Recent content by user44455555

I do measure now from 2 hz instead 20 hz of soundbox loopback. its low level on a 16 bit UMC22 bwehringer USB from 2 hz let the excess group delay at 40 hz much more valid. for rt60 or rt60 decay this not help. from 20 hz

this explain alot. I do soundcard cal then the phase get better, but excess phase go more minus. And when i measure a speaker i compare and i see now that in bass the excess phase go in that value lower as the excess phase from soundcard go lower. so better dont use excess phase for speaker...

then the question is how can get correct results in phase and RT 60 ?. if there is no exact way then i think it is more near the truth how much phase correction need when subtract in my case 67 degree when i want see real phase at 50 hz

I do always estimate IR delay and it shift 0,1 sample. do you think this is correct ? . the step response look strange is this because the sounddevice is very bad or is that normal look ? . it is a focusrite scarlett which is test good the impulse response

or a better idea . is it possible when do a calibrate soundcard that this is the reference and the result of the soundbox loopback is subtract from the measurement automatic ?.

I choose third octave this give more delay in low. I choose time reversed filtering it reduce to 200 ms at 50 hz and then filter order 6 then i get 120 ms. have the time reversed filtering any disadvantages ?. At least good to see that it is not only my room that increased that so much. is it...

Hello I think there is no feedback, because distortion is very low and FR is very linear. I choose estimate IR and create minimum phase. attach is the mdat file too . maybe a setting is not good or REW problem ?

here i upload the rew measures. in this measure speakers are on desktop with tilt foam 6 degreee up. the microphone was on same place on stative for both measures on ear height. with sweep or pink random it sound not good. but with periodic it sound after correction in compare to headphone and...

Hello pink random (with a rta snapshot) sound and look very simular to sweep. but in the rta the periodic pink give better results.it have much less random variations during rta. you can see in my screenshot or try yourself. the speakers stand on a desktop on foam microphone is on ear height...

I notice now if a cal file did not have a sensitivity or sensfactor inside it do not set back the value that was in the spl meter calibrate choose. I better add in all my 3 cal files sensivity values, then i can set my input gain of soundbox to middle and change sensitivity facter in cal file...

yes i edit it with a editor and save it . i try reload , or start rew new, switch to other cal file and then reload the edit. stay always same. i use the spl meter long time ago for this input. maybe it can disable something ?. I use asio device. .I know from the usb ears device that it accept...

I have three microphones(all XLR) and i record a measure and depend on microphone i change later the cal file fo compare.microphone results. I try this line in cal file on top to see if anything change Sens Factor =4.0dB or Sensitivity -12.34 dBFS as explained here...

hello I think best sound get when smothing is 1/1 . rew use now LS filter because i enable it for -6 - + 6 db. there is no Q in LS. is it always 1 ?. DSP use for default 0.74. I notice that always the filter curve is mirror in X in eq setting. so a + LS got to - . see the compare that 1/1...

so when i do 4 measure sweeps i get 6 db less noise floor which mean half. when noise floor is at a freuency 0.1% then i get on this frequency only 0.05% noise floor level ?. high harmonic 7. th and 9. th sound really bad even if it is only 0.1% 0.05% is then good for my ears.

I understand that in RTA with constant frequency noise floor is lower as a single sweep.. but i use 4 sweeps(set in measure dialog). it reduce noise floor not much. see screenshots for compare here can see clear 5 and 7 harmonic here noise floor is much too heigh. how can get this less noise...