- 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.
improved ADCsample processing, including replacement of "rogue" samples from other channels (this happens at least once in 5 seconds !!).
It compiles, don't ship it yet - needs more testing.
- 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.
- change arduinoFFT to float (custom)
- update audioreactive to use float
- update effects to use float
- info slider (usermod)
- hide Peek in 2D
- minor bugfixes
