* fixed a few typo's in comments
* fixed 8266 specific warning about 'comparison of integer expressions of different signedness'
based on recommendations made by @willmmiles:
* make sure that audioSyncPacket is the same size (44bytes) on all platforms
* use static buffer for receiving (avoids heap fragmentation)
* copy receive buffer to local audioSyncPacket struct - avoids alignment problems
* esp32 only: to stay in sync with UDP, Udp.flush() is needed when Udp.parsePacket() is _not_ followed by Udp.read()
audioSyncPacket contains four "invisible" padding bytes added by the compiler. These need to be initialized to zero, as future versions of the protocol will make use of these fields.
Many bad quality analog mics are not centered properly at 1.6V, but stuck at 0V or stuck at 3.3V in silence. The bandpass filter removes DC offsets and improve signal quality.
Remove sync receive
Disallow 2D effects on non-2D segments
Optimisations
Sync clarification
AR palettes
Return of 2 audio simulations
Bugfix in sync #3344
- remove excessive segments
- ignore inactive segments if not syncing bounds
- send UDP/WS on segment change
- pop_back() when removing last segment
Add pairing support for ESP-NOW sync
Reduce string RAM footprint
UDP parse optimisation
Make WizMote work with sync.
ESP-NOW wireless sync POC.
- caveat: devices have to be on the same channel
- clashes with WizMote handling ATM
* remove "not supported" warning on -S2
* minor tweaking of beat detector
* optimized parameters for arduinoFFT
* fixed a few implicit "double" promotions
* minor UDP protocol optimizations
* small optimization for fft task create / suspend
* completely remove analogmic settings for -S3/-S2/-C3 where analog is not supported
changes were already tested extensively in the MM fork.
* audioreactive driver update
- Better handling of PDM and I2S Line-in
- Bugfixes for ES7243 (allocateMultiplePins)
- More error messages for ES7243
- sample scaling (needed for sources that use full scale of samples)
* audiorective update
* align SR_DEBUG with WLED_DEBUG
* optional bandpass filter (needed for PDM mics)
* sample scaling for PDM and Line-In
* small improvements for analog input
* bugfixes and small performance improvements
* code for FFT task refactored, for better readablity. Introduces separate functions for filtering and post-processing
* small improvement for beat detection
* default mic settings can be configured at compile time
* correct mic type if MCU does not support PDM or ADC
* hide analog PIN config if not supported by MCU
* audioreactive updates
- minor updates to source code (see discussion in PR #2907)
- usermod readme improvements
* small readme update
* one think I overlooked
* ok, another edit. Now its final. Hopefully.
* small upps
wrong parameter order in debug message.
* Grouping um settings, add pre and post Info, update SR & 4LD settings
Settings_um.htm:
- modify addField (grouping)
- addInfo (pre and post texts)
Add preInfo to audio reactive and 4ld usermod
Extra:
platformio: wemos_shield: add audio reactive usermod and update to alt display
* um settings: cpp: lowercase, js: initcap
* um settings: txt only pre, txt2 only post and initCap as function
* Fix rotary encoder info string
Co-authored-by: Blaž Kristan <blaz@kristan-sp.si>
it's possible that volume samples become negative. In this scenario, our simple noise gate does stupid things, and it looks like "effects temporarily lost the sound".
This fix improves the situation, and makes sure that volume samples are always >= 0.
16kHz might be a good compromise on small MCU's:
- GEQ will show frequencies up to ~6Khz
- FFT process may use up to 32millis (-> 100% CPU load). Try to stay below <60% so FreeRTOS can schedule all tasks properly.
- more CPU time is left for other WLED, so it maintains LED FPS and stays responsive on web UI.
Only run FFT when the result will be used.
Please note that this also means that `FFTTime` shown in Info is only meaningful when there is sound input and not silence.
--> To get exact FFT times, the optimization can be disabled by compiling with `-D SR_DEBUG`
- the connected() method only get called once a Wifi STA connection is established. UDP Sound Sync should also work when sender is in AP Mode.
- added a few comments that should help to understand the code structure.
Frank