(non Audacity) Post-processing trouble

[Shaky]

Thanks very much Steve, that is very helpful indeed.

I get how an analogue filter (emulation) wouldn’t produce the same results as a purely digital manipulation like Fast Fourier, and now feel comfortable in accepting the outcomes on your say so.

From my old clubbing days I always though Turbosound had some of the best sounding systems around, and a quick scan of their site seems to show their subs go down to around 25Hz.

So I’ll try the high pass filter again with that as the cut-off, and would like to use Koz’s suggestion of 24db as the roll off, to break the back of that possible hump on the spectrum I posted above around 12-15Hz.

However, having searched and found Steve’s post here http://forum.audacityteam.org/viewtopic ... 20&t=58247
on setting Q, it seems that the 24db roll-off requires a non-default q setting.

Is there any guidance on what that should be?

[steve]

From my old clubbing days I always though Turbosound had some of the best sounding systems around, and a quick scan of their site seems to show their subs go down to around 25Hz.

They also make some great out-door rigs.
The low frequency range varies depending on the model, but for big subs it's usually somewhere in the region of 25-40 Hz. Anything that goes below 20 Hz is an engineering research tool rather than something for entertainment (though vibrational stress testing can be quite entertaining )

However, having searched and found Steve’s post here viewtopic.php?f=20&t=58247
on setting Q, it seems that the 24db roll-off requires a non-default q setting.

No. the 24 dB roll-off does not use the Q setting.
The only rolloff setting that uses the Q setting is "12 dB per octave".
All other rolloff settings ignore the Q value.
(This is why I think the setting is confusing and should be removed.)

When rolloff is set to 12 dB per octave, the default (Q = 0.7071) is the optimum setting and produces the expected response for a 12 dB/oct filter (-3 dB at the cutoff frequency then 12 dB/oct).

12 dB/oct 1000 Hz Q=0.701

Frequency Analysis_005.png (41.82 KiB) Viewed 1082 times

If the Q is set higher than the default, there is a peak in the filter response just before the set frequency.

12 dB/oct 1000 Hz Q=10

Frequency Analysis_003.png (39.35 KiB) Viewed 1082 times

If Q is less than the default then the filter "knee" is more gradual so the cutoff begins before the set frequency and is less than 12 dB/octave.

12 dB/oct 1000 Hz Q=0.2

Frequency Analysis_004.png (41.61 KiB) Viewed 1082 times

[Shaky]

Steve, I am really, really grateful for the time you are taking on this, it is much appreciated.

However, having searched and found Steve’s post here viewtopic.php?f=20&t=58247
on setting Q, it seems that the 24db roll-off requires a non-default q setting.

No. the 24 dB roll-off does not use the Q setting.
The only rolloff setting that uses the Q setting is "12 dB per octave".
All other rolloff settings ignore the Q value.
(This is why I think the setting is confusing and should be removed.)

When rolloff is set to 12 dB per octave, the default (Q = 0.7071) is the optimum setting and produces the expected response for a 12 dB/oct filter (-3 dB at the cutoff frequency then 12 dB/oct).

OK, I’ll take your word for this. But when I compare the results of the 12db roll off with q=0.7071 versus a 24db roll off (and the same q you say does nothing) there appears to be a materially greater resulting magnitude in the new waveform peaks using the latter settings.

Based on your examples I would have expected them to be roughly the same if the Q implied in the latter calculation were optimised, and as such it could appear this is analogous to the situation you show where Q is greater than the optimum ‘default’.

However, I’ll just put this down to the vagaries of the analogue style filter, if you’re sure there is no error in the internal calculations.

In any case many, many thanks,

[steve]

I’ll just put this down to the vagaries of the analogue style filter

It perhaps does look a little unexpected, but actually produce a very close match with the results that would be predicted for this type of filter. The mathematics are rather complicated (over my head), but if you want to look it up, these filters are "biquad filters" (biquadratic) http://en.wikipedia.org/wiki/Digital_biquad_filter

Here's an illustration of the phase shift that occurs.
This is a 1kHz sine wave. The first track is unfiltered and the other tracks are each filtered with a high-pass filter. The rolloff is set to:
Track 2: 6 dB/oct
Track 3: 12 dB/oct
Track 4: 24 dB/oct
Track 5: 36 dB/oct

You can see how, as the steepness (number of dB's per octave) increases, the phase waveform is gradually shifted backward (to the left)

tracks002.png (29.89 KiB) Viewed 1071 times

The actual amount of phase shift depends on the frequency of the waveform being processed and the filter settings being applied. Because some frequencies are shifted more than others the frequency components in a sound will combine differently after filtering to produce different (a bit higher or lower) peak levels at any specific point in time. Human hearing is largely insensitive to phase shift, but is highly sensitive to frequency content, and the overall effect of these filters is to produce a precise and predictable change in the frequency content (as illustrated by the spectrum plots in my previous post).

I don't understand the mathematics well enough to say if the phase shift is exactly as the theory says that it should be, but the frequency response is certainly a very good match with an idealised filter.

[kozikowski]

Anything that goes below 20 Hz is an engineering research tool rather than something for entertainment

The Crown DC 300 'product' is a shake table driver. Not a speaker amplifier. The audio people found it and ran with it. True, it didn't sound like anything else on the market.

Nowhere is it written that the bass in the club has to be in the vinyl. I have the early version of this...

http://www.dbxpro.com/120A/

And it produces lovely 16' pipes that weren't there in the original organ performance.

Koz

[kozikowski]

While the ear doesn't respond well to phase problems, the effect isn't zero. Sound system with low phase shift are frequently described as more "open" and "clear" and less "restricted."

This is also the reason square wave testing keeps going into and out of favor. It goes into favor when people find it can accurately describe their equipment shortcomings. It goes out of favor when the manufacturers find it can accurately describe their equipment shortcomings.

And just to be clear, Fourier described the number, frequency and phase of different pure sine waves need to make up any other waveform including, notably, square waves. In the best case of "everything is hooked to everything else," mess with the response anywhere and the effects ripple all over. Particularly since FFT is Best Guess. It's not perfect.

Koz

[steve]

Sound system with low phase shift are frequently described as more "open" and "clear" and less "restricted."

That's a really vague statement most likely to be found in audio equipment adverts.
Phase shift within the crossover of a speaker system can cause comb filtering due to the phase difference between the speaker drivers (tweeter and woofer) which will interfere with each other in the crossover frequency band. One of the things that make PA speakers by Turbosound so good is that they have gone to great lengths to ensure that all the speaker drivers are in phase with each other.
Phase shift is often deliberately introduced in high quality live sound reinforcement systems to avoid acoustic feedback.

[Shaky]

Many thanks to all for the further comments.

Steve, I think I’m a bit like you in that when I see a complex looking equation my eyes glaze over and the mind goes numb.

Considering the entire discussion I am therefore more inclined to revert to the 12db roll off - firstly because the parameters seem more clear cut and secondly to minimise the new peak amplitude - but instead up the cut-off to maybe 30Hz or even a little more.

I keep turning Koz’s comments about the lower frequencies I am capturing not coming from the vinyl. In understand that position, but I can see many waveforms recorded with the same setup up where the spectrum falls off the chart at around 20db as expected, which seems to suggest it is not the overall equipment setup.

Secondly the waveforms with all the action at the low frequencies look similar regardless of whether I am sampling for the spectrum at the start or the end of tracks suggesting to me some external environmental factor is not the cause.

However, just by guesstimation it seems to me that these issues are largely cropping up on 90s dance type records. Is it not possible that associated - shall we say - informal methods of production and mastering eg in the bedroom are responsible for this?

In any case it might be wise to get rid of more low frequency stuff in the frankly unlikely event I ever again get to spin some of these tunes on a monster Turbosound system

[kozikowski]

Is it not possible that associated - shall we say - informal methods of production and mastering eg in the bedroom are responsible for this?

Before the RIAA forced the adoption of one vinyl customization curve, everybody had their own and each one was far better than the others. I wouldn't be shocked to find dance records tailored to the Stanton 680 phono cartridge (for example).

http://www.copystars.com/stanton_d_6800 ... 4-prd1.htm

It was the one that came, or was caused to come with a majority of Techniques SL1200 turntables (in the US).

http://en.wikipedia.org/wiki/Technics_SL-1200

It sounded OK, was hard to break, and you could back cue with it. That back-cue talent caused it to have "restrained" low frequency response.

And everybody knew that. The SL1200 was well damped and had all those ropes and pulleys (or modern equivalent) that would keep the needle in the groove in the face of a small, intimate thermonuclear event, so it was open season on bass music.

Vinyl cutters have "look ahead" to make sure that bass-heavy grooves don't damage each other. They widen the cutter groove pitch before the music gets there. The joke was Disco/Trance/Electro disks had one song on each side and the grooves were wide enough to lose your car keys in there.

It had terrific bass line, though.

Koz

[Shaky]

Thanks Koz, very, very interesting.

I am in fact using Technics 1210 MKIIs but equipped with Ortofon Nightclub carts rather that the Stantons that came with them.

Does this shed further light on where the awesome raw sound they produce starts to distort? Or a solution?