diff options
Diffstat (limited to 'quantum/audio')
| -rw-r--r-- | quantum/audio/audio.h | 13 | ||||
| -rw-r--r-- | quantum/audio/driver_avr_pwm.h | 17 | ||||
| -rw-r--r-- | quantum/audio/driver_avr_pwm_hardware.c | 332 | ||||
| -rw-r--r-- | quantum/audio/driver_chibios_dac.h | 126 | ||||
| -rw-r--r-- | quantum/audio/driver_chibios_dac_additive.c | 335 | ||||
| -rw-r--r-- | quantum/audio/driver_chibios_dac_basic.c | 245 | ||||
| -rw-r--r-- | quantum/audio/driver_chibios_pwm.h | 40 | ||||
| -rw-r--r-- | quantum/audio/driver_chibios_pwm_hardware.c | 144 | ||||
| -rw-r--r-- | quantum/audio/driver_chibios_pwm_software.c | 164 | ||||
| -rw-r--r-- | quantum/audio/song_list.h | 8 |
10 files changed, 7 insertions, 1417 deletions
diff --git a/quantum/audio/audio.h b/quantum/audio/audio.h index 56b9158a1a..290d461f5a 100644 --- a/quantum/audio/audio.h +++ b/quantum/audio/audio.h @@ -26,17 +26,12 @@ #if defined(__AVR__) # include <avr/io.h> -# if defined(AUDIO_DRIVER_PWM) -# include "driver_avr_pwm.h" -# endif #endif -#if defined(PROTOCOL_CHIBIOS) -# if defined(AUDIO_DRIVER_PWM) -# include "driver_chibios_pwm.h" -# elif defined(AUDIO_DRIVER_DAC) -# include "driver_chibios_dac.h" -# endif +#if defined(AUDIO_DRIVER_PWM) +# include "audio_pwm.h" +#elif defined(AUDIO_DRIVER_DAC) +# include "audio_dac.h" #endif typedef union { diff --git a/quantum/audio/driver_avr_pwm.h b/quantum/audio/driver_avr_pwm.h deleted file mode 100644 index d6eb3571da..0000000000 --- a/quantum/audio/driver_avr_pwm.h +++ /dev/null @@ -1,17 +0,0 @@ -/* Copyright 2020 Jack Humbert - * Copyright 2020 JohSchneider - * - * This program is free software: you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation, either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses/>. - */ -#pragma once diff --git a/quantum/audio/driver_avr_pwm_hardware.c b/quantum/audio/driver_avr_pwm_hardware.c deleted file mode 100644 index df03a4558c..0000000000 --- a/quantum/audio/driver_avr_pwm_hardware.c +++ /dev/null @@ -1,332 +0,0 @@ -/* Copyright 2016 Jack Humbert - * Copyright 2020 JohSchneider - * - * This program is free software: you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation, either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses/>. - */ - -#if defined(__AVR__) -# include <avr/pgmspace.h> -# include <avr/interrupt.h> -# include <avr/io.h> -#endif - -#include "audio.h" - -extern bool playing_note; -extern bool playing_melody; -extern uint8_t note_timbre; - -#define CPU_PRESCALER 8 - -/* - Audio Driver: PWM - - drive up to two speakers through the AVR PWM hardware-peripheral, using timer1 and/or timer3 on Atmega32U4. - - the primary channel_1 can be connected to either pin PC4 PC5 or PC6 (the later being used by most AVR based keyboards) with a PMW signal generated by timer3 - and an optional secondary channel_2 on either pin PB5, PB6 or PB7, with a PWM signal from timer1 - - alternatively, the PWM pins on PORTB can be used as only/primary speaker -*/ - -#if defined(AUDIO_PIN) && (AUDIO_PIN != C4) && (AUDIO_PIN != C5) && (AUDIO_PIN != C6) && (AUDIO_PIN != B5) && (AUDIO_PIN != B6) && (AUDIO_PIN != B7) && (AUDIO_PIN != D5) -# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under the AVR settings for available options." -#endif - -#if (AUDIO_PIN == C4) || (AUDIO_PIN == C5) || (AUDIO_PIN == C6) -# define AUDIO1_PIN_SET -# define AUDIO1_TIMSKx TIMSK3 -# define AUDIO1_TCCRxA TCCR3A -# define AUDIO1_TCCRxB TCCR3B -# define AUDIO1_ICRx ICR3 -# define AUDIO1_WGMx0 WGM30 -# define AUDIO1_WGMx1 WGM31 -# define AUDIO1_WGMx2 WGM32 -# define AUDIO1_WGMx3 WGM33 -# define AUDIO1_CSx0 CS30 -# define AUDIO1_CSx1 CS31 -# define AUDIO1_CSx2 CS32 - -# if (AUDIO_PIN == C6) -# define AUDIO1_COMxy0 COM3A0 -# define AUDIO1_COMxy1 COM3A1 -# define AUDIO1_OCIExy OCIE3A -# define AUDIO1_OCRxy OCR3A -# define AUDIO1_PIN C6 -# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPA_vect -# elif (AUDIO_PIN == C5) -# define AUDIO1_COMxy0 COM3B0 -# define AUDIO1_COMxy1 COM3B1 -# define AUDIO1_OCIExy OCIE3B -# define AUDIO1_OCRxy OCR3B -# define AUDIO1_PIN C5 -# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPB_vect -# elif (AUDIO_PIN == C4) -# define AUDIO1_COMxy0 COM3C0 -# define AUDIO1_COMxy1 COM3C1 -# define AUDIO1_OCIExy OCIE3C -# define AUDIO1_OCRxy OCR3C -# define AUDIO1_PIN C4 -# define AUDIO1_TIMERx_COMPy_vect TIMER3_COMPC_vect -# endif -#endif - -#if defined(AUDIO_PIN) && defined(AUDIO_PIN_ALT) && (AUDIO_PIN == AUDIO_PIN_ALT) -# error "Audio feature: AUDIO_PIN and AUDIO_PIN_ALT on the same pin makes no sense." -#endif - -#if ((AUDIO_PIN == B5) && ((AUDIO_PIN_ALT == B6) || (AUDIO_PIN_ALT == B7))) || ((AUDIO_PIN == B6) && ((AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B7))) || ((AUDIO_PIN == B7) && ((AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B6))) -# error "Audio feature: PORTB as AUDIO_PIN and AUDIO_PIN_ALT at the same time is not supported." -#endif - -#if defined(AUDIO_PIN_ALT) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7) -# error "Audio feature: the pin selected as AUDIO_PIN_ALT is not supported." -#endif - -#if (AUDIO_PIN == B5) || (AUDIO_PIN == B6) || (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B5) || (AUDIO_PIN_ALT == B6) || (AUDIO_PIN_ALT == B7) || (AUDIO_PIN == D5) -# define AUDIO2_PIN_SET -# define AUDIO2_TIMSKx TIMSK1 -# define AUDIO2_TCCRxA TCCR1A -# define AUDIO2_TCCRxB TCCR1B -# define AUDIO2_ICRx ICR1 -# define AUDIO2_WGMx0 WGM10 -# define AUDIO2_WGMx1 WGM11 -# define AUDIO2_WGMx2 WGM12 -# define AUDIO2_WGMx3 WGM13 -# define AUDIO2_CSx0 CS10 -# define AUDIO2_CSx1 CS11 -# define AUDIO2_CSx2 CS12 - -# if (AUDIO_PIN == B5) || (AUDIO_PIN_ALT == B5) -# define AUDIO2_COMxy0 COM1A0 -# define AUDIO2_COMxy1 COM1A1 -# define AUDIO2_OCIExy OCIE1A -# define AUDIO2_OCRxy OCR1A -# define AUDIO2_PIN B5 -# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect -# elif (AUDIO_PIN == B6) || (AUDIO_PIN_ALT == B6) -# define AUDIO2_COMxy0 COM1B0 -# define AUDIO2_COMxy1 COM1B1 -# define AUDIO2_OCIExy OCIE1B -# define AUDIO2_OCRxy OCR1B -# define AUDIO2_PIN B6 -# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPB_vect -# elif (AUDIO_PIN == B7) || (AUDIO_PIN_ALT == B7) -# define AUDIO2_COMxy0 COM1C0 -# define AUDIO2_COMxy1 COM1C1 -# define AUDIO2_OCIExy OCIE1C -# define AUDIO2_OCRxy OCR1C -# define AUDIO2_PIN B7 -# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPC_vect -# elif (AUDIO_PIN == D5) && defined(__AVR_ATmega32A__) -# pragma message "Audio support for ATmega32A is experimental and can cause crashes." -# undef AUDIO2_TIMSKx -# define AUDIO2_TIMSKx TIMSK -# define AUDIO2_COMxy0 COM1A0 -# define AUDIO2_COMxy1 COM1A1 -# define AUDIO2_OCIExy OCIE1A -# define AUDIO2_OCRxy OCR1A -# define AUDIO2_PIN D5 -# define AUDIO2_TIMERx_COMPy_vect TIMER1_COMPA_vect -# endif -#endif - -// C6 seems to be the assumed default by many existing keyboard - but sill warn the user -#if !defined(AUDIO1_PIN_SET) && !defined(AUDIO2_PIN_SET) -# pragma message "Audio feature enabled, but no suitable pin selected - see docs/feature_audio under the AVR settings for available options. Don't expect to hear anything... :-)" -// TODO: make this an error - go through the breaking-change-process and change all keyboards to the new define -#endif -// ----------------------------------------------------------------------------- - -#ifdef AUDIO1_PIN_SET -static float channel_1_frequency = 0.0f; -void channel_1_set_frequency(float freq) { - if (freq == 0.0f) // a pause/rest is a valid "note" with freq=0 - { - // disable the output, but keep the pwm-ISR going (with the previous - // frequency) so the audio-state keeps getting updated - // Note: setting the duty-cycle 0 is not possible on non-inverting PWM mode - see the AVR data-sheet - AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0)); - return; - } else { - AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); // enable output, PWM mode - } - - channel_1_frequency = freq; - - // set pwm period - AUDIO1_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER)); - // and duty cycle - AUDIO1_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100); -} - -void channel_1_start(void) { - // enable timer-counter ISR - AUDIO1_TIMSKx |= _BV(AUDIO1_OCIExy); - // enable timer-counter output - AUDIO1_TCCRxA |= _BV(AUDIO1_COMxy1); -} - -void channel_1_stop(void) { - // disable timer-counter ISR - AUDIO1_TIMSKx &= ~_BV(AUDIO1_OCIExy); - // disable timer-counter output - AUDIO1_TCCRxA &= ~(_BV(AUDIO1_COMxy1) | _BV(AUDIO1_COMxy0)); -} -#endif - -#ifdef AUDIO2_PIN_SET -static float channel_2_frequency = 0.0f; -void channel_2_set_frequency(float freq) { - if (freq == 0.0f) { - AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0)); - return; - } else { - AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1); - } - - channel_2_frequency = freq; - - AUDIO2_ICRx = (uint16_t)(((float)F_CPU) / (freq * CPU_PRESCALER)); - AUDIO2_OCRxy = (uint16_t)((((float)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre / 100); -} - -float channel_2_get_frequency(void) { return channel_2_frequency; } - -void channel_2_start(void) { - AUDIO2_TIMSKx |= _BV(AUDIO2_OCIExy); - AUDIO2_TCCRxA |= _BV(AUDIO2_COMxy1); -} - -void channel_2_stop(void) { - AUDIO2_TIMSKx &= ~_BV(AUDIO2_OCIExy); - AUDIO2_TCCRxA &= ~(_BV(AUDIO2_COMxy1) | _BV(AUDIO2_COMxy0)); -} -#endif - -void audio_driver_initialize() { -#ifdef AUDIO1_PIN_SET - channel_1_stop(); - setPinOutput(AUDIO1_PIN); -#endif - -#ifdef AUDIO2_PIN_SET - channel_2_stop(); - setPinOutput(AUDIO2_PIN); -#endif - - // TCCR3A / TCCR3B: Timer/Counter #3 Control Registers TCCR3A/TCCR3B, TCCR1A/TCCR1B - // Compare Output Mode (COM3An and COM1An) = 0b00 = Normal port operation - // OC3A -- PC6 - // OC3B -- PC5 - // OC3C -- PC4 - // OC1A -- PB5 - // OC1B -- PB6 - // OC1C -- PB7 - - // Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14. Period = ICR3, Duty Cycle OCR3A) - // OCR3A - PC6 - // OCR3B - PC5 - // OCR3C - PC4 - // OCR1A - PB5 - // OCR1B - PB6 - // OCR1C - PB7 - - // Clock Select (CS3n) = 0b010 = Clock / 8 -#ifdef AUDIO1_PIN_SET - // initialize timer-counter - AUDIO1_TCCRxA = (0 << AUDIO1_COMxy1) | (0 << AUDIO1_COMxy0) | (1 << AUDIO1_WGMx1) | (0 << AUDIO1_WGMx0); - AUDIO1_TCCRxB = (1 << AUDIO1_WGMx3) | (1 << AUDIO1_WGMx2) | (0 << AUDIO1_CSx2) | (1 << AUDIO1_CSx1) | (0 << AUDIO1_CSx0); -#endif - -#ifdef AUDIO2_PIN_SET - AUDIO2_TCCRxA = (0 << AUDIO2_COMxy1) | (0 << AUDIO2_COMxy0) | (1 << AUDIO2_WGMx1) | (0 << AUDIO2_WGMx0); - AUDIO2_TCCRxB = (1 << AUDIO2_WGMx3) | (1 << AUDIO2_WGMx2) | (0 << AUDIO2_CSx2) | (1 << AUDIO2_CSx1) | (0 << AUDIO2_CSx0); -#endif -} - -void audio_driver_stop() { -#ifdef AUDIO1_PIN_SET - channel_1_stop(); -#endif - -#ifdef AUDIO2_PIN_SET - channel_2_stop(); -#endif -} - -void audio_driver_start(void) { -#ifdef AUDIO1_PIN_SET - channel_1_start(); - if (playing_note) { - channel_1_set_frequency(audio_get_processed_frequency(0)); - } -#endif - -#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET) - channel_2_start(); - if (playing_note) { - channel_2_set_frequency(audio_get_processed_frequency(0)); - } -#endif -} - -static volatile uint32_t isr_counter = 0; -#ifdef AUDIO1_PIN_SET -ISR(AUDIO1_TIMERx_COMPy_vect) { - isr_counter++; - if (isr_counter < channel_1_frequency / (CPU_PRESCALER * 8)) return; - - isr_counter = 0; - bool state_changed = audio_update_state(); - - if (!playing_note && !playing_melody) { - channel_1_stop(); -# ifdef AUDIO2_PIN_SET - channel_2_stop(); -# endif - return; - } - - if (state_changed) { - channel_1_set_frequency(audio_get_processed_frequency(0)); -# ifdef AUDIO2_PIN_SET - if (audio_get_number_of_active_tones() > 1) { - channel_2_set_frequency(audio_get_processed_frequency(1)); - } else { - channel_2_stop(); - } -# endif - } -} -#endif - -#if !defined(AUDIO1_PIN_SET) && defined(AUDIO2_PIN_SET) -ISR(AUDIO2_TIMERx_COMPy_vect) { - isr_counter++; - if (isr_counter < channel_2_frequency / (CPU_PRESCALER * 8)) return; - - isr_counter = 0; - bool state_changed = audio_update_state(); - - if (!playing_note && !playing_melody) { - channel_2_stop(); - return; - } - - if (state_changed) { - channel_2_set_frequency(audio_get_processed_frequency(0)); - } -} -#endif diff --git a/quantum/audio/driver_chibios_dac.h b/quantum/audio/driver_chibios_dac.h deleted file mode 100644 index 07cd622ead..0000000000 --- a/quantum/audio/driver_chibios_dac.h +++ /dev/null @@ -1,126 +0,0 @@ -/* Copyright 2019 Jack Humbert - * Copyright 2020 JohSchneider - * - * This program is free software: you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation, either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses/>. - */ -#pragma once - -#ifndef A4 -# define A4 PAL_LINE(GPIOA, 4) -#endif -#ifndef A5 -# define A5 PAL_LINE(GPIOA, 5) -#endif - -/** - * Size of the dac_buffer arrays. All must be the same size. - */ -#define AUDIO_DAC_BUFFER_SIZE 256U - -/** - * Highest value allowed sample value. - - * since the DAC is limited to 12 bit, the absolute max is 0xfff = 4095U; - * lower values adjust the peak-voltage aka volume down. - * adjusting this value has only an effect on a sample-buffer whose values are - * are NOT pregenerated - see square-wave - */ -#ifndef AUDIO_DAC_SAMPLE_MAX -# define AUDIO_DAC_SAMPLE_MAX 4095U -#endif - -#if !defined(AUDIO_DAC_SAMPLE_RATE) && !defined(AUDIO_MAX_SIMULTANEOUS_TONES) && !defined(AUDIO_DAC_QUALITY_VERY_LOW) && !defined(AUDIO_DAC_QUALITY_LOW) && !defined(AUDIO_DAC_QUALITY_HIGH) && !defined(AUDIO_DAC_QUALITY_VERY_HIGH) -# define AUDIO_DAC_QUALITY_SANE_MINIMUM -#endif - -/** - * These presets allow you to quickly switch between quality settings for - * the DAC. The sample rate and maximum number of simultaneous tones roughly - * has an inverse relationship - slightly higher sample rates may be possible. - * - * NOTE: a high sample-rate results in a higher cpu-load, which might lead to - * (audible) discontinuities and/or starve other processes of cpu-time - * (like RGB-led back-lighting, ...) - */ -#ifdef AUDIO_DAC_QUALITY_VERY_LOW -# define AUDIO_DAC_SAMPLE_RATE 11025U -# define AUDIO_MAX_SIMULTANEOUS_TONES 8 -#endif - -#ifdef AUDIO_DAC_QUALITY_LOW -# define AUDIO_DAC_SAMPLE_RATE 22050U -# define AUDIO_MAX_SIMULTANEOUS_TONES 4 -#endif - -#ifdef AUDIO_DAC_QUALITY_HIGH -# define AUDIO_DAC_SAMPLE_RATE 44100U -# define AUDIO_MAX_SIMULTANEOUS_TONES 2 -#endif - -#ifdef AUDIO_DAC_QUALITY_VERY_HIGH -# define AUDIO_DAC_SAMPLE_RATE 88200U -# define AUDIO_MAX_SIMULTANEOUS_TONES 1 -#endif - -#ifdef AUDIO_DAC_QUALITY_SANE_MINIMUM -/* a sane-minimum config: with a trade-off between cpu-load and tone-range - * - * the (currently) highest defined note is NOTE_B8 with 7902Hz; if we now - * aim for an even even multiple of the buffer-size, we end up with: - * ( roundUptoPow2(highest note / AUDIO_DAC_BUFFER_SIZE) * nyquist-rate * AUDIO_DAC_BUFFER_SIZE) - * 7902/256 = 30.867 * 2 * 256 ~= 16384 - * which works out (but the 'scope shows some sampling artifacts with lower harmonics :-P) - */ -# define AUDIO_DAC_SAMPLE_RATE 16384U -# define AUDIO_MAX_SIMULTANEOUS_TONES 8 -#endif - -/** - * Effective bit-rate of the DAC. 44.1khz is the standard for most audio - any - * lower will sacrifice perceptible audio quality. Any higher will limit the - * number of simultaneous tones. In most situations, a tenth (1/10) of the - * sample rate is where notes become unbearable. - */ -#ifndef AUDIO_DAC_SAMPLE_RATE -# define AUDIO_DAC_SAMPLE_RATE 44100U -#endif - -/** - * The number of tones that can be played simultaneously. If too high a value - * is used here, the keyboard will freeze and glitch-out when that many tones - * are being played. - */ -#ifndef AUDIO_MAX_SIMULTANEOUS_TONES -# define AUDIO_MAX_SIMULTANEOUS_TONES 2 -#endif - -/** - * The default value of the DAC when not playing anything. Certain hardware - * setups may require a high (AUDIO_DAC_SAMPLE_MAX) or low (0) value here. - * Since multiple added sine waves tend to oscillate around the midpoint, - * and possibly never/rarely reach either 0 of MAX, 1/2 MAX can be a - * reasonable default value. - */ -#ifndef AUDIO_DAC_OFF_VALUE -# define AUDIO_DAC_OFF_VALUE AUDIO_DAC_SAMPLE_MAX / 2 -#endif - -#if AUDIO_DAC_OFF_VALUE > AUDIO_DAC_SAMPLE_MAX -# error "AUDIO_DAC: OFF_VALUE may not be larger than SAMPLE_MAX" -#endif - -/** - *user overridable sample generation/processing - */ -uint16_t dac_value_generate(void); diff --git a/quantum/audio/driver_chibios_dac_additive.c b/quantum/audio/driver_chibios_dac_additive.c deleted file mode 100644 index db304adb87..0000000000 --- a/quantum/audio/driver_chibios_dac_additive.c +++ /dev/null @@ -1,335 +0,0 @@ -/* Copyright 2016-2019 Jack Humbert - * Copyright 2020 JohSchneider - * - * This program is free software: you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation, either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses/>. - */ - -#include "audio.h" -#include <ch.h> -#include <hal.h> - -/* - Audio Driver: DAC - - which utilizes the dac unit many STM32 are equipped with, to output a modulated waveform from samples stored in the dac_buffer_* array who are passed to the hardware through DMA - - it is also possible to have a custom sample-LUT by implementing/overriding 'dac_value_generate' - - this driver allows for multiple simultaneous tones to be played through one single channel by doing additive wave-synthesis -*/ - -#if !defined(AUDIO_PIN) -# error "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC additive)' for available options." -#endif -#if defined(AUDIO_PIN_ALT) && !defined(AUDIO_PIN_ALT_AS_NEGATIVE) -# pragma message "Audio feature: AUDIO_PIN_ALT set, but not AUDIO_PIN_ALT_AS_NEGATIVE - pin will be left unused; audio might still work though." -#endif - -#if !defined(AUDIO_PIN_ALT) -// no ALT pin defined is valid, but the c-ifs below need some value set -# define AUDIO_PIN_ALT PAL_NOLINE -#endif - -#if !defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE) && !defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID) -# define AUDIO_DAC_SAMPLE_WAVEFORM_SINE -#endif - -#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SINE -/* one full sine wave over [0,2*pi], but shifted up one amplitude and left pi/4; for the samples to start at 0 - */ -static const dacsample_t dac_buffer_sine[AUDIO_DAC_BUFFER_SIZE] = { - // 256 values, max 4095 - 0x0, 0x1, 0x2, 0x6, 0xa, 0xf, 0x16, 0x1e, 0x27, 0x32, 0x3d, 0x4a, 0x58, 0x67, 0x78, 0x89, 0x9c, 0xb0, 0xc5, 0xdb, 0xf2, 0x10a, 0x123, 0x13e, 0x159, 0x175, 0x193, 0x1b1, 0x1d1, 0x1f1, 0x212, 0x235, 0x258, 0x27c, 0x2a0, 0x2c6, 0x2ed, 0x314, 0x33c, 0x365, 0x38e, 0x3b8, 0x3e3, 0x40e, 0x43a, 0x467, 0x494, 0x4c2, 0x4f0, 0x51f, 0x54e, 0x57d, 0x5ad, 0x5dd, 0x60e, 0x63f, 0x670, 0x6a1, 0x6d3, 0x705, 0x737, 0x769, 0x79b, 0x7cd, 0x800, 0x832, 0x864, 0x896, 0x8c8, 0x8fa, 0x92c, 0x95e, 0x98f, 0x9c0, 0x9f1, 0xa22, 0xa52, 0xa82, 0xab1, 0xae0, 0xb0f, 0xb3d, 0xb6b, 0xb98, 0xbc5, 0xbf1, 0xc1c, 0xc47, 0xc71, 0xc9a, 0xcc3, 0xceb, 0xd12, 0xd39, 0xd5f, 0xd83, 0xda7, 0xdca, 0xded, 0xe0e, 0xe2e, 0xe4e, 0xe6c, 0xe8a, 0xea6, 0xec1, 0xedc, 0xef5, 0xf0d, 0xf24, 0xf3a, 0xf4f, 0xf63, 0xf76, 0xf87, 0xf98, 0xfa7, 0xfb5, 0xfc2, 0xfcd, 0xfd8, 0xfe1, 0xfe9, 0xff0, 0xff5, 0xff9, 0xffd, 0xffe, - 0xfff, 0xffe, 0xffd, 0xff9, 0xff5, 0xff0, 0xfe9, 0xfe1, 0xfd8, 0xfcd, 0xfc2, 0xfb5, 0xfa7, 0xf98, 0xf87, 0xf76, 0xf63, 0xf4f, 0xf3a, 0xf24, 0xf0d, 0xef5, 0xedc, 0xec1, 0xea6, 0xe8a, 0xe6c, 0xe4e, 0xe2e, 0xe0e, 0xded, 0xdca, 0xda7, 0xd83, 0xd5f, 0xd39, 0xd12, 0xceb, 0xcc3, 0xc9a, 0xc71, 0xc47, 0xc1c, 0xbf1, 0xbc5, 0xb98, 0xb6b, 0xb3d, 0xb0f, 0xae0, 0xab1, 0xa82, 0xa52, 0xa22, 0x9f1, 0x9c0, 0x98f, 0x95e, 0x92c, 0x8fa, 0x8c8, 0x896, 0x864, 0x832, 0x800, 0x7cd, 0x79b, 0x769, 0x737, 0x705, 0x6d3, 0x6a1, 0x670, 0x63f, 0x60e, 0x5dd, 0x5ad, 0x57d, 0x54e, 0x51f, 0x4f0, 0x4c2, 0x494, 0x467, 0x43a, 0x40e, 0x3e3, 0x3b8, 0x38e, 0x365, 0x33c, 0x314, 0x2ed, 0x2c6, 0x2a0, 0x27c, 0x258, 0x235, 0x212, 0x1f1, 0x1d1, 0x1b1, 0x193, 0x175, 0x159, 0x13e, 0x123, 0x10a, 0xf2, 0xdb, 0xc5, 0xb0, 0x9c, 0x89, 0x78, 0x67, 0x58, 0x4a, 0x3d, 0x32, 0x27, 0x1e, 0x16, 0xf, 0xa, 0x6, 0x2, 0x1}; -#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SINE -#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE -static const dacsample_t dac_buffer_triangle[AUDIO_DAC_BUFFER_SIZE] = { - // 256 values, max 4095 - 0x0, 0x20, 0x40, 0x60, 0x80, 0xa0, 0xc0, 0xe0, 0x100, 0x120, 0x140, 0x160, 0x180, 0x1a0, 0x1c0, 0x1e0, 0x200, 0x220, 0x240, 0x260, 0x280, 0x2a0, 0x2c0, 0x2e0, 0x300, 0x320, 0x340, 0x360, 0x380, 0x3a0, 0x3c0, 0x3e0, 0x400, 0x420, 0x440, 0x460, 0x480, 0x4a0, 0x4c0, 0x4e0, 0x500, 0x520, 0x540, 0x560, 0x580, 0x5a0, 0x5c0, 0x5e0, 0x600, 0x620, 0x640, 0x660, 0x680, 0x6a0, 0x6c0, 0x6e0, 0x700, 0x720, 0x740, 0x760, 0x780, 0x7a0, 0x7c0, 0x7e0, 0x800, 0x81f, 0x83f, 0x85f, 0x87f, 0x89f, 0x8bf, 0x8df, 0x8ff, 0x91f, 0x93f, 0x95f, 0x97f, 0x99f, 0x9bf, 0x9df, 0x9ff, 0xa1f, 0xa3f, 0xa5f, 0xa7f, 0xa9f, 0xabf, 0xadf, 0xaff, 0xb1f, 0xb3f, 0xb5f, 0xb7f, 0xb9f, 0xbbf, 0xbdf, 0xbff, 0xc1f, 0xc3f, 0xc5f, 0xc7f, 0xc9f, 0xcbf, 0xcdf, 0xcff, 0xd1f, 0xd3f, 0xd5f, 0xd7f, 0xd9f, 0xdbf, 0xddf, 0xdff, 0xe1f, 0xe3f, 0xe5f, 0xe7f, 0xe9f, 0xebf, 0xedf, 0xeff, 0xf1f, 0xf3f, 0xf5f, 0xf7f, 0xf9f, 0xfbf, 0xfdf, - 0xfff, 0xfdf, 0xfbf, 0xf9f, 0xf7f, 0xf5f, 0xf3f, 0xf1f, 0xeff, 0xedf, 0xebf, 0xe9f, 0xe7f, 0xe5f, 0xe3f, 0xe1f, 0xdff, 0xddf, 0xdbf, 0xd9f, 0xd7f, 0xd5f, 0xd3f, 0xd1f, 0xcff, 0xcdf, 0xcbf, 0xc9f, 0xc7f, 0xc5f, 0xc3f, 0xc1f, 0xbff, 0xbdf, 0xbbf, 0xb9f, 0xb7f, 0xb5f, 0xb3f, 0xb1f, 0xaff, 0xadf, 0xabf, 0xa9f, 0xa7f, 0xa5f, 0xa3f, 0xa1f, 0x9ff, 0x9df, 0x9bf, 0x99f, 0x97f, 0x95f, 0x93f, 0x91f, 0x8ff, 0x8df, 0x8bf, 0x89f, 0x87f, 0x85f, 0x83f, 0x81f, 0x800, 0x7e0, 0x7c0, 0x7a0, 0x780, 0x760, 0x740, 0x720, 0x700, 0x6e0, 0x6c0, 0x6a0, 0x680, 0x660, 0x640, 0x620, 0x600, 0x5e0, 0x5c0, 0x5a0, 0x580, 0x560, 0x540, 0x520, 0x500, 0x4e0, 0x4c0, 0x4a0, 0x480, 0x460, 0x440, 0x420, 0x400, 0x3e0, 0x3c0, 0x3a0, 0x380, 0x360, 0x340, 0x320, 0x300, 0x2e0, 0x2c0, 0x2a0, 0x280, 0x260, 0x240, 0x220, 0x200, 0x1e0, 0x1c0, 0x1a0, 0x180, 0x160, 0x140, 0x120, 0x100, 0xe0, 0xc0, 0xa0, 0x80, 0x60, 0x40, 0x20}; -#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE -#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE -static const dacsample_t dac_buffer_square[AUDIO_DAC_BUFFER_SIZE] = { - [0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1] = 0, // first and - [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX, // second half -}; -#endif // AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE -/* -// four steps: 0, 1/3, 2/3 and 1 -static const dacsample_t dac_buffer_staircase[AUDIO_DAC_BUFFER_SIZE] = { - [0 ... AUDIO_DAC_BUFFER_SIZE/3 -1 ] = 0, - [AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE / 2 -1 ] = AUDIO_DAC_SAMPLE_MAX / 3, - [AUDIO_DAC_BUFFER_SIZE / 2 ... 3 * AUDIO_DAC_BUFFER_SIZE / 4 -1 ] = 2 * AUDIO_DAC_SAMPLE_MAX / 3, - [3 * AUDIO_DAC_BUFFER_SIZE / 4 ... AUDIO_DAC_BUFFER_SIZE -1 ] = AUDIO_DAC_SAMPLE_MAX, -} -*/ -#ifdef AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID -static const dacsample_t dac_buffer_trapezoid[AUDIO_DAC_BUFFER_SIZE] = {0x0, 0x1f, 0x7f, 0xdf, 0x13f, 0x19f, 0x1ff, 0x25f, 0x2bf, 0x31f, 0x37f, 0x3df, 0x43f, 0x49f, 0x4ff, 0x55f, 0x5bf, 0x61f, 0x67f, 0x6df, 0x73f, 0x79f, 0x7ff, 0x85f, 0x8bf, 0x91f, 0x97f, 0x9df, 0xa3f, 0xa9f, 0xaff, 0xb5f, 0xbbf, 0xc1f, 0xc7f, 0xcdf, 0xd3f, 0xd9f, 0xdff, 0xe5f, 0xebf, 0xf1f, 0xf7f, 0xfdf, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, 0xfff, - 0xfff, 0xfdf, 0xf7f, 0xf1f, 0xebf, 0xe5f, 0xdff, 0xd9f, 0xd3f, 0xcdf, 0xc7f, 0xc1f, 0xbbf, 0xb5f, 0xaff, 0xa9f, 0xa3f, 0x9df, 0x97f, 0x91f, 0x8bf, 0x85f, 0x7ff, 0x79f, 0x73f, 0x6df, 0x67f, 0x61f, 0x5bf, 0x55f, 0x4ff, 0x49f, 0x43f, 0x3df, 0x37f, 0x31f, 0x2bf, 0x25f, 0x1ff, 0x19f, 0x13f, 0xdf, 0x7f, 0x1f, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0}; -#endif // AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID - -static dacsample_t dac_buffer_empty[AUDIO_DAC_BUFFER_SIZE] = {AUDIO_DAC_OFF_VALUE}; - -/* keep track of the sample position for for each frequency */ -static float dac_if[AUDIO_MAX_SIMULTANEOUS_TONES] = {0.0}; - -static float active_tones_snapshot[AUDIO_MAX_SIMULTANEOUS_TONES] = {0, 0}; -static uint8_t active_tones_snapshot_length = 0; - -typedef enum { - OUTPUT_SHOULD_START, - OUTPUT_RUN_NORMALLY, - // path 1: wait for zero, then change/update active tones - OUTPUT_TONES_CHANGED, - OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE, - // path 2: hardware should stop, wait for zero then turn output off = stop the timer - OUTPUT_SHOULD_STOP, - OUTPUT_REACHED_ZERO_BEFORE_OFF, - OUTPUT_OFF, - OUTPUT_OFF_1, - OUTPUT_OFF_2, // trailing off: giving the DAC two more conversion cycles until the AUDIO_DAC_OFF_VALUE reaches the output, then turn the timer off, which leaves the output at that level - number_of_output_states -} output_states_t; -output_states_t state = OUTPUT_OFF_2; - -/** - * Generation of the waveform being passed to the callback. Declared weak so users - * can override it with their own wave-forms/noises. - */ -__attribute__((weak)) uint16_t dac_value_generate(void) { - // DAC is running/asking for values but snapshot length is zero -> must be playing a pause - if (active_tones_snapshot_length == 0) { - return AUDIO_DAC_OFF_VALUE; - } - - /* doing additive wave synthesis over all currently playing tones = adding up - * sine-wave-samples for each frequency, scaled by the number of active tones - */ - uint16_t value = 0; - float frequency = 0.0f; - - for (uint8_t i = 0; i < active_tones_snapshot_length; i++) { - /* Note: a user implementation does not have to rely on the active_tones_snapshot, but - * could directly query the active frequencies through audio_get_processed_frequency */ - frequency = active_tones_snapshot[i]; - - dac_if[i] = dac_if[i] + ((frequency * AUDIO_DAC_BUFFER_SIZE) / AUDIO_DAC_SAMPLE_RATE) * 2 / 3; - /*Note: the 2/3 are necessary to get the correct frequencies on the - * DAC output (as measured with an oscilloscope), since the gpt - * timer runs with 3*AUDIO_DAC_SAMPLE_RATE; and the DAC callback - * is called twice per conversion.*/ - - dac_if[i] = fmod(dac_if[i], AUDIO_DAC_BUFFER_SIZE); - - // Wavetable generation/lookup - uint16_t dac_i = (uint16_t)dac_if[i]; - -#if defined(AUDIO_DAC_SAMPLE_WAVEFORM_SINE) - value += dac_buffer_sine[dac_i] / active_tones_snapshot_length; -#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE) - value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length; -#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID) - value += dac_buffer_trapezoid[dac_i] / active_tones_snapshot_length; -#elif defined(AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE) - value += dac_buffer_square[dac_i] / active_tones_snapshot_length; -#endif - /* - // SINE - value += dac_buffer_sine[dac_i] / active_tones_snapshot_length / 3; - // TRIANGLE - value += dac_buffer_triangle[dac_i] / active_tones_snapshot_length / 3; - // SQUARE - value += dac_buffer_square[dac_i] / active_tones_snapshot_length / 3; - //NOTE: combination of these three wave-forms is more exemplary - and doesn't sound particularly good :-P - */ - - // STAIRS (mostly usefully as test-pattern) - // value_avg = dac_buffer_staircase[dac_i] / active_tones_snapshot_length; - } - - return value; -} - -/** - * DAC streaming callback. Does all of the main computing for playing songs. - * - * Note: chibios calls this CB twice: during the 'half buffer event', and the 'full buffer event'. - */ -static void dac_end(DACDriver *dacp) { - dacsample_t *sample_p = (dacp)->samples; - - // work on the other half of the buffer - if (dacIsBufferComplete(dacp)) { - sample_p += AUDIO_DAC_BUFFER_SIZE / 2; // 'half_index' - } - - for (uint8_t s = 0; s < AUDIO_DAC_BUFFER_SIZE / 2; s++) { - if (OUTPUT_OFF <= state) { - sample_p[s] = AUDIO_DAC_OFF_VALUE; - continue; - } else { - sample_p[s] = dac_value_generate(); - } - - /* zero crossing (or approach, whereas zero == DAC_OFF_VALUE, which can be configured to anything from 0 to DAC_SAMPLE_MAX) - * ============================*=*========================== AUDIO_DAC_SAMPLE_MAX - * * * - * * * - * --------------------------------------------------------- - * * * } AUDIO_DAC_SAMPLE_MAX/100 - * --------------------------------------------------------- AUDIO_DAC_OFF_VALUE - * * * } AUDIO_DAC_SAMPLE_MAX/100 - * --------------------------------------------------------- - * * - * * * - * * * - * =====*=*================================================= 0x0 - */ - if (((sample_p[s] + (AUDIO_DAC_SAMPLE_MAX / 100)) > AUDIO_DAC_OFF_VALUE) && // value approaches from below - (sample_p[s] < (AUDIO_DAC_OFF_VALUE + (AUDIO_DAC_SAMPLE_MAX / 100))) // or above - ) { - if ((OUTPUT_SHOULD_START == state) && (active_tones_snapshot_length > 0)) { - state = OUTPUT_RUN_NORMALLY; - } else if (OUTPUT_TONES_CHANGED == state) { - state = OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE; - } else if (OUTPUT_SHOULD_STOP == state) { - state = OUTPUT_REACHED_ZERO_BEFORE_OFF; - } - } - - // still 'ramping up', reset the output to OFF_VALUE until the generated values reach that value, to do a smooth handover - if (OUTPUT_SHOULD_START == state) { - sample_p[s] = AUDIO_DAC_OFF_VALUE; - } - - if ((OUTPUT_SHOULD_START == state) || (OUTPUT_REACHED_ZERO_BEFORE_OFF == state) || (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state)) { - uint8_t active_tones = MIN(AUDIO_MAX_SIMULTANEOUS_TONES, audio_get_number_of_active_tones()); - active_tones_snapshot_length = 0; - // update the snapshot - once, and only on occasion that something changed; - // -> saves cpu cycles (?) - for (uint8_t i = 0; i < active_tones; i++) { - float freq = audio_get_processed_frequency(i); - if (freq > 0) { // disregard 'rest' notes, with valid frequency 0.0f; which would only lower the resulting waveform volume during the additive synthesis step - active_tones_snapshot[active_tones_snapshot_length++] = freq; - } - } - - if ((0 == active_tones_snapshot_length) && (OUTPUT_REACHED_ZERO_BEFORE_OFF == state)) { - state = OUTPUT_OFF; - } - if (OUTPUT_REACHED_ZERO_BEFORE_TONE_CHANGE == state) { - state = OUTPUT_RUN_NORMALLY; - } - } - } - - // update audio internal state (note position, current_note, ...) - if (audio_update_state()) { - if (OUTPUT_SHOULD_STOP != state) { - state = OUTPUT_TONES_CHANGED; - } - } - - if (OUTPUT_OFF <= state) { - if (OUTPUT_OFF_2 == state) { - // stopping timer6 = stopping the DAC at whatever value it is currently pushing to the output = AUDIO_DAC_OFF_VALUE - gptStopTimer(&GPTD6); - } else { - state++; - } - } -} - -static void dac_error(DACDriver *dacp, dacerror_t err) { - (void)dacp; - (void)err; - - chSysHalt("DAC failure. halp"); -} - -static const GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE * 3, - .callback = NULL, - .cr2 = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event. */ - .dier = 0U}; - -static const DACConfig dac_conf = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT}; - -/** - * @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered - * on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency - * to be a third of what we expect. - * - * Here are all the values for DAC_TRG (TSEL in the ref manual) - * TIM15_TRGO 0b011 - * TIM2_TRGO 0b100 - * TIM3_TRGO 0b001 - * TIM6_TRGO 0b000 - * TIM7_TRGO 0b010 - * EXTI9 0b110 - * SWTRIG 0b111 - */ -static const DACConversionGroup dac_conv_cfg = {.num_channels = 1U, .end_cb = dac_end, .error_cb = dac_error, .trigger = DAC_TRG(0b000)}; - -void audio_driver_initialize() { - if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) { - palSetLineMode(A4, PAL_MODE_INPUT_ANALOG); - dacStart(&DACD1, &dac_conf); - } - if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) { - palSetLineMode(A5, PAL_MODE_INPUT_ANALOG); - dacStart(&DACD2, &dac_conf); - } |