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Posts by cjwhitehouse

My statement was indeed a generalisation and it is always possible to cite exceptions, but as a rule, most subwoofer distortion behaviour is dominated by the excursion issue down low and this is what soho54 was trying to emulate. I thought the idea was to try and prove/disprove whether the CEA2010 distortion thresholds are a good measure of distortion audibility and I seem to recall that it was your request to do the test with a distorted music signal rather than sine tones.
OK, but if you drop the peak input signal to the distortion program to -6dBfs then by my reckoning, your 10% distortion pattern will then yield less than 2% on the peaks and correspondingly less over the rest of the signal.
Thanks for doing this. When I tried this, the dips were even more pronounced, especially with the 6th order version. I'm not sure exactly what the vertical scale is on your plots but the point I was seeking to illustrate is that the dips in the frequency response around the crossovers are not insubstantial in relation to the distortion effects you are trying to simulate.
Thinking more about this, I am surprised you are hitting a clipping issue. I think Keith's program should automatic attenuate the output to avoid any clipping. The test.wav file he provides as an example is recorded at 0dBfs and you can happily apply distortion to that without clipping. So I am puzzled by your problem. To illustrate what I was saying about harmonic polarity, attached are a couple of charts based on my own measurement of the PB13 in 10Hz tune, delivering...
The point of using the very low crossover and shallow slope is to apply a differential gain to the bass. That way, content at 15Hz, 30Hz and 60Hz will be presented to the distortion process at 0dB, -6dB and -12dB respectively. Because the effect of the static distortion pattern diminishes as the amplitude reduces, distortion will occur mostly at the bottom end as in a real subwoofer. When stitching back with the high-passed version of the signal, the proper level of 30Hz...
I am wondering if you might be better to apply a low pass filter only at, say 15Hz with a shallow slope like 1st (or maybe 2nd) order, put that into your distortion program and use the shallow rolloff to provide the distortion "emphasis" to the lower frequencies. This is probably a better model of a typical subwoofer distortion profile than trying to split it into two bands like you are. As you will no doubt have found, Keith's program tends to increase the signal length...
OK, since you seem to be pretty accomplished with Cool Edit Pro, humour me and try this experiment. Generate yourself a -5dB log swept sine sweep from, say 10Hz -> 200Hz using the Tone Generate function. Then put it through your Cool Edit Pro filter regime, stitch it back together and look at the waveform envelope you end up with. Try it with 6, 4 and 2 order filters if you have time. I'm intrigued to see what you get by this process without even applying any distortion....
I was referring to the Cool Edit Pro filters you are using to split the signal into frequency bands rather than the filters used in the upsampling process within Keith's program. What kind of filters are you using? Butterworth, Bessel, Chebychev 1 or 2? What slopes? I'm not saying you are wrong but naysayers will likely treat this as another variable in play.
It could be done but it's a bit of a fiddle and the additional filtering to split the signal and stitch it back together afterwards may start to add other unwanted artefacts and noise.
Another thing to bear in mind when adding distortion using these programs is that with a single tone signal you will get harmonic distortion only. If you start with a signal consisting of more than one tone you will also get intermodulation products generated. With a music signal you will get a mass of such intermodulation components. These intermodulation products will be at frequencies that will tend to be discordant. Although auditory masking comes into play to some...
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