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authorDrashna Jaelre <drashna@live.com>2021-02-14 14:40:38 -0800
committerGitHub <noreply@github.com>2021-02-15 09:40:38 +1100
commitc80e5f9f8868ccaa8cb990be6f4da3f1011c2b78 (patch)
tree146b27153da876b0d068c512691536e8aa9a3769 /quantum/audio/audio_pwm.c
parentf53e41ac81662a560a299a23c7863dd2f618a1f8 (diff)
downloadqmk_firmware-c80e5f9f8868ccaa8cb990be6f4da3f1011c2b78.tar.gz
qmk_firmware-c80e5f9f8868ccaa8cb990be6f4da3f1011c2b78.zip
Audio system overhaul (#11820)
* Redo Arm DAC implementation for additive, wavetable synthesis, sample playback changes by Jack Humbert on an implementation for DAC audio on arm/chibios platforms this commits bundles the changes from the arm-dac-work branch focused on audio/audio_arm.* into one commit (leaving out the test-keyboard) f52faeb5d (origin/arm-dac-work) add sample and wavetable examples, parsers for both -> only the changes on audio_arm_.*, the keyboard related parts are split off to a separate commit bfe468ef1 start morphing wavetable 474d100b5 refined a bit 208bee10f play_notes working 3e6478b0b start in-place documentation of dac settings 3e1826a33 fixed blip (rounding error), other waves, added key selection (left/right) 73853d651 5 voices at 44.1khz dfb401b95 limit voices to working number 9632b3379 configuration for the ez 6241f3f3b notes working in a new way * Redo Arm DAC implementation for additive, wavetable synthesis, sample playback changes by Jack Humbert on an implementation for DAC audio on arm/chibios platforms this commit splits off the plank example keymap from commit f52faeb5d (origin/arm-dac-work) add sample and wavetable examples, parsers for both * refactoring: rename audio_ to reflect their supported hardware-platform and audio-generation method: avr vs arm, and pwm vs dac * refactoring: deducplicate ISR code to update the pwm duty-cycle and period in the avr-pwm-implementation pulls three copies of the same code into one function which should improve readability and maintainability :-) * refactoring: move common code of arm and avr implementation into a separate/new file * refactoring: audio_avr_pwm, renaming defines to decouple them from actually used timers, registers and ISRs * refactoring: audio_avr_pwm - replacing function defines with plain register defines aligns better with other existing qmk code (and the new audio_arm_pwm) doing similar pwm thing * add audio-arm-pwm since not all STM32 have a DAC onboard (STM32F2xx and STM32F3xx), pwm-audio is an alternative (STM32F1xx) this code works on a "BluePill" clone, with an STM32F103C8B * clang-format changes on quantum/audio/* only * audio_arm_dac: stopping the notes caused screeching when using the DAC audio paths * audio_arm_pwm: use pushpull on the pin; so that a piezzo can be hooked up direclty without additional components (opendrain would require an external pullup) * refactoring: remove unused file from/for atmel-avr chips * refactoring: remove unused (avr) wavetable file * audio_arm_dac: adapt dac_end callback to changed chibios DAC api the previous chibios (17.6.0) passed along a pointer into the buffer plus a sample_count (which are/already where included in the DACDrivre object) - the current chibios (19.1.0) only passes the driver object. this patch ports more or less exactly what the previous chibios ISR code did: either have the user-callback work the first or second half of the buffer (dacsample_t pointer, with half the DAC_BUFFER_SIZE samples) by adjusting the pointer and sample count * audio-arm-dac: show a compile-warning on undefined audio-pins Co-Authored-By: Drashna Jaelre <drashna@live.com> * audio_arm_dac: switch from exemplary wavetable generation to sine only sine+triangle+squrare is exemplary, and not realy fit for "production" use 'stairs' are usefull for debugging (hardware, with an oscilloscope) * audio_arm_dac: enable output buffers in the STM32 to drive external loads without any additional ciruitry - external opamps and such * audio: prevent out-of-bounds array access * audio_arm_dac: add output-frequency correcting factor * audio_arm_pwm: get both the alternate-function and pm-callback variants back into working condition and do some code-cleanup, refine documentation, ... * audio_arm_pwm: increase pwm frequency for "higher fidelity" on the previous .frequency=100000 higher frequency musical notes came out wrong (frequency measured on a Tektronix TDS2014B) note | freq | arm-pwm C2 | 65.4 | 65.491 C5 | 523.25 | 523.93 C6 | 1046.5 | 1053.38 C7 | 2093 | 2129 C8 | 4186 | 4350.91 with .frequency = 500000 C8 | 4186 | 4204.6 * audio refactoring: remove unused variables * audio_arm_dac: calibrate note tempo: with a tempo of 60beats-per-second a whole-note should last for exactly one second * audio: allow feature selection in rules.mk so the user can switch the audio driver between DAC and PWM on STM32 boards which support both (STM32F2 and up) or select the "pin alternate" pwm mode, for example on STM32F103 * audio-refactoring: move codeblocks in audio.[ch] into more coherent groups and add some inline documentation * audio-refactoring: cleanup and streamline common code between audio_arm_[dac|pwm] untangeling the relation between audio.c and the two drivers and adding more documenting comments :-) * audio_avr_pwm: getting it back into working condition, and cleanup+refactor * audio-refactoring: documentation and typo fixes Co-Authored-By: Nick Brassel <nick@tzarc.org> * audio-refactoring: cleanup defines, inludes and remove debug-prints * audio_chibios_dac: define&use a minimal sampling rate, based on the available tone-range to ease up on the cpu-load, while still rendering the higher notes/tones sufficiently also reenable the lower tones, since with the new implementation there is no evidence of them still beeing 'bugged' * audio-refactoring: one common AUDIO_MAX_VOICES define for all audio-drivers * audio-chibios-pwm: pwm-pin-allternate: make the the timer, timer-channel and alternate function user-#definable * audio_chibios_dac: math.h has fmod for this * Redo Arm DAC implementation for additive, wavetable synthesis, sample playback update Jack Humberts dac-example keymaps for the slight changes in the audio-dac interface * audio-refactoring: use a common AUDIO_PIN configuration switch instead of defines have the user select a pin by configuration in rules.mk instead of a define in config.h has the advantage of beeing in a common form/pattern across all audio-driver implementations * audio-refactoring: switch backlight_avr.c to the new AUDIO_PIN defines * audio-common: have advance_note return a boolean if the note changed, to the next one in the melody beeing played * audio-chibios-pwm: fix issue with ~130ms silence between note/frequency changes while playing a SONG through trial,error and a scope/logic analyzer figured out Chibios-PWMDriver (at least in the current version) misbehaves if the initial period is set to zero (or one; two seems to work); when thats the case subsequent calls to 'pwmChhangePeriod' + pwmEnableChannel took ~135ms of silence, before the PWM continued with the new frequency... * audio-refactoring: get 'play_note' working again with a limited number of available voices (say AUDIO_VOICES_MAX=1) allow new frequencies to be played, by discarding the oldest one in the 'frequencies' queue * audio: set the fallback driver to DAC for chibios and PWM for all others (==avr at the moment) * audio-refactoring: moore documentation and some cleanup * audio-avr-pwm: no fallback on unset AUDIO_PIN this seems to be the expected behaviour by some keyboards (looking at ckeys/handwire_101:default) which otherwise fail to build because the firmware-image ends up beeing too large for the atmega... so we fail silently instead to keep travis happy * audio-refactoring: untangling terminology: voice->tone the code actually was working on tones (combination of pitch/frequency, duration, timbre, intensity/volume) and not voices (characteristic sound of an instrument; think piano vs guitar, which can be played together, each having its own "track" = voice on a music sheet) * audio-pwm: allow freq=0 aka a pause/rest in a SONG continue processing, but do not enable pwm units, since freq=0 wouldn't produce any sound anyway (and lead to division by zero on that occasion) * audio-refactoring: audio_advance_note -> audio_advance_state since it does not only affect 'one note', but the internally kept state as a whole * audio-refactoring: untangling terminology: polyphony the feature om the "inherited" avr code has little to do with polyphony (see wikipedia), but is more a time-multiplexing feature, to work around hardware limitations - like only having one pwm channel, that could on its own only reproduce one voice/instrument at a time * audio-chibios-dac: add zero-crossing feature have tones only change/stop when the waveform approaches zero - to avoid audible clicks note that this also requires the samples to start at zero, since the internally kept index into the samples is reset to zero too * audio-refactoring: feature: time-multiplexing of tones on a single output channel this feature was in the original avr-pwm implementation misnomed as "polyphony" with polyphony_rate and so on; did the same thing though: time-multiplexing multiple active notes so that a single output channel could reproduce more than one note at a time (which is not the same as a polyphony - see wikipedia :-) ) * audio-avr-pwm: get music-mode working (again) on AVRs with both pwm channels, or either one of the two :-) play_notes worked already - but music_mode uses play_note * audio-refactoring: split define MAX_SIMULTANEOUS_TONES -> TONE_STACKSIZE since the two cases are independant from one another, the hardware might impose limitations on the number of simultaneously reproducable tones, but the audio state should be able to track an unrelated number of notes recently started by play_note * audio-arm-dac: per define selectable sample-luts plus generation script in ./util * audio-refactoring: heh, avr has a MIN... * audio-refactoring: add basic dac audio-driver based on the current/master implementation whereas current=d96380e65496912e0f68e6531565f4b45efd1623 which is the state of things before this whole audio-refactoring branch boiled down to interface with the refactored audio system = removing all redundant state-managing and frequency calculation * audio-refactoring: rename audio-drivers to driver_$PLATFORM_$DRIVER * audio-arm-pwm: split the software/hardware implementations into separate files which saves us partially from a 'define hell', with the tradeoff that now two somewhat similar chibios_pwm implementations have to be maintained * audio-refactoring: update documentation * audio-arm-dac: apply AUDIO_PIN defines to driver_chibios_dac_basic * audio-arm-dac: dac_additive: stop the hardware when the last sample completed the audio system calls for a driver_stop, which is delayed until the current sample conversion finishes * audio-refactoring: make function-namespace consistent - all (public) audio functions start with audio_ - also refactoring play*_notes/tones to play*_melody, to visually distance it a bit from play*_tone/_note * audio-refactoring: consistent define namespace: DAC_ -> AUDIO_DAC_ * audio-arm-dac: update (inline) documentation regarding MAX for sample values * audio-chibios-dac: remove zero-crossing feature didn't quite work as intended anyway, and stopping the hardware on close-to-zero seems to be enought anyway * audio-arm-dac: dac_basic: respect the configured sample-rate * audio-arm-pwm: have 'note_timbre' influence the pwm-duty cycle like it already does in the avr implementation * audio-refactoring: get VIBRATO working (again) with all drivers (verified with chibios_[dac|pwm]) * audio-arm-dac: zero-crossing feature (Mk II) wait for the generated waveform to approach 'zero' before either turning off the output+timer or switching to the current set of active_tones * audio-refactoring: re-add note-resting -> introduce short_rest inbetween - introduce a short pause/rest between two notes of the same frequency, to separate them audibly - also updating the refactoring comments * audio-refactoring: cleanup refactoring remnants remove the former avr-isr code block - since all its features are now refactored into the different parts of the current system also updates the TODOS * audio-refactoring: reserve negative numbers as unitialized frequencies to allow the valid tone/frequency f=0Hz == rest/pause * audio-refactoring: FIX: first note of melody was missing the first note was missing because 'goto_next_note'=false overrode a state_change=true of the initial play_tone and some code-indentations/cleanup of related parts * audio-arm-dac: fix hardware init-click due to wron .init= value * audio-refactoring: new conveniance function: audio_play_click which can be used to further refactor/remove fauxclicky (avr only) and/or the 'clicky' features * audio-refactoring: clang-format on quantum/audio/* * audio-avr-pwm: consecutive notes of the same frequency get a pause inserted inbetween by audio.c * audio-refactoring: use milliseconds instead of seconds for 'click' parameters clicks are supposed to be short, seconds make little sense * audio-refactoring: use timer ticks instead of counters local counters were used in the original (avr)ISR to advance an index into the lookup tables (for vibrato), and something similar was used for the tone-multiplexing feature decoupling these from the (possibly irregular) calls to advance_state made sesne, since those counters/lookups need to be in relation to a wall-time anyway * audio-refactoring: voices.c: drop 'envelope_index' counter in favour of timer ticks * audio-refactoring: move vibrato and timbre related parts from audio.c to voices.c also drops the now (globally) unused AUDIO_VIBRATO/AUDIO_ENABLE_VIBRATO defines * audio.c: use system-ticks instead of counters the drivers have to take care of for the internal state posision since there already is a system-tick with ms resolution, keeping count separatly with each driver implementation makes little sense; especially since they had to take special care to call audio_advance_state with the correct step/end parameters for the audio state to advance regularly and with the correct pace * audio.c: stop notes after new ones have been started avoids brief states of with no notes playing that would otherwise stop the hardware and might lead to clicks * audio.c: bugfix: actually play a pause instead of just idling/stopping which lead the pwm drivers to stop entirely... * audio-arm-pwm: pwm-software: add inverted output new define AUDIO_PIN_ALT_AS_NEGATIVE will generate an inverted signal on the alternate pin, which boosts the volume if a piezo is connected to both AUDIO_PIN and AUDIO_PIN_ALT * audio-arm-dac: basic: handle piezo configured&wired to both audio pins * audio-refactoring: docs: update for AUDIO_PIN_ALT_AS_NEGATIVE and piezo wiring * audio.c: bugfix: use timer_elapsed32 instad of keeping timestamps avoids running into issues when the uint32 of the timer overflows * audio-refactoring: add 'pragma once' and remove deprecated NOTE_REST * audio_arm_dac: basic: add missing bracket * audio.c: fix delta calculation was in the wrong place, needs to use the 'last_timestamp' before it was reset * audio-refactoring: buildfix: wrong legacy macro for set_timbre * audio.c: 16bit timerstamps suffice * audio-refactoring: separate includes for AVR and chibios * audio-refactoring: timbre: use uint8 instead of float * audio-refactoring: duration: use uint16 for internal per-tone/note state * audio-refactoring: tonemultiplexing: use uint16 instead of float * audio-arm-dac: additive: set second pin output-low used when a piezo is connected to AUDIO_PIN and AUDIO_PIN_ALT, with PIN_ALT_AS_NEGATIVE * audio-refactoring: move AUDIO_PIN selection from rules.mk to config.h to be consistent with how other features are handled in QMK * audio-refactoring: buildfix: wrong legacy macro for set_tempo * audio-arm-dac: additive: set second pin output-low -- FIXUP * audio.c: do duration<>ms conversion in uint instead of float on AVR, to save a couple of bytes in the firmware size * audio-refactoring: cleanup eeprom defines/usage for ARM, avr is handled automagically through the avr libc and common_features.mk Co-Authored-By: Drashna Jaelre <drashna@live.com> * audio.h: throw an error if OFF is larger than MAX * audio-arm-dac: basic: actually stop the dac-conversion on a audio_driver_stop to put the output pin in a known state == AUDIO_DAC_OFF_VALUE, instead of just leaving them where the last conversion was... with AUDIO_PIN_ALT_AS_NEGATIVE this meant one output was left HIGH while the other was left LOW one CAVEAT: due to this change the opposing squarewave when using both A4 and A5 with AUDIO_PIN_ALT_AS_NEGATIVE show extra pulses at the beginning/end on one of the outputs, the two waveforms are in sync otherwise. the extra pusles probably matter little, since this is no high-fidelity sound generation :P * audio-arm-dac: additive: move zero-crossing code out of dac_value_generate which is/should be user-overridable == simple, and doing one thing: providing sample values state-transitions necessary for the zero crossing are better handled in the surrounding loop in the dac_end callback * audio-arm-dac: dac-additive: zero-crossing: ramping up or down after a start trigger ramp up: generate values until zero=OFF_VALUE is reached, then continue normally same in reverse for strop trigger: output values until zero is reached/crossed, then keep OFF_VALUE on the output * audio-arm-dac: dac-additive: BUGFIX: return OFF_VALUE when a pause is playing fixes a bug during SONG playback, which suddenly stopped when it encoutnered a pause * audio-arm-dac: set a sensible default for AUDIO_DAC_VALUE_OFF 1/2 MAX was probably exemplary, can't think of a setup where that would make sense :-P * audio-arm-dac: update synth_sample/_wavetable for new pin-defines * audio-arm-dac: default for AUDIO_DAC_VALUE_OFF turned out that zero or max are bad default choices: when multiple tones are played (>>5) and released at the same time (!), due to the complex waveform never reaching 'zero' the output can take quite a while to reach zero, and hence the zero-crossing code only "releases" the output waaay to late * audio-arm-dac: additive: use DAC for negative pin instead of PAL, which only allows the pin to be configured as output; LOW or HIGH * audio-arm-dac: more compile-time configuration checks * audio-refactoring: typo fixed * audio-refactoring: clang-format on quantum/audio/* * audio-avr-pwm: add defines for B-pin as primary/only speaker also updates documentation. * audio-refactoring: update documentation with proton-c config.h example * audio-refactoring: move glissando (TODO) to voices.c refactored/saved from the original glissando implementation in then upstream-master:audio_avr.c still needs some work though, as it is now the calculation *should* work, but the start-frequency needs to be tracked somewhere/somehow; not only during a SONG playback but also with user input? * audio-refactoring: cleanup: one round of aspell -c * audio-avr-pwm: back to AUDIO_PIN since config_common.h expands them to plain integers, the AUDIO_PIN define can directly be compared to e.g. B5 so there is no need to deal with separate defines like AUDIO_PIN_B5 * audio-refactoring: add technical documentation audio_driver.md which moves some in-code documentation there * audio-arm-dac: move AUDIO_PIN checks into c-code instead of doing everything with the preprocessor, since A4/A5 do not expand to simple integers, preprocessor int-comparison is not possible. but necessary to get a consistent configuration scheme going throughout the audio-code... solution: let c-code handle the different AUDIO_PIN configurations instead (and leave code/size optimizations to the compiler) * audio-arm-dac: compile-fix: set AUDIO_PIN if unset workaround to get the build going again, and be backwarts compatible to arm-keyboards which not yet set the AUDIO_PIN define. until the define is enforced through an '#error" * audio-refactoring: document tone-multiplexing feature * audio-refactoring: Apply suggestions from documentation review Co-authored-by: James Young <18669334+noroadsleft@users.noreply.github.com> * audio-refactoring: Update docs/audio_driver.md * audio-refactoring: docs: fix markdown newlines Terminating a line in Markdown with <space>-<space>-<linebreak> creates an HTML single-line break (<br>). Co-authored-by: James Young <18669334+noroadsleft@users.noreply.github.com> * audio-arm-dac: additive: fix AUDIO_PIN_ALT handling * audio-arm-pwm: align define naming with other drivers Co-authored-by: Joel Challis <git@zvecr.com> * audio-refactoring: set detault tempo to 120 and add documentation for the override * audio-refactoring: update backlight define checks to new AUDIO_PIN names * audio-refactoring: reworking PWM related defines to be more consistent with other QMK code Co-authored-by: Joel Challis <git@zvecr.com> * audio-arm: have the state-update-timer user configurable defaulting to GPTD6 or GPTD8 for stm32f2+ (=proton-c) stm32f1 might need to set this to GPTD4, since 6 and 8 are not available * audio-refactoring: PLAY_NOTE_ARRAY was already removed in master * Add prototype for startup * Update chibiOS dac basic to disable pins on stop * Add defaults for Proton C * avoid hanging audio if note is completely missed * Don't redefine pins if they're already defined * Define A4 and A5 for CTPC support * Add license headers to keymap files * Remove figlet? comments * Add DAC config to audio driver docs * Apply suggestions from code review Co-authored-by: Jack Humbert <jack.humb@gmail.com> * Add license header to py files * correct license header * Add JohSchneider's name to modified files AKA credit where credit's due * Set executable permission and change interpeter * Add 'wave' to pip requirements * Improve documentation * Add some settings I missed * Strip AUDIO_DRIVER to parse the name correctly * fix depreciated * Update util/audio_generate_dac_lut.py Co-authored-by: Jack Humbert <jack.humb@gmail.com> * Fix type in clueboard config * Apply suggestions from tzarc Co-authored-by: Nick Brassel <nick@tzarc.org> Co-authored-by: Johannes <you@example.com> Co-authored-by: JohSchneider <JohSchneider@googlemail.com> Co-authored-by: Nick Brassel <nick@tzarc.org> Co-authored-by: James Young <18669334+noroadsleft@users.noreply.github.com> Co-authored-by: Joel Challis <git@zvecr.com> Co-authored-by: Joshua Diamond <josh@windowoffire.com> Co-authored-by: Jack Humbert <jack.humb@gmail.com>
Diffstat (limited to 'quantum/audio/audio_pwm.c')
-rw-r--r--quantum/audio/audio_pwm.c606
1 files changed, 0 insertions, 606 deletions
diff --git a/quantum/audio/audio_pwm.c b/quantum/audio/audio_pwm.c
deleted file mode 100644
index d93ac4bb4..000000000
--- a/quantum/audio/audio_pwm.c
+++ /dev/null
@@ -1,606 +0,0 @@
1/* Copyright 2016 Jack Humbert
2 *
3 * This program is free software: you can redistribute it and/or modify
4 * it under the terms of the GNU General Public License as published by
5 * the Free Software Foundation, either version 2 of the License, or
6 * (at your option) any later version.
7 *
8 * This program is distributed in the hope that it will be useful,
9 * but WITHOUT ANY WARRANTY; without even the implied warranty of
10 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
11 * GNU General Public License for more details.
12 *
13 * You should have received a copy of the GNU General Public License
14 * along with this program. If not, see <http://www.gnu.org/licenses/>.
15 */
16#include <stdio.h>
17#include <string.h>
18//#include <math.h>
19#include <avr/pgmspace.h>
20#include <avr/interrupt.h>
21#include <avr/io.h>
22#include "print.h"
23#include "audio.h"
24#include "keymap.h"
25
26#include "eeconfig.h"
27
28#define PI 3.14159265
29
30#define CPU_PRESCALER 8
31
32#ifndef STARTUP_SONG
33# define STARTUP_SONG SONG(STARTUP_SOUND)
34#endif
35float startup_song[][2] = STARTUP_SONG;
36
37// Timer Abstractions
38
39// TIMSK3 - Timer/Counter #3 Interrupt Mask Register
40// Turn on/off 3A interputs, stopping/enabling the ISR calls
41#define ENABLE_AUDIO_COUNTER_3_ISR TIMSK3 |= _BV(OCIE3A)
42#define DISABLE_AUDIO_COUNTER_3_ISR TIMSK3 &= ~_BV(OCIE3A)
43
44// TCCR3A: Timer/Counter #3 Control Register
45// Compare Output Mode (COM3An) = 0b00 = Normal port operation, OC3A disconnected from PC6
46#define ENABLE_AUDIO_COUNTER_3_OUTPUT TCCR3A |= _BV(COM3A1);
47#define DISABLE_AUDIO_COUNTER_3_OUTPUT TCCR3A &= ~(_BV(COM3A1) | _BV(COM3A0));
48
49#define NOTE_PERIOD ICR3
50#define NOTE_DUTY_CYCLE OCR3A
51
52#ifdef PWM_AUDIO
53# include "wave.h"
54# define SAMPLE_DIVIDER 39
55# define SAMPLE_RATE (2000000.0 / SAMPLE_DIVIDER / 2048)
56// Resistor value of 1/ (2 * PI * 10nF * (2000000 hertz / SAMPLE_DIVIDER / 10)) for 10nF cap
57
58float places[8] = {0, 0, 0, 0, 0, 0, 0, 0};
59uint16_t place_int = 0;
60bool repeat = true;
61#endif
62
63void delay_us(int count) {
64 while (count--) {
65 _delay_us(1);
66 }
67}
68
69int voices = 0;
70int voice_place = 0;
71float frequency = 0;
72int volume = 0;
73long position = 0;
74
75float frequencies[8] = {0, 0, 0, 0, 0, 0, 0, 0};
76int volumes[8] = {0, 0, 0, 0, 0, 0, 0, 0};
77bool sliding = false;
78
79float place = 0;
80
81uint8_t* sample;
82uint16_t sample_length = 0;
83// float freq = 0;
84
85bool playing_notes = false;
86bool playing_note = false;
87float note_frequency = 0;
88float note_length = 0;
89uint8_t note_tempo = TEMPO_DEFAULT;
90float note_timbre = TIMBRE_DEFAULT;
91uint16_t note_position = 0;
92float (*notes_pointer)[][2];
93uint16_t notes_count;
94bool notes_repeat;
95float notes_rest;
96bool note_resting = false;
97
98uint16_t current_note = 0;
99uint8_t rest_counter = 0;
100
101#ifdef VIBRATO_ENABLE
102float vibrato_counter = 0;
103float vibrato_strength = .5;
104float vibrato_rate = 0.125;
105#endif
106
107float polyphony_rate = 0;
108
109static bool audio_initialized = false;
110
111audio_config_t audio_config;
112
113uint16_t envelope_index = 0;
114
115void audio_init() {
116 // Check EEPROM
117 if (!eeconfig_is_enabled()) {
118 eeconfig_init();
119 }
120 audio_config.raw = eeconfig_read_audio();
121
122#ifdef PWM_AUDIO
123
124 PLLFRQ = _BV(PDIV2);
125 PLLCSR = _BV(PLLE);
126 while (!(PLLCSR & _BV(PLOCK)))
127 ;
128 PLLFRQ |= _BV(PLLTM0); /* PCK 48MHz */
129
130 /* Init a fast PWM on Timer4 */
131 TCCR4A = _BV(COM4A0) | _BV(PWM4A); /* Clear OC4A on Compare Match */
132 TCCR4B = _BV(CS40); /* No prescaling => f = PCK/256 = 187500Hz */
133 OCR4A = 0;
134
135 /* Enable the OC4A output */
136 DDRC |= _BV(PORTC6);
137
138 DISABLE_AUDIO_COUNTER_3_ISR; // Turn off 3A interputs
139
140 TCCR3A = 0x0; // Options not needed
141 TCCR3B = _BV(CS31) | _BV(CS30) | _BV(WGM32); // 64th prescaling and CTC
142 OCR3A = SAMPLE_DIVIDER - 1; // Correct count/compare, related to sample playback
143
144#else
145
146 // Set port PC6 (OC3A and /OC4A) as output
147 DDRC |= _BV(PORTC6);
148
149 DISABLE_AUDIO_COUNTER_3_ISR;
150
151 // TCCR3A / TCCR3B: Timer/Counter #3 Control Registers
152 // Compare Output Mode (COM3An) = 0b00 = Normal port operation, OC3A disconnected from PC6
153 // Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14 (Period = ICR3, Duty Cycle = OCR3A)
154 // Clock Select (CS3n) = 0b010 = Clock / 8
155 TCCR3A = (0 << COM3A1) | (0 << COM3A0) | (1 << WGM31) | (0 << WGM30);
156 TCCR3B = (1 << WGM33) | (1 << WGM32) | (0 << CS32) | (1 << CS31) | (0 << CS30);
157
158#endif
159
160 audio_initialized = true;
161}
162
163void audio_startup() {
164 if (audio_config.enable) {
165 PLAY_SONG(startup_song);
166 }
167}
168
169void stop_all_notes() {
170 if (!audio_initialized) {
171 audio_init();
172 }
173 voices = 0;
174#ifdef PWM_AUDIO
175 DISABLE_AUDIO_COUNTER_3_ISR;
176#else
177 DISABLE_AUDIO_COUNTER_3_ISR;
178 DISABLE_AUDIO_COUNTER_3_OUTPUT;
179#endif
180
181 playing_notes = false;
182 playing_note = false;
183 frequency = 0;
184 volume = 0;
185
186 for (uint8_t i = 0; i < 8; i++) {
187 frequencies[i] = 0;
188 volumes[i] = 0;
189 }
190}
191
192void stop_note(float freq) {
193 if (playing_note) {
194 if (!audio_initialized) {
195 audio_init();
196 }
197#ifdef PWM_AUDIO
198 freq = freq / SAMPLE_RATE;
199#endif
200 for (int i = 7; i >= 0; i--) {
201 if (frequencies[i] == freq) {
202 frequencies[i] = 0;
203 volumes[i] = 0;
204 for (int j = i; (j < 7); j++) {
205 frequencies[j] = frequencies[j + 1];
206 frequencies[j + 1] = 0;
207 volumes[j] = volumes[j + 1];
208 volumes[j + 1] = 0;
209 }
210 break;
211 }
212 }
213 voices--;
214 if (voices < 0) voices = 0;
215 if (voice_place >= voices) {
216 voice_place = 0;
217 }
218 if (voices == 0) {
219#ifdef PWM_AUDIO
220 DISABLE_AUDIO_COUNTER_3_ISR;
221#else
222 DISABLE_AUDIO_COUNTER_3_ISR;
223 DISABLE_AUDIO_COUNTER_3_OUTPUT;
224#endif
225 frequency = 0;
226 volume = 0;
227 playing_note = false;
228 }
229 }
230}
231
232#ifdef VIBRATO_ENABLE
233
234float mod(float a, int b) {
235 float r = fmod(a, b);
236 return r < 0 ? r + b : r;
237}
238
239float vibrato(float average_freq) {
240# ifdef VIBRATO_STRENGTH_ENABLE
241 float vibrated_freq = average_freq * pow(vibrato_lut[(int)vibrato_counter], vibrato_strength);
242# else
243 float vibrated_freq = average_freq * vibrato_lut[(int)vibrato_counter];
244# endif
245 vibrato_counter = mod((vibrato_counter + vibrato_rate * (1.0 + 440.0 / average_freq)), VIBRATO_LUT_LENGTH);
246 return vibrated_freq;
247}
248
249#endif
250
251ISR(TIMER3_COMPA_vect) {
252 if (playing_note) {
253#ifdef PWM_AUDIO
254 if (voices == 1) {
255 // SINE
256 OCR4A = pgm_read_byte(&sinewave[(uint16_t)place]) >> 2;
257
258 // SQUARE
259 // if (((int)place) >= 1024){
260 // OCR4A = 0xFF >> 2;
261 // } else {
262 // OCR4A = 0x00;
263 // }
264
265 // SAWTOOTH
266 // OCR4A = (int)place / 4;
267
268 // TRIANGLE
269 // if (((int)place) >= 1024) {
270 // OCR4A = (int)place / 2;
271 // } else {
272 // OCR4A = 2048 - (int)place / 2;
273 // }
274
275 place += frequency;
276
277 if (place >= SINE_LENGTH) place -= SINE_LENGTH;
278
279 } else {
280 int sum = 0;
281 for (int i = 0; i < voices; i++) {
282 // SINE
283 sum += pgm_read_byte(&sinewave[(uint16_t)places[i]]) >> 2;
284
285 // SQUARE
286 // if (((int)places[i]) >= 1024){
287 // sum += 0xFF >> 2;
288 // } else {
289 // sum += 0x00;
290 // }
291
292 places[i] += frequencies[i];
293
294 if (places[i] >= SINE_LENGTH) places[i] -= SINE_LENGTH;
295 }
296 OCR4A = sum;
297 }
298#else
299 if (voices > 0) {
300 float freq;
301 if (polyphony_rate > 0) {
302 if (voices > 1) {
303 voice_place %= voices;
304 if (place++ > (frequencies[voice_place] / polyphony_rate / CPU_PRESCALER)) {
305 voice_place = (voice_place + 1) % voices;
306 place = 0.0;
307 }
308 }
309# ifdef VIBRATO_ENABLE
310 if (vibrato_strength > 0) {
311 freq = vibrato(frequencies[voice_place]);
312 } else {
313# else
314 {
315# endif
316 freq = frequencies[voice_place];
317 }
318 } else {
319 if (frequency != 0 && frequency < frequencies[voices - 1] && frequency < frequencies[voices - 1] * pow(2, -440 / frequencies[voices - 1] / 12 / 2)) {
320 frequency = frequency * pow(2, 440 / frequency / 12 / 2);
321 } else if (frequency != 0 && frequency > frequencies[voices - 1] && frequency > frequencies[voices - 1] * pow(2, 440 / frequencies[voices - 1] / 12 / 2)) {
322 frequency = frequency * pow(2, -440 / frequency / 12 / 2);
323 } else {
324 frequency = frequencies[voices - 1];
325 }
326
327# ifdef VIBRATO_ENABLE
328 if (vibrato_strength > 0) {
329 freq = vibrato(frequency);
330 } else {
331# else
332 {
333# endif
334 freq = frequency;
335 }
336 }
337
338 if (envelope_index < 65535) {
339 envelope_index++;
340 }
341 freq = voice_envelope(freq);
342
343 if (freq < 30.517578125) freq = 30.52;
344 NOTE_PERIOD = (int)(((double)F_CPU) / (freq * CPU_PRESCALER)); // Set max to the period
345 NOTE_DUTY_CYCLE = (int)((((double)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre); // Set compare to half the period
346 }
347#endif
348 }
349
350 // SAMPLE
351 // OCR4A = pgm_read_byte(&sample[(uint16_t)place_int]);
352
353 // place_int++;
354
355 // if (place_int >= sample_length)
356 // if (repeat)
357 // place_int -= sample_length;
358 // else
359 // DISABLE_AUDIO_COUNTER_3_ISR;
360
361 if (playing_notes) {
362#ifdef PWM_AUDIO
363 OCR4A = pgm_read_byte(&sinewave[(uint16_t)place]) >> 0;
364
365 place += note_frequency;
366 if (place >= SINE_LENGTH) place -= SINE_LENGTH;
367#else
368 if (note_frequency > 0) {
369 float freq;
370
371# ifdef VIBRATO_ENABLE
372 if (vibrato_strength > 0) {
373 freq = vibrato(note_frequency);
374 } else {
375# else
376 {
377# endif
378 freq = note_frequency;
379 }
380
381 if (envelope_index < 65535) {
382 envelope_index++;
383 }
384 freq = voice_envelope(freq);
385
386 NOTE_PERIOD = (int)(((double)F_CPU) / (freq * CPU_PRESCALER)); // Set max to the period
387 NOTE_DUTY_CYCLE = (int)((((double)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre); // Set compare to half the period
388 } else {
389 NOTE_PERIOD = 0;
390 NOTE_DUTY_CYCLE = 0;
391 }
392#endif
393
394 note_position++;
395 bool end_of_note = false;
396 if (NOTE_PERIOD > 0)
397 end_of_note = (note_position >= (note_length / NOTE_PERIOD * 0xFFFF));
398 else
399 end_of_note = (note_position >= (note_length * 0x7FF));
400 if (end_of_note) {
401 current_note++;
402 if (current_note >= notes_count) {
403 if (notes_repeat) {
404 current_note = 0;
405 } else {
406#ifdef PWM_AUDIO
407 DISABLE_AUDIO_COUNTER_3_ISR;
408#else
409 DISABLE_AUDIO_COUNTER_3_ISR;
410 DISABLE_AUDIO_COUNTER_3_OUTPUT;
411#endif
412 playing_notes = false;
413 return;
414 }
415 }
416 if (!note_resting && (notes_rest > 0)) {
417 note_resting = true;
418 note_frequency = 0;
419 note_length = notes_rest;
420 current_note--;
421 } else {
422 note_resting = false;
423#ifdef PWM_AUDIO
424 note_frequency = (*notes_pointer)[current_note][0] / SAMPLE_RATE;
425 note_length = (*notes_pointer)[current_note][1] * (((float)note_tempo) / 100);
426#else
427 envelope_index = 0;
428 note_frequency = (*notes_pointer)[current_note][0];
429 note_length = ((*notes_pointer)[current_note][1] / 4) * (((float)note_tempo) / 100);
430#endif
431 }
432 note_position = 0;
433 }
434 }
435
436 if (!audio_config.enable) {
437 playing_notes = false;
438 playing_note = false;
439 }
440}
441
442void play_note(float freq, int vol) {
443 if (!audio_initialized) {
444 audio_init();
445 }
446
447 if (audio_config.enable && voices < 8) {
448 DISABLE_AUDIO_COUNTER_3_ISR;
449
450 // Cancel notes if notes are playing
451 if (playing_notes) stop_all_notes();
452
453 playing_note = true;
454
455 envelope_index = 0;
456
457#ifdef PWM_AUDIO
458 freq = freq / SAMPLE_RATE;
459#endif
460 if (freq > 0) {
461 frequencies[voices] = freq;
462 volumes[voices] = vol;
463 voices++;
464 }
465
466#ifdef PWM_AUDIO
467 ENABLE_AUDIO_COUNTER_3_ISR;
468#else
469 ENABLE_AUDIO_COUNTER_3_ISR;
470 ENABLE_AUDIO_COUNTER_3_OUTPUT;
471#endif
472 }
473}
474
475void play_notes(float (*np)[][2], uint16_t n_count, bool n_repeat, float n_rest) {
476 if (!audio_initialized) {
477 audio_init();
478 }
479
480 if (audio_config.enable) {
481 DISABLE_AUDIO_COUNTER_3_ISR;
482
483 // Cancel note if a note is playing
484 if (playing_note) stop_all_notes();
485
486 playing_notes = true;
487
488 notes_pointer = np;
489 notes_count = n_count;
490 notes_repeat = n_repeat;
491 notes_rest = n_rest;
492
493 place = 0;
494 current_note = 0;
495
496#ifdef PWM_AUDIO
497 note_frequency = (*notes_pointer)[current_note][0] / SAMPLE_RATE;
498 note_length = (*notes_pointer)[current_note][1] * (((float)note_tempo) / 100);
499#else
500 note_frequency = (*notes_pointer)[current_note][0];
501 note_length = ((*notes_pointer)[current_note][1] / 4) * (((float)note_tempo) / 100);
502#endif
503 note_position = 0;
504
505#ifdef PWM_AUDIO
506 ENABLE_AUDIO_COUNTER_3_ISR;
507#else
508 ENABLE_AUDIO_COUNTER_3_ISR;
509 ENABLE_AUDIO_COUNTER_3_OUTPUT;
510#endif
511 }
512}
513
514#ifdef PWM_AUDIO
515void play_sample(uint8_t* s, uint16_t l, bool r) {
516 if (!audio_initialized) {
517 audio_init();
518 }
519
520 if (audio_config.enable) {
521 DISABLE_AUDIO_COUNTER_3_ISR;
522 stop_all_notes();
523 place_int = 0;
524 sample = s;
525 sample_length = l;
526 repeat = r;
527
528 ENABLE_AUDIO_COUNTER_3_ISR;
529 }
530}
531#endif
532
533void audio_toggle(void) {
534 audio_config.enable ^= 1;
535 eeconfig_update_audio(audio_config.raw);
536}
537
538void audio_on(void) {
539 audio_config.enable = 1;
540 eeconfig_update_audio(audio_config.raw);
541}
542
543void audio_off(void) {
544 audio_config.enable = 0;
545 eeconfig_update_audio(audio_config.raw);
546}
547
548#ifdef VIBRATO_ENABLE
549
550// Vibrato rate functions
551
552void set_vibrato_rate(float rate) { vibrato_rate = rate; }
553
554void increase_vibrato_rate(float change) { vibrato_rate *= change; }
555
556void decrease_vibrato_rate(float change) { vibrato_rate /= change; }
557
558# ifdef VIBRATO_STRENGTH_ENABLE
559
560void set_vibrato_strength(float strength) { vibrato_strength = strength; }
561
562void increase_vibrato_strength(float change) { vibrato_strength *= change; }
563
564void decrease_vibrato_strength(float change) { vibrato_strength /= change; }
565
566# endif /* VIBRATO_STRENGTH_ENABLE */
567
568#endif /* VIBRATO_ENABLE */
569
570// Polyphony functions
571
572void set_polyphony_rate(float rate) { polyphony_rate = rate; }
573
574void enable_polyphony() { polyphony_rate = 5; }
575
576void disable_polyphony() { polyphony_rate = 0; }
577
578void increase_polyphony_rate(float change) { polyphony_rate *= change; }
579
580void decrease_polyphony_rate(float change) { polyphony_rate /= change; }
581
582// Timbre function
583
584void set_timbre(float timbre) { note_timbre = timbre; }
585
586// Tempo functions
587
588void set_tempo(uint8_t tempo) { note_tempo = tempo; }
589
590void decrease_tempo(uint8_t tempo_change) { note_tempo += tempo_change; }
591
592void increase_tempo(uint8_t tempo_change) {
593 if (note_tempo - tempo_change < 10) {
594 note_tempo = 10;
595 } else {
596 note_tempo -= tempo_change;
597 }
598}
599
600//------------------------------------------------------------------------------
601// Override these functions in your keymap file to play different tunes on
602// startup and bootloader jump
603__attribute__((weak)) void play_startup_tone() {}
604
605__attribute__((weak)) void play_goodbye_tone() {}
606//------------------------------------------------------------------------------
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