Audio system overhaul (#11820)

* Redo Arm DAC implementation for additive, wavetable synthesis, sample playback

changes by Jack Humbert on an implementation for DAC audio on arm/chibios platforms
this commits bundles the changes from the arm-dac-work branch focused on audio/audio_arm.* into one commit (leaving out the test-keyboard)

f52faeb5d (origin/arm-dac-work) add sample and wavetable examples, parsers for both
  -> only the changes on audio_arm_.*, the keyboard related parts are split off to a separate commit
bfe468ef1 start morphing wavetable
474d100b5 refined a bit
208bee10f play_notes working
3e6478b0b start in-place documentation of dac settings
3e1826a33 fixed blip (rounding error), other waves, added key selection (left/right)
73853d651 5 voices at 44.1khz
dfb401b95 limit voices to working number
9632b3379 configuration for the ez
6241f3f3b notes working in a new way

* Redo Arm DAC implementation for additive, wavetable synthesis, sample playback

changes by Jack Humbert on an implementation for DAC audio on arm/chibios platforms

this commit splits off the plank example keymap from commit
    f52faeb5d (origin/arm-dac-work) add sample and wavetable examples, parsers for both

* refactoring: rename audio_ to reflect their supported hardware-platform and audio-generation method: avr vs arm, and pwm vs dac

* refactoring: deducplicate ISR code to update the pwm duty-cycle and period in the avr-pwm-implementation

pulls three copies of the same code into one function
which should improve readability and maintainability :-)

* refactoring: move common code of arm and avr implementation into a separate/new file

* refactoring: audio_avr_pwm, renaming defines to decouple them from actually used timers, registers and ISRs

* refactoring: audio_avr_pwm - replacing function defines with plain register defines

aligns better with other existing qmk code (and the new audio_arm_pwm) doing similar pwm thing

* add audio-arm-pwm

since not all STM32 have a DAC onboard (STM32F2xx and STM32F3xx), pwm-audio is an alternative (STM32F1xx)
this code works on a "BluePill" clone, with an STM32F103C8B

* clang-format changes on quantum/audio/* only

* audio_arm_dac: stopping the notes caused screeching when using the DAC audio paths

* audio_arm_pwm: use pushpull on the pin; so that a piezzo can be hooked up direclty without additional components (opendrain would require an external pullup)

* refactoring: remove unused file from/for atmel-avr chips

* refactoring: remove unused (avr) wavetable file

* audio_arm_dac: adapt dac_end callback to changed chibios DAC api

the previous chibios (17.6.0) passed along a pointer into the buffer plus a sample_count (which are/already where included in the DACDrivre object) - the current chibios (19.1.0) only passes the driver object.
this patch ports more or less exactly what the previous chibios ISR code did: either have the user-callback work the first or second half of the buffer (dacsample_t pointer, with half the DAC_BUFFER_SIZE samples) by adjusting the pointer and sample count

* audio-arm-dac: show a compile-warning on undefined audio-pins

Co-Authored-By: Drashna Jaelre <drashna@live.com>

* audio_arm_dac: switch from exemplary wavetable generation to sine only

sine+triangle+squrare is exemplary, and not realy fit for "production" use
'stairs' are usefull for debugging (hardware, with an oscilloscope)

* audio_arm_dac: enable output buffers in the STM32

to drive external loads without any additional ciruitry - external opamps and such

* audio: prevent out-of-bounds array access

* audio_arm_dac: add output-frequency correcting factor

* audio_arm_pwm: get both the alternate-function and pm-callback variants back into working condition

and do some code-cleanup, refine documentation, ...

* audio_arm_pwm: increase pwm frequency for "higher fidelity"

on the previous .frequency=100000 higher frequency musical notes came out wrong
(frequency measured on a Tektronix TDS2014B)
note | freq | arm-pwm
C2 | 65.4 | 65.491
C5 | 523.25 | 523.93
C6 | 1046.5 | 1053.38
C7 | 2093 | 2129
C8 | 4186 | 4350.91

with .frequency = 500000
C8 | 4186 | 4204.6

* audio refactoring: remove unused variables

* audio_arm_dac: calibrate note tempo: with a tempo of 60beats-per-second a whole-note should last for exactly one second

* audio: allow feature selection in rules.mk

so the user can switch the audio driver between DAC and PWM on STM32 boards which support both (STM32F2 and up)
or select the "pin alternate" pwm mode, for example on STM32F103

* audio-refactoring: move codeblocks in audio.[ch] into more coherent groups

and add some inline documentation

* audio-refactoring: cleanup and streamline common code between audio_arm_[dac|pwm]

untangeling the relation between audio.c and the two drivers
and adding more documenting comments :-)

* audio_avr_pwm: getting it back into working condition, and cleanup+refactor

* audio-refactoring: documentation and typo fixes

Co-Authored-By: Nick Brassel <nick@tzarc.org>

* audio-refactoring: cleanup defines, inludes and remove debug-prints

* audio_chibios_dac: define&use a minimal sampling rate, based on the available tone-range

to ease up on the cpu-load, while still rendering the higher notes/tones sufficiently
also reenable the lower tones, since with the new implementation there is no evidence of them still beeing 'bugged'

* audio-refactoring: one common AUDIO_MAX_VOICES define for all audio-drivers

* audio-chibios-pwm: pwm-pin-allternate: make the the timer, timer-channel and alternate function user-#definable

* audio_chibios_dac: math.h has fmod for this

* Redo Arm DAC implementation for additive, wavetable synthesis, sample playback

update Jack Humberts dac-example keymaps for the slight changes in the audio-dac interface

* audio-refactoring: use a common AUDIO_PIN configuration switch instead of defines

have the user select a pin by configuration in rules.mk instead of a define in config.h
has the advantage of beeing in a common form/pattern across all audio-driver implementations

* audio-refactoring: switch backlight_avr.c to the new AUDIO_PIN defines

* audio-common: have advance_note return a boolean if the note changed, to the next one in the melody beeing played

* audio-chibios-pwm: fix issue with ~130ms silence between note/frequency changes while playing a SONG

through trial,error and a scope/logic analyzer figured out Chibios-PWMDriver (at least in the current version) misbehaves if the initial period is set to zero (or one; two seems to work); when thats the case subsequent calls to 'pwmChhangePeriod' + pwmEnableChannel took ~135ms of silence, before the PWM continued with the new frequency...

* audio-refactoring: get 'play_note' working again

with a limited number of available voices (say AUDIO_VOICES_MAX=1) allow new frequencies to be played, by discarding the oldest one in the 'frequencies' queue

* audio: set the fallback driver to DAC for chibios and PWM for all others (==avr at the moment)

* audio-refactoring: moore documentation

and some cleanup

* audio-avr-pwm: no fallback on unset AUDIO_PIN

this seems to be the expected behaviour by some keyboards (looking at ckeys/handwire_101:default) which otherwise fail to build because the firmware-image ends up beeing too large for the atmega... so we fail silently instead to keep travis happy

* audio-refactoring: untangling terminology: voice->tone

the code actually was working on tones (combination of pitch/frequency, duration, timbre, intensity/volume) and not voices (characteristic sound of an instrument; think piano vs guitar, which can be played together, each having its own "track" = voice on a music sheet)

* audio-pwm: allow freq=0 aka a pause/rest in a SONG

continue processing, but do not enable pwm units, since freq=0 wouldn't produce any sound anyway (and lead to division by zero on that occasion)

* audio-refactoring: audio_advance_note -> audio_advance_state

since it does not only affect 'one note', but the internally kept state as a whole

* audio-refactoring: untangling terminology: polyphony

the feature om the "inherited" avr code has little to do with polyphony (see wikipedia), but is more a time-multiplexing feature, to work around hardware limitations - like only having one pwm channel, that could on its own only reproduce one voice/instrument at a time

* audio-chibios-dac: add zero-crossing feature

have tones only change/stop when the waveform approaches zero - to avoid audible clicks
note that this also requires the samples to start at zero, since the internally kept index into the samples is reset to zero too

* audio-refactoring: feature: time-multiplexing of tones on a single output channel

this feature was in the original avr-pwm implementation misnomed as "polyphony"
with polyphony_rate and so on; did the same thing though: time-multiplexing multiple active notes so that a single output channel could reproduce more than one note at a time (which is not the same as a polyphony - see wikipedia :-) )

* audio-avr-pwm: get music-mode working (again) on AVRs

with both pwm channels, or either one of the two :-)
play_notes worked already - but music_mode uses play_note

* audio-refactoring: split define MAX_SIMULTANEOUS_TONES -> TONE_STACKSIZE

since the two cases are independant from one another, the hardware might impose limitations on the number of simultaneously reproducable tones, but the audio state should be able to track an unrelated number of notes recently started by play_note

* audio-arm-dac: per define selectable sample-luts

plus generation script in ./util

* audio-refactoring: heh, avr has a MIN...

* audio-refactoring: add basic dac audio-driver based on the current/master implementation

whereas current=d96380e65496912e0f68e6531565f4b45efd1623
which is the state of things before this whole audio-refactoring branch

boiled down to interface with the refactored audio system = removing all
redundant state-managing and frequency calculation

* audio-refactoring: rename audio-drivers to driver_$PLATFORM_$DRIVER

* audio-arm-pwm: split the software/hardware implementations into separate files

which saves us partially from a 'define hell', with the tradeoff that now two somewhat similar chibios_pwm implementations have to be maintained

* audio-refactoring: update documentation

* audio-arm-dac: apply AUDIO_PIN defines to driver_chibios_dac_basic

* audio-arm-dac: dac_additive: stop the hardware when the last sample completed

the audio system calls for a driver_stop, which is delayed until the current sample conversion finishes

* audio-refactoring: make function-namespace consistent

- all (public) audio functions start with audio_
- also refactoring play*_notes/tones to play*_melody, to visually distance it a bit from play*_tone/_note

* audio-refactoring: consistent define namespace: DAC_ -> AUDIO_DAC_

* audio-arm-dac: update (inline) documentation regarding MAX for sample values

* audio-chibios-dac: remove zero-crossing feature

didn't quite work as intended anyway, and stopping the hardware on close-to-zero seems to be enought anyway

* audio-arm-dac: dac_basic: respect the configured sample-rate

* audio-arm-pwm: have 'note_timbre' influence the pwm-duty cycle

like it already does in the avr implementation

* audio-refactoring: get VIBRATO working (again)

with all drivers (verified with chibios_[dac|pwm])

* audio-arm-dac: zero-crossing feature (Mk II)

wait for the generated waveform to approach 'zero' before either turning off the output+timer or switching to the current set of active_tones

* audio-refactoring: re-add note-resting -> introduce short_rest inbetween

- introduce a short pause/rest between two notes of the same frequency, to separate them audibly
- also updating the refactoring comments

* audio-refactoring: cleanup refactoring remnants

remove the former avr-isr code block - since all its features are now refactored into the different parts of the current system

also updates the TODOS

* audio-refactoring: reserve negative numbers as unitialized frequencies

to allow the valid tone/frequency f=0Hz == rest/pause

* audio-refactoring: FIX: first note of melody was missing

the first note was missing because 'goto_next_note'=false overrode a state_change=true of the initial play_tone
and some code-indentations/cleanup of related parts

* audio-arm-dac: fix hardware init-click

due to wron .init= value

* audio-refactoring: new conveniance function: audio_play_click

which can be used to further refactor/remove fauxclicky (avr only) and/or the 'clicky' features

* audio-refactoring: clang-format on quantum/audio/*

* audio-avr-pwm: consecutive notes of the same frequency get a pause inserted inbetween by audio.c

* audio-refactoring: use milliseconds instead of seconds for 'click' parameters

clicks are supposed to be short, seconds make little sense

* audio-refactoring: use timer ticks instead of counters

local counters were used in the original (avr)ISR to advance an index into the lookup tables (for vibrato), and something similar was used for the tone-multiplexing feature
decoupling these from the (possibly irregular) calls to advance_state made sesne, since those counters/lookups need to be in relation to a wall-time anyway

* audio-refactoring: voices.c: drop 'envelope_index' counter in favour of timer ticks

* audio-refactoring: move vibrato and timbre related parts from audio.c to voices.c

also drops the now (globally) unused AUDIO_VIBRATO/AUDIO_ENABLE_VIBRATO defines

* audio.c: use system-ticks instead of counters the drivers have to take care of for the internal state posision

since there already is a system-tick with ms resolution, keeping count separatly with each driver implementation makes little sense; especially since they had to take special care to call audio_advance_state with the correct step/end parameters for the audio state to advance regularly and with the correct pace

* audio.c: stop notes after new ones have been started

avoids brief states of with no notes playing that would otherwise stop the hardware and might lead to clicks

* audio.c: bugfix: actually play a pause

instead of just idling/stopping which lead the pwm drivers to stop entirely...

* audio-arm-pwm: pwm-software: add inverted output

new define AUDIO_PIN_ALT_AS_NEGATIVE will generate an inverted signal on the alternate pin, which boosts the volume if a piezo is connected to both AUDIO_PIN and AUDIO_PIN_ALT

* audio-arm-dac: basic: handle piezo configured&wired to both audio pins

* audio-refactoring: docs: update for AUDIO_PIN_ALT_AS_NEGATIVE and piezo wiring

* audio.c: bugfix: use timer_elapsed32 instad of keeping timestamps

avoids running into issues when the uint32 of the timer overflows

* audio-refactoring: add 'pragma once' and remove deprecated NOTE_REST

* audio_arm_dac: basic: add missing bracket

* audio.c: fix delta calculation

was in the wrong place, needs to use the 'last_timestamp' before it was reset

* audio-refactoring: buildfix: wrong legacy macro for set_timbre

* audio.c: 16bit timerstamps suffice

* audio-refactoring: separate includes for AVR and chibios

* audio-refactoring: timbre: use uint8 instead of float

* audio-refactoring: duration: use uint16 for internal per-tone/note state

* audio-refactoring: tonemultiplexing: use uint16 instead of float

* audio-arm-dac: additive: set second pin output-low

used when a piezo is connected to AUDIO_PIN and AUDIO_PIN_ALT, with PIN_ALT_AS_NEGATIVE

* audio-refactoring: move AUDIO_PIN selection from rules.mk to config.h

to be consistent with how other features are handled in QMK

* audio-refactoring: buildfix: wrong legacy macro for set_tempo

* audio-arm-dac: additive: set second pin output-low -- FIXUP

* audio.c: do duration<>ms conversion in uint instead of float

on AVR, to save a couple of bytes in the firmware size

* audio-refactoring: cleanup eeprom defines/usage

for ARM, avr is handled automagically through the avr libc and common_features.mk

Co-Authored-By: Drashna Jaelre <drashna@live.com>

* audio.h: throw an error if OFF is larger than MAX

* audio-arm-dac: basic: actually stop the dac-conversion on a audio_driver_stop

to put the output pin in a known state == AUDIO_DAC_OFF_VALUE, instead of just leaving them where the last conversion was... with AUDIO_PIN_ALT_AS_NEGATIVE this meant one output was left HIGH while the other was left LOW

one CAVEAT: due to this change the opposing squarewave when using both A4 and A5 with AUDIO_PIN_ALT_AS_NEGATIVE
show extra pulses at the beginning/end on one of the outputs, the two waveforms are in sync otherwise.
the extra pusles probably matter little, since this is no high-fidelity sound generation :P

* audio-arm-dac: additive: move zero-crossing code out of dac_value_generate

which is/should be user-overridable == simple, and doing one thing: providing sample values
state-transitions necessary for the zero crossing are better handled in the surrounding loop in the dac_end callback

* audio-arm-dac: dac-additive: zero-crossing: ramping up or down

after a start trigger ramp up: generate values until zero=OFF_VALUE is reached, then continue normally
same in reverse for strop trigger: output values until zero is reached/crossed, then keep OFF_VALUE on the output

* audio-arm-dac: dac-additive: BUGFIX: return OFF_VALUE when a pause is playing

fixes a bug during SONG playback, which suddenly stopped when it encoutnered a pause

* audio-arm-dac: set a sensible default for AUDIO_DAC_VALUE_OFF

1/2 MAX was probably exemplary, can't think of  a setup where that would make sense :-P

* audio-arm-dac: update synth_sample/_wavetable for new pin-defines

* audio-arm-dac:  default for AUDIO_DAC_VALUE_OFF

turned out that zero or max are bad default choices:
when multiple tones are played (>>5) and released at the same time (!), due to the complex waveform never reaching 'zero' the output can take quite a while to reach zero, and hence the zero-crossing code only "releases" the output waaay to late

* audio-arm-dac: additive: use DAC for negative pin

instead of PAL, which only allows the pin to be configured as output; LOW or HIGH

* audio-arm-dac: more compile-time configuration checks

* audio-refactoring: typo fixed

* audio-refactoring: clang-format on quantum/audio/*

* audio-avr-pwm: add defines for B-pin as primary/only speaker

also updates documentation.

* audio-refactoring: update documentation with proton-c config.h example

* audio-refactoring: move glissando (TODO) to voices.c

refactored/saved from the original glissando implementation in then upstream-master:audio_avr.c

still needs some work though, as it is now the calculation *should* work, but the start-frequency needs to be tracked somewhere/somehow; not only during a SONG playback but also with user input?

* audio-refactoring: cleanup: one round of aspell -c

* audio-avr-pwm: back to AUDIO_PIN

since config_common.h expands them to plain integers, the AUDIO_PIN define can directly be compared to e.g. B5
so there is no need to deal with separate defines like AUDIO_PIN_B5

* audio-refactoring: add technical documentation audio_driver.md

which moves some in-code documentation there

* audio-arm-dac: move AUDIO_PIN checks into c-code

instead of doing everything with the preprocessor, since A4/A5 do not expand to simple integers, preprocessor int-comparison is not possible. but necessary to get a consistent configuration scheme going throughout the audio-code... solution: let c-code handle the different AUDIO_PIN configurations instead (and leave code/size optimizations to the compiler)

* audio-arm-dac: compile-fix: set AUDIO_PIN if unset

workaround to get the build going again, and be backwarts compatible to arm-keyboards which not yet set the AUDIO_PIN define. until the define is enforced through an '#error"

* audio-refactoring: document tone-multiplexing feature

* audio-refactoring: Apply suggestions from documentation review

Co-authored-by: James Young <18669334+noroadsleft@users.noreply.github.com>

* audio-refactoring: Update docs/audio_driver.md

* audio-refactoring: docs: fix markdown newlines

Terminating a line in Markdown with <space>-<space>-<linebreak> creates an HTML single-line break (<br>).

Co-authored-by: James Young <18669334+noroadsleft@users.noreply.github.com>

* audio-arm-dac: additive: fix AUDIO_PIN_ALT handling

* audio-arm-pwm: align define naming with other drivers

Co-authored-by: Joel Challis <git@zvecr.com>

* audio-refactoring: set detault tempo to 120

and add documentation for the override

* audio-refactoring: update backlight define checks to new AUDIO_PIN names

* audio-refactoring: reworking PWM related defines

to be more consistent with other QMK code

Co-authored-by: Joel Challis <git@zvecr.com>

* audio-arm: have the state-update-timer user configurable

defaulting to GPTD6 or GPTD8 for stm32f2+ (=proton-c)
stm32f1 might need to set this to GPTD4, since 6 and 8 are not available

* audio-refactoring: PLAY_NOTE_ARRAY was already removed in master

* Add prototype for startup

* Update chibiOS dac basic to disable pins on stop

* Add defaults for Proton C

* avoid hanging audio if note is completely missed

* Don't redefine pins if they're already defined

* Define A4 and A5 for CTPC support

* Add license headers to keymap files

* Remove figlet? comments

* Add DAC config to audio driver docs

* Apply suggestions from code review

Co-authored-by: Jack Humbert <jack.humb@gmail.com>

* Add license header to py files

* correct license header

* Add JohSchneider's name to modified files

AKA credit where credit's due

* Set executable permission and change interpeter

* Add 'wave' to pip requirements

* Improve documentation

* Add some settings I missed

* Strip AUDIO_DRIVER to parse the name correctly

* fix depreciated

* Update util/audio_generate_dac_lut.py

Co-authored-by: Jack Humbert <jack.humb@gmail.com>

* Fix type in clueboard config

* Apply suggestions from tzarc

Co-authored-by: Nick Brassel <nick@tzarc.org>

Co-authored-by: Johannes <you@example.com>
Co-authored-by: JohSchneider <JohSchneider@googlemail.com>
Co-authored-by: Nick Brassel <nick@tzarc.org>
Co-authored-by: James Young <18669334+noroadsleft@users.noreply.github.com>
Co-authored-by: Joel Challis <git@zvecr.com>
Co-authored-by: Joshua Diamond <josh@windowoffire.com>
Co-authored-by: Jack Humbert <jack.humb@gmail.com>

4 files changed, 360 insertions, 17 deletions

diff --git a/docs/_summary.md b/docs/_summary.md
index 526caf926f..acbfcfaeda 100644
--- a/docs/_summary.md
+++ b/docs/_summary.md
@@ -133,6 +133,7 @@
     * [Compatible Microcontrollers](compatible_microcontrollers.md)
     * [Drivers](hardware_drivers.md)
       * [ADC Driver](adc_driver.md)
+      * [Audio Driver](audio_driver.md)
       * [I2C Driver](i2c_driver.md)
       * [SPI Driver](spi_driver.md)
       * [WS2812 Driver](ws2812_driver.md)
diff --git a/docs/audio_driver.md b/docs/audio_driver.md
new file mode 100644
index 0000000000..7cd5a98d9f
--- /dev/null
+++ b/docs/audio_driver.md
@@ -0,0 +1,221 @@
+# Audio Driver :id=audio-driver
+
+The [Audio feature](feature_audio.md) breaks the hardware specifics out into separate, exchangeable driver units, with a common interface to the audio-"core" - which itself handles playing songs and notes while tracking their progress in an internal state, initializing/starting/stopping the driver as needed.
+
+Not all MCUs support every available driver, either the platform-support is not there (yet?) or the MCU simply does not have the required hardware peripheral.
+
+
+## AVR :id=avr
+
+Boards built around an Atmega32U4 can use two sets of PWM capable pins, each driving a separate speaker.
+The possible configurations are:
+
+|              | Timer3      | Timer1       |
+|--------------|-------------|--------------|
+| one speaker  | C4,C5 or C6 |              |
+| one speaker  |             | B4, B5 or B7 |
+| two speakers | C4,C5 or C6 | B4, B5 or B7 |
+
+Currently there is only one/default driver for AVR based boards, which is automatically configured to:
+
+```make
+AUDIO_DRIVER = pwm_hardware
+```
+
+
+## ARM :id=arm
+
+For Arm based boards, QMK depends on ChibiOS - hence any MCU supported by the later is likely usable, as long as certain hardware peripherals are available.
+
+Supported wiring configurations, with their ChibiOS/MCU peripheral requirement are listed below;
+piezo speakers are marked with :one: for the first/primary and :two: for the secondary.
+
+  | driver       | GPTD6<br>Tim6                            | GPTD7<br>Tim7          | GPTD8<br>Tim8 | PWMD1<sup>1</sup><br>Tim1_Ch1 |
+  |--------------|------------------------------------------|------------------------|---------------|-------------------------------|
+  | dac_basic    | A4+DACD1 = :one:                         | A5+DACD2 = :one:       | state         |                               |
+  |              | A4+DACD1 = :one: + Gnd                   | A5+DACD2 = :two: + Gnd | state         |                               |
+  |              | A4+DACD1 = :two: + Gnd                   | A5+DACD2 = :one: + Gnd | state         |                               |
+  |              | A4+DACD1 = :one: + Gnd                   |                        | state         |                               |
+  |              |                                          | A5+DACD2 = :one: + Gnd | state         |                               |
+  | dac_additive | A4+DACD1 = :one: + Gnd                   |                        |               |                               |
+  |              | A5+DACD2 = :one: + Gnd                   |                        |               |                               |
+  |              | A4+DACD1 + A5+DACD2 = :one: <sup>2</sup> |                        |               |                               |
+  | pwm_software | state-update                             |                        |               | any = :one:                   |
+  | pwm hardware | state-update                             |                        |               | A8 = :one: <sup>3</sup>       |
+
+
+<sup>1</sup>: the routing and alternate functions for PWM differ sometimes between STM32 MCUs, if in doubt consult the data-sheet  
+<sup>2</sup>: one piezo connected to A4 and A5, with AUDIO_PIN_ALT_AS_NEGATIVE set  
+<sup>3</sup>: TIM1_CH1 = A8 on STM32F103C8, other combinations are possible, see Data-sheet. configured with: AUDIO_PWM_DRIVER and AUDIO_PWM_CHANNEL
+
+
+
+### DAC basic :id=dac-basic
+
+The default driver for ARM boards, in absence of an overriding configuration.
+This driver needs one Timer per enabled/used DAC channel, to trigger conversion; and a third timer to trigger state updates with the audio-core.
+
+Additionally, in the board config, you'll want to make changes to enable the DACs, GPT for Timers 6, 7 and 8:
+
+``` c
+//halconf.h:
+#define HAL_USE_DAC                 TRUE
+#define HAL_USE_GPT                 TRUE
+#include_next <halconf.h>
+```
+
+``` c
+// mcuconf.h:
+#include_next <mcuconf.h>
+#undef STM32_DAC_USE_DAC1_CH1
+#define STM32_DAC_USE_DAC1_CH1              TRUE
+#undef STM32_DAC_USE_DAC1_CH2
+#define STM32_DAC_USE_DAC1_CH2              TRUE
+#undef STM32_GPT_USE_TIM6
+#define STM32_GPT_USE_TIM6                  TRUE
+#undef STM32_GPT_USE_TIM7
+#define STM32_GPT_USE_TIM7                  TRUE
+#undef STM32_GPT_USE_TIM8
+#define STM32_GPT_USE_TIM8                  TRUE
+```
+
+?> Note: DAC1 (A4) uses TIM6, DAC2 (A5) uses TIM7, and the audio state timer uses TIM8 (configurable). 
+
+You can also change the timer used for the overall audio state by defining the driver.  For instance: 
+
+```c
+#define AUDIO_STATE_TIMER GPTD9
+```
+
+### DAC additive :id=dac-additive
+
+only needs one timer (GPTD6, Tim6) to trigger the DAC unit to do a conversion; the audio state updates are in turn triggered during the DAC callback.
+
+Additionally, in the board config, you'll want to make changes to enable the DACs, GPT for Timer 6:
+
+``` c
+//halconf.h:
+#define HAL_USE_DAC                 TRUE
+#define HAL_USE_GPT                 TRUE
+#include_next <halconf.h>
+```
+
+``` c
+// mcuconf.h:
+#include_next <mcuconf.h>
+#undef STM32_DAC_USE_DAC1_CH1
+#define STM32_DAC_USE_DAC1_CH1              TRUE
+#undef STM32_DAC_USE_DAC1_CH2
+#define STM32_DAC_USE_DAC1_CH2              TRUE
+#undef STM32_GPT_USE_TIM6
+#define STM32_GPT_USE_TIM6                  TRUE
+```
+
+### DAC Config
+
+| Define                           | Defaults                   | Description                                                --------------------------------------------------------------------------------------------- |
+| `AUDIO_DAC_SAMPLE_MAX`           | `4095U`                    | Highest value allowed. Lower value means lower volume. And 4095U is the upper limit, since this is limited to a 12 bit value. Only effects non-pregenerated samples.  |
+| `AUDIO_DAC_OFF_VALUE`            | `AUDIO_DAC_SAMPLE_MAX / 2` | The value of the DAC when notplaying anything. Some setups may require a high (`AUDIO_DAC_SAMPLE_MAX`) or low (`0`) value here.                                        |
+| `AUDIO_MAX_SIMULTANEOUS_TONES`   | __see next table__         | The number of tones that can be played simultaneously.  A value that is too high may freeze the controller or glitch out when too many tones are being played.        |
+| `AUDIO_DAC_SAMPLE_RATE`          | __see next table__         | Effective bit rate of the DAC (in hertz), higher limits simultaneous tones, and lower sacrifices quality.                                                          |
+
+There are a number of predefined quality settings that you can use, with "sane minimum" being the default.  You can use custom values by simply defining the sample rate and number of simultaneous tones, instead of using one of the listed presets. 
+
+| Define                            | Sample Rate | Simultaneous tones  |
+| `AUDIO_DAC_QUALITY_VERY_LOW`      | `11025U`    | `8`                 |
+| `AUDIO_DAC_QUALITY_LOW`           | `22040U`    | `4`                 |
+| `AUDIO_DAC_QUALITY_HIGH`          | `44100U`    | `2`                 |
+| `AUDIO_DAC_QUALITY_VERY_HIGH`     | `88200U`    | `1`                 |
+| `AUDIO_DAC_QUALITY_SANE_MINIMUM`  | `16384U`    | `8`                 |
+
+
+```c
+        /* 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
+         */
+```
+
+
+### PWM hardware :id=pwm-hardware
+
+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 MCU's data-sheet for which pin can be driven by what timer - looking for TIMx_CHy and the corresponding alternate function.
+
+A configuration example for the STM32F103C8 would be:
+``` c
+//halconf.h:
+#define HAL_USE_PWM                 TRUE
+#define HAL_USE_PAL                 TRUE
+#define HAL_USE_GPT                 TRUE
+#include_next <halconf.h>
+```
+
+``` c
+// mcuconf.h:
+#include_next <mcuconf.h>
+#undef STM32_PWM_USE_TIM1
+#define STM32_PWM_USE_TIM1                  TRUE
+#undef STM32_GPT_USE_TIM4
+#define STM32_GPT_USE_TIM4                  TRUE
+```
+
+If we now target pin A8, looking through the data-sheet of the STM32F103C8, for the timers and alternate functions
+- TIM1_CH1 = PA8 <- alternate0
+- TIM1_CH2 = PA9
+- TIM1_CH3 = PA10
+- TIM1_CH4 = PA11
+
+with all this information, the configuration would contain these lines:
+``` c
+//config.h:
+#define AUDIO_PIN A8
+#define AUDIO_PWM_DRIVER PWMD1
+#define AUDIO_PWM_CHANNEL 1
+#define AUDIO_STATE_TIMER GPTD4
+```
+
+ChibiOS uses GPIOv1 for the F103, which only knows of one alternate function.
+On 'larger' STM32s, GPIOv2 or GPIOv3 are used; with them it is also necessary to configure `AUDIO_PWM_PAL_MODE` to the correct alternate function for the selected pin, timer and timer-channel.
+
+
+### PWM software :id=pwm-software
+
+This driver uses the PWM callbacks from PWMD1 with TIM1_CH1 to toggle the selected AUDIO_PIN in software.
+During the same callback, with AUDIO_PIN_ALT_AS_NEGATIVE set, the AUDIO_PIN_ALT is toggled inversely to AUDIO_PIN. This is useful for setups that drive a piezo from two pins (instead of one and Gnd).
+
+You can also change the timer used for software PWM by defining the driver.  For instance: 
+
+```c
+#define AUDIO_STATE_TIMER GPTD8
+```
+
+
+### Testing Notes :id=testing-notes
+
+While not an exhaustive list, the following table provides the scenarios that have been partially validated:
+
+|                          | DAC basic          | DAC additive       | PWM hardware       | PWM software       |
+|--------------------------|--------------------|--------------------|--------------------|--------------------|
+| Atmega32U4               | :o:                | :o:                | :heavy_check_mark: | :o:                |
+| STM32F103C8 (bluepill)   | :x:                | :x:                | :heavy_check_mark: | :heavy_check_mark: |
+| STM32F303CCT6 (proton-c) | :heavy_check_mark: | :heavy_check_mark: | ?                  | :heavy_check_mark: |
+| STM32F405VG              | :heavy_check_mark: | :heavy_check_mark: | :heavy_check_mark: | :heavy_check_mark: |
+| L0xx                     | :x: (no Tim8)      | ?                  | ?                  | ?                  |
+
+
+:heavy_check_mark: : works and was tested  
+:o: : does not apply  
+:x: : not supported by MCU
+
+*Other supported ChibiOS boards and/or pins may function, it will be highly chip and configuration dependent.*
diff --git a/docs/config_options.md b/docs/config_options.md
index 9a64b9b3d2..aeaaf47aaf 100644
--- a/docs/config_options.md
+++ b/docs/config_options.md
@@ -67,16 +67,22 @@ This is a C header file that is one of the first things included, and will persi
   * turns on the alternate audio voices (to cycle through)
 * `#define C4_AUDIO`
   * enables audio on pin C4
+  * Deprecated. Use `#define AUDIO_PIN C4`
 * `#define C5_AUDIO`
   * enables audio on pin C5
+  * Deprecated. Use `#define AUDIO_PIN C5`
 * `#define C6_AUDIO`
   * enables audio on pin C6
+  * Deprecated. Use `#define AUDIO_PIN C6`
 * `#define B5_AUDIO`
-  * enables audio on pin B5 (duophony is enables if one of B[5-7]\_AUDIO is enabled along with one of C[4-6]\_AUDIO)
+  * enables audio on pin B5 (duophony is enabled if one of B pins is enabled along with one of C pins)
+  * Deprecated. Use `#define AUDIO_PIN B5`, or use `#define AUDIO_PIN_ALT B5` if a `C` pin is enabled with `AUDIO_PIN`
 * `#define B6_AUDIO`
-  * enables audio on pin B6 (duophony is enables if one of B[5-7]\_AUDIO is enabled along with one of C[4-6]\_AUDIO)
+  * enables audio on pin B5 (duophony is enabled if one of B pins is enabled along with one of C pins)
+  * Deprecated. Use `#define AUDIO_PIN B6`, or use `#define AUDIO_PIN_ALT B6` if a `C` pin is enabled with `AUDIO_PIN`
 * `#define B7_AUDIO`
-  * enables audio on pin B7 (duophony is enables if one of B[5-7]\_AUDIO is enabled along with one of C[4-6]\_AUDIO)
+  * enables audio on pin B5 (duophony is enabled if one of B pins is enabled along with one of C pins)
+  * Deprecated. Use `#define AUDIO_PIN B7`, or use `#define AUDIO_PIN_ALT B7` if a `C` pin is enabled with `AUDIO_PIN`
 * `#define BACKLIGHT_PIN B7`
   * pin of the backlight
 * `#define BACKLIGHT_LEVELS 3`
diff --git a/docs/feature_audio.md b/docs/feature_audio.md
index 68ba4477a5..9e7ba75f52 100644
--- a/docs/feature_audio.md
+++ b/docs/feature_audio.md
@@ -1,21 +1,117 @@
 # Audio
 
-Your keyboard can make sounds! If you've got a Planck, Preonic, or basically any AVR keyboard that allows access to certain PWM-capable pins, you can hook up a simple speaker and make it beep. You can use those beeps to indicate layer transitions, modifiers, special keys, or just to play some funky 8bit tunes.
+Your keyboard can make sounds! If you've got a spare pin you can hook up a simple speaker and make it beep. You can use those beeps to indicate layer transitions, modifiers, special keys, or just to play some funky 8bit tunes.
 
-Up to two simultaneous audio voices are supported, one driven by timer 1 and another driven by timer 3.  The following pins can be defined as audio outputs in config.h:
+To activate this feature, add `AUDIO_ENABLE = yes` to your `rules.mk`.
 
-Timer 1:
-`#define B5_AUDIO`
-`#define B6_AUDIO`
-`#define B7_AUDIO`
+## AVR based boards
+On Atmega32U4 based boards, up to two simultaneous tones can be rendered.
+With one speaker connected to a PWM capable pin on PORTC driven by timer 3 and the other on one of the PWM pins on PORTB driven by timer 1.
 
-Timer 3:
-`#define C4_AUDIO`
-`#define C5_AUDIO`
-`#define C6_AUDIO`
+The following pins can be configured as audio outputs in `config.h` - for one speaker set eiter one out of:
 
-If you add `AUDIO_ENABLE = yes` to your `rules.mk`, there's a couple different sounds that will automatically be enabled without any other configuration:
+* `#define AUDIO_PIN C4`
+* `#define AUDIO_PIN C5`
+* `#define AUDIO_PIN C6`
+* `#define AUDIO_PIN B5`
+* `#define AUDIO_PIN B6`
+* `#define AUDIO_PIN B7`
 
+and *optionally*, for a second speaker, one of:
+* `#define AUDIO_PIN_ALT B5`
+* `#define AUDIO_PIN_ALT B6`
+* `#define AUDIO_PIN_ALT B7`
+
+### Wiring
+per speaker is - for example with a piezo buzzer - the black lead to Ground, and the red lead connected to the selected AUDIO_PIN for the primary; and similarly with AUDIO_PIN_ALT for the secondary.
+
+
+## ARM based boards
+for more technical details, see the notes on [Audio driver](audio_driver.md).
+
+<!-- because I'm not sure where to fit this in: https://waveeditonline.com/ -->
+### DAC (basic)
+Most STM32 MCUs have DAC peripherals, with a notable exception of the STM32F1xx series. Generally, the DAC peripheral drives pins A4 or A5. To enable DAC-based audio output on STM32 devices, add `AUDIO_DRIVER = dac_basic` to `rules.mk` and set in `config.h` either:
+
+`#define AUDIO_PIN A4` or `#define AUDIO_PIN A5`
+
+the other DAC channel can optionally be used with a secondary speaker, just set:
+
+`#define AUDIO_PIN_ALT A4` or `#define AUDIO_PIN_ALT A5`
+
+Do note though that the dac_basic driver is only capable of reproducing one tone per speaker/channel at a time, for more tones simultaneously, try the dac_additive driver.
+
+#### Wiring:
+for two piezos, for example configured as `AUDIO_PIN A4` and `AUDIO_PIN_ALT A5` would be: red lead to A4 and black to Ground, and similarly with the second one: A5 = red, and Ground = black
+
+another alternative is to drive *one* piezo with both DAC pins - for an extra "push".
+wiring red to A4 and black to A5 (or the other way round) and add `#define AUDIO_PIN_ALT_AS_NEGATIVE` to `config.h`
+
+##### Proton-C Example:
+The Proton-C comes (optionally) with one 'builtin' piezo, which is wired to A4+A5.
+For this board `config.h` would include these defines:
+
+```c
+#define AUDIO_PIN A5
+#define AUDIO_PIN_ALT A4
+#define AUDIO_PIN_ALT_AS_NEGATIVE
+```
+
+### DAC (additive)
+Another option, besides dac_basic (which produces sound through a square-wave), is to use the DAC to do additive wave synthesis.
+With a number of predefined wave-forms or by providing your own implementation to generate samples on the fly.
+To use this feature set `AUDIO_DRIVER = dac_additive` in your `rules.mk`, and select in `config.h` EITHER `#define AUDIO_PIN A4` or `#define AUDIO_PIN A5`.
+
+The used waveform *defaults* to sine, but others can be selected by adding one of the following defines to `config.h`:
+
+* `#define AUDIO_DAC_SAMPLE_WAVEFORM_SINE`
+* `#define AUDIO_DAC_SAMPLE_WAVEFORM_TRIANGLE`
+* `#define AUDIO_DAC_SAMPLE_WAVEFORM_TRAPEZOID`
+* `#define AUDIO_DAC_SAMPLE_WAVEFORM_SQUARE`
+
+Should you rather choose to generate and use your own sample-table with the DAC unit, implement `uint16_t dac_value_generate(void)` with your keyboard - for an example implementation see keyboards/planck/keymaps/synth_sample or keyboards/planck/keymaps/synth_wavetable
+
+
+### PWM (software)
+if the DAC pins are unavailable (or the MCU has no usable DAC at all, like STM32F1xx); PWM can be an alternative.
+Note that there is currently only one speaker/pin supported.
+
+set in `rules.mk`:
+
+`AUDIO_DRIVER = pwm_software` and in `config.h`: 
+`#define AUDIO_PIN C13` (can be any pin) to have the selected pin output a pwm signal, generated from a timer callback which toggles the pin in software.
+
+#### Wiring
+the usual piezo wiring: red goes to the selected AUDIO_PIN, black goes to ground.
+
+OR if you can chose to drive one piezo with two pins, for example `#define AUDIO_PIN B1`, `#define AUDIO_PIN_ALT B2` in `config.h`, with `#define AUDIO_PIN_ALT_AS_NEGATIVE` - then the red lead could go to B1, the black to B2.
+
+### PWM (hardware)
+STM32F1xx have to fall back to using PWM, but can do so in hardware; but again on currently only one speaker/pin.
+
+`AUDIO_DRIVER = pwm_hardware` in `rules.mk`, and in `config.h`:
+`#define AUDIO_PIN A8`
+`#define AUDIO_PWM_DRIVER PWMD1`
+`#define AUDIO_PWM_CHANNEL 1`
+(as well as `#define AUDIO_PWM_PAL_MODE 42` if you are on STM32F2 or larger)
+which will use Timer 1 to directly drive pin PA8 through the PWM hardware (TIM1_CH1 = PA8).
+Should you want to use the pwm-hardware on another pin and timer - be ready to dig into the STM32 data-sheet to pick the right TIMx_CHy and pin-alternate function.
+
+
+## Tone Multiplexing
+Since most drivers can only render one tone per speaker at a time (with the one exception: arm dac-additive) there also exists a "workaround-feature" that does time-slicing/multiplexing - which does what the name implies: cycle through a set of active tones (e.g. when playing chords in Music Mode) at a given rate, and put one tone at a time out through the one/few speakers that are available.
+
+To enable this feature, and configure a starting-rate, add the following defines to `config.h`:
+```c
+#define AUDIO_ENABLE_TONE_MULTIPLEXING
+#define AUDIO_TONE_MULTIPLEXING_RATE_DEFAULT 10
+```
+
+The audio core offers interface functions to get/set/change the tone multiplexing rate from within `keymap.c`.
+
+
+## Songs
+There's a couple of different sounds that will automatically be enabled without any other configuration:
 ```
 STARTUP_SONG // plays when the keyboard starts up (audio.c)
 GOODBYE_SONG // plays when you press the RESET key (quantum.c)
@@ -67,15 +163,34 @@ The available keycodes for audio are:
 * `AU_OFF` - Turn Audio Feature off
 * `AU_TOG` - Toggle Audio Feature state
 
-!> These keycodes turn all of the audio functionality on and off.  Turning it off means that audio feedback, audio clicky, music mode, etc. are disabled, completely. 
+!> These keycodes turn all of the audio functionality on and off.  Turning it off means that audio feedback, audio clicky, music mode, etc. are disabled, completely.
+
+## Tempo
+the 'speed' at which SONGs are played is dictated by the set Tempo, which is measured in beats-per-minute. Note lenghts are defined relative to that.
+The initial/default tempo is set to 120 bpm, but can be configured by setting `TEMPO_DEFAULT` in `config.c`.
+There is also a set of functions to modify the tempo from within the user/keymap code:
+```c
+void audio_set_tempo(uint8_t tempo);
+void audio_increase_tempo(uint8_t tempo_change);
+void audio_decrease_tempo(uint8_t tempo_change);
+```
 
 ## ARM Audio Volume
 
-For ARM devices, you can adjust the DAC sample values. If your board is too loud for you or your coworkers, you can set the max using `DAC_SAMPLE_MAX` in your `config.h`:
+For ARM devices, you can adjust the DAC sample values. If your board is too loud for you or your coworkers, you can set the max using `AUDIO_DAC_SAMPLE_MAX` in your `config.h`:
 
 ```c
-#define DAC_SAMPLE_MAX 65535U
+#define AUDIO_DAC_SAMPLE_MAX 4095U
 ```
+the DAC usually runs in 12Bit mode, hence a volume of 100% = 4095U
+
+Note: this only adjusts the volume aka 'works' if you stick to WAVEFORM_SQUARE, since its samples are generated on the fly - any other waveform uses a hardcoded/precomputed sample-buffer.
+
+## Voices
+Aka "audio effects", different ones can be enabled by setting in `config.h` these defines:
+`#define AUDIO_VOICES` to enable the feature, and `#define AUDIO_VOICE_DEFAULT something` to select a specific effect
+for details see quantum/audio/voices.h and .c
+
 
 ## Music Mode