From 996a19ee7ba3308e17fd347afde0b135852835cc Mon Sep 17 00:00:00 2001 From: Drashna Jael're Date: Tue, 29 Jun 2021 15:36:35 -0700 Subject: Revert "Audio system overhaul (#11820)" due to freezing issues This reverts commit c80e5f9f8868ccaa8cb990be6f4da3f1011c2b78. --- common_features.mk | 21 +- keyboards/planck/config.h | 2 +- keyboards/planck/ez/config.h | 5 +- quantum/audio/audio.c | 539 ------------------------- quantum/audio/audio.h | 281 +++---------- quantum/audio/audio_chibios.c | 20 +- quantum/audio/audio_pwm.c | 606 ++++++++++++++++++++++++++++ quantum/audio/driver_avr_pwm.h | 17 - quantum/audio/driver_avr_pwm_hardware.c | 332 --------------- quantum/audio/driver_chibios_dac.h | 126 ------ quantum/audio/driver_chibios_dac_additive.c | 335 --------------- quantum/audio/driver_chibios_dac_basic.c | 245 ----------- quantum/audio/driver_chibios_pwm.h | 40 -- quantum/audio/driver_chibios_pwm_hardware.c | 144 ------- quantum/audio/driver_chibios_pwm_software.c | 164 -------- quantum/audio/musical_notes.h | 77 ++-- quantum/audio/voices.c | 170 +++----- quantum/audio/voices.h | 20 - quantum/audio/wave.h | 36 ++ quantum/backlight/backlight_avr.c | 4 +- util/audio_generate_dac_lut.py | 67 --- util/sample_parser.py | 39 -- util/wavetable_parser.py | 40 -- 23 files changed, 798 insertions(+), 2532 deletions(-) delete mode 100644 quantum/audio/audio.c create mode 100644 quantum/audio/audio_pwm.c delete mode 100644 quantum/audio/driver_avr_pwm.h delete mode 100644 quantum/audio/driver_avr_pwm_hardware.c delete mode 100644 quantum/audio/driver_chibios_dac.h delete mode 100644 quantum/audio/driver_chibios_dac_additive.c delete mode 100644 quantum/audio/driver_chibios_dac_basic.c delete mode 100644 quantum/audio/driver_chibios_pwm.h delete mode 100644 quantum/audio/driver_chibios_pwm_hardware.c delete mode 100644 quantum/audio/driver_chibios_pwm_software.c create mode 100644 quantum/audio/wave.h delete mode 100755 util/audio_generate_dac_lut.py delete mode 100755 util/sample_parser.py delete mode 100755 util/wavetable_parser.py 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 . #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 . - */ -#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 . */ + #pragma once #include #include +#if defined(__AVR__) +# include +#endif +#include "wait.h" #include "musical_notes.h" #include "song_list.h" #include "voices.h" #include "quantum.h" #include -#if defined(__AVR__) -# include -# 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(¬e_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(¬e_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(¬e_array, NOTE_ARRAY_SIZE((note_array)), false) +#define PLAY_LOOP(note_array) play_notes(¬e_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 . + */ +#include +#include +//#include +#include +#include +#include +#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 . - */ -#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 . - */ - -#if defined(__AVR__) -# include -# include -# include -#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 . - */ -#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 . - */ - -#include "audio.h" -#include -#include - -/* - 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 . - */ - -#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 . - */ -#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 . - */ - -/* -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 . - */ - -/* -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 . */ + #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 -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 . + */ + +#include +#include +#include + +#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, 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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 . -# - -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 . -# - -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(". -# - -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("