* resynced with upstream, and adjusted keymap layout for planck
* updated keyboard layout
Signed-off-by: Sean Johnson <sean@ttys0.net>
* swapped out bspc for del on symb layer
Signed-off-by: Sean Johnson <sean@ttys0.net>
* fixed typo with brightness media keys
* turns out my brightness config was correct, it was macOS that had gone sideways
* updated to bring in line with requirements for merging into upstream
* removed redundant config from rules.mk
moved media controls to FUNC layer for Planck layout
* added GPL2+ compatible license header
Signed-off-by: Sean Johnson <sean@skj.dev>
* removed unused MIDI comment
Signed-off-by: Sean Johnson <sean@skj.dev>
* removed extraneous MIDI comments
* removed extraneous comments
* Add RGB matrix suspend wake function for Planck/rev6
* Update suggested definition to allow user override.
Co-authored-by: Nick Brassel <nick@tzarc.org>
Co-authored-by: filterpaper <filterpaper@localhost>
Co-authored-by: Nick Brassel <nick@tzarc.org>
Co-authored-by: Joel Challis <git@zvecr.com>
Co-authored-by: George Wietor <george@LIGHT-GRAY.local>
Co-authored-by: Jack Humbert <jack.humb@gmail.com>
* 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>
* added default as a starting point
* added readme per guidelines
* made necessary changed to keymap to follow guidelines
* changed debounce mode for more smooth usage at the cost of mem
* increased polling interval
* Revert "made necessary changed to keymap to follow guidelines"
yikes! I modified the WRONG directory. reverted.
This reverts commit 8ba3d97af538796d765c47b48537dbcf55c3dbc1.
* reverted commit that modified file in incorrect directory. modified intended file.
* added license tp config.h
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