dac.h
+++ /dev/null
@@ -1,126 +0,0 @@
-/* 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);
diff --git a/quantum/audio/driver_chibios_dac_additive.c b/quantum/audio/driver_chibios_dac_additive.c
deleted file mode 100644
index db304adb87..0000000000
--- a/quantum/audio/driver_chibios_dac_additive.c
+++ /dev/null
@@ -1,335 +0,0 @@
-/* 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;
-}
diff --git a/quantum/audio/driver_chibios_dac_basic.c b/quantum/audio/driver_chibios_dac_basic.c
deleted file mode 100644
index fac6513506..0000000000
--- a/quantum/audio/driver_chibios_dac_basic.c
+++ /dev/null
@@ -1,245 +0,0 @@
-/* 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);
-}
diff --git a/quantum/audio/driver_chibios_pwm.h b/quantum/audio/driver_chibios_pwm.h
deleted file mode 100644
index 86cab916e1..0000000000
--- a/quantum/audio/driver_chibios_pwm.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/* 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
diff --git a/quantum/audio/driver_chibios_pwm_hardware.c b/quantum/audio/driver_chibios_pwm_hardware.c
deleted file mode 100644
index 3c7d89b290..0000000000
--- a/quantum/audio/driver_chibios_pwm_hardware.c
+++ /dev/null
@@ -1,144 +0,0 @@
-/* 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);
- }
-}
diff --git a/quantum/audio/driver_chibios_pwm_software.c b/quantum/audio/driver_chibios_pwm_software.c
deleted file mode 100644
index 15c3e98b6a..0000000000
--- a/quantum/audio/driver_chibios_pwm_software.c
+++ /dev/null
@@ -1,164 +0,0 @@
-/* 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);
- }
-}
diff --git a/quantum/audio/musical_notes.h b/quantum/audio/musical_notes.h
index ddd7d374f5..0ba572c346 100644
--- a/quantum/audio/musical_notes.h
+++ b/quantum/audio/musical_notes.h
@@ -1,5 +1,4 @@
/* 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
@@ -14,11 +13,12 @@
* 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
+// Tempo Placeholder
#ifndef TEMPO_DEFAULT
-# define TEMPO_DEFAULT 120
-// in beats-per-minute
+# define TEMPO_DEFAULT 100
#endif
#define SONG(notes...) \
@@ -27,14 +27,12 @@
// Note Types
#define MUSICAL_NOTE(note, duration) \
{ (NOTE##note), duration }
-
#define BREVE_NOTE(note) MUSICAL_NOTE(note, 128)
#define WHOLE_NOTE(note) MUSICAL_NOTE(note, 64)
#define HALF_NOTE(note) MUSICAL_NOTE(note, 32)
#define QUARTER_NOTE(note) MUSICAL_NOTE(note, 16)
#define EIGHTH_NOTE(note) MUSICAL_NOTE(note, 8)
#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 WHOLE_DOT_NOTE(note) MUSICAL_NOTE(note, 64 + 32)
@@ -42,9 +40,6 @@
#define QUARTER_DOT_NOTE(note) MUSICAL_NOTE(note, 16 + 8)
#define EIGHTH_DOT_NOTE(note) MUSICAL_NOTE(note, 8 + 4)
#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
#define M__NOTE(note, duration) MUSICAL_NOTE(note, duration)
@@ -54,52 +49,56 @@
#define Q__NOTE(n) QUARTER_NOTE(n)
#define E__NOTE(n) EIGHTH_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 WD_NOTE(n) WHOLE_DOT_NOTE(n)
#define HD_NOTE(n) HALF_DOT_NOTE(n)
#define QD_NOTE(n) QUARTER_DOT_NOTE(n)
#define ED_NOTE(n) EIGHTH_DOT_NOTE(n)
#define SD_NOTE(n) SIXTEENTH_DOT_NOTE(n)
-#define TD_NOTE(n) THIRTYSECOND_DOT_NOTE(n)
// Note Timbre
// Changes how the notes sound
-#define TIMBRE_12 12
-#define TIMBRE_25 25
-#define TIMBRE_50 50
-#define TIMBRE_75 75
+#define TIMBRE_12 0.125f
+#define TIMBRE_25 0.250f
+#define TIMBRE_50 0.500f
+#define TIMBRE_75 0.750f
#ifndef TIMBRE_DEFAULT
# define TIMBRE_DEFAULT TIMBRE_50
#endif
-
// Notes - # = Octave
-#define NOTE_REST 0.00f
+#ifdef __arm__
+# define NOTE_REST 1.00f
+#else
+# define NOTE_REST 0.00f
+#endif
+
+/* These notes are currently bugged
+#define NOTE_C0 16.35f
+#define NOTE_CS0 17.32f
+#define NOTE_D0 18.35f
+#define NOTE_DS0 19.45f
+#define NOTE_E0 20.60f
+#define NOTE_F0 21.83f
+#define NOTE_FS0 23.12f
+#define NOTE_G0 24.50f
+#define NOTE_GS0 25.96f
+#define NOTE_A0 27.50f
+#define NOTE_AS0 29.14f
+#define NOTE_B0 30.87f
+#define NOTE_C1 32.70f
+#define NOTE_CS1 34.65f
+#define NOTE_D1 36.71f
+#define NOTE_DS1 38.89f
+#define NOTE_E1 41.20f
+#define NOTE_F1 43.65f
+#define NOTE_FS1 46.25f
+#define NOTE_G1 49.00f
+#define NOTE_GS1 51.91f
+#define NOTE_A1 55.00f
+#define NOTE_AS1 58.27f
+*/
-#define NOTE_C0 16.35f
-#define NOTE_CS0 17.32f
-#define NOTE_D0 18.35f
-#define NOTE_DS0 19.45f
-#define NOTE_E0 20.60f
-#define NOTE_F0 21.83f
-#define NOTE_FS0 23.12f
-#define NOTE_G0 24.50f
-#define NOTE_GS0 25.96f
-#define NOTE_A0 27.50f
-#define NOTE_AS0 29.14f
-#define NOTE_B0 30.87f
-#define NOTE_C1 32.70f
-#define NOTE_CS1 34.65f
-#define NOTE_D1 36.71f
-#define NOTE_DS1 38.89f
-#define NOTE_E1 41.20f
-#define NOTE_F1 43.65f
-#define NOTE_FS1 46.25f
-#define NOTE_G1 49.00f
-#define NOTE_GS1 51.91f
-#define NOTE_A1 55.00f
-#define NOTE_AS1 58.27f
#define NOTE_B1 61.74f
#define NOTE_C2 65.41f
#define NOTE_CS2 69.30f
diff --git a/quantum/audio/voices.c b/quantum/audio/voices.c
index d43fb8d169..d412ad5057 100644
--- a/quantum/audio/voices.c
+++ b/quantum/audio/voices.c
@@ -1,5 +1,4 @@
/* 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
@@ -18,73 +17,35 @@
#include "audio.h"
#include <stdlib.h>
-uint8_t note_timbre = TIMBRE_DEFAULT;
-bool glissando = false;
-bool vibrato = false;
-float vibrato_strength = 0.5;
-float vibrato_rate = 0.125;
+// these are imported from audio.c
+extern uint16_t envelope_index;
+extern float note_timbre;
+extern float polyphony_rate;
+extern bool glissando;
-uint16_t voices_timer = 0;
-
-#ifdef AUDIO_VOICE_DEFAULT
-voice_type voice = AUDIO_VOICE_DEFAULT;
-#else
voice_type voice = default_voice;
-#endif
void set_voice(voice_type v) { voice = v; }
void voice_iterate() { voice = (voice + 1) % 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) {
// 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));
-#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
+ __attribute__((unused)) uint16_t compensated_index = (uint16_t)((float)envelope_index * (880.0 / frequency));
switch (voice) {
case default_voice:
- glissando = false;
- // note_timbre = TIMBRE_50; //Note: leave the user the possibility to adjust the timbre with 'audio_set_timbre'
+ glissando = false;
+ note_timbre = TIMBRE_50;
+ polyphony_rate = 0;
break;
#ifdef AUDIO_VOICES
- case vibrating:
- glissando = false;
- vibrato = true;
- break;
-
case something:
- glissando = false;
+ glissando = false;
+ polyphony_rate = 0;
switch (compensated_index) {
case 0 ... 9:
note_timbre = TIMBRE_12;
@@ -95,23 +56,24 @@ float voice_envelope(float frequency) {
break;
case 20 ... 200:
- note_timbre = 12 + 12;
+ note_timbre = .125 + .125;
break;
default:
- note_timbre = 12;
+ note_timbre = .125;
break;
}
break;
case drums:
- glissando = false;
+ glissando = false;
+ polyphony_rate = 0;
// switch (compensated_index) {
// case 0 ... 10:
- // note_timbre = 50;
+ // note_timbre = 0.5;
// break;
// case 11 ... 20:
- // note_timbre = 50 * (21 - compensated_index) / 10;
+ // note_timbre = 0.5 * (21 - compensated_index) / 10;
// break;
// default:
// note_timbre = 0;
@@ -125,10 +87,10 @@ float voice_envelope(float frequency) {
frequency = (rand() % (int)(40)) + 60;
switch (envelope_index) {
case 0 ... 10:
- note_timbre = 50;
+ note_timbre = 0.5;
break;
case 11 ... 20:
- note_timbre = 50 * (21 - envelope_index) / 10;
+ note_timbre = 0.5 * (21 - envelope_index) / 10;
break;
default:
note_timbre = 0;
@@ -140,10 +102,10 @@ float voice_envelope(float frequency) {
frequency = (rand() % (int)(1000)) + 1000;
switch (envelope_index) {
case 0 ... 5:
- note_timbre = 50;
+ note_timbre = 0.5;
break;
case 6 ... 20:
- note_timbre = 50 * (21 - envelope_index) / 15;
+ note_timbre = 0.5 * (21 - envelope_index) / 15;
break;
default:
note_timbre = 0;
@@ -155,10 +117,10 @@ float voice_envelope(float frequency) {
frequency = (rand() % (int)(2000)) + 3000;
switch (envelope_index) {
case 0 ... 15:
- note_timbre = 50;
+ note_timbre = 0.5;
break;
case 16 ... 20:
- note_timbre = 50 * (21 - envelope_index) / 5;
+ note_timbre = 0.5 * (21 - envelope_index) / 5;
break;
default:
note_timbre = 0;
@@ -170,10 +132,10 @@ float voice_envelope(float frequency) {
frequency = (rand() % (int)(2000)) + 3000;
switch (envelope_index) {
case 0 ... 35:
- note_timbre = 50;
+ note_timbre = 0.5;
break;
case 36 ... 50:
- note_timbre = 50 * (51 - envelope_index) / 15;
+ note_timbre = 0.5 * (51 - envelope_index) / 15;
break;
default:
note_timbre = 0;
@@ -182,7 +144,8 @@ float voice_envelope(float frequency) {
}
break;
case butts_fader:
- glissando = true;
+ glissando = true;
+ polyphony_rate = 0;
switch (compensated_index) {
case 0 ... 9:
frequency = frequency / 4;
@@ -195,7 +158,7 @@ float voice_envelope(float frequency) {
break;
case 20 ... 200:
- note_timbre = 12 - (uint8_t)(pow(((float)compensated_index - 20) / (200 - 20), 2) * 12.5);
+ note_timbre = .125 - pow(((float)compensated_index - 20) / (200 - 20), 2) * .125;
break;
default:
@@ -205,6 +168,7 @@ float voice_envelope(float frequency) {
break;
// case octave_crunch:
+ // polyphony_rate = 0;
// switch (compensated_index) {
// case 0 ... 9:
// case 20 ... 24:
@@ -222,13 +186,14 @@ float voice_envelope(float frequency) {
// default:
// note_timbre = TIMBRE_12;
- // break;
+ // break;
// }
// break;
case duty_osc:
// This slows the loop down a substantial amount, so higher notes may freeze
- glissando = true;
+ glissando = true;
+ polyphony_rate = 0;
switch (compensated_index) {
default:
# define OCS_SPEED 10
@@ -236,36 +201,38 @@ float voice_envelope(float frequency) {
// sine wave is slow
// note_timbre = (sin((float)compensated_index/10000*OCS_SPEED) * OCS_AMP / 2) + .5;
// triangle wave is a bit faster
- note_timbre = (uint8_t)abs((compensated_index * OCS_SPEED % 3000) - 1500) * (OCS_AMP / 1500) + (1 - OCS_AMP) / 2;
+ note_timbre = (float)abs((compensated_index * OCS_SPEED % 3000) - 1500) * (OCS_AMP / 1500) + (1 - OCS_AMP) / 2;
break;
}
break;
case duty_octave_down:
- glissando = true;
- note_timbre = (uint8_t)(100 * (envelope_index % 2) * .125 + .375 * 2);
- if ((envelope_index % 4) == 0) note_timbre = 50;
+ glissando = true;
+ polyphony_rate = 0;
+ note_timbre = (envelope_index % 2) * .125 + .375 * 2;
+ if ((envelope_index % 4) == 0) note_timbre = 0.5;
if ((envelope_index % 8) == 0) note_timbre = 0;
break;
case delayed_vibrato:
- glissando = true;
- note_timbre = TIMBRE_50;
+ glissando = true;
+ polyphony_rate = 0;
+ note_timbre = TIMBRE_50;
# define VOICE_VIBRATO_DELAY 150
# define VOICE_VIBRATO_SPEED 50
switch (compensated_index) {
case 0 ... VOICE_VIBRATO_DELAY:
break;
default:
-
frequency = frequency * vibrato_lut[(int)fmod((((float)compensated_index - (VOICE_VIBRATO_DELAY + 1)) / 1000 * VOICE_VIBRATO_SPEED), VIBRATO_LUT_LENGTH)];
break;
}
break;
// case delayed_vibrato_octave:
+ // polyphony_rate = 0;
// if ((envelope_index % 2) == 1) {
- // note_timbre = 55;
+ // note_timbre = 0.55;
// } else {
- // note_timbre = 45;
+ // note_timbre = 0.45;
// }
// #define VOICE_VIBRATO_DELAY 150
// #define VOICE_VIBRATO_SPEED 50
@@ -278,64 +245,35 @@ float voice_envelope(float frequency) {
// }
// break;
// case duty_fifth_down:
- // note_timbre = TIMBRE_50;
+ // note_timbre = 0.5;
// if ((envelope_index % 3) == 0)
- // note_timbre = TIMBRE_75;
+ // note_timbre = 0.75;
// break;
// case duty_fourth_down:
- // note_timbre = 0;
+ // note_timbre = 0.0;
// if ((envelope_index % 12) == 0)
- // note_timbre = TIMBRE_75;
+ // note_timbre = 0.75;
// if (((envelope_index % 12) % 4) != 1)
- // note_timbre = TIMBRE_75;
+ // note_timbre = 0.75;
// break;
// case duty_third_down:
- // note_timbre = TIMBRE_50;
+ // note_timbre = 0.5;
// if ((envelope_index % 5) == 0)
- // note_timbre = TIMBRE_75;
+ // note_timbre = 0.75;
// break;
// case duty_fifth_third_down:
- // note_timbre = TIMBRE_50;
+ // note_timbre = 0.5;
// if ((envelope_index % 5) == 0)
- // note_timbre = TIMBRE_75;
+ // note_timbre = 0.75;
// if ((envelope_index % 3) == 0)
- // note_timbre = TIMBRE_25;
+ // note_timbre = 0.25;
// break;
-#endif // AUDIO_VOICES
+#endif
default:
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;
}
-
-// 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; }
diff --git a/quantum/audio/voices.h b/quantum/audio/voices.h
index 7bd26b461f..abafa2b404 100644
--- a/quantum/audio/voices.h
+++ b/quantum/audio/voices.h
@@ -1,5 +1,4 @@
/* 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
@@ -30,7 +29,6 @@ float voice_envelope(float frequency);
typedef enum {
default_voice,
#ifdef AUDIO_VOICES
- vibrating,
something,
drums,
butts_fader,
@@ -50,21 +48,3 @@ typedef enum {
void set_voice(voice_type v);
void voice_iterate(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);
diff --git a/quantum/audio/wave.h b/quantum/audio/wave.h
new file mode 100644
index 0000000000..48210a944e
--- /dev/null
+++ b/quantum/audio/wave.h
@@ -0,0 +1,36 @@
+/* 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
+ {0x80, 0x80, 0x80, 0x81, 0x81, 0x81, 0x82, 0x82, 0x83, 0x83, 0x83, 0x84, 0x84, 0x85, 0x85, 0x85, 0x86, 0x86, 0x87, 0x87, 0x87, 0x88, 0x88, 0x88, 0x89, 0x89, 0x8a, 0x8a, 0x8a, 0x8b, 0x8b, 0x8c, 0x8c, 0x8c, 0x8d, 0x8d, 0x8e, 0x8e, 0x8e, 0x8f, 0x8f, 0x8f, 0x90, 0x90, 0x91, 0x91, 0x91, 0x92, 0x92, 0x93, 0x93, 0x93, 0x94, 0x94, 0x95, 0x95, 0x95, 0x96, 0x96, 0x96, 0x97, 0x97, 0x98, 0x98, 0x98, 0x99, 0x99, 0x9a, 0x9a, 0x9a, 0x9b, 0x9b, 0x9b, 0x9c, 0x9c, 0x9d, 0x9d, 0x9d, 0x9e, 0x9e, 0x9e, 0x9f, 0x9f, 0xa0, 0xa0, 0xa0, 0xa1, 0xa1, 0xa2, 0xa2, 0xa2, 0xa3, 0xa3, 0xa3, 0xa4, 0xa4, 0xa5, 0xa5, 0xa5, 0xa6, 0xa6, 0xa6, 0xa7, 0xa7, 0xa7, 0xa8, 0xa8, 0xa9, 0xa9, 0xa9, 0xaa, 0xaa, 0xaa, 0xab, 0xab, 0xac, 0xac, 0xac, 0xad, 0xad, 0xad, 0xae, 0xae, 0xae, 0xaf, 0xaf, 0xb0, 0xb0, 0xb0, 0xb1, 0xb1, 0xb1, 0xb2, 0xb2, 0xb2, 0xb3, 0xb3, 0xb4, 0xb4, 0xb4, 0xb5, 0xb5, 0xb5, 0xb6, 0xb6, 0xb6, 0xb7, 0xb7, 0xb7, 0xb8, 0xb8, 0xb8, 0xb9, 0xb9, 0xba, 0xba, 0xba, 0xbb,
+ 0xbb, 0xbb, 0xbc, 0xbc, 0xbc, 0xbd, 0xbd, 0xbd, 0xbe, 0xbe, 0xbe, 0xbf, 0xbf, 0xbf, 0xc0, 0xc0, 0xc0, 0xc1, 0xc1, 0xc1, 0xc2, 0xc2, 0xc2, 0xc3, 0xc3, 0xc3, 0xc4, 0xc4, 0xc4, 0xc5, 0xc5, 0xc5, 0xc6, 0xc6, 0xc6, 0xc7, 0xc7, 0xc7, 0xc8, 0xc8, 0xc8, 0xc9, 0xc9, 0xc9, 0xca, 0xca, 0xca, 0xcb, 0xcb, 0xcb, 0xcb, 0xcc, 0xcc, 0xcc, 0xcd, 0xcd, 0xcd, 0xce, 0xce, 0xce, 0xcf, 0xcf, 0xcf, 0xcf, 0xd0, 0xd0, 0xd0, 0xd1, 0xd1, 0xd1, 0xd2, 0xd2, 0xd2, 0xd2, 0xd3, 0xd3, 0xd3, 0xd4, 0xd4, 0xd4, 0xd5, 0xd5, 0xd5, 0xd5, 0xd6, 0xd6, 0xd6, 0xd7, 0xd7, 0xd7, 0xd7, 0xd8, 0xd8, 0xd8, 0xd9, 0xd9, 0xd9, 0xd9, 0xda, 0xda, 0xda, 0xda, 0xdb, 0xdb, 0xdb, 0xdc, 0xdc, 0xdc, 0xdc, 0xdd, 0xdd, 0xdd, 0xdd, 0xde, 0xde, 0xde, 0xde, 0xdf, 0xdf, 0xdf, 0xe0, 0xe0, 0xe0, 0xe0, 0xe1, 0xe1, 0xe1, 0xe1, 0xe2, 0xe2, 0xe2, 0xe2, 0xe3, 0xe3, 0xe3, 0xe3, 0xe4, 0xe4, 0xe4, 0xe4, 0xe4, 0xe5, 0xe5, 0xe5, 0xe5, 0xe6, 0xe6, 0xe6, 0xe6, 0xe7, 0xe7, 0xe7, 0xe7, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8,
+ 0xe9, 0xe9, 0xe9, 0xe9, 0xea, 0xea, 0xea, 0xea, 0xea, 0xeb, 0xeb, 0xeb, 0xeb, 0xeb, 0xec, 0xec, 0xec, 0xec, 0xec, 0xed, 0xed, 0xed, 0xed, 0xed, 0xee, 0xee, 0xee, 0xee, 0xee, 0xef, 0xef, 0xef, 0xef, 0xef, 0xf0, 0xf0, 0xf0, 0xf0, 0xf0, 0xf0, 0xf1, 0xf1, 0xf1, 0xf1, 0xf1, 0xf2, 0xf2, 0xf2, 0xf2, 0xf2, 0xf2, 0xf3, 0xf3, 0xf3, 0xf3, 0xf3, 0xf3, 0xf4, 0xf4, 0xf4, 0xf4, 0xf4, 0xf4, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe,
+ 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7,
+ 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf4, 0xf4, 0xf4, 0xf4, 0xf4, 0xf4, 0xf3, 0xf3, 0xf3, 0xf3, 0xf3, 0xf3, 0xf2, 0xf2, 0xf2, 0xf2, 0xf2, 0xf2, 0xf1, 0xf1, 0xf1, 0xf1, 0xf1, 0xf0, 0xf0, 0xf0, 0xf0, 0xf0, 0xf0, 0xef, 0xef, 0xef, 0xef, 0xef, 0xee, 0xee, 0xee, 0xee, 0xee, 0xed, 0xed, 0xed, 0xed, 0xed, 0xec, 0xec, 0xec, 0xec, 0xec, 0xeb, 0xeb, 0xeb, 0xeb, 0xeb, 0xea, 0xea, 0xea, 0xea, 0xea, 0xe9, 0xe9, 0xe9, 0xe9, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0xe7, 0xe7, 0xe7, 0xe7, 0xe6, 0xe6, 0xe6, 0xe6, 0xe5, 0xe5, 0xe5, 0xe5, 0xe4, 0xe4, 0xe4, 0xe4, 0xe4, 0xe3, 0xe3, 0xe3, 0xe3, 0xe2, 0xe2, 0xe2, 0xe2, 0xe1, 0xe1, 0xe1, 0xe1, 0xe0, 0xe0, 0xe0, 0xe0, 0xdf, 0xdf, 0xdf, 0xde, 0xde, 0xde, 0xde, 0xdd, 0xdd, 0xdd, 0xdd, 0xdc, 0xdc, 0xdc, 0xdc, 0xdb, 0xdb, 0xdb, 0xda, 0xda, 0xda, 0xda, 0xd9, 0xd9, 0xd9, 0xd9, 0xd8, 0xd8, 0xd8, 0xd7, 0xd7, 0xd7, 0xd7, 0xd6, 0xd6, 0xd6, 0xd5, 0xd5, 0xd5, 0xd5, 0xd4, 0xd4, 0xd4,
+ 0xd3, 0xd3, 0xd3, 0xd2, 0xd2, 0xd2, 0xd2, 0xd1, 0xd1, 0xd1, 0xd0, 0xd0, 0xd0, 0xcf, 0xcf, 0xcf, 0xcf, 0xce, 0xce, 0xce, 0xcd, 0xcd, 0xcd, 0xcc, 0xcc, 0xcc, 0xcb, 0xcb, 0xcb, 0xcb, 0xca, 0xca, 0xca, 0xc9, 0xc9, 0xc9, 0xc8, 0xc8, 0xc8, 0xc7, 0xc7, 0xc7, 0xc6, 0xc6, 0xc6, 0xc5, 0xc5, 0xc5, 0xc4, 0xc4, 0xc4, 0xc3, 0xc3, 0xc3, 0xc2, 0xc2, 0xc2, 0xc1, 0xc1, 0xc1, 0xc0, 0xc0, 0xc0, 0xbf, 0xbf, 0xbf, 0xbe, 0xbe, 0xbe, 0xbd, 0xbd, 0xbd, 0xbc, 0xbc, 0xbc, 0xbb, 0xbb, 0xbb, 0xba, 0xba, 0xba, 0xb9, 0xb9, 0xb8, 0xb8, 0xb8, 0xb7, 0xb7, 0xb7, 0xb6, 0xb6, 0xb6, 0xb5, 0xb5, 0xb5, 0xb4, 0xb4, 0xb4, 0xb3, 0xb3, 0xb2, 0xb2, 0xb2, 0xb1, 0xb1, 0xb1, 0xb0, 0xb0, 0xb0, 0xaf, 0xaf, 0xae, 0xae, 0xae, 0xad, 0xad, 0xad, 0xac, 0xac, 0xac, 0xab, 0xab, 0xaa, 0xaa, 0xaa, 0xa9, 0xa9, 0xa9, 0xa8, 0xa8, 0xa7, 0xa7, 0xa7, 0xa6, 0xa6, 0xa6, 0xa5, 0xa5, 0xa5, 0xa4, 0xa4, 0xa3, 0xa3, 0xa3, 0xa2, 0xa2, 0xa2, 0xa1, 0xa1, 0xa0, 0xa0, 0xa0, 0x9f, 0x9f, 0x9e, 0x9e, 0x9e, 0x9d,
+ 0x9d, 0x9d, 0x9c, 0x9c, 0x9b, 0x9b, 0x9b, 0x9a, 0x9a, 0x9a, 0x99, 0x99, 0x98, 0x98, 0x98, 0x97, 0x97, 0x96, 0x96, 0x96, 0x95, 0x95, 0x95, 0x94, 0x94, 0x93, 0x93, 0x93, 0x92, 0x92, 0x91, 0x91, 0x91, 0x90, 0x90, 0x8f, 0x8f, 0x8f, 0x8e, 0x8e, 0x8e, 0x8d, 0x8d, 0x8c, 0x8c, 0x8c, 0x8b, 0x8b, 0x8a, 0x8a, 0x8a, 0x89, 0x89, 0x88, 0x88, 0x88, 0x87, 0x87, 0x87, 0x86, 0x86, 0x85, 0x85, 0x85, 0x84, 0x84, 0x83, 0x83, 0x83, 0x82, 0x82, 0x81, 0x81, 0x81, 0x80, 0x80, 0x80, 0x7f, 0x7f, 0x7e, 0x7e, 0x7e, 0x7d, 0x7d, 0x7c, 0x7c, 0x7c, 0x7b, 0x7b, 0x7a, 0x7a, 0x7a, 0x79, 0x79, 0x78, 0x78, 0x78, 0x77, 0x77, 0x77, 0x76, 0x76, 0x75, 0x75, 0x75, 0x74, 0x74, 0x73, 0x73, 0x73, 0x72, 0x72, 0x71, 0x71, 0x71, 0x70, 0x70, 0x70, 0x6f, 0x6f, 0x6e, 0x6e, 0x6e, 0x6d, 0x6d, 0x6c, 0x6c, 0x6c, 0x6b, 0x6b, 0x6a, 0x6a, 0x6a, 0x69, 0x69, 0x69, 0x68, 0x68, 0x67, 0x67, 0x67, 0x66, 0x66, 0x65, 0x65, 0x65, 0x64, 0x64, 0x64, 0x63, 0x63, 0x62, 0x62, 0x62, 0x61, 0x61, 0x61, 0x60,
+ 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,
+ 0x2a, 0x2a, 0x29, 0x29, 0x29, 0x28, 0x28, 0x28, 0x28, 0x27, 0x27, 0x27, 0x26, 0x26, 0x26, 0x26, 0x25, 0x25, 0x25, 0x25, 0x24, 0x24, 0x24, 0x23, 0x23, 0x23, 0x23, 0x22, 0x22, 0x22, 0x22, 0x21, 0x21, 0x21, 0x21, 0x20, 0x20, 0x20, 0x1f, 0x1f, 0x1f, 0x1f, 0x1e, 0x1e, 0x1e, 0x1e, 0x1d, 0x1d, 0x1d, 0x1d, 0x1c, 0x1c, 0x1c, 0x1c, 0x1b, 0x1b, 0x1b, 0x1b, 0x1b, 0x1a, 0x1a, 0x1a, 0x1a, 0x19, 0x19, 0x19, 0x19, 0x18, 0x18, 0x18, 0x18, 0x17, 0x17, 0x17, 0x17, 0x17, 0x16, 0x16, 0x16, 0x16, 0x15, 0x15, 0x15, 0x15, 0x15, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x13, 0x13, 0x13, 0x13, 0x12, 0x12, 0x12, 0x12, 0x12, 0x11, 0x11, 0x11, 0x11, 0x11, 0x10, 0x10, 0x10, 0x10, 0x10, 0xf, 0xf, 0xf, 0xf, 0xf, 0xf, 0xe, 0xe, 0xe, 0xe, 0xe, 0xd, 0xd, 0xd, 0xd, 0xd, 0xd, 0xc, 0xc, 0xc, 0xc, 0xc, 0xc, 0xb, 0xb, 0xb, 0xb, 0xb, 0xb, 0xa, 0xa, 0xa, 0xa, 0xa, 0xa, 0xa, 0x9, 0x9, 0x9, 0x9, 0x9, 0x9, 0x9, 0x8, 0x8, 0x8, 0x8, 0x8,
+ 0x8, 0x8, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x6, 0x6, 0x6, 0x6, 0x6, 0x6, 0x6, 0x6, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x5, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x4, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x3, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x2, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 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, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1, 0x1,
+ 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};
diff --git a/quantum/backlight/backlight_avr.c b/quantum/backlight/backlight_avr.c
index e47192de34..2ecdd4f2c4 100644
--- a/quantum/backlight/backlight_avr.c
+++ b/quantum/backlight/backlight_avr.c
@@ -126,7 +126,7 @@
# define COMxx1 COM1B1
# define OCRxx OCR1B
# endif
-#elif (AUDIO_PIN != B5) && (AUDIO_PIN != B6) && (AUDIO_PIN != B7) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7)
+#elif !defined(B5_AUDIO) && !defined(B6_AUDIO) && !defined(B7_AUDIO)
// Timer 1 is not in use by Audio feature, Backlight can use it
# pragma message "Using hardware timer 1 with software PWM"
# define HARDWARE_PWM
@@ -145,7 +145,7 @@
# define OCIExA OCIE1A
# define OCRxx OCR1A
-#elif (AUDIO_PIN != C4) && (AUDIO_PIN != C5) && (AUDIO_PIN != C6)
+#elif !defined(C6_AUDIO) && !defined(C5_AUDIO) && !defined(C4_AUDIO)
# pragma message "Using hardware timer 3 with software PWM"
// Timer 3 is not in use by Audio feature, Backlight can use it
# define HARDWARE_PWM
diff --git a/util/audio_generate_dac_lut.py b/util/audio_generate_dac_lut.py
deleted file mode 100755
index c31ba3d7ee..0000000000
--- a/util/audio_generate_dac_lut.py
+++ /dev/null
@@ -1,67 +0,0 @@
-#!/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)
diff --git a/util/sample_parser.py b/util/sample_parser.py
deleted file mode 100755
index 70e193aee7..0000000000
--- a/util/sample_parser.py
+++ /dev/null
@@ -1,39 +0,0 @@
-#!/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)
diff --git a/util/wavetable_parser.py b/util/wavetable_parser.py
deleted file mode 100755
index be0f01f7b4..0000000000
--- a/util/wavetable_parser.py
+++ /dev/null
@@ -1,40 +0,0 @@
-#!/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)