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* Extensible split data sync capability through transactions.
- Split common transport has been split up between the transport layer
and data layer.
- Split "transactions" model used, with convergence between I2C and
serial data definitions.
- Slave matrix "generation count" is used to determine if the full slave
matrix needs to be retrieved.
- Encoders get the same "generation count" treatment.
- All other blocks of data are synchronised when a change is detected.
- All transmissions have a globally-configurable deadline before a
transmission is forced (`FORCED_SYNC_THROTTLE_MS`, default 100ms).
- Added atomicity for all core-synced data, preventing partial updates
- Added retries to AVR i2c_master's i2c_start, to minimise the number of
failed transactions when interrupts are disabled on the slave due to
atomicity checks.
- Some keyboards have had slight modifications made in order to ensure
that they still build due to firmware size restrictions.
* Fixup LED_MATRIX compile.
* Parameterise ERROR_DISCONNECT_COUNT.
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* 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 no |
