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.
- Info Page: add a small horizontal line below usermod specific part. Improves readability.
- updated 2D mapping mode of some 1D soundreactive effects
- alllow some effects to fade slowly, even slower that possible with SEGMENT.fade_out(). Looks nice.
not sure why - most effects only fade when using SEGMENT.fade_out(), while some need SEGMENT.fadeToBlackBy().
- save 1K of RAM by optimizing out
fftBin[].
- moved several copies of the peak reset code into a single function
- moved peak detection out of getSample().
- call peak detection function as last step of FFTcode. More optimal, and we can be sure that fresh FFT result are available.
Peak detection/reset are now called from both tasks, so I had to move some peak-related vars out of AudioReactive class and make them global (static).
- put variables with same context next to each other.
- removed a few vars that are not needed any more.
- replaced "16" by a more descriptive constant
... found that stupid commit messages get more attention ;-)
- use 22050 Hz for sampling, as it is a standard frequency. I think this is the best choise.
- redesigned the GEQ channels (fftResult[]) for 22Khz, based on channels found on old HiFi equalizer equipment. 1Kzh is now at the center; Bass/Trebble channels are using 1/4 on left/right side respectively - similar to real equalizers. Looks nice :-)
- adjusted effects that use FFT_MajorPeak so that the maximum frequency is supported.
- new feature: "Input Level" (info page) can be used as global "GEQ gain" - only when AGC is ON (was already possible when AGC=off)
- some parameter tweaking in FFT function
- hidden feature: FFT decay is slower when setting a high "dynamics Limiter Fall time" (steps: <1000, <2000, <3000, >3000)
- FFT_MajorPeak default 1.0f (as log(0.0) is invalid)
- FX.cppp: ensure that fftResult[] is always used inside array bounds
- removed broken FFTResult "squelch" feature. It was completely broken, and caused flashes in GEQ.
- added Frequency scaling options: linear and logarithmic
- fixed a few numerical accidents in FX.cpp (bouncing_balls, ripplepeak, freqmap, gravfreq, waterfall)
- On/Off controls the complete feature
- Rise Time and Fall Time are the minimum times (in milliseconds) for "volume" to go from 0% to 80% and back.
- when "On" we also use some filtering to smooth FFTResults[]. Rise and Fall Times do not affect Frequency reactive effects otherwise.
* Header checking for sound sync receiver: removed wrong "!"
* make sure all member vars have initial values
* some robustness improvements in case of receiving bad UDP data.
- smoothing FFTResult (don't have a matrix to test)
- UDP sound sync improvements
- some bugfixes from SR WLED
- button.cpp: avoid starvation: strip.isUpdating() can be true for a long time.
work in progress - still needs testing!!
-new methods: getType(), isInitailized(), postProcessSample()
- allow users to compile for RIGHT audio channel (-D I2S_USE_RIGHT_CHANNEL)
- better handling in case audio input driver failed to initialize
- removed some unneeded code and unneeded parameters
UDP audio sync: introduced new header version, because the new struct (without myvals[]) is not compatible with the previous struct. Also optimized structure size.
UDP audio sync: sender decides is AGC or non-AGC samples are transmitted.
getsamples: move volumeSmth/volumeRaw code out of AGC core function.
gain =1 does not make much senses, at it means "0.0825"; 40 internally translates to "1". 60 seems to be a good start.
- Don't use ADC analog microphone as default, to avoid well-known conflicts with other stuff hooked up onto ADC1,
- re-enabled a forgotten delay (overlooked that in my last commit)
- same fix as in SR-WLED upstream
- if strip.isupdating() is true for more than 12ms, run audio filter loop regardlessly. The userloop is very fast, so I'm expect no bad side-effects from this.
