Revert "Revert "Audio system overhaul (#11820)" due to freezing issues"

This reverts commit 996a19ee7ba3308e17fd347afde0b135852835cc.

22 files changed, 1993 insertions, 798 deletions

diff --git a/common_features.mk b/common_features.mk
index 926eef4c99..aea79789df 100644
--- a/common_features.mk
+++ b/common_features.mk
@@ -58,12 +58,31 @@ ifeq ($(strip $(COMMAND_ENABLE)), yes)
     OPT_DEFS += -DCOMMAND_ENABLE
 endif
 
+AUDIO_ENABLE ?= no
 ifeq ($(strip $(AUDIO_ENABLE)), yes)
+    ifeq ($(PLATFORM),CHIBIOS)
+        AUDIO_DRIVER ?= dac_basic
+        ifeq ($(strip $(AUDIO_DRIVER)), dac_basic)
+            OPT_DEFS += -DAUDIO_DRIVER_DAC
+        else ifeq ($(strip $(AUDIO_DRIVER)), dac_additive)
+            OPT_DEFS += -DAUDIO_DRIVER_DAC
+        ## stm32f2 and above have a usable DAC unit, f1 do not, and need to use pwm instead
+        else ifeq ($(strip $(AUDIO_DRIVER)), pwm_software)
+            OPT_DEFS += -DAUDIO_DRIVER_PWM
+        else ifeq ($(strip $(AUDIO_DRIVER)), pwm_hardware)
+            OPT_DEFS += -DAUDIO_DRIVER_PWM
+        endif
+    else
+        # fallback for all other platforms is pwm
+        AUDIO_DRIVER ?= pwm_hardware
+        OPT_DEFS += -DAUDIO_DRIVER_PWM
+    endif
     OPT_DEFS += -DAUDIO_ENABLE
     MUSIC_ENABLE = yes
     SRC += $(QUANTUM_DIR)/process_keycode/process_audio.c
     SRC += $(QUANTUM_DIR)/process_keycode/process_clicky.c
-    SRC += $(QUANTUM_DIR)/audio/audio_$(PLATFORM_KEY).c
+    SRC += $(QUANTUM_DIR)/audio/audio.c ## common audio code, hardware agnostic
+    SRC += $(QUANTUM_DIR)/audio/driver_$(PLATFORM_KEY)_$(strip $(AUDIO_DRIVER)).c
     SRC += $(QUANTUM_DIR)/audio/voices.c
     SRC += $(QUANTUM_DIR)/audio/luts.c
 endif
diff --git a/keyboards/planck/config.h b/keyboards/planck/config.h
index 9ef2b0b0dd..71111eca21 100644
--- a/keyboards/planck/config.h
+++ b/keyboards/planck/config.h
@@ -40,7 +40,7 @@ along with this program.  If not, see <http://www.gnu.org/licenses/>.
 #define QMK_SPEAKER C6
 
 #define AUDIO_VOICES
-#define C6_AUDIO
+#define AUDIO_PIN C6
 
 #define BACKLIGHT_PIN B7
 
diff --git a/keyboards/planck/ez/config.h b/keyboards/planck/ez/config.h
index e924d077d2..41abb00808 100644
--- a/keyboards/planck/ez/config.h
+++ b/keyboards/planck/ez/config.h
@@ -58,7 +58,10 @@
 
 #define MUSIC_MAP
 #undef AUDIO_VOICES
-#undef C6_AUDIO
+#undef AUDIO_PIN
+#define AUDIO_PIN A5
+#define AUDIO_PIN_ALT A4
+#define AUDIO_PIN_ALT_AS_NEGATIVE
 
 /* Debounce reduces chatter (unintended double-presses) - set 0 if debouncing is not needed */
 // #define DEBOUNCE 6
diff --git a/quantum/audio/audio.h b/quantum/audio/audio.h
index dccf03d5f6..56b9158a1a 100644
--- a/quantum/audio/audio.h
+++ b/quantum/audio/audio.h
@@ -1,4 +1,5 @@
-/* Copyright 2016 Jack Humbert
+/* Copyright 2016-2020 Jack Humbert
+ * Copyright 2020 JohSchneider
  *
  * This program is free software: you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
@@ -13,28 +14,30 @@
  * 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
 
 #include <stdint.h>
 #include <stdbool.h>
-#if defined(__AVR__)
-#    include <avr/io.h>
-#endif
-#include "wait.h"
 #include "musical_notes.h"
 #include "song_list.h"
 #include "voices.h"
 #include "quantum.h"
 #include <math.h>
 
-// Largely untested PWM audio mode (doesn't sound as good)
-// #define PWM_AUDIO
-
-// #define VIBRATO_ENABLE
+#if defined(__AVR__)
+#    include <avr/io.h>
+#    if defined(AUDIO_DRIVER_PWM)
+#        include "driver_avr_pwm.h"
+#    endif
+#endif
 
-// Enable vibrato strength/amplitude - slows down ISR too much
-// #define VIBRATO_STRENGTH_ENABLE
+#if defined(PROTOCOL_CHIBIOS)
+#    if defined(AUDIO_DRIVER_PWM)
+#        include "driver_chibios_pwm.h"
+#    elif defined(AUDIO_DRIVER_DAC)
+#        include "driver_chibios_dac.h"
+#    endif
+#endif
 
 typedef union {
     uint8_t raw;
@@ -45,62 +48,238 @@ typedef union {
     };
 } audio_config_t;
 
-bool is_audio_on(void);
+// AVR/LUFA has a MIN, arm/chibios does not
+#ifndef MIN
+#    define MIN(a, b) (((a) < (b)) ? (a) : (b))
+#endif
+
+/*
+ * a 'musical note' is represented by pitch and duration; a 'musical tone' adds intensity and timbre
+ * https://en.wikipedia.org/wiki/Musical_tone
+ * "A musical tone is characterized by its duration, pitch, intensity (or loudness), and timbre (or quality)"
+ */
+typedef struct {
+    uint16_t time_started;  // timestamp the tone/note was started, system time runs with 1ms resolution -> 16bit timer overflows every ~64 seconds, long enough under normal circumstances; but might be too soon for long-duration notes when the note_tempo is set to a very low value
+    float    pitch;         // aka frequency, in Hz
+    uint16_t duration;      // in ms, converted from the musical_notes.h unit which has 64parts to a beat, factoring in the current tempo in beats-per-minute
+    // float intensity;    // aka volume [0,1] TODO: not used at the moment; pwm drivers can't handle it
+    // uint8_t timbre;     // range: [0,100] TODO: this currently kept track of globally, should we do this per tone instead?
+} musical_tone_t;
+
+// public interface
+
+/**
+ * @brief one-time initialization called by quantum/quantum.c
+ * @details usually done lazy, when some tones are to be played
+ *
+ * @post audio system (and hardware) initialized and ready to play tones
+ */
+void audio_init(void);
+void audio_startup(void);
+
+/**
+ * @brief en-/disable audio output, save this choice to the eeprom
+ */
 void audio_toggle(void);
+/**
+ * @brief enable audio output, save this choice to the eeprom
+ */
 void audio_on(void);
+/**
+ * @brief disable audio output, save this choice to the eeprom
+ */
 void audio_off(void);
+/**
+ * @brief query the if audio output is enabled
+ */
+bool audio_is_on(void);
 
-// Vibrato rate functions
+/**
+ * @brief start playback of a tone with the given frequency and duration
+ *
+ * @details starts the playback of a given note, which is automatically stopped
+ *          at the the end of its duration = fire&forget
+ *
+ * @param[in] pitch frequency of the tone be played
+ * @param[in] duration in milliseconds, use 'audio_duration_to_ms' to convert
+ *                     from the musical_notes.h unit to ms
+ */
+void audio_play_note(float pitch, uint16_t duration);
+// TODO: audio_play_note(float pitch, uint16_t duration, float intensity, float timbre);
+// audio_play_note_with_instrument ifdef AUDIO_ENABLE_VOICES
 
-#ifdef VIBRATO_ENABLE
+/**
+ * @brief start playback of a tone with the given frequency
+ *
+ * @details the 'frequency' is put on-top the internal stack of active tones,
+ *          as a new tone with indefinite duration. this tone is played by
+ *          the hardware until a call to 'audio_stop_tone'.
+ *          should a tone with that frequency already be active, its entry
+ *          is put on the top of said internal stack - so no duplicate
+ *          entries are kept.
+ *          'hardware_start' is called upon the first note.
+ *
+ * @param[in] pitch frequency of the tone be played
+ */
+void audio_play_tone(float pitch);
 
-void set_vibrato_rate(float rate);
-void increase_vibrato_rate(float change);
-void decrease_vibrato_rate(float change);
+/**
+ * @brief stop a given tone/frequency
+ *
+ * @details removes a tone matching the given frequency from the internal
+ *          playback stack
+ *          the hardware is stopped in case this was the last/only frequency
+ *          being played.
+ *
+ * @param[in] pitch tone/frequency to be stopped
+ */
+void audio_stop_tone(float pitch);
 
-#    ifdef VIBRATO_STRENGTH_ENABLE
+/**
+ * @brief play a melody
+ *
+ * @details starts playback of a melody passed in from a SONG definition - an
+ *          array of {pitch, duration} float-tuples
+ *
+ * @param[in] np note-pointer to the SONG array
+ * @param[in] n_count number of MUSICAL_NOTES of the SONG
+ * @param[in] n_repeat false for onetime, true for looped playback
+ */
+void audio_play_melody(float (*np)[][2], uint16_t n_count, bool n_repeat);
 
-void set_vibrato_strength(float strength);
-void increase_vibrato_strength(float change);
-void decrease_vibrato_strength(float change);
+/**
+ * @brief play a short tone of a specific frequency to emulate a 'click'
+ *
+ * @details constructs a two-note melody (one pause plus a note) and plays it through
+ *          audio_play_melody. very short durations might not quite work due to
+ *          hardware limitations (DAC: added pulses from zero-crossing feature;...)
+ *
+ * @param[in] delay in milliseconds, length for the pause before the pulses, can be zero
+ * @param[in] pitch
+ * @param[in] duration in milliseconds, length of the 'click'
+ */
+void audio_play_click(uint16_t delay, float pitch, uint16_t duration);
 
-#    endif
+/**
+ * @brief stops all playback
+ *
+ * @details stops playback of both a melody as well as single tones, resetting
+ *          the internal state
+ */
+void audio_stop_all(void);
 
-#endif
+/**
+ * @brief query if one/multiple tones are playing
+ */
+bool audio_is_playing_note(void);
 
-// Polyphony functions
+/**
+ * @brief query if a melody/SONG is playing
+ */
+bool audio_is_playing_melody(void);
 
-void set_polyphony_rate(float rate);
-void enable_polyphony(void);
-void disable_polyphony(void);
-void increase_polyphony_rate(float change);
-void decrease_polyphony_rate(float change);
+// These macros are used to allow audio_play_melody to play an array of indeterminate
+// length. This works around the limitation of C's sizeof operation on pointers.
+// The global float array for the song must be used here.
+#define NOTE_ARRAY_SIZE(x) ((int16_t)(sizeof(x) / (sizeof(x[0]))))
 
-void set_timbre(float timbre);
-void set_tempo(uint8_t tempo);
+/**
+ * @brief convenience macro, to play a melody/SONG once
+ */
+#define PLAY_SONG(note_array) audio_play_melody(&note_array, NOTE_ARRAY_SIZE((note_array)), false)
+// TODO: a 'song' is a melody plus singing/vocals -> PLAY_MELODY
+/**
+ * @brief convenience macro, to play a melody/SONG in a loop, until stopped by 'audio_stop_all'
+ */
+#define PLAY_LOOP(note_array) audio_play_melody(&note_array, NOTE_ARRAY_SIZE((note_array)), true)
 
-void increase_tempo(uint8_t tempo_change);
-void decrease_tempo(uint8_t tempo_change);
+// Tone-Multiplexing functions
+// this feature only makes sense for hardware setups which can't do proper
+// audio-wave synthesis = have no DAC and need to use PWM for tone generation
+#ifdef AUDIO_ENABLE_TONE_MULTIPLEXING
+#    ifndef AUDIO_TONE_MULTIPLEXING_RATE_DEFAULT
+#        define AUDIO_TONE_MULTIPLEXING_RATE_DEFAULT 0
+//       0=off, good starting value is 4; the lower the value the higher the cpu-load
+#    endif
+void audio_set_tone_multiplexing_rate(uint16_t rate);
+void audio_enable_tone_multiplexing(void);
+void audio_disable_tone_multiplexing(void);
+void audio_increase_tone_multiplexing_rate(uint16_t change);
+void audio_decrease_tone_multiplexing_rate(uint16_t change);
+#endif
+
+// Tempo functions
+
+void audio_set_tempo(uint8_t tempo);
+void audio_increase_tempo(uint8_t tempo_change);
+void audio_decrease_tempo(uint8_t tempo_change);
+
+// conversion macros, from 64parts-to-a-beat to milliseconds and back
+uint16_t audio_duration_to_ms(uint16_t duration_bpm);
+uint16_t audio_ms_to_duration(uint16_t duration_ms);
 
-void audio_init(void);
 void audio_startup(void);
 
-#ifdef PWM_AUDIO
-void play_sample(uint8_t* s, uint16_t l, bool r);
-#endif
-void play_note(float freq, int vol);
-void stop_note(float freq);
-void stop_all_notes(void);
-void play_notes(float (*np)[][2], uint16_t n_count, bool n_repeat);
+// hardware interface
 
-#define SCALE \
-    (int8_t[]) { 0 + (12 * 0), 2 + (12 * 0), 4 + (12 * 0), 5 + (12 * 0), 7 + (12 * 0), 9 + (12 * 0), 11 + (12 * 0), 0 + (12 * 1), 2 + (12 * 1), 4 + (12 * 1), 5 + (12 * 1), 7 + (12 * 1), 9 + (12 * 1), 11 + (12 * 1), 0 + (12 * 2), 2 + (12 * 2), 4 + (12 * 2), 5 + (12 * 2), 7 + (12 * 2), 9 + (12 * 2), 11 + (12 * 2), 0 + (12 * 3), 2 + (12 * 3), 4 + (12 * 3), 5 + (12 * 3), 7 + (12 * 3), 9 + (12 * 3), 11 + (12 * 3), 0 + (12 * 4), 2 + (12 * 4), 4 + (12 * 4), 5 + (12 * 4), 7 + (12 * 4), 9 + (12 * 4), 11 + (12 * 4), }
+// implementation in the driver_avr/arm_* respective parts
+void audio_driver_initialize(void);
+void audio_driver_start(void);
+void audio_driver_stop(void);
 
-// These macros are used to allow play_notes to play an array of indeterminate
-// length. This works around the limitation of C's sizeof operation on pointers.
-// The global float array for the song must be used here.
-#define NOTE_ARRAY_SIZE(x) ((int16_t)(sizeof(x) / (sizeof(x[0]))))
-#define PLAY_SONG(note_array) play_notes(&note_array, NOTE_ARRAY_SIZE((note_array)), false)
-#define PLAY_LOOP(note_array) play_notes(&note_array, NOTE_ARRAY_SIZE((note_array)), true)
+/**
+ * @brief get the number of currently active tones
+ * @return number, 0=none active
+ */
+uint8_t audio_get_number_of_active_tones(void);
+
+/**
+ * @brief access to the raw/unprocessed frequency for a specific tone
+ * @details each active tone has a frequency associated with it, which
+ *          the internal state keeps track of, and is usually influenced
+ *          by various effects
+ * @param[in] tone_index, ranging from 0 to number_of_active_tones-1, with the
+ *            first being the most recent and each increment yielding the next
+ *            older one
+ * @return a positive frequency, in Hz; or zero if the tone is a pause
+ */
+float audio_get_frequency(uint8_t tone_index);
+
+/**
+ * @brief calculate and return the frequency for the requested tone
+ * @details effects like glissando, vibrato, ... are post-processed onto the
+ *          each active tones 'base'-frequency; this function returns the
+ *          post-processed result.
+ * @param[in] tone_index, ranging from 0 to number_of_active_tones-1, with the
+ *            first being the most recent and each increment yielding the next
+ *            older one
+ * @return a positive frequency, in Hz; or zero if the tone is a pause
+ */
+float audio_get_processed_frequency(uint8_t tone_index);
+
+/**
+ * @brief   update audio internal state: currently playing and active tones,...
+ * @details This function is intended to be called by the audio-hardware
+ *          specific implementation on a somewhat regular basis while a SONG
+ *          or notes (pitch+duration) are playing to 'advance' the internal
+ *          state (current playing notes, position in the melody, ...)
+ *
+ * @return true if something changed in the currently active tones, which the
+ *         hardware might need to react to
+ */
+bool audio_update_state(void);
+
+// legacy and back-warts compatibility stuff
+
+#define is_audio_on() audio_is_on()
+#define is_playing_notes() audio_is_playing_melody()
+#define is_playing_note() audio_is_playing_note()
+#define stop_all_notes() audio_stop_all()
+#define stop_note(f) audio_stop_tone(f)
+#define play_note(f, v) audio_play_tone(f)
 
-bool is_playing_notes(void);
+#define set_timbre(t) voice_set_timbre(t)
+#define set_tempo(t) audio_set_tempo(t)
+#define increase_tempo(t) audio_increase_tempo(t)
+#define decrease_tempo(t) audio_decrease_tempo(t)
+// vibrato functions are not used in any keyboards
diff --git a/quantum/audio/audio_chibios.c b/quantum/audio/audio_chibios.c
index 3640423e91..377f93de5d 100644
--- a/quantum/audio/audio_chibios.c
+++ b/quantum/audio/audio_chibios.c
@@ -84,23 +84,27 @@ static void gpt_cb8(GPTDriver *gptp);
 #    define DAC_SAMPLE_MAX 65535U
 #endif
 
-#define START_CHANNEL_1()           \
-    gptStart(&GPTD6, &gpt6cfg1);    \
+#define START_CHANNEL_1()        \
+    gptStart(&GPTD6, &gpt6cfg1); \
     gptStartContinuous(&GPTD6, 2U); \
     palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG)
-#define START_CHANNEL_2()           \
-    gptStart(&GPTD7, &gpt7cfg1);    \
+
+#define START_CHANNEL_2()        \
+    gptStart(&GPTD7, &gpt7cfg1); \
     gptStartContinuous(&GPTD7, 2U); \
     palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG)
-#define STOP_CHANNEL_1()                               \
-    gptStopTimer(&GPTD6);                              \
+
+#define STOP_CHANNEL_1()          \
+    gptStopTimer(&GPTD6);         \
     palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL); \
     palSetPad(GPIOA, 4)
-#define STOP_CHANNEL_2()                               \
-    gptStopTimer(&GPTD7);                              \
+
+#define STOP_CHANNEL_2()         \
+    gptStopTimer(&GPTD7);        \
     palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL); \
     palSetPad(GPIOA, 5)
 
+
 #define RESTART_CHANNEL_1() \
     STOP_CHANNEL_1();       \
     START_CHANNEL_1()
diff --git a/quantum/audio/audio_pwm.c b/quantum/audio/audio_pwm.c
deleted file mode 100644
index d93ac4bb40..0000000000
--- a/quantum/audio/audio_pwm.c
+++ /dev/null
@@ -1,606 +0,0 @@
-/* Copyright 2016 Jack Humbert
- *
- * This program is free software: you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation, either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program.  If not, see <http://www.gnu.org/licenses/>.
- */
-#include <stdio.h>
-#include <string.h>
-//#include <math.h>
-#include <avr/pgmspace.h>
-#include <avr/interrupt.h>
-#include <avr/io.h>
-#include "print.h"
-#include "audio.h"
-#include "keymap.h"
-
-#include "eeconfig.h"
-
-#define PI 3.14159265
-
-#define CPU_PRESCALER 8
-
-#ifndef STARTUP_SONG
-#    define STARTUP_SONG SONG(STARTUP_SOUND)
-#endif
-float startup_song[][2] = STARTUP_SONG;
-
-// Timer Abstractions
-
-// TIMSK3 - Timer/Counter #3 Interrupt Mask Register
-// Turn on/off 3A interputs, stopping/enabling the ISR calls
-#define ENABLE_AUDIO_COUNTER_3_ISR TIMSK3 |= _BV(OCIE3A)
-#define DISABLE_AUDIO_COUNTER_3_ISR TIMSK3 &= ~_BV(OCIE3A)
-
-// TCCR3A: Timer/Counter #3 Control Register
-// Compare Output Mode (COM3An) = 0b00 = Normal port operation, OC3A disconnected from PC6
-#define ENABLE_AUDIO_COUNTER_3_OUTPUT TCCR3A |= _BV(COM3A1);
-#define DISABLE_AUDIO_COUNTER_3_OUTPUT TCCR3A &= ~(_BV(COM3A1) | _BV(COM3A0));
-
-#define NOTE_PERIOD ICR3
-#define NOTE_DUTY_CYCLE OCR3A
-
-#ifdef PWM_AUDIO
-#    include "wave.h"
-#    define SAMPLE_DIVIDER 39
-#    define SAMPLE_RATE (2000000.0 / SAMPLE_DIVIDER / 2048)
-// Resistor value of 1/ (2 * PI * 10nF * (2000000 hertz / SAMPLE_DIVIDER / 10)) for 10nF cap
-
-float    places[8] = {0, 0, 0, 0, 0, 0, 0, 0};
-uint16_t place_int = 0;
-bool     repeat    = true;
-#endif
-
-void delay_us(int count) {
-    while (count--) {
-        _delay_us(1);
-    }
-}
-
-int   voices      = 0;
-int   voice_place = 0;
-float frequency   = 0;
-int   volume      = 0;
-long  position    = 0;
-
-float frequencies[8] = {0, 0, 0, 0, 0, 0, 0, 0};
-int   volumes[8]     = {0, 0, 0, 0, 0, 0, 0, 0};
-bool  sliding        = false;
-
-float place = 0;
-
-uint8_t* sample;
-uint16_t sample_length = 0;
-// float freq = 0;
-
-bool     playing_notes  = false;
-bool     playing_note   = false;
-float    note_frequency = 0;
-float    note_length    = 0;
-uint8_t  note_tempo     = TEMPO_DEFAULT;
-float    note_timbre    = TIMBRE_DEFAULT;
-uint16_t note_position  = 0;
-float (*notes_pointer)[][2];
-uint16_t notes_count;
-bool     notes_repeat;
-float    notes_rest;
-bool     note_resting = false;
-
-uint16_t current_note = 0;
-uint8_t  rest_counter = 0;
-
-#ifdef VIBRATO_ENABLE
-float vibrato_counter  = 0;
-float vibrato_strength = .5;
-float vibrato_rate     = 0.125;
-#endif
-
-float polyphony_rate = 0;
-
-static bool audio_initialized = false;
-
-audio_config_t audio_config;
-
-uint16_t envelope_index = 0;
-
-void audio_init() {
-    // Check EEPROM
-    if (!eeconfig_is_enabled()) {
-        eeconfig_init();
-    }
-    audio_config.raw = eeconfig_read_audio();
-
-#ifdef PWM_AUDIO
-
-    PLLFRQ = _BV(PDIV2);
-    PLLCSR = _BV(PLLE);
-    while (!(PLLCSR & _BV(PLOCK)))
-        ;
-    PLLFRQ |= _BV(PLLTM0); /* PCK 48MHz */
-
-    /* Init a fast PWM on Timer4 */
-    TCCR4A = _BV(COM4A0) | _BV(PWM4A); /* Clear OC4A on Compare Match */
-    TCCR4B = _BV(CS40);                /* No prescaling => f = PCK/256 = 187500Hz */
-    OCR4A  = 0;
-
-    /* Enable the OC4A output */
-    DDRC |= _BV(PORTC6);
-
-    DISABLE_AUDIO_COUNTER_3_ISR;  // Turn off 3A interputs
-
-    TCCR3A = 0x0;                                 // Options not needed
-    TCCR3B = _BV(CS31) | _BV(CS30) | _BV(WGM32);  // 64th prescaling and CTC
-    OCR3A  = SAMPLE_DIVIDER - 1;                  // Correct count/compare, related to sample playback
-
-#else
-
-    // Set port PC6 (OC3A and /OC4A) as output
-    DDRC |= _BV(PORTC6);
-
-    DISABLE_AUDIO_COUNTER_3_ISR;
-
-    // TCCR3A / TCCR3B: Timer/Counter #3 Control Registers
-    // Compare Output Mode (COM3An) = 0b00 = Normal port operation, OC3A disconnected from PC6
-    // Waveform Generation Mode (WGM3n) = 0b1110 = Fast PWM Mode 14 (Period = ICR3, Duty Cycle = OCR3A)
-    // Clock Select (CS3n) = 0b010 = Clock / 8
-    TCCR3A = (0 << COM3A1) | (0 << COM3A0) | (1 << WGM31) | (0 << WGM30);
-    TCCR3B = (1 << WGM33) | (1 << WGM32) | (0 << CS32) | (1 << CS31) | (0 << CS30);
-
-#endif
-
-    audio_initialized = true;
-}
-
-void audio_startup() {
-    if (audio_config.enable) {
-        PLAY_SONG(startup_song);
-    }
-}
-
-void stop_all_notes() {
-    if (!audio_initialized) {
-        audio_init();
-    }
-    voices = 0;
-#ifdef PWM_AUDIO
-    DISABLE_AUDIO_COUNTER_3_ISR;
-#else
-    DISABLE_AUDIO_COUNTER_3_ISR;
-    DISABLE_AUDIO_COUNTER_3_OUTPUT;
-#endif
-
-    playing_notes = false;
-    playing_note  = false;
-    frequency     = 0;
-    volume        = 0;
-
-    for (uint8_t i = 0; i < 8; i++) {
-        frequencies[i] = 0;
-        volumes[i]     = 0;
-    }
-}
-
-void stop_note(float freq) {
-    if (playing_note) {
-        if (!audio_initialized) {
-            audio_init();
-        }
-#ifdef PWM_AUDIO
-        freq = freq / SAMPLE_RATE;
-#endif
-        for (int i = 7; i >= 0; i--) {
-            if (frequencies[i] == freq) {
-                frequencies[i] = 0;
-                volumes[i]     = 0;
-                for (int j = i; (j < 7); j++) {
-                    frequencies[j]     = frequencies[j + 1];
-                    frequencies[j + 1] = 0;
-                    volumes[j]         = volumes[j + 1];
-                    volumes[j + 1]     = 0;
-                }
-                break;
-            }
-        }
-        voices--;
-        if (voices < 0) voices = 0;
-        if (voice_place >= voices) {
-            voice_place = 0;
-        }
-        if (voices == 0) {
-#ifdef PWM_AUDIO
-            DISABLE_AUDIO_COUNTER_3_ISR;
-#else
-            DISABLE_AUDIO_COUNTER_3_ISR;
-            DISABLE_AUDIO_COUNTER_3_OUTPUT;
-#endif
-            frequency    = 0;
-            volume       = 0;
-            playing_note = false;
-        }
-    }
-}
-
-#ifdef VIBRATO_ENABLE
-
-float mod(float a, int b) {
-    float r = fmod(a, b);
-    return r < 0 ? r + b : r;
-}
-
-float vibrato(float average_freq) {
-#    ifdef VIBRATO_STRENGTH_ENABLE
-    float vibrated_freq = average_freq * pow(vibrato_lut[(int)vibrato_counter], vibrato_strength);
-#    else
-    float vibrated_freq = average_freq * vibrato_lut[(int)vibrato_counter];
-#    endif
-    vibrato_counter = mod((vibrato_counter + vibrato_rate * (1.0 + 440.0 / average_freq)), VIBRATO_LUT_LENGTH);
-    return vibrated_freq;
-}
-
-#endif
-
-ISR(TIMER3_COMPA_vect) {
-    if (playing_note) {
-#ifdef PWM_AUDIO
-        if (voices == 1) {
-            // SINE
-            OCR4A = pgm_read_byte(&sinewave[(uint16_t)place]) >> 2;
-
-            // SQUARE
-            // if (((int)place) >= 1024){
-            //     OCR4A = 0xFF >> 2;
-            // } else {
-            //     OCR4A = 0x00;
-            // }
-
-            // SAWTOOTH
-            // OCR4A = (int)place / 4;
-
-            // TRIANGLE
-            // if (((int)place) >= 1024) {
-            //     OCR4A = (int)place / 2;
-            // } else {
-            //     OCR4A = 2048 - (int)place / 2;
-            // }
-
-            place += frequency;
-
-            if (place >= SINE_LENGTH) place -= SINE_LENGTH;
-
-        } else {
-            int sum = 0;
-            for (int i = 0; i < voices; i++) {
-                // SINE
-                sum += pgm_read_byte(&sinewave[(uint16_t)places[i]]) >> 2;
-
-                // SQUARE
-                // if (((int)places[i]) >= 1024){
-                //     sum += 0xFF >> 2;
-                // } else {
-                //     sum += 0x00;
-                // }
-
-                places[i] += frequencies[i];
-
-                if (places[i] >= SINE_LENGTH) places[i] -= SINE_LENGTH;
-            }
-            OCR4A = sum;
-        }
-#else
-        if (voices > 0) {
-            float freq;
-            if (polyphony_rate > 0) {
-                if (voices > 1) {
-                    voice_place %= voices;
-                    if (place++ > (frequencies[voice_place] / polyphony_rate / CPU_PRESCALER)) {
-                        voice_place = (voice_place + 1) % voices;
-                        place       = 0.0;
-                    }
-                }
-#    ifdef VIBRATO_ENABLE
-                if (vibrato_strength > 0) {
-                    freq = vibrato(frequencies[voice_place]);
-                } else {
-#    else
-                {
-#    endif
-                    freq = frequencies[voice_place];
-                }
-            } else {
-                if (frequency != 0 && frequency < frequencies[voices - 1] && frequency < frequencies[voices - 1] * pow(2, -440 / frequencies[voices - 1] / 12 / 2)) {
-                    frequency = frequency * pow(2, 440 / frequency / 12 / 2);
-                } else if (frequency != 0 && frequency > frequencies[voices - 1] && frequency > frequencies[voices - 1] * pow(2, 440 / frequencies[voices - 1] / 12 / 2)) {
-                    frequency = frequency * pow(2, -440 / frequency / 12 / 2);
-                } else {
-                    frequency = frequencies[voices - 1];
-                }
-
-#    ifdef VIBRATO_ENABLE
-                if (vibrato_strength > 0) {
-                    freq = vibrato(frequency);
-                } else {
-#    else
-                {
-#    endif
-                    freq = frequency;
-                }
-            }
-
-            if (envelope_index < 65535) {
-                envelope_index++;
-            }
-            freq = voice_envelope(freq);
-
-            if (freq < 30.517578125) freq = 30.52;
-            NOTE_PERIOD     = (int)(((double)F_CPU) / (freq * CPU_PRESCALER));                  // Set max to the period
-            NOTE_DUTY_CYCLE = (int)((((double)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre);  // Set compare to half the period
-        }
-#endif
-    }
-
-    // SAMPLE
-    // OCR4A = pgm_read_byte(&sample[(uint16_t)place_int]);
-
-    // place_int++;
-
-    // if (place_int >= sample_length)
-    //     if (repeat)
-    //         place_int -= sample_length;
-    //     else
-    //         DISABLE_AUDIO_COUNTER_3_ISR;
-
-    if (playing_notes) {
-#ifdef PWM_AUDIO
-        OCR4A = pgm_read_byte(&sinewave[(uint16_t)place]) >> 0;
-
-        place += note_frequency;
-        if (place >= SINE_LENGTH) place -= SINE_LENGTH;
-#else
-        if (note_frequency > 0) {
-            float freq;
-
-#    ifdef VIBRATO_ENABLE
-            if (vibrato_strength > 0) {
-                freq = vibrato(note_frequency);
-            } else {
-#    else
-            {
-#    endif
-                freq = note_frequency;
-            }
-
-            if (envelope_index < 65535) {
-                envelope_index++;
-            }
-            freq = voice_envelope(freq);
-
-            NOTE_PERIOD     = (int)(((double)F_CPU) / (freq * CPU_PRESCALER));                  // Set max to the period
-            NOTE_DUTY_CYCLE = (int)((((double)F_CPU) / (freq * CPU_PRESCALER)) * note_timbre);  // Set compare to half the period
-        } else {
-            NOTE_PERIOD     = 0;
-            NOTE_DUTY_CYCLE = 0;
-        }
-#endif
-
-        note_position++;
-        bool end_of_note = false;
-        if (NOTE_PERIOD > 0)
-            end_of_note = (note_position >= (note_length / NOTE_PERIOD * 0xFFFF));
-        else
-            end_of_note = (note_position >= (note_length * 0x7FF));
-        if (end_of_note) {
-            current_note++;
-            if (current_note >= notes_count) {
-                if (notes_repeat) {
-                    current_note = 0;
-                } else {
-#ifdef PWM_AUDIO
-                    DISABLE_AUDIO_COUNTER_3_ISR;
-#else
-                    DISABLE_AUDIO_COUNTER_3_ISR;
-                    DISABLE_AUDIO_COUNTER_3_OUTPUT;
-#endif
-                    playing_notes = false;
-                    return;
-                }
-            }
-            if (!note_resting && (notes_rest > 0)) {
-                note_resting   = true;
-                note_frequency = 0;
-                note_length    = notes_rest;
-                current_note--;
-            } else {
-                note_resting = false;
-#ifdef PWM_AUDIO
-                note_frequency = (*notes_pointer)[current_note][0] / SAMPLE_RATE;
-                note_length    = (*notes_pointer)[current_note][1] * (((float)note_tempo) / 100);
-#else
-                envelope_index = 0;
-                note_frequency = (*notes_pointer)[current_note][0];
-                note_length    = ((*notes_pointer)[current_note][1] / 4) * (((float)note_tempo) / 100);
-#endif
-            }
-            note_position = 0;
-        }
-    }
-
-    if (!audio_config.enable) {
-        playing_notes = false;
-        playing_note  = false;
-    }
-}
-
-void play_note(float freq, int vol) {
-    if (!audio_initialized) {
-        audio_init();
-    }
-
-    if (audio_config.enable && voices < 8) {
-        DISABLE_AUDIO_COUNTER_3_ISR;
-
-        // Cancel notes if notes are playing
-        if (playing_notes) stop_all_notes();
-
-        playing_note = true;
-
-        envelope_index = 0;
-
-#ifdef PWM_AUDIO
-        freq = freq / SAMPLE_RATE;
-#endif
-        if (freq > 0) {
-            frequencies[voices] = freq;
-            volumes[voices]     = vol;
-            voices++;
-        }
-
-#ifdef PWM_AUDIO
-        ENABLE_AUDIO_COUNTER_3_ISR;
-#else
-        ENABLE_AUDIO_COUNTER_3_ISR;
-        ENABLE_AUDIO_COUNTER_3_OUTPUT;
-#endif
-    }
-}
-
-void play_notes(float (*np)[][2], uint16_t n_count, bool n_repeat, float n_rest) {
-    if (!audio_initialized) {
-        audio_init();
-    }
-
-    if (audio_config.enable) {
-        DISABLE_AUDIO_COUNTER_3_ISR;
-
-        // Cancel note if a note is playing
-        if (playing_note) stop_all_notes();
-
-        playing_notes = true;
-
-        notes_pointer = np;
-        notes_count   = n_count;
-        notes_repeat  = n_repeat;
-        notes_rest    = n_rest;
-
-        place        = 0;
-        current_note = 0;
-
-#ifdef PWM_AUDIO
-        note_frequency = (*notes_pointer)[current_note][0] / SAMPLE_RATE;
-        note_length    = (*notes_pointer)[current_note][1] * (((float)note_tempo) / 100);
-#else
-        note_frequency = (*notes_pointer)[current_note][0];
-        note_length    = ((*notes_pointer)[current_note][1] / 4) * (((float)note_tempo) / 100);
-#endif
-        note_position = 0;
-
-#ifdef PWM_AUDIO
-        ENABLE_AUDIO_COUNTER_3_ISR;
-#else
-        ENABLE_AUDIO_COUNTER_3_ISR;
-        ENABLE_AUDIO_COUNTER_3_OUTPUT;
-#endif
-    }
-}
-
-#ifdef PWM_AUDIO
-void play_sample(uint8_t* s, uint16_t l, bool r) {
-    if (!audio_initialized) {
-        audio_init();
-    }
-
-    if (audio_config.enable) {
-        DISABLE_AUDIO_COUNTER_3_ISR;
-        stop_all_notes();
-        place_int     = 0;
-        sample        = s;
-        sample_length = l;
-        repeat        = r;
-
-        ENABLE_AUDIO_COUNTER_3_ISR;
-    }
-}
-#endif
-
-void audio_toggle(void) {
-    audio_config.enable ^= 1;
-    eeconfig_update_audio(audio_config.raw);
-}
-
-void audio_on(void) {
-    audio_config.enable = 1;
-    eeconfig_update_audio(audio_config.raw);
-}
-
-void audio_off(void) {
-    audio_config.enable = 0;
-    eeconfig_update_audio(audio_config.raw);
-}
-
-#ifdef VIBRATO_ENABLE
-
-// Vibrato rate functions
-
-void set_vibrato_rate(float rate) { vibrato_rate = rate; }
-
-void increase_vibrato_rate(float change) { vibrato_rate *= change; }
-
-void decrease_vibrato_rate(float change) { vibrato_rate /= change; }
-
-#    ifdef VIBRATO_STRENGTH_ENABLE
-
-void set_vibrato_strength(float strength) { vibrato_strength = strength; }
-
-void increase_vibrato_strength(float change) { vibrato_strength *= change; }
-
-void decrease_vibrato_strength(float change) { vibrato_strength /= change; }
-
-#    endif /* VIBRATO_STRENGTH_ENABLE */
-
-#endif /* VIBRATO_ENABLE */
-
-// Polyphony functions
-
-void set_polyphony_rate(float rate) { polyphony_rate = rate; }
-
-void enable_polyphony() { polyphony_rate = 5; }
-
-void disable_polyphony() { polyphony_rate = 0; }
-
-void increase_polyphony_rate(float change) { polyphony_rate *= change; }
-
-void decrease_polyphony_rate(float change) { polyphony_rate /= change; }
-
-// Timbre function
-
-void set_timbre(float timbre) { note_timbre = timbre; }
-
-// Tempo functions
-
-void set_tempo(uint8_t tempo) { note_tempo = tempo; }
-
-void decrease_tempo(uint8_t tempo_change) { note_tempo += tempo_change; }
-
-void increase_tempo(uint8_t tempo_change) {
-    if (note_tempo - tempo_change < 10) {
-        note_tempo = 10;
-    } else {
-        note_tempo -= tempo_change;
-    }
-}
-
-//------------------------------------------------------------------------------
-// Override these functions in your keymap file to play different tunes on
-// startup and bootloader jump
-__attribute__((weak)) void play_startup_tone() {}
-
-__attribute__((weak)) void play_goodbye_tone() {}
-//------------------------------------------------------------------------------
diff --git a/quantum/audio/driver_avr_pwm.h b/quantum/audio/driver_avr_pwm.h
new file mode 100644
index 0000000000..d6eb3571da
--- /dev/null
+++ b/quantum/audio/driver_avr_pwm.h
@@ -0,0 +1,17 @@
+/* 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
new file mode 100644
index 0000000000..df03a4558c
--- /dev/null
+++ b/quantum/audio/driver_avr_pwm_hardware.c
@@ -0,0 +1,332 @@
+/* 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
new file mode 100644
index 0000000000..07cd622ead
--- /dev/null
+++ b/quantum/audio/driver_chibios_dac.h
@@ -0,0 +1,126 @@
+/* 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
new file mode 100644
index 0000000000..db304adb87
--- /dev/null
+++ b/quantum/audio/driver_chibios_dac_additive.c
@@ -0,0 +1,335 @@
+/* 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
+     * Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
+     * Note: enabling the output buffer imparts an additional dc-offset of a couple mV
+     *
+     * this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
+     * (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
+     */
+    DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
+    DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
+
+    if (AUDIO_PIN == A4) {
+        dacStartConversion(&DACD1, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
+    } else if (AUDIO_PIN == A5) {
+        dacStartConversion(&DACD2, &dac_conv_cfg, dac_buffer_empty, AUDIO_DAC_BUFFER_SIZE);
+    }
+
+    // no inverted/out-of-phase waveform (yet?), only pulling AUDIO_PIN_ALT to AUDIO_DAC_OFF_VALUE
+#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+    if (AUDIO_PIN_ALT == A4) {
+        dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
+    } else if (AUDIO_PIN_ALT == A5) {
+        dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
+    }
+#endif
+
+    gptStart(&GPTD6, &gpt6cfg1);
+}
+
+void audio_driver_stop(void) { state = OUTPUT_SHOULD_STOP; }
+
+void audio_driver_start(void) {
+    gptStartContinuous(&GPTD6, 2U);
+
+    for (uint8_t i = 0; i < AUDIO_MAX_SIMULTANEOUS_TONES; i++) {
+        dac_if[i]                = 0.0f;
+        active_tones_snapshot[i] = 0.0f;
+    }
+    active_tones_snapshot_length = 0;
+    state                        = OUTPUT_SHOULD_START;
+}
diff --git a/quantum/audio/driver_chibios_dac_basic.c b/quantum/audio/driver_chibios_dac_basic.c
new file mode 100644
index 0000000000..fac6513506
--- /dev/null
+++ b/quantum/audio/driver_chibios_dac_basic.c
@@ -0,0 +1,245 @@
+/* Copyright 2016-2020 Jack Humbert
+ * Copyright 2020 JohSchneider
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "audio.h"
+#include "ch.h"
+#include "hal.h"
+
+/*
+  Audio Driver: DAC
+
+  which utilizes both channels of the DAC unit many STM32 are equipped with to output a modulated square-wave, from precomputed samples stored in a buffer, which is passed to the hardware through DMA
+
+  this driver can either be used to drive to separate speakers, wired to A4+Gnd and A5+Gnd, which allows two tones to be played simultaneously
+  OR
+  one speaker wired to A4+A5 with the AUDIO_PIN_ALT_AS_NEGATIVE define set - see docs/feature_audio
+
+*/
+
+#if !defined(AUDIO_PIN)
+#    pragma message "Audio feature enabled, but no suitable pin selected as AUDIO_PIN - see docs/feature_audio under 'ARM (DAC basic)' for available options."
+// TODO: make this an 'error' instead; go through a breaking change, and add AUDIO_PIN A5 to all keyboards currently using AUDIO on STM32 based boards? - for now: set the define here
+#    define AUDIO_PIN A5
+#endif
+// check configuration for ONE speaker, connected to both DAC pins
+#if defined(AUDIO_PIN_ALT_AS_NEGATIVE) && !defined(AUDIO_PIN_ALT)
+#    error "Audio feature: AUDIO_PIN_ALT_AS_NEGATIVE set, but no pin configured as AUDIO_PIN_ALT"
+#endif
+
+#ifndef AUDIO_PIN_ALT
+// no ALT pin defined is valid, but the c-ifs below need some value set
+#    define AUDIO_PIN_ALT -1
+#endif
+
+#if !defined(AUDIO_STATE_TIMER)
+#    define AUDIO_STATE_TIMER GPTD8
+#endif
+
+// square-wave
+static const dacsample_t dac_buffer_1[AUDIO_DAC_BUFFER_SIZE] = {
+    // First half is max, second half is 0
+    [0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1]                     = AUDIO_DAC_SAMPLE_MAX,
+    [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = 0,
+};
+
+// square-wave
+static const dacsample_t dac_buffer_2[AUDIO_DAC_BUFFER_SIZE] = {
+    // opposite of dac_buffer above
+    [0 ... AUDIO_DAC_BUFFER_SIZE / 2 - 1]                     = 0,
+    [AUDIO_DAC_BUFFER_SIZE / 2 ... AUDIO_DAC_BUFFER_SIZE - 1] = AUDIO_DAC_SAMPLE_MAX,
+};
+
+GPTConfig gpt6cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
+                      .callback  = NULL,
+                      .cr2       = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event.    */
+                      .dier      = 0U};
+GPTConfig gpt7cfg1 = {.frequency = AUDIO_DAC_SAMPLE_RATE,
+                      .callback  = NULL,
+                      .cr2       = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event.    */
+                      .dier      = 0U};
+
+static void gpt_audio_state_cb(GPTDriver *gptp);
+GPTConfig   gptStateUpdateCfg = {.frequency = 10,
+                               .callback  = gpt_audio_state_cb,
+                               .cr2       = TIM_CR2_MMS_1, /* MMS = 010 = TRGO on Update Event.    */
+                               .dier      = 0U};
+
+static const DACConfig dac_conf_ch1 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
+static const DACConfig dac_conf_ch2 = {.init = AUDIO_DAC_OFF_VALUE, .datamode = DAC_DHRM_12BIT_RIGHT};
+
+/**
+ * @note The DAC_TRG(0) here selects the Timer 6 TRGO event, which is triggered
+ * on the rising edge after 3 APB1 clock cycles, causing our gpt6cfg1.frequency
+ * to be a third of what we expect.
+ *
+ * Here are all the values for DAC_TRG (TSEL in the ref manual)
+ * TIM15_TRGO 0b011
+ * TIM2_TRGO  0b100
+ * TIM3_TRGO  0b001
+ * TIM6_TRGO  0b000
+ * TIM7_TRGO  0b010
+ * EXTI9      0b110
+ * SWTRIG     0b111
+ */
+static const DACConversionGroup dac_conv_grp_ch1 = {.num_channels = 1U, .trigger = DAC_TRG(0b000)};
+static const DACConversionGroup dac_conv_grp_ch2 = {.num_channels = 1U, .trigger = DAC_TRG(0b010)};
+
+void channel_1_start(void) {
+    gptStart(&GPTD6, &gpt6cfg1);
+    gptStartContinuous(&GPTD6, 2U);
+    palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
+}
+
+void channel_1_stop(void) {
+    gptStopTimer(&GPTD6);
+    palSetPadMode(GPIOA, 4, PAL_MODE_OUTPUT_PUSHPULL);
+    palSetPad(GPIOA, 4);
+}
+
+static float channel_1_frequency = 0.0f;
+void         channel_1_set_frequency(float freq) {
+    channel_1_frequency = freq;
+
+    channel_1_stop();
+    if (freq <= 0.0)  // a pause/rest has freq=0
+        return;
+
+    gpt6cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
+    channel_1_start();
+}
+float channel_1_get_frequency(void) { return channel_1_frequency; }
+
+void channel_2_start(void) {
+    gptStart(&GPTD7, &gpt7cfg1);
+    gptStartContinuous(&GPTD7, 2U);
+    palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
+}
+
+void channel_2_stop(void) {
+    gptStopTimer(&GPTD7);
+    palSetPadMode(GPIOA, 5, PAL_MODE_OUTPUT_PUSHPULL);
+    palSetPad(GPIOA, 5);
+}
+
+static float channel_2_frequency = 0.0f;
+void         channel_2_set_frequency(float freq) {
+    channel_2_frequency = freq;
+
+    channel_2_stop();
+    if (freq <= 0.0)  // a pause/rest has freq=0
+        return;
+
+    gpt7cfg1.frequency = 2 * freq * AUDIO_DAC_BUFFER_SIZE;
+    channel_2_start();
+}
+float channel_2_get_frequency(void) { return channel_2_frequency; }
+
+static void gpt_audio_state_cb(GPTDriver *gptp) {
+    if (audio_update_state()) {
+#if defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+        // one piezo/speaker connected to both audio pins, the generated square-waves are inverted
+        channel_1_set_frequency(audio_get_processed_frequency(0));
+        channel_2_set_frequency(audio_get_processed_frequency(0));
+
+#else  // two separate audio outputs/speakers
+       // primary speaker on A4, optional secondary on A5
+        if (AUDIO_PIN == A4) {
+            channel_1_set_frequency(audio_get_processed_frequency(0));
+            if (AUDIO_PIN_ALT == A5) {
+                if (audio_get_number_of_active_tones() > 1) {
+                    channel_2_set_frequency(audio_get_processed_frequency(1));
+                } else {
+                    channel_2_stop();
+                }
+            }
+        }
+
+        // primary speaker on A5, optional secondary on A4
+        if (AUDIO_PIN == A5) {
+            channel_2_set_frequency(audio_get_processed_frequency(0));
+            if (AUDIO_PIN_ALT == A4) {
+                if (audio_get_number_of_active_tones() > 1) {
+                    channel_1_set_frequency(audio_get_processed_frequency(1));
+                } else {
+                    channel_1_stop();
+                }
+            }
+        }
+#endif
+    }
+}
+
+void audio_driver_initialize() {
+    if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
+        palSetPadMode(GPIOA, 4, PAL_MODE_INPUT_ANALOG);
+        dacStart(&DACD1, &dac_conf_ch1);
+
+        // initial setup of the dac-triggering timer is still required, even
+        // though it gets reconfigured and restarted later on
+        gptStart(&GPTD6, &gpt6cfg1);
+    }
+
+    if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
+        palSetPadMode(GPIOA, 5, PAL_MODE_INPUT_ANALOG);
+        dacStart(&DACD2, &dac_conf_ch2);
+
+        gptStart(&GPTD7, &gpt7cfg1);
+    }
+
+    /* enable the output buffer, to directly drive external loads with no additional circuitry
+     *
+     * see: AN4566 Application note: Extending the DAC performance of STM32 microcontrollers
+     * Note: Buffer-Off bit -> has to be set 0 to enable the output buffer
+     * Note: enabling the output buffer imparts an additional dc-offset of a couple mV
+     *
+     * this is done here, reaching directly into the stm32 registers since chibios has not implemented BOFF handling yet
+     * (see: chibios/os/hal/ports/STM32/todo.txt '- BOFF handling in DACv1.'
+     */
+    DACD1.params->dac->CR &= ~DAC_CR_BOFF1;
+    DACD2.params->dac->CR &= ~DAC_CR_BOFF2;
+
+    // start state-updater
+    gptStart(&AUDIO_STATE_TIMER, &gptStateUpdateCfg);
+}
+
+void audio_driver_stop(void) {
+    if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
+        gptStopTimer(&GPTD6);
+
+        // stop the ongoing conversion and put the output in a known state
+        dacStopConversion(&DACD1);
+        dacPutChannelX(&DACD1, 0, AUDIO_DAC_OFF_VALUE);
+    }
+
+    if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
+        gptStopTimer(&GPTD7);
+
+        dacStopConversion(&DACD2);
+        dacPutChannelX(&DACD2, 0, AUDIO_DAC_OFF_VALUE);
+    }
+    gptStopTimer(&AUDIO_STATE_TIMER);
+}
+
+void audio_driver_start(void) {
+    if ((AUDIO_PIN == A4) || (AUDIO_PIN_ALT == A4)) {
+        dacStartConversion(&DACD1, &dac_conv_grp_ch1, (dacsample_t *)dac_buffer_1, AUDIO_DAC_BUFFER_SIZE);
+    }
+    if ((AUDIO_PIN == A5) || (AUDIO_PIN_ALT == A5)) {
+        dacStartConversion(&DACD2, &dac_conv_grp_ch2, (dacsample_t *)dac_buffer_2, AUDIO_DAC_BUFFER_SIZE);
+    }
+    gptStartContinuous(&AUDIO_STATE_TIMER, 2U);
+}
diff --git a/quantum/audio/driver_chibios_pwm.h b/quantum/audio/driver_chibios_pwm.h
new file mode 100644
index 0000000000..86cab916e1
--- /dev/null
+++ b/quantum/audio/driver_chibios_pwm.h
@@ -0,0 +1,40 @@
+/* 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
+
+#if !defined(AUDIO_PWM_DRIVER)
+// NOTE: Timer2 seems to be used otherwise in QMK, otherwise we could default to A5 (= TIM2_CH1, with PWMD2 and alternate-function(1))
+#    define AUDIO_PWM_DRIVER PWMD1
+#endif
+
+#if !defined(AUDIO_PWM_CHANNEL)
+// NOTE: sticking to the STM data-sheet numbering: TIMxCH1 to TIMxCH4
+// default: STM32F303CC PA8+TIM1_CH1 -> 1
+#    define AUDIO_PWM_CHANNEL 1
+#endif
+
+#if !defined(AUDIO_PWM_PAL_MODE)
+// pin-alternate function: see the data-sheet for which pin needs what AF to connect to TIMx_CHy
+// default: STM32F303CC PA8+TIM1_CH1 -> 6
+#    define AUDIO_PWM_PAL_MODE 6
+#endif
+
+#if !defined(AUDIO_STATE_TIMER)
+// timer used to trigger updates in the audio-system, configured/enabled in chibios mcuconf.
+// Tim6 is the default for "larger" STMs, smaller ones might not have this one (enabled) and need to switch to a different one (e.g.: STM32F103 has only Tim1-Tim4)
+#    define AUDIO_STATE_TIMER GPTD6
+#endif
diff --git a/quantum/audio/driver_chibios_pwm_hardware.c b/quantum/audio/driver_chibios_pwm_hardware.c
new file mode 100644
index 0000000000..3c7d89b290
--- /dev/null
+++ b/quantum/audio/driver_chibios_pwm_hardware.c
@@ -0,0 +1,144 @@
+/* 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/>.
+ */
+
+/*
+Audio Driver: PWM
+
+the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
+
+this driver uses the chibios-PWM system to produce a square-wave on specific output pins that are connected to the PWM hardware.
+The hardware directly toggles the pin via its alternate function. see your MCUs data-sheet for which pin can be driven by what timer - looking for TIMx_CHy and the corresponding alternate function.
+
+ */
+
+#include "audio.h"
+#include "ch.h"
+#include "hal.h"
+
+#if !defined(AUDIO_PIN)
+#    error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
+#endif
+
+extern bool    playing_note;
+extern bool    playing_melody;
+extern uint8_t note_timbre;
+
+static PWMConfig pwmCFG = {
+    .frequency = 100000, /* PWM clock frequency  */
+    // CHIBIOS-BUG? can't set the initial period to <2, or the pwm (hard or software) takes ~130ms with .frequency=500000 for a pwmChangePeriod to take effect; with no output=silence in the meantime
+    .period   = 2,    /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
+    .callback = NULL, /* no callback, the hardware directly toggles the pin */
+    .channels =
+        {
+#if AUDIO_PWM_CHANNEL == 4
+            {PWM_OUTPUT_DISABLED, NULL},   /* channel 0 -> TIMx_CH1 */
+            {PWM_OUTPUT_DISABLED, NULL},   /* channel 1 -> TIMx_CH2 */
+            {PWM_OUTPUT_DISABLED, NULL},   /* channel 2 -> TIMx_CH3 */
+            {PWM_OUTPUT_ACTIVE_HIGH, NULL} /* channel 3 -> TIMx_CH4 */
+#elif AUDIO_PWM_CHANNEL == 3
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH3 */
+            {PWM_OUTPUT_DISABLED, NULL}
+#elif AUDIO_PWM_CHANNEL == 2
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH2 */
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_DISABLED, NULL}
+#else /*fallback to CH1 */
+            {PWM_OUTPUT_ACTIVE_HIGH, NULL}, /* TIMx_CH1 */
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_DISABLED, NULL},
+            {PWM_OUTPUT_DISABLED, NULL}
+#endif
+        },
+};
+
+static float channel_1_frequency = 0.0f;
+void         channel_1_set_frequency(float freq) {
+    channel_1_frequency = freq;
+
+    if (freq <= 0.0)  // a pause/rest has freq=0
+        return;
+
+    pwmcnt_t period = (pwmCFG.frequency / freq);
+    pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
+    pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
+                     // adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
+                     PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
+}
+
+float channel_1_get_frequency(void) { return channel_1_frequency; }
+
+void channel_1_start(void) {
+    pwmStop(&AUDIO_PWM_DRIVER);
+    pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
+}
+
+void channel_1_stop(void) { pwmStop(&AUDIO_PWM_DRIVER); }
+
+static void gpt_callback(GPTDriver *gptp);
+GPTConfig   gptCFG = {
+    /* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
+       the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
+       the tempo (which might vary!) is in bpm (beats per minute)
+       therefore: if the timer ticks away at .frequency = (60*64)Hz,
+       and the .interval counts from 64 downwards - audio_update_state is
+       called just often enough to not miss any notes
+    */
+    .frequency = 60 * 64,
+    .callback  = gpt_callback,
+};
+
+void audio_driver_initialize(void) {
+    pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
+
+    // connect the AUDIO_PIN to the PWM hardware
+#if defined(USE_GPIOV1)  // STM32F103C8
+    palSetLineMode(AUDIO_PIN, PAL_MODE_STM32_ALTERNATE_PUSHPULL);
+#else  // GPIOv2 (or GPIOv3 for f4xx, which is the same/compatible at this command)
+    palSetLineMode(AUDIO_PIN, PAL_STM32_MODE_ALTERNATE | PAL_STM32_ALTERNATE(AUDIO_PWM_PAL_MODE));
+#endif
+
+    gptStart(&AUDIO_STATE_TIMER, &gptCFG);
+}
+
+void audio_driver_start(void) {
+    channel_1_stop();
+    channel_1_start();
+
+    if (playing_note || playing_melody) {
+        gptStartContinuous(&AUDIO_STATE_TIMER, 64);
+    }
+}
+
+void audio_driver_stop(void) {
+    channel_1_stop();
+    gptStopTimer(&AUDIO_STATE_TIMER);
+}
+
+/* a regular timer task, that checks the note to be currently played
+ * and updates the pwm to output that frequency
+ */
+static void gpt_callback(GPTDriver *gptp) {
+    float freq;  // TODO: freq_alt
+
+    if (audio_update_state()) {
+        freq = audio_get_processed_frequency(0);  // freq_alt would be index=1
+        channel_1_set_frequency(freq);
+    }
+}
diff --git a/quantum/audio/driver_chibios_pwm_software.c b/quantum/audio/driver_chibios_pwm_software.c
new file mode 100644
index 0000000000..15c3e98b6a
--- /dev/null
+++ b/quantum/audio/driver_chibios_pwm_software.c
@@ -0,0 +1,164 @@
+/* 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/>.
+ */
+
+/*
+Audio Driver: PWM
+
+the duty-cycle is always kept at 50%, and the pwm-period is adjusted to match the frequency of a note to be played back.
+
+this driver uses the chibios-PWM system to produce a square-wave on any given output pin in software
+- a pwm callback is used to set/clear the configured pin.
+
+ */
+#include "audio.h"
+#include "ch.h"
+#include "hal.h"
+
+#if !defined(AUDIO_PIN)
+#    error "Audio feature enabled, but no pin selected - see docs/feature_audio under the ARM PWM settings"
+#endif
+extern bool    playing_note;
+extern bool    playing_melody;
+extern uint8_t note_timbre;
+
+static void pwm_audio_period_callback(PWMDriver *pwmp);
+static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp);
+
+static PWMConfig pwmCFG = {
+    .frequency = 100000, /* PWM clock frequency  */
+    // CHIBIOS-BUG? can't set the initial period to <2, or the pwm (hard or software) takes ~130ms with .frequency=500000 for a pwmChangePeriod to take effect; with no output=silence in the meantime
+    .period   = 2, /* initial PWM period (in ticks) 1S (1/10kHz=0.1mS 0.1ms*10000 ticks=1S) */
+    .callback = pwm_audio_period_callback,
+    .channels =
+        {
+            // software-PWM just needs another callback on any channel
+            {PWM_OUTPUT_ACTIVE_HIGH, pwm_audio_channel_interrupt_callback}, /* channel 0 -> TIMx_CH1 */
+            {PWM_OUTPUT_DISABLED, NULL},                                    /* channel 1 -> TIMx_CH2 */
+            {PWM_OUTPUT_DISABLED, NULL},                                    /* channel 2 -> TIMx_CH3 */
+            {PWM_OUTPUT_DISABLED, NULL}                                     /* channel 3 -> TIMx_CH4 */
+        },
+};
+
+static float channel_1_frequency = 0.0f;
+void         channel_1_set_frequency(float freq) {
+    channel_1_frequency = freq;
+
+    if (freq <= 0.0)  // a pause/rest has freq=0
+        return;
+
+    pwmcnt_t period = (pwmCFG.frequency / freq);
+    pwmChangePeriod(&AUDIO_PWM_DRIVER, period);
+
+    pwmEnableChannel(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1,
+                     // adjust the duty-cycle so that the output is for 'note_timbre' duration HIGH
+                     PWM_PERCENTAGE_TO_WIDTH(&AUDIO_PWM_DRIVER, (100 - note_timbre) * 100));
+}
+
+float channel_1_get_frequency(void) { return channel_1_frequency; }
+
+void channel_1_start(void) {
+    pwmStop(&AUDIO_PWM_DRIVER);
+    pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
+
+    pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER);
+    pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
+}
+
+void channel_1_stop(void) {
+    pwmStop(&AUDIO_PWM_DRIVER);
+
+    palClearLine(AUDIO_PIN);  // leave the line low, after last note was played
+
+#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+    palClearLine(AUDIO_PIN_ALT);  // leave the line low, after last note was played
+#endif
+}
+
+// generate a PWM signal on any pin, not necessarily the one connected to the timer
+static void pwm_audio_period_callback(PWMDriver *pwmp) {
+    (void)pwmp;
+    palClearLine(AUDIO_PIN);
+
+#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+    palSetLine(AUDIO_PIN_ALT);
+#endif
+}
+static void pwm_audio_channel_interrupt_callback(PWMDriver *pwmp) {
+    (void)pwmp;
+    if (channel_1_frequency > 0) {
+        palSetLine(AUDIO_PIN);  // generate a PWM signal on any pin, not necessarily the one connected to the timer
+#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+        palClearLine(AUDIO_PIN_ALT);
+#endif
+    }
+}
+
+static void gpt_callback(GPTDriver *gptp);
+GPTConfig   gptCFG = {
+    /* a whole note is one beat, which is - per definition in musical_notes.h - set to 64
+       the longest note is BREAVE_DOT=128+64=192, the shortest SIXTEENTH=4
+       the tempo (which might vary!) is in bpm (beats per minute)
+       therefore: if the timer ticks away at .frequency = (60*64)Hz,
+       and the .interval counts from 64 downwards - audio_update_state is
+       called just often enough to not miss anything
+    */
+    .frequency = 60 * 64,
+    .callback  = gpt_callback,
+};
+
+void audio_driver_initialize(void) {
+    pwmStart(&AUDIO_PWM_DRIVER, &pwmCFG);
+
+    palSetLineMode(AUDIO_PIN, PAL_MODE_OUTPUT_PUSHPULL);
+    palClearLine(AUDIO_PIN);
+
+#if defined(AUDIO_PIN_ALT) && defined(AUDIO_PIN_ALT_AS_NEGATIVE)
+    palSetLineMode(AUDIO_PIN_ALT, PAL_MODE_OUTPUT_PUSHPULL);
+    palClearLine(AUDIO_PIN_ALT);
+#endif
+
+    pwmEnablePeriodicNotification(&AUDIO_PWM_DRIVER);  // enable pwm callbacks
+    pwmEnableChannelNotification(&AUDIO_PWM_DRIVER, AUDIO_PWM_CHANNEL - 1);
+
+    gptStart(&AUDIO_STATE_TIMER, &gptCFG);
+}
+
+void audio_driver_start(void) {
+    channel_1_stop();
+    channel_1_start();
+
+    if (playing_note || playing_melody) {
+        gptStartContinuous(&AUDIO_STATE_TIMER, 64);
+    }
+}
+
+void audio_driver_stop(void) {
+    channel_1_stop();
+    gptStopTimer(&AUDIO_STATE_TIMER);
+}
+
+/* a regular timer task, that checks the note to be currently played
+ * and updates the pwm to output that frequency
+ */
+static void gpt_callback(GPTDriver *gptp) {
+    float freq;  // TODO: freq_alt
+
+    if (audio_update_state()) {
+        freq = audio_get_processed_frequency(0);  // freq_alt would be index=1
+        channel_1_set_frequency(freq);
+    }
+}
diff --git a/quantum/audio/musical_notes.h b/quantum/audio/musical_notes.h
index 0ba572c346..ddd7d374f5 100644
--- a/quantum/audio/musical_notes.h
+++ b/quantum/audio/musical_notes.h
@@ -1,4 +1,5 @@
 /* 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
@@ -13,12 +14,11 @@
  * 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
 
-// Tempo Placeholder
 #ifndef TEMPO_DEFAULT
-#    define TEMPO_DEFAULT 100
+#    define TEMPO_DEFAULT 120
+// in beats-per-minute
 #endif
 
 #define SONG(notes...) \
@@ -27,12 +27,14 @@
 // Note Types
 #define MUSICAL_NOTE(note, duration) \
     { (NOTE##note), duration }
+
 #define BREVE_NOTE(note) MUSICAL_NOTE(note, 128)
 #define WHOLE_NOTE(note) MUSICAL_NOTE(note, 64)
 #define HALF_NOTE(note) MUSICAL_NOTE(note, 32)
 #define QUARTER_NOTE(note) MUSICAL_NOTE(note, 16)
 #define EIGHTH_NOTE(note) MUSICAL_NOTE(note, 8)
 #define SIXTEENTH_NOTE(note) MUSICAL_NOTE(note, 4)
+#define THIRTYSECOND_NOTE(note) MUSICAL_NOTE(note, 2)
 
 #define BREVE_DOT_NOTE(note) MUSICAL_NOTE(note, 128 + 64)
 #define WHOLE_DOT_NOTE(note) MUSICAL_NOTE(note, 64 + 32)
@@ -40,6 +42,9 @@
 #define QUARTER_DOT_NOTE(note) MUSICAL_NOTE(note, 16 + 8)
 #define EIGHTH_DOT_NOTE(note) MUSICAL_NOTE(note, 8 + 4)
 #define SIXTEENTH_DOT_NOTE(note) MUSICAL_NOTE(note, 4 + 2)
+#define THIRTYSECOND_DOT_NOTE(note) MUSICAL_NOTE(note, 2 + 1)
+// duration of 64 units == one beat == one whole note
+// with a tempo of 60bpm this comes to a length of one second
 
 // Note Type Shortcuts
 #define M__NOTE(note, duration) MUSICAL_NOTE(note, duration)
@@ -49,56 +54,52 @@
 #define Q__NOTE(n) QUARTER_NOTE(n)
 #define E__NOTE(n) EIGHTH_NOTE(n)
 #define S__NOTE(n) SIXTEENTH_NOTE(n)
+#define T__NOTE(n) THIRTYSECOND_NOTE(n)
 #define BD_NOTE(n) BREVE_DOT_NOTE(n)
 #define WD_NOTE(n) WHOLE_DOT_NOTE(n)
 #define HD_NOTE(n) HALF_DOT_NOTE(n)
 #define QD_NOTE(n) QUARTER_DOT_NOTE(n)
 #define ED_NOTE(n) EIGHTH_DOT_NOTE(n)
 #define SD_NOTE(n) SIXTEENTH_DOT_NOTE(n)
+#define TD_NOTE(n) THIRTYSECOND_DOT_NOTE(n)
 
 // Note Timbre
 // Changes how the notes sound
-#define TIMBRE_12 0.125f
-#define TIMBRE_25 0.250f
-#define TIMBRE_50 0.500f
-#define TIMBRE_75 0.750f
+#define TIMBRE_12 12
+#define TIMBRE_25 25
+#define TIMBRE_50 50
+#define TIMBRE_75 75
 #ifndef TIMBRE_DEFAULT
 #    define TIMBRE_DEFAULT TIMBRE_50
 #endif
-// Notes - # = Octave
 
-#ifdef __arm__
-#    define NOTE_REST 1.00f
-#else
-#    define NOTE_REST 0.00f
-#endif
+// Notes - # = Octave
 
-/* These notes are currently bugged
-#define NOTE_C0          16.35f
-#define NOTE_CS0         17.32f
-#define NOTE_D0          18.35f
-#define NOTE_DS0         19.45f
-#define NOTE_E0          20.60f
-#define NOTE_F0          21.83f
-#define NOTE_FS0         23.12f
-#define NOTE_G0          24.50f
-#define NOTE_GS0         25.96f
-#define NOTE_A0          27.50f
-#define NOTE_AS0         29.14f
-#define NOTE_B0          30.87f
-#define NOTE_C1          32.70f
-#define NOTE_CS1         34.65f
-#define NOTE_D1          36.71f
-#define NOTE_DS1         38.89f
-#define NOTE_E1          41.20f
-#define NOTE_F1          43.65f
-#define NOTE_FS1         46.25f
-#define NOTE_G1          49.00f
-#define NOTE_GS1         51.91f
-#define NOTE_A1          55.00f
-#define NOTE_AS1         58.27f
-*/
+#define NOTE_REST 0.00f
 
+#define NOTE_C0 16.35f
+#define NOTE_CS0 17.32f
+#define NOTE_D0 18.35f
+#define NOTE_DS0 19.45f
+#define NOTE_E0 20.60f
+#define NOTE_F0 21.83f
+#define NOTE_FS0 23.12f
+#define NOTE_G0 24.50f
+#define NOTE_GS0 25.96f
+#define NOTE_A0 27.50f
+#define NOTE_AS0 29.14f
+#define NOTE_B0 30.87f
+#define NOTE_C1 32.70f
+#define NOTE_CS1 34.65f
+#define NOTE_D1 36.71f
+#define NOTE_DS1 38.89f
+#define NOTE_E1 41.20f
+#define NOTE_F1 43.65f
+#define NOTE_FS1 46.25f
+#define NOTE_G1 49.00f
+#define NOTE_GS1 51.91f
+#define NOTE_A1 55.00f
+#define NOTE_AS1 58.27f
 #define NOTE_B1 61.74f
 #define NOTE_C2 65.41f
 #define NOTE_CS2 69.30f
diff --git a/quantum/audio/voices.c b/quantum/audio/voices.c
index d412ad5057..d43fb8d169 100644
--- a/quantum/audio/voices.c
+++ b/quantum/audio/voices.c
@@ -1,4 +1,5 @@
 /* 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
@@ -17,35 +18,73 @@
 #include "audio.h"
 #include <stdlib.h>
 
-// these are imported from audio.c
-extern uint16_t envelope_index;
-extern float    note_timbre;
-extern float    polyphony_rate;
-extern bool     glissando;
+uint8_t note_timbre      = TIMBRE_DEFAULT;
+bool    glissando        = false;
+bool    vibrato          = false;
+float   vibrato_strength = 0.5;
+float   vibrato_rate     = 0.125;
 
+uint16_t voices_timer = 0;
+
+#ifdef AUDIO_VOICE_DEFAULT
+voice_type voice = AUDIO_VOICE_DEFAULT;
+#else
 voice_type voice = default_voice;
+#endif
 
 void set_voice(voice_type v) { voice = v; }
 
 void voice_iterate() { voice = (voice + 1) % number_of_voices; }
 void voice_deiterate() { voice = (voice - 1 + number_of_voices) % number_of_voices; }
 
+#ifdef AUDIO_VOICES
+float mod(float a, int b) {
+    float r = fmod(a, b);
+    return r < 0 ? r + b : r;
+}
+
+// Effect: 'vibrate' a given target frequency slightly above/below its initial value
+float voice_add_vibrato(float average_freq) {
+    float vibrato_counter = mod(timer_read() / (100 * vibrato_rate), VIBRATO_LUT_LENGTH);
+
+    return average_freq * pow(vibrato_lut[(int)vibrato_counter], vibrato_strength);
+}
+
+// Effect: 'slides' the 'frequency' from the starting-point, to the target frequency
+float voice_add_glissando(float from_freq, float to_freq) {
+    if (to_freq != 0 && from_freq < to_freq && from_freq < to_freq * pow(2, -440 / to_freq / 12 / 2)) {
+        return from_freq * pow(2, 440 / from_freq / 12 / 2);
+    } else if (to_freq != 0 && from_freq > to_freq && from_freq > to_freq * pow(2, 440 / to_freq / 12 / 2)) {
+        return from_freq * pow(2, -440 / from_freq / 12 / 2);
+    } else {
+        return to_freq;
+    }
+}
+#endif
+
 float voice_envelope(float frequency) {
     // envelope_index ranges from 0 to 0xFFFF, which is preserved at 880.0 Hz
-    __attribute__((unused)) uint16_t compensated_index = (uint16_t)((float)envelope_index * (880.0 / frequency));
+//    __attribute__((unused)) uint16_t compensated_index = (uint16_t)((float)envelope_index * (880.0 / frequency));
+#ifdef AUDIO_VOICES
+    uint16_t envelope_index    = timer_elapsed(voices_timer);  // TODO: multiply in some factor?
+    uint16_t compensated_index = envelope_index / 100;         // TODO: correct factor would be?
+#endif
 
     switch (voice) {
         case default_voice:
-            glissando      = false;
-            note_timbre    = TIMBRE_50;
-            polyphony_rate = 0;
+            glissando = false;
+            // note_timbre    = TIMBRE_50; //Note: leave the user the possibility to adjust the timbre with 'audio_set_timbre'
             break;
 
 #ifdef AUDIO_VOICES
 
+        case vibrating:
+            glissando = false;
+            vibrato   = true;
+            break;
+
         case something:
-            glissando      = false;
-            polyphony_rate = 0;
+            glissando = false;
             switch (compensated_index) {
                 case 0 ... 9:
                     note_timbre = TIMBRE_12;
@@ -56,24 +95,23 @@ float voice_envelope(float frequency) {
                     break;
 
                 case 20 ... 200:
-                    note_timbre = .125 + .125;
+                    note_timbre = 12 + 12;
                     break;
 
                 default:
-                    note_timbre = .125;
+                    note_timbre = 12;
                     break;
             }
             break;
 
         case drums:
-            glissando      = false;
-            polyphony_rate = 0;
+            glissando = false;
             // switch (compensated_index) {
             //     case 0 ... 10:
-            //         note_timbre = 0.5;
+            //         note_timbre = 50;
             //         break;
             //     case 11 ... 20:
-            //         note_timbre = 0.5 * (21 - compensated_index) / 10;
+            //         note_timbre = 50 * (21 - compensated_index) / 10;
             //         break;
             //     default:
             //         note_timbre = 0;
@@ -87,10 +125,10 @@ float voice_envelope(float frequency) {
                 frequency = (rand() % (int)(40)) + 60;
                 switch (envelope_index) {
                     case 0 ... 10:
-                        note_timbre = 0.5;
+                        note_timbre = 50;
                         break;
                     case 11 ... 20:
-                        note_timbre = 0.5 * (21 - envelope_index) / 10;
+                        note_timbre = 50 * (21 - envelope_index) / 10;
                         break;
                     default:
                         note_timbre = 0;
@@ -102,10 +140,10 @@ float voice_envelope(float frequency) {
                 frequency = (rand() % (int)(1000)) + 1000;
                 switch (envelope_index) {
                     case 0 ... 5:
-                        note_timbre = 0.5;
+                        note_timbre = 50;
                         break;
                     case 6 ... 20:
-                        note_timbre = 0.5 * (21 - envelope_index) / 15;
+                        note_timbre = 50 * (21 - envelope_index) / 15;
                         break;
                     default:
                         note_timbre = 0;
@@ -117,10 +155,10 @@ float voice_envelope(float frequency) {
                 frequency = (rand() % (int)(2000)) + 3000;
                 switch (envelope_index) {
                     case 0 ... 15:
-                        note_timbre = 0.5;
+                        note_timbre = 50;
                         break;
                     case 16 ... 20:
-                        note_timbre = 0.5 * (21 - envelope_index) / 5;
+                        note_timbre = 50 * (21 - envelope_index) / 5;
                         break;
                     default:
                         note_timbre = 0;
@@ -132,10 +170,10 @@ float voice_envelope(float frequency) {
                 frequency = (rand() % (int)(2000)) + 3000;
                 switch (envelope_index) {
                     case 0 ... 35:
-                        note_timbre = 0.5;
+                        note_timbre = 50;
                         break;
                     case 36 ... 50:
-                        note_timbre = 0.5 * (51 - envelope_index) / 15;
+                        note_timbre = 50 * (51 - envelope_index) / 15;
                         break;
                     default:
                         note_timbre = 0;
@@ -144,8 +182,7 @@ float voice_envelope(float frequency) {
             }
             break;
         case butts_fader:
-            glissando      = true;
-            polyphony_rate = 0;
+            glissando = true;
             switch (compensated_index) {
                 case 0 ... 9:
                     frequency   = frequency / 4;
@@ -158,7 +195,7 @@ float voice_envelope(float frequency) {
                     break;
 
                 case 20 ... 200:
-                    note_timbre = .125 - pow(((float)compensated_index - 20) / (200 - 20), 2) * .125;
+                    note_timbre = 12 - (uint8_t)(pow(((float)compensated_index - 20) / (200 - 20), 2) * 12.5);
                     break;
 
                 default:
@@ -168,7 +205,6 @@ float voice_envelope(float frequency) {
             break;
 
             // case octave_crunch:
-            //     polyphony_rate = 0;
             //     switch (compensated_index) {
             //         case 0 ... 9:
             //         case 20 ... 24:
@@ -186,14 +222,13 @@ float voice_envelope(float frequency) {
 
             //         default:
             //             note_timbre = TIMBRE_12;
-            //         	break;
+            //          break;
             //     }
             //  break;
 
         case duty_osc:
             // This slows the loop down a substantial amount, so higher notes may freeze
-            glissando      = true;
-            polyphony_rate = 0;
+            glissando = true;
             switch (compensated_index) {
                 default:
 #    define OCS_SPEED 10
@@ -201,38 +236,36 @@ float voice_envelope(float frequency) {
                     // sine wave is slow
                     // note_timbre = (sin((float)compensated_index/10000*OCS_SPEED) * OCS_AMP / 2) + .5;
                     // triangle wave is a bit faster
-                    note_timbre = (float)abs((compensated_index * OCS_SPEED % 3000) - 1500) * (OCS_AMP / 1500) + (1 - OCS_AMP) / 2;
+                    note_timbre = (uint8_t)abs((compensated_index * OCS_SPEED % 3000) - 1500) * (OCS_AMP / 1500) + (1 - OCS_AMP) / 2;
                     break;
             }
             break;
 
         case duty_octave_down:
-            glissando      = true;
-            polyphony_rate = 0;
-            note_timbre    = (envelope_index % 2) * .125 + .375 * 2;
-            if ((envelope_index % 4) == 0) note_timbre = 0.5;
+            glissando   = true;
+            note_timbre = (uint8_t)(100 * (envelope_index % 2) * .125 + .375 * 2);
+            if ((envelope_index % 4) == 0) note_timbre = 50;
             if ((envelope_index % 8) == 0) note_timbre = 0;
             break;
         case delayed_vibrato:
-            glissando      = true;
-            polyphony_rate = 0;
-            note_timbre    = TIMBRE_50;
+            glissando   = true;
+            note_timbre = TIMBRE_50;
 #    define VOICE_VIBRATO_DELAY 150
 #    define VOICE_VIBRATO_SPEED 50
             switch (compensated_index) {
                 case 0 ... VOICE_VIBRATO_DELAY:
                     break;
                 default:
+
                     frequency = frequency * vibrato_lut[(int)fmod((((float)compensated_index - (VOICE_VIBRATO_DELAY + 1)) / 1000 * VOICE_VIBRATO_SPEED), VIBRATO_LUT_LENGTH)];
                     break;
             }
             break;
             // case delayed_vibrato_octave:
-            //     polyphony_rate = 0;
             //     if ((envelope_index % 2) == 1) {
-            //         note_timbre = 0.55;
+            //         note_timbre = 55;
             //     } else {
-            //         note_timbre = 0.45;
+            //         note_timbre = 45;
             //     }
             //     #define VOICE_VIBRATO_DELAY 150
             //     #define VOICE_VIBRATO_SPEED 50
@@ -245,35 +278,64 @@ float voice_envelope(float frequency) {
             //     }
             //     break;
             // case duty_fifth_down:
-            //     note_timbre = 0.5;
+            //     note_timbre = TIMBRE_50;
             //     if ((envelope_index % 3) == 0)
-            //         note_timbre = 0.75;
+            //         note_timbre = TIMBRE_75;
             //     break;
             // case duty_fourth_down:
-            //     note_timbre = 0.0;
+            //     note_timbre = 0;
             //     if ((envelope_index % 12) == 0)
-            //         note_timbre = 0.75;
+            //         note_timbre = TIMBRE_75;
             //     if (((envelope_index % 12) % 4) != 1)
-            //         note_timbre = 0.75;
+            //         note_timbre = TIMBRE_75;
             //     break;
             // case duty_third_down:
-            //     note_timbre = 0.5;
+            //     note_timbre = TIMBRE_50;
             //     if ((envelope_index % 5) == 0)
-            //         note_timbre = 0.75;
+            //         note_timbre = TIMBRE_75;
             //     break;
             // case duty_fifth_third_down:
-            //     note_timbre = 0.5;
+            //     note_timbre = TIMBRE_50;
             //     if ((envelope_index % 5) == 0)
-            //         note_timbre = 0.75;
+            //         note_timbre = TIMBRE_75;
             //     if ((envelope_index % 3) == 0)
-            //         note_timbre = 0.25;
+            //         note_timbre = TIMBRE_25;
             //     break;
 
-#endif
+#endif  // AUDIO_VOICES
 
         default:
             break;
     }
 
+#ifdef AUDIO_VOICES
+    if (vibrato && (vibrato_strength > 0)) {
+        frequency = voice_add_vibrato(frequency);
+    }
+
+    if (glissando) {
+        // TODO: where to keep track of the start-frequency?
+        // frequency = voice_add_glissando(??, frequency);
+    }
+#endif  // AUDIO_VOICES
+
     return frequency;
 }
+
+// Vibrato functions
+
+void voice_set_vibrato_rate(float rate) { vibrato_rate = rate; }
+void voice_increase_vibrato_rate(float change) { vibrato_rate *= change; }
+void voice_decrease_vibrato_rate(float change) { vibrato_rate /= change; }
+void voice_set_vibrato_strength(float strength) { vibrato_strength = strength; }
+void voice_increase_vibrato_strength(float change) { vibrato_strength *= change; }
+void voice_decrease_vibrato_strength(float change) { vibrato_strength /= change; }
+
+// Timbre functions
+
+void voice_set_timbre(uint8_t timbre) {
+    if ((timbre > 0) && (timbre < 100)) {
+        note_timbre = timbre;
+    }
+}
+uint8_t voice_get_timbre(void) { return note_timbre; }
diff --git a/quantum/audio/voices.h b/quantum/audio/voices.h
index 478cb1ef0b..d3fd62dc3f 100644
--- a/quantum/audio/voices.h
+++ b/quantum/audio/voices.h
@@ -1,4 +1,5 @@
 /* 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
@@ -26,6 +27,7 @@ float voice_envelope(float frequency);
 typedef enum {
     default_voice,
 #ifdef AUDIO_VOICES
+    vibrating,
     something,
     drums,
     butts_fader,
@@ -45,3 +47,21 @@ typedef enum {
 void set_voice(voice_type v);
 void voice_iterate(void);
 void voice_deiterate(void);
+
+// Vibrato functions
+void voice_set_vibrato_rate(float rate);
+void voice_increase_vibrato_rate(float change);
+void voice_decrease_vibrato_rate(float change);
+void voice_set_vibrato_strength(float strength);
+void voice_increase_vibrato_strength(float change);
+void voice_decrease_vibrato_strength(float change);
+
+// Timbre functions
+/**
+ * @brief set the global timbre for tones to be played
+ * @note: only applies to pwm implementations - where it adjusts the duty-cycle
+ * @note: using any instrument from voices.[ch] other than 'default' may override the set value
+ * @param[in]: timbre: valid range is (0,100)
+ */
+void    voice_set_timbre(uint8_t timbre);
+uint8_t voice_get_timbre(void);
diff --git a/quantum/audio/wave.h b/quantum/audio/wave.h
deleted file mode 100644
index 48210a944e..0000000000
--- a/quantum/audio/wave.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/* Copyright 2016 Jack Humbert
- *
- * This program is free software: you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation, either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <avr/io.h>
-#include <avr/interrupt.h>
-#include <avr/pgmspace.h>
-
-#define SINE_LENGTH 2048
-
-const uint8_t sinewave[] PROGMEM =  // 2048 values
-    {0x80, 0x80, 0x80, 0x81, 0x81, 0x81, 0x82, 0x82, 0x83, 0x83, 0x83, 0x84, 0x84, 0x85, 0x85, 0x85, 0x86, 0x86, 0x87, 0x87, 0x87, 0x88, 0x88, 0x88, 0x89, 0x89, 0x8a, 0x8a, 0x8a, 0x8b, 0x8b, 0x8c, 0x8c, 0x8c, 0x8d, 0x8d, 0x8e, 0x8e, 0x8e, 0x8f, 0x8f, 0x8f, 0x90, 0x90, 0x91, 0x91, 0x91, 0x92, 0x92, 0x93, 0x93, 0x93, 0x94, 0x94, 0x95, 0x95, 0x95, 0x96, 0x96, 0x96, 0x97, 0x97, 0x98, 0x98, 0x98, 0x99, 0x99, 0x9a, 0x9a, 0x9a, 0x9b, 0x9b, 0x9b, 0x9c, 0x9c, 0x9d, 0x9d, 0x9d, 0x9e, 0x9e, 0x9e, 0x9f, 0x9f, 0xa0, 0xa0, 0xa0, 0xa1, 0xa1, 0xa2, 0xa2, 0xa2, 0xa3, 0xa3, 0xa3, 0xa4, 0xa4, 0xa5, 0xa5, 0xa5, 0xa6, 0xa6, 0xa6, 0xa7, 0xa7, 0xa7, 0xa8, 0xa8, 0xa9, 0xa9, 0xa9, 0xaa, 0xaa, 0xaa, 0xab, 0xab, 0xac, 0xac, 0xac, 0xad, 0xad, 0xad, 0xae, 0xae, 0xae, 0xaf, 0xaf, 0xb0, 0xb0, 0xb0, 0xb1, 0xb1, 0xb1, 0xb2, 0xb2, 0xb2, 0xb3, 0xb3, 0xb4, 0xb4, 0xb4, 0xb5, 0xb5, 0xb5, 0xb6, 0xb6, 0xb6, 0xb7, 0xb7, 0xb7, 0xb8, 0xb8, 0xb8, 0xb9, 0xb9, 0xba, 0xba, 0xba, 0xbb,
-     0xbb, 0xbb, 0xbc, 0xbc, 0xbc, 0xbd, 0xbd, 0xbd, 0xbe, 0xbe, 0xbe, 0xbf, 0xbf, 0xbf, 0xc0, 0xc0, 0xc0, 0xc1, 0xc1, 0xc1, 0xc2, 0xc2, 0xc2, 0xc3, 0xc3, 0xc3, 0xc4, 0xc4, 0xc4, 0xc5, 0xc5, 0xc5, 0xc6, 0xc6, 0xc6, 0xc7, 0xc7, 0xc7, 0xc8, 0xc8, 0xc8, 0xc9, 0xc9, 0xc9, 0xca, 0xca, 0xca, 0xcb, 0xcb, 0xcb, 0xcb, 0xcc, 0xcc, 0xcc, 0xcd, 0xcd, 0xcd, 0xce, 0xce, 0xce, 0xcf, 0xcf, 0xcf, 0xcf, 0xd0, 0xd0, 0xd0, 0xd1, 0xd1, 0xd1, 0xd2, 0xd2, 0xd2, 0xd2, 0xd3, 0xd3, 0xd3, 0xd4, 0xd4, 0xd4, 0xd5, 0xd5, 0xd5, 0xd5, 0xd6, 0xd6, 0xd6, 0xd7, 0xd7, 0xd7, 0xd7, 0xd8, 0xd8, 0xd8, 0xd9, 0xd9, 0xd9, 0xd9, 0xda, 0xda, 0xda, 0xda, 0xdb, 0xdb, 0xdb, 0xdc, 0xdc, 0xdc, 0xdc, 0xdd, 0xdd, 0xdd, 0xdd, 0xde, 0xde, 0xde, 0xde, 0xdf, 0xdf, 0xdf, 0xe0, 0xe0, 0xe0, 0xe0, 0xe1, 0xe1, 0xe1, 0xe1, 0xe2, 0xe2, 0xe2, 0xe2, 0xe3, 0xe3, 0xe3, 0xe3, 0xe4, 0xe4, 0xe4, 0xe4, 0xe4, 0xe5, 0xe5, 0xe5, 0xe5, 0xe6, 0xe6, 0xe6, 0xe6, 0xe7, 0xe7, 0xe7, 0xe7, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8,
-     0xe9, 0xe9, 0xe9, 0xe9, 0xea, 0xea, 0xea, 0xea, 0xea, 0xeb, 0xeb, 0xeb, 0xeb, 0xeb, 0xec, 0xec, 0xec, 0xec, 0xec, 0xed, 0xed, 0xed, 0xed, 0xed, 0xee, 0xee, 0xee, 0xee, 0xee, 0xef, 0xef, 0xef, 0xef, 0xef, 0xf0, 0xf0, 0xf0, 0xf0, 0xf0, 0xf0, 0xf1, 0xf1, 0xf1, 0xf1, 0xf1, 0xf2, 0xf2, 0xf2, 0xf2, 0xf2, 0xf2, 0xf3, 0xf3, 0xf3, 0xf3, 0xf3, 0xf3, 0xf4, 0xf4, 0xf4, 0xf4, 0xf4, 0xf4, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe,
-     0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf9, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7, 0xf7,
-     0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf6, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf5, 0xf4, 0xf4, 0xf4, 0xf4, 0xf4, 0xf4, 0xf3, 0xf3, 0xf3, 0xf3, 0xf3, 0xf3, 0xf2, 0xf2, 0xf2, 0xf2, 0xf2, 0xf2, 0xf1, 0xf1, 0xf1, 0xf1, 0xf1, 0xf0, 0xf0, 0xf0, 0xf0, 0xf0, 0xf0, 0xef, 0xef, 0xef, 0xef, 0xef, 0xee, 0xee, 0xee, 0xee, 0xee, 0xed, 0xed, 0xed, 0xed, 0xed, 0xec, 0xec, 0xec, 0xec, 0xec, 0xeb, 0xeb, 0xeb, 0xeb, 0xeb, 0xea, 0xea, 0xea, 0xea, 0xea, 0xe9, 0xe9, 0xe9, 0xe9, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0xe7, 0xe7, 0xe7, 0xe7, 0xe6, 0xe6, 0xe6, 0xe6, 0xe5, 0xe5, 0xe5, 0xe5, 0xe4, 0xe4, 0xe4, 0xe4, 0xe4, 0xe3, 0xe3, 0xe3, 0xe3, 0xe2, 0xe2, 0xe2, 0xe2, 0xe1, 0xe1, 0xe1, 0xe1, 0xe0, 0xe0, 0xe0, 0xe0, 0xdf, 0xdf, 0xdf, 0xde, 0xde, 0xde, 0xde, 0xdd, 0xdd, 0xdd, 0xdd, 0xdc, 0xdc, 0xdc, 0xdc, 0xdb, 0xdb, 0xdb, 0xda, 0xda, 0xda, 0xda, 0xd9, 0xd9, 0xd9, 0xd9, 0xd8, 0xd8, 0xd8, 0xd7, 0xd7, 0xd7, 0xd7, 0xd6, 0xd6, 0xd6, 0xd5, 0xd5, 0xd5, 0xd5, 0xd4, 0xd4, 0xd4,
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-     0x2a, 0x2a, 0x29, 0x29, 0x29, 0x28, 0x28, 0x28, 0x28, 0x27, 0x27, 0x27, 0x26, 0x26, 0x26, 0x26, 0x25, 0x25, 0x25, 0x25, 0x24, 0x24, 0x24, 0x23, 0x23, 0x23, 0x23, 0x22, 0x22, 0x22, 0x22, 0x21, 0x21, 0x21, 0x21, 0x20, 0x20, 0x20, 0x1f, 0x1f, 0x1f, 0x1f, 0x1e, 0x1e, 0x1e, 0x1e, 0x1d, 0x1d, 0x1d, 0x1d, 0x1c, 0x1c, 0x1c, 0x1c, 0x1b, 0x1b, 0x1b, 0x1b, 0x1b, 0x1a, 0x1a, 0x1a, 0x1a, 0x19, 0x19, 0x19, 0x19, 0x18, 0x18, 0x18, 0x18, 0x17, 0x17, 0x17, 0x17, 0x17, 0x16, 0x16, 0x16, 0x16, 0x15, 0x15, 0x15, 0x15, 0x15, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x13, 0x13, 0x13, 0x13, 0x12, 0x12, 0x12, 0x12, 0x12, 0x11, 0x11, 0x11, 0x11, 0x11, 0x10, 0x10, 0x10, 0x10, 0x10, 0xf,  0xf,  0xf,  0xf,  0xf,  0xf,  0xe,  0xe,  0xe,  0xe,  0xe,  0xd,  0xd,  0xd,  0xd,  0xd,  0xd,  0xc,  0xc,  0xc,  0xc,  0xc,  0xc,  0xb,  0xb,  0xb,  0xb,  0xb,  0xb,  0xa,  0xa,  0xa,  0xa,  0xa,  0xa,  0xa,  0x9,  0x9,  0x9,  0x9,  0x9,  0x9,  0x9,  0x8,  0x8,  0x8,  0x8,  0x8,
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-     0x18, 0x18, 0x18, 0x18, 0x19, 0x19, 0x19, 0x19, 0x1a, 0x1a, 0x1a, 0x1a, 0x1b, 0x1b, 0x1b, 0x1b, 0x1b, 0x1c, 0x1c, 0x1c, 0x1c, 0x1d, 0x1d, 0x1d, 0x1d, 0x1e, 0x1e, 0x1e, 0x1e, 0x1f, 0x1f, 0x1f, 0x1f, 0x20, 0x20, 0x20, 0x21, 0x21, 0x21, 0x21, 0x22, 0x22, 0x22, 0x22, 0x23, 0x23, 0x23, 0x23, 0x24, 0x24, 0x24, 0x25, 0x25, 0x25, 0x25, 0x26, 0x26, 0x26, 0x26, 0x27, 0x27, 0x27, 0x28, 0x28, 0x28, 0x28, 0x29, 0x29, 0x29, 0x2a, 0x2a, 0x2a, 0x2a, 0x2b, 0x2b, 0x2b, 0x2c, 0x2c, 0x2c, 0x2d, 0x2d, 0x2d, 0x2d, 0x2e, 0x2e, 0x2e, 0x2f, 0x2f, 0x2f, 0x30, 0x30, 0x30, 0x30, 0x31, 0x31, 0x31, 0x32, 0x32, 0x32, 0x33, 0x33, 0x33, 0x34, 0x34, 0x34, 0x34, 0x35, 0x35, 0x35, 0x36, 0x36, 0x36, 0x37, 0x37, 0x37, 0x38, 0x38, 0x38, 0x39, 0x39, 0x39, 0x3a, 0x3a, 0x3a, 0x3b, 0x3b, 0x3b, 0x3c, 0x3c, 0x3c, 0x3d, 0x3d, 0x3d, 0x3e, 0x3e, 0x3e, 0x3f, 0x3f, 0x3f, 0x40, 0x40, 0x40, 0x41, 0x41, 0x41, 0x42, 0x42, 0x42, 0x43, 0x43, 0x43, 0x44, 0x44, 0x44, 0x45, 0x45, 0x45, 0x46,
-     0x46, 0x47, 0x47, 0x47, 0x48, 0x48, 0x48, 0x49, 0x49, 0x49, 0x4a, 0x4a, 0x4a, 0x4b, 0x4b, 0x4b, 0x4c, 0x4c, 0x4d, 0x4d, 0x4d, 0x4e, 0x4e, 0x4e, 0x4f, 0x4f, 0x4f, 0x50, 0x50, 0x51, 0x51, 0x51, 0x52, 0x52, 0x52, 0x53, 0x53, 0x53, 0x54, 0x54, 0x55, 0x55, 0x55, 0x56, 0x56, 0x56, 0x57, 0x57, 0x58, 0x58, 0x58, 0x59, 0x59, 0x59, 0x5a, 0x5a, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c, 0x5c, 0x5d, 0x5d, 0x5d, 0x5e, 0x5e, 0x5f, 0x5f, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x61, 0x62, 0x62, 0x62, 0x63, 0x63, 0x64, 0x64, 0x64, 0x65, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x67, 0x68, 0x68, 0x69, 0x69, 0x69, 0x6a, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c, 0x6d, 0x6d, 0x6e, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71, 0x71, 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75, 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79, 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f};
diff --git a/quantum/backlight/backlight_avr.c b/quantum/backlight/backlight_avr.c
index 2ecdd4f2c4..e47192de34 100644
--- a/quantum/backlight/backlight_avr.c
+++ b/quantum/backlight/backlight_avr.c
@@ -126,7 +126,7 @@
 #        define COMxx1 COM1B1
 #        define OCRxx OCR1B
 #    endif
-#elif !defined(B5_AUDIO) && !defined(B6_AUDIO) && !defined(B7_AUDIO)
+#elif (AUDIO_PIN != B5) && (AUDIO_PIN != B6) && (AUDIO_PIN != B7) && (AUDIO_PIN_ALT != B5) && (AUDIO_PIN_ALT != B6) && (AUDIO_PIN_ALT != B7)
 // Timer 1 is not in use by Audio feature, Backlight can use it
 #    pragma message "Using hardware timer 1 with software PWM"
 #    define HARDWARE_PWM
@@ -145,7 +145,7 @@
 
 #    define OCIExA OCIE1A
 #    define OCRxx OCR1A
-#elif !defined(C6_AUDIO) && !defined(C5_AUDIO) && !defined(C4_AUDIO)
+#elif (AUDIO_PIN != C4) && (AUDIO_PIN != C5) && (AUDIO_PIN != C6)
 #    pragma message "Using hardware timer 3 with software PWM"
 // Timer 3 is not in use by Audio feature, Backlight can use it
 #    define HARDWARE_PWM
diff --git a/util/audio_generate_dac_lut.py b/util/audio_generate_dac_lut.py
new file mode 100755
index 0000000000..c31ba3d7ee
--- /dev/null
+++ b/util/audio_generate_dac_lut.py
@@ -0,0 +1,67 @@
+#!/usr/bin/env python3
+#
+# Copyright 2020 JohSchneider
+#
+# This program is free software: you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation, either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program.  If not, see <http://www.gnu.org/licenses/>.
+#
+
+AUDIO_DAC_BUFFER_SIZE=256
+AUDIO_DAC_SAMPLE_MAX=4095
+
+def plot(values):
+    for v in values:
+        print('0'* int(v * 80/AUDIO_DAC_SAMPLE_MAX))
+
+def to_lut(values):
+    for v in values:
+        print(hex(int(v)), end=", ")
+
+
+from math import sin, tau, pi
+
+samples=[]
+
+def sampleSine():
+    for s in range(AUDIO_DAC_BUFFER_SIZE):
+        samples.append((sin((s/AUDIO_DAC_BUFFER_SIZE)*tau - pi/2) + 1 )/2* AUDIO_DAC_SAMPLE_MAX)
+
+def sampleTriangle():
+    for s in range(AUDIO_DAC_BUFFER_SIZE):
+        if s < AUDIO_DAC_BUFFER_SIZE/2:
+            samples.append(s/(AUDIO_DAC_BUFFER_SIZE/2) * AUDIO_DAC_SAMPLE_MAX)
+        else:
+            samples.append(AUDIO_DAC_SAMPLE_MAX - (s-AUDIO_DAC_BUFFER_SIZE/2)/(AUDIO_DAC_BUFFER_SIZE/2) * AUDIO_DAC_SAMPLE_MAX)
+
+#compromise between square and triangle wave,
+def sampleTrapezoidal():
+    for i in range(AUDIO_DAC_BUFFER_SIZE):
+        a=3 #slope/inclination
+        if (i < AUDIO_DAC_BUFFER_SIZE/2):
+            s = a * (i * AUDIO_DAC_SAMPLE_MAX/(AUDIO_DAC_BUFFER_SIZE/2)) + (1-a)*AUDIO_DAC_SAMPLE_MAX/2
+        else:
+            i = i - AUDIO_DAC_BUFFER_SIZE/2
+            s = AUDIO_DAC_SAMPLE_MAX - a * (i * AUDIO_DAC_SAMPLE_MAX/(AUDIO_DAC_BUFFER_SIZE/2)) - (1-a)*AUDIO_DAC_SAMPLE_MAX/2
+
+        if s < 0:
+            s=0
+        if s> AUDIO_DAC_SAMPLE_MAX:
+            s=AUDIO_DAC_SAMPLE_MAX
+        samples.append(s)
+
+
+#sampleSine()
+sampleTrapezoidal()
+#print(samples)
+plot(samples)
+to_lut(samples)
diff --git a/util/sample_parser.py b/util/sample_parser.py
new file mode 100755
index 0000000000..70e193aee7
--- /dev/null
+++ b/util/sample_parser.py
@@ -0,0 +1,39 @@
+#!/usr/bin/env python3
+#
+# Copyright 2019 Jack Humbert
+#
+# This program is free software: you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation, either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program.  If not, see <http://www.gnu.org/licenses/>.
+#
+
+import wave, struct, sys
+
+waveFile = wave.open(sys.argv[1], 'r')
+# print(str(waveFile.getparams()))
+# sys.exit()
+
+if (waveFile.getsampwidth() != 2):
+    raise(Exception("This script currently only works with 16bit audio files"))
+
+length = waveFile.getnframes()
+out = "#define DAC_SAMPLE_CUSTOM_LENGTH " + str(length) + "\n\n"
+out += "static const dacsample_t dac_sample_custom[" + str(length) + "] = {"
+for i in range(0,length):
+    if (i % 8 == 0):
+        out += "\n    "
+    waveData = waveFile.readframes(1)
+    data = struct.unpack("<h", waveData)
+    out += str(int((int(data[0]) + 0x8000) / 16)) + ", "
+out = out[:-2]
+out += "\n};"
+print(out)
diff --git a/util/wavetable_parser.py b/util/wavetable_parser.py
new file mode 100755
index 0000000000..be0f01f7b4
--- /dev/null
+++ b/util/wavetable_parser.py
@@ -0,0 +1,40 @@
+#!/usr/bin/env python3
+#
+# Copyright 2019 Jack Humbert
+#
+# This program is free software: you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation, either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program.  If not, see <http://www.gnu.org/licenses/>.
+#
+
+import wave, struct, sys
+
+waveFile = wave.open(sys.argv[1], 'r')
+
+length = waveFile.getnframes()
+out = "#define DAC_WAVETABLE_CUSTOM_LENGTH " + str(int(length / 256)) + "\n\n"
+out += "static const dacsample_t dac_wavetable_custom[" + str(int(length / 256)) + "][256] = {"
+for i in range(0,length):
+    if (i % 8 == 0):
+        out += "\n    "
+    if (i % 256 == 0):
+        out = out[:-2]
+        out += "{\n    "
+    waveData = waveFile.readframes(1)
+    data = struct.unpack("<h", waveData)
+    out += str(int((int(data[0]) + 0x8000) / 16)) + ", "
+    if (i % 256 == 255):
+        out = out[:-2]
+        out += "\n  },"
+out = out[:-1]
+out += "\n};"
+print(out)