Move Audio drivers from quantum to platform drivers folder (#14308)

* Move Audio drivers from quantum to platform drivers folder

* fix path for audio drivers

Co-authored-by: Ryan <fauxpark@gmail.com>

Co-authored-by: Ryan <fauxpark@gmail.com>

9 files changed, 4 insertions, 1412 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);
-    }
-
-    /* enable the output buffer, to directly drive external loads with no additional circuitry
-     *
-     * see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers</