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-rw-r--r--quantum/audio/audio.h13
-rw-r--r--quantum/audio/driver_avr_pwm.h17
-rw-r--r--quantum/audio/driver_avr_pwm_hardware.c332
-rw-r--r--quantum/audio/driver_chibios_dac.h126
-rw-r--r--quantum/audio/driver_chibios_dac_additive.c335
-rw-r--r--quantum/audio/driver_chibios_dac_basic.c245
-rw-r--r--quantum/audio/driver_chibios_pwm.h40
-rw-r--r--quantum/audio/driver_chibios_pwm_hardware.c144
-rw-r--r--quantum/audio/driver_chibios_pwm_software.c164
-rw-r--r--quantum/audio/song_list.h8
10 files changed, 7 insertions, 1417 deletions
diff --git a/quantum/audio/audio.h b/quantum/audio/audio.h
index 56b9158a1..290d461f5 100644
--- a/quantum/audio/audio.h
+++ b/quantum/audio/audio.h
@@ -26,17 +26,12 @@
26 26
27#if defined(__AVR__) 27#if defined(__AVR__)
28# include <avr/io.h> 28# include <avr/io.h>
29# if defined(AUDIO_DRIVER_PWM)
30# include "driver_avr_pwm.h"
31# endif
32#endif 29#endif
33 30
34#if defined(PROTOCOL_CHIBIOS) 31#if defined(AUDIO_DRIVER_PWM)
35# if defined(AUDIO_DRIVER_PWM) 32# include "audio_pwm.h"
36# include "driver_chibios_pwm.h" 33#elif defined(AUDIO_DRIVER_DAC)
37# elif defined(AUDIO_DRIVER_DAC) 34# include "audio_dac.h"
38# include "driver_chibios_dac.h"
39# endif
40#endif 35#endif
41 36
42typedef union { 37typedef union {
diff --git a/quantum/audio/driver_avr_pwm.h b/quantum/audio/driver_avr_pwm.h
deleted file mode 100644
index d6eb3571d..000000000
--- a/quantum/audio/driver_avr_pwm.h
+++ /dev/null
@@ -1,17 +0,0 @@
1/* Copyright 2020 Jack Humbert
2 * Copyright 2020 JohSchneider
3 *
4 * This program is free software: you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation, either version 2 of the License, or
7 * (at your option) any later version.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program. If not, see <http://www.gnu.org/licenses/>.
16 */
17#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 df03a4558..000000000
--- a/quantum/audio/driver_avr_pwm_hardware.c
+++ /dev/null
@@ -1,332 +0,0 @@
1/* Copyright 2016 Jack Humbert
2 * Copyright 2020 JohSchneider
3 *
4 * This program is free software: you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation, either version 2 of the License, or
7 * (at your option) any later version.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program. If not, see <http://www.gnu.org/licenses/>.
16 */
17
18#if defined(__AVR__)
19# include <avr/pgmspace.h>
20# include <avr/interrupt.h>
21# include <avr/io.h>
22#endif
23
24#include "audio.h"
25
26extern bool playing_note;
27extern bool playing_melody;
28extern uint8_t note_timbre;
29
30#define CPU_PRESCALER 8
31
32/*
33 Audio Driver: PWM
34
35 drive up to two speakers through the AVR PWM hardware-peripheral, using timer1 and/or timer3 on Atmega32U4.
36
37 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
38 and an optional secondary channel_2 on either pin PB5, PB6 or PB7, with a PWM signal from timer1
39
40 alternatively, the PWM pins on PORTB can be used as only/primary speaker
41*/
42
43#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)
44# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under the AVR settings for available options."
45#endif
46
47#if (AUDIO_PIN == C4) || (AUDIO_PIN == C5) || (AUDIO_PIN == C6)
48# define AUDIO1_PIN_SET
49# define AUDIO1_TIMSKx TIMSK3
50# define AUDIO1_TCCRxA TCCR3A
51# define AUDIO1_TCCRxB TCCR3B
52# define AUDIO1_ICRx ICR3
53# define AUDIO1_WGMx0 WGM30
54# define AUDIO1_WGMx1 WGM31
55# define AUDIO1_WGMx2 WGM32
56# define AUDIO1_WGMx3 WGM33
57# define AUDIO1_CSx0 CS30
58# define AUDIO1_CSx1 CS31
59# define AUDIO1_CSx2 CS32
60
61# if (AUDIO_PIN == C6)
62# define AUDIO1_COMxy0 COM3A0
63# define AUDIO1_COMxy1 COM3A1
64# define AUDIO1_OCIExy OCIE3A
65# define AUDIO1_OCRxy OCR3A
66# define AUDIO1_PIN C6
67# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPA_vect
68# elif (AUDIO_PIN == C5)
69# define AUDIO1_COMxy0 COM3B0
70# define AUDIO1_COMxy1 COM3B1
71# define AUDIO1_OCIExy OCIE3B
72# define AUDIO1_OCRxy OCR3B
73# define AUDIO1_PIN C5
74# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPB_vect
75# elif (AUDIO_PIN == C4)
76# define AUDIO1_COMxy0 COM3C0
77# define AUDIO1_COMxy1 COM3C1
78# define AUDIO1_OCIExy OCIE3C
79# define AUDIO1_OCRxy OCR3C
80# define AUDIO1_PIN C4
81# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPC_vect
82# endif
83#endif
84
85#if defined(AUDIO_PIN) && defined(AUDIO_PIN_ALT) && (AUDIO_PIN == AUDIO_PIN_ALT)
86# error "Audio feature: AUDIO_PIN and AUDIO_PIN_ALT on the same pin makes no sense."
87#endif
88
89#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)))
90# error "Audio feature: PORTB as AUDIO_PIN and AUDIO_PIN_ALT at the same time is not supported."
91#endif
92
93#if defined(AUDIO_PIN_ALT) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7)
94# error "Audio feature: the pin selected as AUDIO_PIN_ALT is not supported."
95#endif
96
97#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)
98# define AUDIO2_PIN_SET
99# define AUDIO2_TIMSKx TIMSK1
100# define AUDIO2_TCCRxA TCCR1A
101# define AUDIO2_TCCRxB TCCR1B
102# define AUDIO2_ICRx ICR1
103# define AUDIO2_WGMx0 WGM10
104# define AUDIO2_WGMx1 WGM11
105# define AUDIO2_WGMx2 WGM12
106# define AUDIO2_WGMx3 WGM13
107# define AUDIO2_CSx0 CS10
108# define AUDIO2_CSx1 CS11
109# define AUDIO2_CSx2 CS12
110
111# if (AUDIO_PIN == B5) || (AUDIO_PIN_ALT == B5)
112# define AUDIO2_COMxy0 COM1A0
113# define AUDIO2_COMxy1 COM1A1
114# define AUDIO2_OCIExy OCIE1A
115# define AUDIO2_OCRxy OCR1A
116# define AUDIO2_PIN B5
117# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect
118# elif (AUDIO_PIN == B6) || (AUDIO_PIN_ALT == B6)
119# define AUDIO2_COMxy0 COM1B0
120# define AUDIO2_COMxy1 COM1B1
121# define AUDIO2_OCIExy OCIE1B
122# define AUDIO2_OCRxy OCR1B
123# define AUDIO2_PIN B6
124# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPB_vect
125# elif (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B7)
126# define AUDIO2_COMxy0 COM1C0
127# define AUDIO2_COMxy1 COM1C1
128# define AUDIO2_OCIExy OCIE1C
129# define AUDIO2_OCRxy OCR1C
130# define AUDIO2_PIN B7
131# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPC_vect
132# elif (AUDIO_PIN == D5) && defined(__AVR_ATmega32A__)
133# pragma message "Audio support for ATmega32A is experimental and can cause crashes."
134# undef AUDIO2_TIMSKx
135# define AUDIO2_TIMSKx TIMSK
136# define AUDIO2_COMxy0 COM1A0
137# define AUDIO2_COMxy1 COM1A1
138# define AUDIO2_OCIExy OCIE1A
139# define AUDIO2_OCRxy OCR1A
140# define AUDIO2_PIN D5
141# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect
142# endif
143#endif
144
145// C6 seems to be the assumed default by many existing keyboard - but sill warn the user
146#if !defined(AUDIO1_PIN_SET) && !defined(AUDIO2_PIN_SET)
147# 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... :-)"
148// TODO: make this an error - go through the breaking-change-process and change all keyboards to the new define
149#endif
150// -----------------------------------------------------------------------------
151
152#ifdef AUDIO1_PIN_SET
153static float channel_1_frequency = 0.0f;
154void channel_1_set_frequency(float freq) {
155 if (freq == 0.0f) // a pause/rest is a valid "note" with freq=0
156 {
157 // disable the output, but keep the pwm-ISR going (with the previous
158 // frequency) so the audio-state keeps getting updated
159 // Note: setting the duty-cycle 0 is not possible on non-inverting PWM mode - see the AVR data-sheet
160 AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0));
161 return;
162 } else {
163 AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); // enable output, PWM mode
164 }
165
166 channel_1_frequency = freq;
167
168 // set pwm period
169 AUDIO1_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER));
170 // and duty cycle
171 AUDIO1_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100);
172}
173
174void channel_1_start(void) {
175 // enable timer-counter ISR
176 AUDIO1_TIMSKx |= _BV(AUDIO1_OCIExy);
177 // enable timer-counter output
178 AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1);
179}
180
181void channel_1_stop(void) {
182 // disable timer-counter ISR
183 AUDIO1_TIMSKx &= ~_BV(AUDIO1_OCIExy);
184 // disable timer-counter output
185 AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0));
186}
187#endif
188
189#ifdef AUDIO2_PIN_SET
190static float channel_2_frequency = 0.0f;
191void channel_2_set_frequency(float freq) {
192 if (freq == 0.0f) {
193 AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0));
194 return;
195 } else {
196 AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1);
197 }
198
199 channel_2_frequency = freq;
200
201 AUDIO2_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER));
202 AUDIO2_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100);
203}
204
205float channel_2_get_frequency(void) { return channel_2_frequency; }
206
207void channel_2_start(void) {
208 AUDIO2_TIMSKx |= _BV(AUDIO2_OCIExy);
209 AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1);
210}
211
212void channel_2_stop(void) {
213 AUDIO2_TIMSKx &= ~_BV(AUDIO2_OCIExy);
214 AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0));
215}
216#endif
217
218void audio_driver_initialize() {
219#ifdef AUDIO1_PIN_SET
220 channel_1_stop();
221 setPinOutput(AUDIO1_PIN);
222#endif
223
224#ifdef AUDIO2_PIN_SET
225 channel_2_stop();
226 setPinOutput(AUDIO2_PIN);
227#endif
228
229 // TCCR3A / TCCR3B: Timer/Counter #3 Control Registers TCCR3A/TCCR3B, TCCR1A/TCCR1B
230 // Compare Output Mode (COM3An and COM1An) = 0b00 = Normal port operation
231 // OC3A -- PC6
232 // OC3B -- PC5
233 // OC3C -- PC4
234 // OC1A -- PB5
235 // OC1B -- PB6
236 // OC1C -- PB7
237
238 // Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14. Period = ICR3, Duty Cycle OCR3A)
239 // OCR3A - PC6
240 // OCR3B - PC5
241 // OCR3C - PC4
242 // OCR1A - PB5
243 // OCR1B - PB6
244 // OCR1C - PB7
245
246 // Clock Select (CS3n) = 0b010 = Clock / 8
247#ifdef AUDIO1_PIN_SET
248 // initialize timer-counter
249 AUDIO1_TCCRxA = (0 << AUDIO1_COMxy1) | (0 << AUDIO1_COMxy0) | (1 << AUDIO1_WGMx1) | (0 << AUDIO1_WGMx0);
250 AUDIO1_TCCRxB = (1 << AUDIO1_WGMx3) | (1 << AUDIO1_WGMx2) | (0 << AUDIO1_CSx2) | (1 << AUDIO1_CSx1) | (0 << AUDIO1_CSx0);
251#endif
252
253#ifdef AUDIO2_PIN_SET
254 AUDIO2_TCCRxA = (0 << AUDIO2_COMxy1) | (0 << AUDIO2_COMxy0) | (1 << AUDIO2_WGMx1) | (0 << AUDIO2_WGMx0);
255 AUDIO2_TCCRxB = (1 << AUDIO2_WGMx3) | (1 << AUDIO2_WGMx2) | (0 << AUDIO2_CSx2) | (1 << AUDIO2_CSx1) | (0 << AUDIO2_CSx0);
256#endif
257}
258
259void audio_driver_stop() {
260#ifdef AUDIO1_PIN_SET
261 channel_1_stop();
262#endif
263
264#ifdef AUDIO2_PIN_SET
265 channel_2_stop();
266#endif
267}
268
269void audio_driver_start(void) {
270#ifdef AUDIO1_PIN_SET
271 channel_1_start();
272 if (playing_note) {
273 channel_1_set_frequency(audio_get_processed_frequency(0));
274 }
275#endif
276
277#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET)
278 channel_2_start();
279 if (playing_note) {
280 channel_2_set_frequency(audio_get_processed_frequency(0));
281 }
282#endif
283}
284
285static volatile uint32_t isr_counter = 0;
286#ifdef AUDIO1_PIN_SET
287ISR(AUDIO1_TIMERx_COMPy_vect) {
288 isr_counter++;
289 if (isr_counter < channel_1_frequency / (CPU_PRESCALER * 8)) return;
290
291 isr_counter = 0;
292 bool state_changed = audio_update_state();
293
294 if (!playing_note && !playing_melody) {
295 channel_1_stop();
296# ifdef AUDIO2_PIN_SET
297 channel_2_stop();
298# endif
299 return;
300 }
301
302 if (state_changed) {
303 channel_1_set_frequency(audio_get_processed_frequency(0));
304# ifdef AUDIO2_PIN_SET
305 if (audio_get_number_of_active_tones() > 1) {
306 channel_2_set_frequency(audio_get_processed_frequency(1));
307 } else {
308 channel_2_stop();
309 }
310# endif
311 }
312}
313#endif
314
315#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET)
316ISR(AUDIO2_TIMERx_COMPy_vect) {
317 isr_counter++;
318 if (isr_counter < channel_2_frequency / (CPU_PRESCALER * 8)) return;
319
320 isr_counter = 0;
321 bool state_changed = audio_update_state();
322
323 if (!playing_note && !playing_melody) {
324 channel_2_stop();
325 return;
326 }
327
328 if (state_changed) {
329 channel_2_set_frequency(audio_get_processed_frequency(0));
330 }
331}
332#endif
diff --git a/quantum/audio/driver_chibios_dac.h b/quantum/audio/driver_chibios_dac.h
deleted file mode 100644
index 07cd622ea..000000000
--- a/quantum/audio/driver_chibios_dac.h
+++ /dev/null
@@ -1,126 +0,0 @@
1/* Copyright 2019 Jack Humbert
2 * Copyright 2020 JohSchneider
3 *
4 * This program is free software: you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation, either version 2 of the License, or
7 * (at your option) any later version.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program. If not, see <http://www.gnu.org/licenses/>.
16 */
17#pragma once
18
19#ifndef A4
20# define A4 PAL_LINE(GPIOA, 4)
21#endif
22#ifndef A5
23# define A5 PAL_LINE(GPIOA, 5)
24#endif
25
26/**
27 * Size of the dac_buffer arrays. All must be the same size.
28 */
29#define AUDIO_DAC_BUFFER_SIZE 256U
30
31/**
32 * Highest value allowed sample value.
33
34 * since the DAC is limited to 12 bit, the absolute max is 0xfff = 4095U;
35 * lower values adjust the peak-voltage aka volume down.
36 * adjusting this value has only an effect on a sample-buffer whose values are
37 * are NOT pregenerated - see square-wave
38 */
39#ifndef AUDIO_DAC_SAMPLE_MAX
40# define AUDIO_DAC_SAMPLE_MAX 4095U
41#endif
42
43#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)
44# define AUDIO_DAC_QUALITY_SANE_MINIMUM
45#endif
46
47/**
48 * These presets allow you to quickly switch between quality settings for
49 * the DAC. The sample rate and maximum number of simultaneous tones roughly
50 * has an inverse relationship - slightly higher sample rates may be possible.
51 *
52 * NOTE: a high sample-rate results in a higher cpu-load, which might lead to
53 * (audible) discontinuities and/or starve other processes of cpu-time
54 * (like RGB-led back-lighting, ...)
55 */
56#ifdef AUDIO_DAC_QUALITY_VERY_LOW
57# define AUDIO_DAC_SAMPLE_RATE 11025U
58# define AUDIO_MAX_SIMULTANEOUS_TONES 8
59#endif
60
61#ifdef AUDIO_DAC_QUALITY_LOW
62# define AUDIO_DAC_SAMPLE_RATE 22050U
63# define AUDIO_MAX_SIMULTANEOUS_TONES 4
64#endif
65
66#ifdef AUDIO_DAC_QUALITY_HIGH
67# define AUDIO_DAC_SAMPLE_RATE 44100U
68# define AUDIO_MAX_SIMULTANEOUS_TONES 2
69#endif
70
71#ifdef AUDIO_DAC_QUALITY_VERY_HIGH
72# define AUDIO_DAC_SAMPLE_RATE 88200U
73# define AUDIO_MAX_SIMULTANEOUS_TONES 1
74#endif
75
76#ifdef AUDIO_DAC_QUALITY_SANE_MINIMUM
77/* a sane-minimum config: with a trade-off between cpu-load and tone-range
78 *
79 * the (currently) highest defined note is NOTE_B8 with 7902Hz; if we now
80 * aim for an even even multiple of the buffer-size, we end up with:
81 * ( roundUptoPow2(highest note / AUDIO_DAC_BUFFER_SIZE) * nyquist-rate * AUDIO_DAC_BUFFER_SIZE)
82 * 7902/256 = 30.867 * 2 * 256 ~= 16384
83 * which works out (but the 'scope shows some sampling artifacts with lower harmonics :-P)
84 */
85# define AUDIO_DAC_SAMPLE_RATE 16384U
86# define AUDIO_MAX_SIMULTANEOUS_TONES 8
87#endif
88
89/**
90 * Effective bit-rate of the DAC. 44.1khz is the standard for most audio - any
91 * lower will sacrifice perceptible audio quality. Any higher will limit the
92 * number of simultaneous tones. In most situations, a tenth (1/10) of the
93 * sample rate is where notes become unbearable.
94 */
95#ifndef AUDIO_DAC_SAMPLE_RATE
96# define AUDIO_DAC_SAMPLE_RATE 44100U
97#endif
98
99/**
100 * The number of tones that can be played simultaneously. If too high a value
101 * is used here, the keyboard will freeze and glitch-out when that many tones
102 * are being played.
103 */
104#ifndef AUDIO_MAX_SIMULTANEOUS_TONES
105# define AUDIO_MAX_SIMULTANEOUS_TONES 2
106#endif
107
108/**
109 * The default value of the DAC when not playing anything. Certain hardware
110 * setups may require a high (AUDIO_DAC_SAMPLE_MAX) or low (0) value here.
111 * Since multiple added sine waves tend to oscillate around the midpoint,
112 * and possibly never/rarely reach either 0 of MAX, 1/2 MAX can be a
113 * reasonable default value.
114 */
115#ifndef AUDIO_DAC_OFF_VALUE
116# define AUDIO_DAC_OFF_VALUE AUDIO_DAC_SAMPLE_MAX / 2
117#endif
118
119#if AUDIO_DAC_OFF_VALUE > AUDIO_DAC_SAMPLE_MAX
120# error "AUDIO_DAC: OFF_VALUE may not be larger than SAMPLE_MAX"
121#endif
122
123/**
124 *user overridable sample generation/processing
125 */
126uint16_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 db304adb8..000000000
--- a/quantum/audio/driver_chibios_dac_additive.c
+++ /dev/null
@@ -1,335 +0,0 @@
1/* Copyright 2016-2019 Jack Humbert
2 * Copyright 2020 JohSchneider
3 *
4 * This program is free software: you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation, either version 2 of the License, or
7 * (at your option) any later version.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program. If not, see <http://www.gnu.org/licenses/>.
16 */
17
18#include "audio.h"
19#include <ch.h>
20#include <hal.h>
21
22/*
23 Audio Driver: DAC
24
25 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
26
27 it is also possible to have a custom sample-LUT by implementing/overriding 'dac_value_generate'
28
29 this driver allows for multiple simultaneous tones to be played through one single channel by doing additive wave-synthesis
30*/
31
32#if !defined(AUDIO_PIN)
33# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC additive)' for available options."
34#endif
35#if defined(AUDIO_PIN_ALT) && !defined(AUDIO_PIN_ALT_AS_NEGATIVE)
36# 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."
37#endif
38
39#if !defined(AUDIO_PIN_ALT)
40// no ALT pin defined is valid, but the c-ifs below need some value set
41# define AUDIO_PIN_ALT PAL_NOLINE
42#endif
43
44#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)
45# define AUDIO_DAC_SAMPLE_WAVEFORM_SINE
46#endif
47
48#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SINE
49/* one full sine wave over [0,2*pi], but shifted up one amplitude and left pi/4; for the samples to start at 0
50 */
51static const dacsample_t dac_buffer_sine[AUDIO_DAC_BUFFER_SIZE] = {
52 // 256 values, max 4095
53 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,
54 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};
55#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SINE
56#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
57static const dacsample_t dac_buffer_triangle[AUDIO_DAC_BUFFER_SIZE] = {
58 // 256 values, max 4095
59 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,
60 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};
61#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE
62#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
63static const dacsample_t dac_buffer_square[AUDIO_DAC_BUFFER_SIZE] = {
64 [0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = 0, // first and
65 [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX, // second half
66};
67#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE
68/*
69// four steps: 0, 1/3, 2/3 and 1
70static const dacsample_t dac_buffer_staircase[AUDIO_DAC_BUFFER_SIZE] = {
71 [0 ... AUDIO_DAC_BUFFER_SIZE/3 -1 ] = 0,
72 [AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE / 2 -1 ] = AUDIO_DAC_SAMPLE_MAX / 3,
73 [AUDIO_DAC_BUFFER_SIZE / 2 ... 3 * AUDIO_DAC_BUFFER_SIZE / 4 -1 ] = 2 * AUDIO_DAC_SAMPLE_MAX / 3,
74 [3 * AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE -1 ] = AUDIO_DAC_SAMPLE_MAX,
75}
76*/
77#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
78static 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,
79 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};
80#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID
81
82static dacsample_t dac_buffer_empty[AUDIO_DAC_BUFFER_SIZE] = {AUDIO_DAC_OFF_VALUE};
83
84/* keep track of the sample position for for each frequency */
85static float dac_if[AUDIO_MAX_SIMULTANEOUS_TONES] = {0.0};
86
87static float active_tones_snapshot[AUDIO_MAX_SIMULTANEOUS_TONES] = {0, 0};
88static uint8_t active_tones_snapshot_length = 0;
89
90typedef enum {
91 OUTPUT_SHOULD_START,
92 OUTPUT_RUN_NORMALLY,
93 // path 1: wait for zero, then change/update active tones
94 OUTPUT_TONES_CHANGED,
95 OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE,
96 // path 2: hardware should stop, wait for zero then turn output off = stop the timer
97 OUTPUT_SHOULD_STOP,
98 OUTPUT_REACHED_ZERO_BEFORE_OFF,
99 OUTPUT_OFF,
100 OUTPUT_OFF_1,
101 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
102 number_of_output_states
103} output_states_t;
104output_states_t state = OUTPUT_OFF_2;
105
106/**
107 * Generation of the waveform being passed to the callback. Declared weak so users
108 * can override it with their own wave-forms/noises.
109 */
110__attribute__((weak)) uint16_t dac_value_generate(void) {
111 // DAC is running/asking for values but snapshot length is zero -> must be playing a pause
112 if (active_tones_snapshot_length == 0) {
113 return AUDIO_DAC_OFF_VALUE;
114 }
115
116 /* doing additive wave synthesis over all currently playing tones = adding up
117 * sine-wave-samples for each frequency, scaled by the number of active tones
118 */
119 uint16_t value = 0;
120 float frequency = 0.0f;
121
122 for (uint8_t i = 0; i < active_tones_snapshot_length; i++) {
123 /* Note: a user implementation does not have to rely on the active_tones_snapshot, but
124 * could directly query the active frequencies through audio_get_processed_frequency */
125 frequency = active_tones_snapshot[i];
126
127 dac_if[i] = dac_if[i] + ((frequency * AUDIO_DAC_BUFFER_SIZE) / AUDIO_DAC_SAMPLE_RATE) * 2 / 3;
128 /*Note: the 2/3 are necessary to get the correct frequencies on the
129 * DAC output (as measured with an oscilloscope), since the gpt
130 * timer runs with 3*AUDIO_DAC_SAMPLE_RATE; and the DAC callback
131 * is called twice per conversion.*/
132
133 dac_if[i] = fmod(dac_if[i], AUDIO_DAC_BUFFER_SIZE);
134
135 // Wavetable generation/lookup
136 uint16_t dac_i = (uint16_t)dac_if[i];
137
138#if defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE)
139 value += dac_buffer_sine[dac_i] / active_tones_snapshot_length;
140#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE)
141 value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length;
142#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID)
143 value += dac_buffer_trapezoid[dac_i] / active_tones_snapshot_length;
144#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE)
145 value += dac_buffer_square[dac_i] / active_tones_snapshot_length;
146#endif
147 /*
148 // SINE
149 value += dac_buffer_sine[dac_i] / active_tones_snapshot_length / 3;
150 // TRIANGLE
151 value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length / 3;
152 // SQUARE
153 value += dac_buffer_square[dac_i] / active_tones_snapshot_length / 3;
154 //NOTE: combination of these three wave-forms is more exemplary - and doesn't sound particularly good :-P
155 */
156
157 // STAIRS (mostly usefully as test-pattern)
158 // value_avg = dac_buffer_staircase[dac_i] / active_tones_snapshot_length;
159 }
160
161 return value;
162}
163
164/**
165 * DAC streaming callback. Does all of the main computing for playing songs.
166 *
167 * Note: chibios calls this CB twice: during the 'half buffer event', and the 'full buffer event'.
168 */
169static void dac_end(DACDriver *dacp) {
170 dacsample_t *sample_p = (dacp)->samples;
171
172 // work on the other half of the buffer
173 if (dacIsBufferComplete(dacp)) {
174 sample_p += AUDIO_DAC_BUFFER_SIZE / 2; // 'half_index'
175 }
176
177 for (uint8_t s = 0; s < AUDIO_DAC_BUFFER_SIZE / 2; s++) {
178 if (OUTPUT_OFF <= state) {
179 sample_p[s] = AUDIO_DAC_OFF_VALUE;
180 continue;
181 } else {
182 sample_p[s] = dac_value_generate();
183 }
184
185 /* zero crossing (or approach, whereas zero == DAC_OFF_VALUE, which can be configured to anything from 0 to DAC_SAMPLE_MAX)
186 * ============================*=*========================== AUDIO_DAC_SAMPLE_MAX
187 * * *
188 * * *
189 * ---------------------------------------------------------
190 * * * } AUDIO_DAC_SAMPLE_MAX/100
191 * --------------------------------------------------------- AUDIO_DAC_OFF_VALUE
192 * * * } AUDIO_DAC_SAMPLE_MAX/100
193 * ---------------------------------------------------------
194 * *
195 * * *
196 * * *
197 * =====*=*================================================= 0x0
198 */
199 if (((sample_p[s] + (AUDIO_DAC_SAMPLE_MAX / 100)) > AUDIO_DAC_OFF_VALUE) && // value approaches from below
200 (sample_p[s] < (AUDIO_DAC_OFF_VALUE + (AUDIO_DAC_SAMPLE_MAX / 100))) // or above
201 ) {
202 if ((OUTPUT_SHOULD_START == state) && (active_tones_snapshot_length > 0)) {
203 state = OUTPUT_RUN_NORMALLY;
204 } else if (OUTPUT_TONES_CHANGED == state) {
205 state = OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE;
206 } else if (OUTPUT_SHOULD_STOP == state) {
207 state = OUTPUT_REACHED_ZERO_BEFORE_OFF;
208 }
209 }
210
211 // still 'ramping up', reset the output to OFF_VALUE until the generated values reach that value, to do a smooth handover
212 if (OUTPUT_SHOULD_START == state) {
213 sample_p[s] = AUDIO_DAC_OFF_VALUE;
214 }
215
216 if ((OUTPUT_SHOULD_START == state) || (OUTPUT_REACHED_ZERO_BEFORE_OFF == state) || (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state)) {
217 uint8_t active_tones = MIN(AUDIO_MAX_SIMULTANEOUS_TONES, audio_get_number_of_active_tones());
218 active_tones_snapshot_length = 0;
219 // update the snapshot - once, and only on occasion that something changed;
220 // -> saves cpu cycles (?)
221 for (uint8_t i = 0; i < active_tones; i++) {
222 float freq = audio_get_processed_frequency(i);
223 if (freq > 0) { // disregard 'rest' notes, with valid frequency 0.0f; which would only lower the resulting waveform volume during the additive synthesis step
224 active_tones_snapshot[active_tones_snapshot_length++] = freq;
225 }
226 }
227
228 if ((0 == active_tones_snapshot_length) && (OUTPUT_REACHED_ZERO_BEFORE_OFF == state)) {
229 state = OUTPUT_OFF;
230 }
231 if (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state) {
232 state = OUTPUT_RUN_NORMALLY;
233 }
234 }
235 }
236
237 // update audio internal state (note position, current_note, ...)
238 if (audio_update_state()) {
239 if (OUTPUT_SHOULD_STOP != state) {
240 state = OUTPUT_TONES_CHANGED;
241 }
242 }
243
244 if (OUTPUT_OFF <= state) {
245 if (OUTPUT_OFF_2 == state) {
246 // stopping timer6 = stopping the DAC at whatever value it is currently pushing to the output = AUDIO_DAC_OFF_VALUE
247 gptStopTimer(&GPTD6);
248 } else {
249 state++;
250 }
251 }
252}
253
254static void dac_error(DACDriver *dacp, dacerror_t err) {
255 (void)dacp;
256 (void)err;
257
258 chSysHalt("DAC failure. halp");
259}
260
261static const GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE * 3,
262 .callback = NULL,
263 .cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
264 .dier = 0U};
265
266static const DACConfig dac_conf = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
267
268/**
269 * @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
270 * on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
271 * to be a third of what we expect.
272 *
273 * Here are all the values for DAC_TRG (TSEL in the ref manual)
274 * TIM15_TRGO 0b011
275 * TIM2_TRGO 0b100
276 * TIM3_TRGO 0b001
277 * TIM6_TRGO 0b000
278 * TIM7_TRGO 0b010
279 * EXTI9 0b110
280 * SWTRIG 0b111
281 */
282static const DACConversionGroup dac_conv_cfg = {.num_channels = 1U, .end_cb = dac_end, .error_cb = dac_error, .trigger = DAC_TRG(0b000)};
283
284void audio_driver_initialize() {
285 if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
286 palSetLineMode(A4, PAL_MODE_INPUT_ANALOG);
287 dacStart(&DACD1, &dac_conf);
288 }
289 if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
290 palSetLineMode(A5, PAL_MODE_INPUT_ANALOG);
291 dacStart(&DACD2, &dac_conf);
292 }
293
294 /* enable the output buffer, to directly drive external loads with no additional circuitry
295 *
296 * see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
297 * Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
298 * Note: enabling the output buffer imparts an additional dc-offset of a couple mV
299 *
300 * this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
301 * (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
302 */
303 DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
304 DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
305
306 if (AUDIO_PIN == A4) {
307 dacStartConversion(&DACD1, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
308 } else if (AUDIO_PIN == A5) {
309 dacStartConversion(&DACD2, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
310 }
311
312 // no inverted/out-of-phase waveform (yet?), only pulling AUDIO_PIN_ALT to AUDIO_DAC_OFF_VALUE
313#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
314 if (AUDIO_PIN_ALT == A4) {
315 dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
316 } else if (AUDIO_PIN_ALT == A5) {
317 dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
318 }
319#endif
320
321 gptStart(&GPTD6, &gpt6cfg1);
322}
323
324void audio_driver_stop(void) { state = OUTPUT_SHOULD_STOP; }
325
326void audio_driver_start(void) {
327 gptStartContinuous(&GPTD6, 2U);
328
329 for (uint8_t i = 0; i < AUDIO_MAX_SIMULTANEOUS_TONES; i++) {
330 dac_if[i] = 0.0f;
331 active_tones_snapshot[i] = 0.0f;
332 }
333 active_tones_snapshot_length = 0;
334 state = OUTPUT_SHOULD_START;
335}
diff --git a/quantum/audio/driver_chibios_dac_basic.c b/quantum/audio/driver_chibios_dac_basic.c
deleted file mode 100644
index fac651350..000000000
--- a/quantum/audio/driver_chibios_dac_basic.c
+++ /dev/null
@@ -1,245 +0,0 @@
1/* Copyright 2016-2020 Jack Humbert
2 * Copyright 2020 JohSchneider
3 *
4 * This program is free software: you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation, either version 2 of the License, or
7 * (at your option) any later version.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program. If not, see <http://www.gnu.org/licenses/>.
16 */
17
18#include "audio.h"
19#include "ch.h"
20#include "hal.h"
21
22/*
23 Audio Driver: DAC
24
25 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
26
27 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
28 OR
29 one speaker wired to A4+A5 with the AUDIO_PIN_ALT_AS_NEGATIVE define set - see docs/feature_audio
30
31*/
32
33#if !defined(AUDIO_PIN)
34# pragma message "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC basic)' for available options."
35// 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
36# define AUDIO_PIN A5
37#endif
38// check configuration for ONE speaker, connected to both DAC pins
39#if defined(AUDIO_PIN_ALT_AS_NEGATIVE) && !defined(AUDIO_PIN_ALT)
40# error "Audio feature: AUDIO_PIN_ALT_AS_NEGATIVE set, but no pin configured as AUDIO_PIN_ALT"
41#endif
42
43#ifndef AUDIO_PIN_ALT
44// no ALT pin defined is valid, but the c-ifs below need some value set
45# define AUDIO_PIN_ALT -1
46#endif
47
48#if !defined(AUDIO_STATE_TIMER)
49# define AUDIO_STATE_TIMER GPTD8
50#endif
51
52// square-wave
53static const dacsample_t dac_buffer_1[AUDIO_DAC_BUFFER_SIZE] = {
54 // First half is max, second half is 0
55 [0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = AUDIO_DAC_SAMPLE_MAX,
56 [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = 0,
57};
58
59// square-wave
60static const dacsample_t dac_buffer_2[AUDIO_DAC_BUFFER_SIZE] = {
61 // opposite of dac_buffer above
62 [0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = 0,
63 [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX,
64};
65
66GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
67 .callback = NULL,
68 .cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
69 .dier = 0U};
70GPTConfig gpt7cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
71 .callback = NULL,
72 .cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
73 .dier = 0U};
74
75static void gpt_audio_state_cb(GPTDriver *gptp);
76GPTConfig gptStateUpdateCfg = {.frequency = 10,
77 .callback = gpt_audio_state_cb,
78 .cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */
79 .dier = 0U};
80
81static const DACConfig dac_conf_ch1 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
82static const DACConfig dac_conf_ch2 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
83
84/**
85 * @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
86 * on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
87 * to be a third of what we expect.
88 *
89 * Here are all the values for DAC_TRG (TSEL in the ref manual)
90 * TIM15_TRGO 0b011
91 * TIM2_TRGO 0b100
92 * TIM3_TRGO 0b001
93 * TIM6_TRGO 0b000
94 * TIM7_TRGO 0b010
95 * EXTI9 0b110
96 * SWTRIG 0b111
97 */
98static const DACConversionGroup dac_conv_grp_ch1 = {.num_channels = 1U, .trigger = DAC_TRG(0b000)};
99static const DACConversionGroup dac_conv_grp_ch2 = {.num_channels = 1U, .trigger = DAC_TRG(0b010)};
100
101void channel_1_start(void) {
102 gptStart(&GPTD6, &gpt6cfg1);
103 gptStartContinuous(&GPTD6, 2U);
104 palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
105}
106
107void channel_1_stop(void) {
108 gptStopTimer(&GPTD6);
109 palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL);
110 palSetPad(GPIOA, 4);
111}
112
113static float channel_1_frequency = 0.0f;
114void channel_1_set_frequency(float freq) {
115 channel_1_frequency = freq;
116
117 channel_1_stop();
118 if (freq <= 0.0) // a pause/rest has freq=0
119 return;
120
121 gpt6cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
122 channel_1_start();
123}
124float channel_1_get_frequency(void) { return channel_1_frequency; }
125
126void channel_2_start(void) {
127 gptStart(&GPTD7, &gpt7cfg1);
128 gptStartContinuous(&GPTD7, 2U);
129 palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
130}
131
132void channel_2_stop(void) {
133 gptStopTimer(&GPTD7);
134 palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL);
135 palSetPad(GPIOA, 5);
136}
137
138static float channel_2_frequency = 0.0f;
139void channel_2_set_frequency(float freq) {
140 channel_2_frequency = freq;
141
142 channel_2_stop();
143 if (freq <= 0.0) // a pause/rest has freq=0
144 return;
145
146 gpt7cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
147 channel_2_start();
148}
149float channel_2_get_frequency(void) { return channel_2_frequency; }
150
151static void gpt_audio_state_cb(GPTDriver *gptp) {
152 if (audio_update_state()) {
153#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
154 // one piezo/speaker connected to both audio pins, the generated square-waves are inverted
155 channel_1_set_frequency(audio_get_processed_frequency(0));
156 channel_2_set_frequency(audio_get_processed_frequency(0));
157
158#else // two separate audio outputs/speakers
159 // primary speaker on A4, optional secondary on A5
160 if (AUDIO_PIN == A4) {
161 channel_1_set_frequency(audio_get_processed_frequency(0));
162 if (AUDIO_PIN_ALT == A5) {
163 if (audio_get_number_of_active_tones() > 1) {
164 channel_2_set_frequency(audio_get_processed_frequency(1));
165 } else {
166 channel_2_stop();
167 }
168 }
169 }
170
171 // primary speaker on A5, optional secondary on A4
172 if (AUDIO_PIN == A5) {
173 channel_2_set_frequency(audio_get_processed_frequency(0));
174 if (AUDIO_PIN_ALT == A4) {
175 if (audio_get_number_of_active_tones() > 1) {
176 channel_1_set_frequency(audio_get_processed_frequency(1));
177 } else {
178 channel_1_stop();
179 }
180 }
181 }
182#endif
183 }
184}
185
186void audio_driver_initialize() {
187 if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
188 palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
189 dacStart(&DACD1, &dac_conf_ch1);
190
191 // initial setup of the dac-triggering timer is still required, even
192 // though it gets reconfigured and restarted later on
193 gptStart(&GPTD6, &gpt6cfg1);
194 }
195
196 if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
197 palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
198 dacStart(&DACD2, &dac_conf_ch2);
199
200 gptStart(&GPTD7, &gpt7cfg1);
201 }
202
203 /* enable the output buffer, to directly drive external loads with no additional circuitry
204 *
205 * see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
206 * Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
207 * Note: enabling the output buffer imparts an additional dc-offset of a couple mV
208 *
209 * this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
210 * (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
211 */
212 DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
213 DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
214
215 // start state-updater
216 gptStart(&AUDIO_STATE_TIMER, &gptStateUpdateCfg);
217}
218
219void audio_driver_stop(void) {
220 if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
221 gptStopTimer(&GPTD6);
222
223 // stop the ongoing conversion and put the output in a known state
224 dacStopConversion(&DACD1);
225 dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
226 }
227
228 if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
229 gptStopTimer(&GPTD7);
230
231 dacStopConversion(&DACD2);
232 dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
233 }
234 gptStopTimer(&AUDIO_STATE_TIMER);
235}
236
237void audio_driver_start(void) {
238 if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
239 dacStartConversion(&DACD1, &dac_conv_grp_ch1, (dacsample_t *)dac_buffer_1, AUDIO_DAC_BUFFER_SIZE);
240 }
241 if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
242 dacStartConversion(&DACD2, &dac_conv_grp_ch2, (dacsample_t *)dac_buffer_2, AUDIO_DAC_BUFFER_SIZE);
243 }
244 gptStartContinuous(&AUDIO_STATE_TIMER, 2U);
245}
diff --git a/quantum/audio/driver_chibios_pwm.h b/quantum/audio/driver_chibios_pwm.h
deleted file mode 100644
index 86cab916e..000000000
--- a/quantum/audio/driver_chibios_pwm.h
+++ /dev/null
@@ -1,40 +0,0 @@
1/* Copyright 2020 Jack Humbert
2 * Copyright 2020 JohSchneider
3 *
4 * This program is free software: you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation, either version 2 of the License, or
7 * (at your option) any later version.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program. If not, see <http://www.gnu.org/licenses/>.
16 */
17#pragma once
18
19#if !defined(AUDIO_PWM_DRIVER)
20// NOTE: Timer2 seems to be used otherwise in QMK, otherwise we could default to A5 (= TIM2_CH1, with PWMD2 and alternate-function(1))
21# define AUDIO_PWM_DRIVER PWMD1
22#endif
23
24#if !defined(AUDIO_PWM_CHANNEL)
25// NOTE: sticking to the STM data-sheet numbering: TIMxCH1 to TIMxCH4
26// default: STM32F303CC PA8+TIM1_CH1 -> 1
27# define AUDIO_PWM_CHANNEL 1
28#endif
29
30#if !defined(AUDIO_PWM_PAL_MODE)
31// pin-alternate function: see the data-sheet for which pin needs what AF to connect to TIMx_CHy
32// default: STM32F303CC PA8+TIM1_CH1 -> 6
33# define AUDIO_PWM_PAL_MODE 6
34#endif
35
36#if !defined(AUDIO_STATE_TIMER)
37// timer used to trigger updates in the audio-system, configured/enabled in chibios mcuconf.
38// 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)
39# define AUDIO_STATE_TIMER GPTD6
40#endif
diff --git a/quantum/audio/driver_chibios_pwm_hardware.c b/quantum/audio/driver_chibios_pwm_hardware.c
deleted file mode 100644
index 3c7d89b29..000000000
--- a/quantum/audio/driver_chibios_pwm_hardware.c
+++ /dev/null
@@ -1,144 +0,0 @@
1/* Copyright 2020 Jack Humbert
2 * Copyright 2020 JohSchneider
3 *
4 * This program is free software: you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation, either version 2 of the License, or
7 * (at your option) any later version.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program. If not, see <http://www.gnu.org/licenses/>.
16 */
17
18/*
19Audio Driver: PWM
20
21the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
22
23this driver uses the chibios-PWM system to produce a square-wave on specific output pins that are connected to the PWM hardware.
24The 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.
25
26 */
27
28#include "audio.h"
29#include "ch.h"
30#include "hal.h"
31
32#if !defined(AUDIO_PIN)
33# error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
34#endif
35
36extern bool playing_note;
37extern bool playing_melody;
38extern uint8_t note_timbre;
39
40static PWMConfig pwmCFG = {
41 .frequency = 100000, /* PWM clock frequency */
42 // 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
43 .period = 2, /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
44 .callback = NULL, /* no callback, the hardware directly toggles the pin */
45 .channels =
46 {
47#if AUDIO_PWM_CHANNEL == 4
48 {PWM_OUTPUT_DISABLED, NULL}, /* channel 0 -> TIMx_CH1 */
49 {PWM_OUTPUT_DISABLED, NULL}, /* channel 1 -> TIMx_CH2 */
50 {PWM_OUTPUT_DISABLED, NULL}, /* channel 2 -> TIMx_CH3 */
51 {PWM_OUTPUT_ACTIVE_HIGH, NULL} /* channel 3 -> TIMx_CH4 */
52#elif AUDIO_PWM_CHANNEL == 3
53 {PWM_OUTPUT_DISABLED, NULL},
54 {PWM_OUTPUT_DISABLED, NULL},
55 {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH3 */
56 {PWM_OUTPUT_DISABLED, NULL}
57#elif AUDIO_PWM_CHANNEL == 2
58 {PWM_OUTPUT_DISABLED, NULL},
59 {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH2 */
60 {PWM_OUTPUT_DISABLED, NULL},
61 {PWM_OUTPUT_DISABLED, NULL}
62#else /*fallback to CH1 */
63 {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH1 */
64 {PWM_OUTPUT_DISABLED, NULL},
65 {PWM_OUTPUT_DISABLED, NULL},
66 {PWM_OUTPUT_DISABLED, NULL}
67#endif
68 },
69};
70
71static float channel_1_frequency = 0.0f;
72void channel_1_set_frequency(float freq) {
73 channel_1_frequency = freq;
74
75 if (freq <= 0.0) // a pause/rest has freq=0
76 return;
77
78 pwmcnt_t period = (pwmCFG.frequency / freq);
79 pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
80 pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
81 // adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
82 PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
83}
84
85float channel_1_get_frequency(void) { return channel_1_frequency; }
86
87void channel_1_start(void) {
88 pwmStop(&AUDIO_PWM_DRIVER);
89 pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
90}
91
92void channel_1_stop(void) { pwmStop(&AUDIO_PWM_DRIVER); }
93
94static void gpt_callback(GPTDriver *gptp);
95GPTConfig gptCFG = {
96 /* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
97 the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
98 the tempo (which might vary!) is in bpm (beats per minute)
99 therefore: if the timer ticks away at .frequency = (60*64)Hz,
100 and the .interval counts from 64 downwards - audio_update_state is
101 called just often enough to not miss any notes
102 */
103 .frequency = 60 * 64,
104 .callback = gpt_callback,
105};
106
107void audio_driver_initialize(void) {
108 pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
109
110 // connect the AUDIO_PIN to the PWM hardware
111#if defined(USE_GPIOV1) // STM32F103C8
112 palSetLineMode(AUDIO_PIN, PAL_MODE_STM32_ALTERNATE_PUSHPULL);
113#else // GPIOv2 (or GPIOv3 for f4xx, which is the same/compatible at this command)
114 palSetLineMode(AUDIO_PIN, PAL_STM32_MODE_ALTERNATE | PAL_STM32_ALTERNATE(AUDIO_PWM_PAL_MODE));
115#endif
116
117 gptStart(&AUDIO_STATE_TIMER, &gptCFG);
118}
119
120void audio_driver_start(void) {
121 channel_1_stop();
122 channel_1_start();
123
124 if (playing_note || playing_melody) {
125 gptStartContinuous(&AUDIO_STATE_TIMER, 64);
126 }
127}
128
129void audio_driver_stop(void) {
130 channel_1_stop();
131 gptStopTimer(&AUDIO_STATE_TIMER);
132}
133
134/* a regular timer task, that checks the note to be currently played
135 * and updates the pwm to output that frequency
136 */
137static void gpt_callback(GPTDriver *gptp) {
138 float freq; // TODO: freq_alt
139
140 if (audio_update_state()) {
141 freq = audio_get_processed_frequency(0); // freq_alt would be index=1
142 channel_1_set_frequency(freq);
143 }
144}
diff --git a/quantum/audio/driver_chibios_pwm_software.c b/quantum/audio/driver_chibios_pwm_software.c
deleted file mode 100644
index 15c3e98b6..000000000
--- a/quantum/audio/driver_chibios_pwm_software.c
+++ /dev/null
@@ -1,164 +0,0 @@
1/* Copyright 2020 Jack Humbert
2 * Copyright 2020 JohSchneider
3 *
4 * This program is free software: you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation, either version 2 of the License, or
7 * (at your option) any later version.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program. If not, see <http://www.gnu.org/licenses/>.
16 */
17
18/*
19Audio Driver: PWM
20
21the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
22
23this driver uses the chibios-PWM system to produce a square-wave on any given output pin in software
24- a pwm callback is used to set/clear the configured pin.
25
26 */
27#include "audio.h"
28#include "ch.h"
29#include "hal.h"
30
31#if !defined(AUDIO_PIN)
32# error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
33#endif
34extern bool playing_note;
35extern bool playing_melody;
36extern uint8_t note_timbre;
37
38static void pwm_audio_period_callback(PWMDriver *pwmp);
39static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp);
40
41static PWMConfig pwmCFG = {
42 .frequency = 100000, /* PWM clock frequency */
43 // 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
44 .period = 2, /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
45 .callback = pwm_audio_period_callback,
46 .channels =
47 {
48 // software-PWM just needs another callback on any channel
49 {PWM_OUTPUT_ACTIVE_HIGH, pwm_audio_channel_interrupt_callback}, /* channel 0 -> TIMx_CH1 */
50 {PWM_OUTPUT_DISABLED, NULL}, /* channel 1 -> TIMx_CH2 */
51 {PWM_OUTPUT_DISABLED, NULL}, /* channel 2 -> TIMx_CH3 */
52 {PWM_OUTPUT_DISABLED, NULL} /* channel 3 -> TIMx_CH4 */
53 },
54};
55
56static float channel_1_frequency = 0.0f;
57void channel_1_set_frequency(float freq) {
58 channel_1_frequency = freq;
59
60 if (freq <= 0.0) // a pause/rest has freq=0
61 return;
62
63 pwmcnt_t period = (pwmCFG.frequency / freq);
64 pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
65
66 pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
67 // adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
68 PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
69}
70
71float channel_1_get_frequency(void) { return channel_1_frequency; }
72
73void channel_1_start(void) {
74 pwmStop(&AUDIO_PWM_DRIVER);
75 pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
76
77 pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER);
78 pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
79}
80
81void channel_1_stop(void) {
82 pwmStop(&AUDIO_PWM_DRIVER);
83
84 palClearLine(AUDIO_PIN); // leave the line low, after last note was played
85
86#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
87 palClearLine(AUDIO_PIN_ALT); // leave the line low, after last note was played
88#endif
89}
90
91// generate a PWM signal on any pin, not necessarily the one connected to the timer
92static void pwm_audio_period_callback(PWMDriver *pwmp) {
93 (void)pwmp;
94 palClearLine(AUDIO_PIN);
95
96#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
97 palSetLine(AUDIO_PIN_ALT);
98#endif
99}
100static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp) {
101 (void)pwmp;
102 if (channel_1_frequency > 0) {
103 palSetLine(AUDIO_PIN); // generate a PWM signal on any pin, not necessarily the one connected to the timer
104#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
105 palClearLine(AUDIO_PIN_ALT);
106#endif
107 }
108}
109
110static void gpt_callback(GPTDriver *gptp);
111GPTConfig gptCFG = {
112 /* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
113 the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
114 the tempo (which might vary!) is in bpm (beats per minute)
115 therefore: if the timer ticks away at .frequency = (60*64)Hz,
116 and the .interval counts from 64 downwards - audio_update_state is
117 called just often enough to not miss anything
118 */
119 .frequency = 60 * 64,
120 .callback = gpt_callback,
121};
122
123void audio_driver_initialize(void) {
124 pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
125
126 palSetLineMode(AUDIO_PIN, PAL_MODE_OUTPUT_PUSHPULL);
127 palClearLine(AUDIO_PIN);
128
129#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
130 palSetLineMode(AUDIO_PIN_ALT, PAL_MODE_OUTPUT_PUSHPULL);
131 palClearLine(AUDIO_PIN_ALT);
132#endif
133
134 pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER); // enable pwm callbacks
135 pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
136
137 gptStart(&AUDIO_STATE_TIMER, &gptCFG);
138}
139
140void audio_driver_start(void) {
141 channel_1_stop();
142 channel_1_start();
143
144 if (playing_note || playing_melody) {
145 gptStartContinuous(&AUDIO_STATE_TIMER, 64);
146 }
147}
148
149void audio_driver_stop(void) {
150 channel_1_stop();
151 gptStopTimer(&AUDIO_STATE_TIMER);
152}
153
154/* a regular timer task, that checks the note to be currently played
155 * and updates the pwm to output that frequency
156 */
157static void gpt_callback(GPTDriver *gptp) {
158 float freq; // TODO: freq_alt
159
160 if (audio_update_state()) {
161 freq = audio_get_processed_frequency(0); // freq_alt would be index=1
162 channel_1_set_frequency(freq);
163 }
164}
diff --git a/quantum/audio/song_list.h b/quantum/audio/song_list.h
index b54b397e1..8e80a016a 100644
--- a/quantum/audio/song_list.h
+++ b/quantum/audio/song_list.h
@@ -20,11 +20,9 @@
20 20
21#include "musical_notes.h" 21#include "musical_notes.h"
22 22
23#if __GNUC__ > 5 // don't use for older gcc compilers since check isn't supported. 23#if __has_include("user_song_list.h")
24# if __has_include("user_song_list.h") 24# include "user_song_list.h"
25# include "user_song_list.h" 25#endif // if file exists
26# endif // if file exists
27#endif // __GNUC__
28 26
29#define NO_SOUND 27#define NO_SOUND
30 28
generated by cgit v1.2.3 (git 2.25.1) at 2025-07-05 21:01:18 +0000