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Mastering Captured Vinyl For CD
audioapprentice
post Nov 29 2008, 01:34
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I want to make (in CEP) a 20Hz to 20kHz bandpass filter to apply to my vinyl captures but I have 3 questions about filtering.

1. What is better suited for this purpose, an FFT filter or one of the scientific filters (e.g. Butterworth)?

2. Is there anything wrong with making it a brickwall filter or should the filter cutoff be gradual, e.g making the lower cutoff transition band start at 30Hz (100%) and finish at 20Hz (0%)?

3. Is it better to do a high pass filter then a low pass filter or a single bandpass filter?

Thanks in advance for any guidance.
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Arnold B. Kruege...
post Nov 29 2008, 03:53
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QUOTE (audioapprentice @ Nov 28 2008, 19:34) *
I want to make (in CEP) a 20Hz to 20kHz bandpass filter to apply to my vinyl captures but I have 3 questions about filtering.

1. What is better suited for this purpose, an FFT filter or one of the scientific filters (e.g. Butterworth)?


The FFT filter is the best choice if you're looking for a one size fits all solution.

Sometimes you run into a filter that is hard to draw on the limited resolution graphic that the FFT provides. Then it the scientific filters can help. For narrow notches, such as removing hum, the DTMF filters really shine.

QUOTE
2. Is there anything wrong with making it a brickwall filter or should the filter cutoff be gradual, e.g making the lower cutoff transition band start at 30Hz (100%) and finish at 20Hz (0%)?


The general rule is do what sounds best.

QUOTE
3. Is it better to do a high pass filter then a low pass filter or a single bandpass filter?


Theroretically, you want to do as much as possible with as few processing steps. So, if you can get what you want with just one pass and a bandpass filter, then this is a good idea.
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Kees de Visser
post Nov 29 2008, 09:44
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What bandlimiting is present in your (digital?) captures ? Perhaps the filtering in your ADC is already good enough.
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DVDdoug
post Dec 3 2008, 01:25
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WARNING - I'm not a DSP expert, so some of this might be wrong. tongue.gif

Why do you want/need to filter? I've never done that with my vinyl transfers. Mostly, I've worked on noise reduction, and sometimes I've used some EQ (high end boost) to fix-up old "dull sounding" recordings.

Any high frequencies (above Nyquist) should be filtered by the soundcard/ADC. (This may not be true for all soundcards, but any good soundcard should have an anti-aliasing filter.) And, if you're downsampling later, the downsampling algorithm/process will also include an anti-aliasing filter.

I'd guess the only reason for low frequency filtering would be subsonic rumble? I've never used a rumble filter, although a simple subsonic filter may have been built into my phono preamp. And, my noise-reduction processing may have removed some subsonic noise. I suppose it wouldn't hurt, but I'd probably set the cuttoff to something like 15Hz, just to preserve any 20Hz sounds... My speakers can't reproduce 20Hz, but I'd "feel better" about it. wink.gif

QUOTE
I want to make (in CEP) a 20Hz to 20kHz bandpass filter...
Keep in mind that the specified cutoff frequency is the -3dB point. So, if you specify a "20 - 20kHz" filter, your signal will be down exactly 3dB at 20Hz and at 20kHz (no matter what type of filter, or how steep it is).

QUOTE
1. What is better suited for this purpose, an FFT filter or one of the scientific filters (e.g. Butterworth)?
A Butterworth filter is probably a good choice. It has the flattest possible passband. This is a good choice whenever its more important not to affect the passband than it is to filter-out stuff outside the passband. A different filter design might be appropriate whenever it's most important to kill some noise (or signal) outside the passband, and in order to accomplish that, you have to accept with some "ripple" in the passband.

You can do amazing things with an FFT filter, but if I understand how FFT filters work, they're "messy" and require lots of approximation and compromise... I think, just the FFT followed by inverse-FFT (without any filtering) is a lossy process... I'm pretty sure you can't get your exact original data back.

QUOTE
2. Is there anything wrong with making it a brickwall filter or should the filter cutoff be gradual, e.g making the lower cutoff transition band start at 30Hz (100%) and finish at 20Hz (0%)?
I think the only "cost' for a steep filter is the processing complexity and processing time (not an issue, since you are not working in real-time). But, there may be some phase/delay issues... I'm not sure.

QUOTE
3. Is it better to do a high pass filter then a low pass filter or a single bandpass filter?
I'm pretty sure the bandpass is made-up of combined high-pass and low-pass filters anyway... So, I wouldn't expect any difference.

This post has been edited by DVDdoug: Dec 3 2008, 01:35
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Glenn Gundlach
post Dec 3 2008, 07:39
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There is intentionally nothing on an LP below 32Hz. Different turntable cartridge combinations have resonances that can cause skips with low frequencies so to avoid the issue, it is simply left out. I had an LP cut with 16-32 Hz classical organ and while it played fine on my Dual 721 with a Shure V-15 type V, other very expensive systems simply couldn't handle it.

A first or second order High pass with the corner at 30Hz would be reasonable for LPs.

Agreed on the HP / LP / bandpass issue. I'd do it in 1 pass also.

G

Moderation: Removed unnecessary quotation of previous post.

This post has been edited by greynol: Dec 3 2008, 18:25
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honestguv
post Dec 3 2008, 09:14
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> 1. What is better suited for this purpose, an FFT filter or one of the
> scientific filters (e.g. Butterworth)?

For my curiousity, what is a scientific filter? Is an FFT an unscientific filter? Are your FFT filters limited to powers of 2 in the number of samples involved?

As mentioned earlier, a high pass Butterworth filter at 30 Hz is a reasonable starting point for removing any low frequency rubbish that may be limiting the real signal. If your find you have quite a lot of it with your particular setup then a bit of experimenting with the cut-off point and slope might be worthwhile.

As mentioned earlier, I am not sure a low pass filter will do much for you since the A->D step should handle that efficiently.

Are you processing at a higher resolution/bit depth and then generating the target resolution/depth as a final step?

> 2. Is there anything wrong with making it a brickwall filter or should the
> filter cutoff be gradual, e.g making the lower cutoff transition band start at
> 30Hz (100%) and finish at 20Hz (0%)?

Brickwall filters will ring and this is often judged to be worse than a less efficient filter which rings less or, perhaps, not all.

> 3. Is it better to do a high pass filter then a low pass filter or a single
> bandpass filter?

In your case, it is unlikely to matter but 1 pass is probably going to be a bit quicker. However, for example, aggressive narrow band pass filters can have stability issues with some types of filters and so the option for more than 1 pass is usually necessary in a general setup.
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Arnold B. Kruege...
post Dec 3 2008, 14:17
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QUOTE (honestguv @ Dec 3 2008, 03:14) *
For my curiousity, what is a scientific filter? Is an FFT an unscientific filter? Are your FFT filters limited to powers of 2 in the number of samples involved?


Scientific filter is just a name that the CEP authors gave to a class of filters. These filters are the classic canonical filters from the days of analog.

QUOTE
As mentioned earlier, a high pass Butterworth filter at 30 Hz is a reasonable starting point for removing any low frequency rubbish that may be limiting the real signal. If your find you have quite a lot of it with your particular setup then a bit of experimenting with the cut-off point and slope might be worthwhile.


Agreed that some kind of high pass filtering will often remove a lot of garbage.

When I filter the low end, I look to the recordng to be a guide as to how high to set the cutoff. If you do a FFT analysis of a whole track, the actual musical tones wiill cause distinct families of clearly visible spikes. Repetitive mechanical and electrical noises cause splkes at multiples of things like power line frequencies. Random mechanical and acoustical noise will cause relatively indistinct broad undulations.

CEP is very nice in that the filters have a preview filter. Click and drag a critical passage, select a filter, click the preview button, and adjust filter paramaters, wait for a second or two for them to become effective, and adjust until things sound right.

QUOTE
As mentioned earlier, I am not sure a low pass filter will do much for you since the A->D step should handle that efficiently.


Low pass filters are usually reserved for really noisy vinyl/

QUOTE
Are you processing at a higher resolution/bit depth and then generating the target resolution/depth as a final step?


IME, a waste of time unless you have a tic/pop algroithm that works better with really sharp-edged tics.

QUOTE
> 2. Is there anything wrong with making it a brickwall filter or should the
> filter cutoff be gradual, e.g making the lower cutoff transition band start at
> 30Hz (100%) and finish at 20Hz (0%)?

Brickwall filters will ring and this is often judged to be worse than a less efficient filter which rings less or, perhaps, not all.


The idea that brickwall filters necessarily ring is an urban myth. The cause of this myth is a misunderstanding that is caused by a misinterpretation of waveform pictures. Just because a picture of a square wave shows something that looks like a damped sine wave near leading and/or trailing edges, is not proof of ringing. This sort of thing can be caused by phase shifts.

QUOTE
> 3. Is it better to do a high pass filter then a low pass filter or a single
> bandpass filter?

In your case, it is unlikely to matter but 1 pass is probably going to be a bit quicker. However, for example, aggressive narrow band pass filters can have stability issues with some types of filters and so the option for more than 1 pass is usually necessary in a general setup.


Agreed.
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2Bdecided
post Dec 3 2008, 17:32
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QUOTE (Arnold B. Krueger @ Dec 3 2008, 14:17) *
The idea that brickwall filters necessarily ring is an urban myth. The cause of this myth is a misunderstanding that is caused by a misinterpretation of waveform pictures. Just because a picture of a square wave shows something that looks like a damped sine wave near leading and/or trailing edges, is not proof of ringing. This sort of thing can be caused by phase shifts.
Steady on - by definition, the sharper you make the cut off of a linear phase filter, the longer it will ring for. If the transition band is in the audible range, and there is original content in the vicinity of the transition band, you will hear the ringing.

If it's not constrained to be linear phase, you can do funky things with the ringing - a usual trick is to minimise pre-ring, while allowing post-ring. It'll still ring though!


FWIW filtering out inaudible frequencies seems a bit of a waste of time, and can be very hit and miss - mainly because until you find the scenario where you do need to remove them (e.g. you get much bigger/better speakers and can suddenly hear 20Hz rumble, or you apply some processing that's sensitive to these frequencies), you won't know whether your filtering did more good than harm.

Cheers,
David.
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2Bdecided
post Dec 3 2008, 18:21
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Oh, I've just noticed the thread title(!).

You don't need to apply a bandpass filter to put a 44.1kHz 16-bit wave file on a CD. Just stick it on there. CD is a 0Hz-22.05kHz medium, whatever the specs for most CD players say.

It won't cause any problems - unless there's a huge DC offset, in which case you might hear a click when skipping tracks.

Cheers,
David.
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Arnold B. Kruege...
post Dec 3 2008, 21:37
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QUOTE (2Bdecided @ Dec 3 2008, 11:32) *
QUOTE (Arnold B. Krueger @ Dec 3 2008, 14:17) *
The idea that brickwall filters necessarily ring is an urban myth. The cause of this myth is a misunderstanding that is caused by a misinterpretation of waveform pictures. Just because a picture of a square wave shows something that looks like a damped sine wave near leading and/or trailing edges, is not proof of ringing. This sort of thing can be caused by phase shifts.


Steady on - by definition, the sharper you make the cut off of a linear phase filter, the longer it will ring for. If the transition band is in the audible range, and there is original content in the vicinity of the transition band, you will hear the ringing.


Not true. There's stuff happening, but there is no ringing.

You can do this for yourself in Cooledit/Audition.

(1) Pick a high sample rate like 24/196 so that any artifacts will be minimal.
(2) Generate a 10 Hz square wave at -20 dB peak, again so that any effects won't go off scale.
(3) Use the FFT filter to put in a brick wall filter (0 dB to -100 dB) low pass in at 6 KHz.

OK so you look at the resulting wave form and there appears to be some *ringing* around the leading edges.

I assert that yes there is what appears to be a damped sinusoid, but it isn't ringing.

If it is ringing, then there will be a peak in the spectral analysis, right?

So run a spectral analyis with 65 k points and show me the peak in response. What we expect to see is the odd harmonics of 10 Hz with amplitude inversely proportional to the frequency, and then they will be attenuated about 100 dB by the brick wall at 6 KHz.

If we look at the square wave, we see what appaears to be a damped rippling effect with a period of about one millisecond.

This is going to be like shooting ducks in a barrel, if it is ringing, there will be a peaking in response around 1 KHz. At one KHz the harmonics of the 10 KHz square wave are going to be coming thick and fast, so the ringing around 1 Khz will be *very* apparent, right?

It ain't there!

The apparent ringing is due to phase shifting of the harmonics of 10 Hz. To add up to be a square wave their phase and amplitude relationships of the harmonics must be very precisely like a square wave.

The brick wall filter messes that up. We don't get a pure square wave.

There is no ringing to hear. What we do hear is the straight-forward effect of the brickwall filter.

QUOTE
If it's not constrained to be linear phase, you can do funky things with the ringing - a usual trick is to minimise pre-ring, while allowing post-ring. It'll still ring though!


A messed up square wave does not necessarily prove the existence of any ringing.
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Axon
post Dec 3 2008, 21:59
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QUOTE (Arnold B. Krueger @ Dec 3 2008, 14:37) *
I assert that yes there is what appears to be a damped sinusoid, but it isn't ringing. If it is ringing, then there will be a peak in the spectral analysis, right?
Absolutely not. If there were a peak, you would have resonance. Resonance is not ringing. Ringing is temporal spreading - and a dampened sinusoid is quite obviously an instance of that. A long FFT is completely insensitive to that sort of thing.

FWIW, the preringing for foobar2000's FIR equalizer is ludicrously audible. I once did a 5db peak followed by a 5db cut and could still ABX the ringing!

QUOTE
The apparent ringing is due to phase shifting of the harmonics of 10 Hz. To add up to be a square wave their phase and amplitude relationships of the harmonics must be very precisely like a square wave. The brick wall filter messes that up. We don't get a pure square wave. There is no ringing to hear. What we do hear is the straight-forward effect of the brickwall filter.
You're specifically referring to the Gibbs phenomenon here - phase shifts have nothing to do with this. The square wave just had its harmonics lopped off.

If the temporal smearing or ringing or dampenened sinusoids (or whatever you want to call them) exceed the temporal masking threshold, no matter what you call the phenomenon, it will be audible. That should not be that hard to reproduce for filters exceeding 4000 samples for khz-range filtering. FIR filtering of rumble is usually a fool's errand, but I could imagine a suitably high-Q filter operating at 30hz to be plausibly audible with a suitable choice of listening equipment. I have no evidence to back that statement up though
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Arnold B. Kruege...
post Dec 4 2008, 00:26
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QUOTE (Arnold B. Krueger @ Dec 3 2008, 14:37) *
I assert that yes there is what appears to be a damped sinusoid, but it isn't ringing. If it is ringing, then there will be a peak in the spectral analysis, right?

QUOTE (Axon) *

Absolutely not. If there were a peak, you would have resonance. Resonance is not ringing. Ringing is temporal spreading - and a dampened sinusoid is quite obviously an instance of that.


Now you are making a semantic argument, and established authories go against you:

http://en.wikipedia.org/wiki/Ringing

"In electrical circuits, ringing is an unwanted oscillation of a voltage or current."

When bells ring, it is due to resonances, but now you want to say that bells don't ring?

I'm not buying any! ;-)

QUOTE
A long FFT is completely insensitive to that sort of thing.


A long FFT of a complex wave, when processed by the inverse FFT transform, gives back a good reproduction of the original wave Ringing, and all. Making FFTs long does not make them inaccurate.

QUOTE
FWIW, the preringing for foobar2000's FIR equalizer is ludicrously audible. I once did a 5db peak followed by a 5db cut and could still ABX the ringing!


Unfortunately I was unable to download the referenced files. They appear to have been deleted.

However, the presence of an audible difference does not guarantee that the difference exists for the reason hypothesized.


QUOTE
The apparent ringing is due to phase shifting of the harmonics of 10 Hz. To add up to be a square wave their phase and amplitude relationships of the harmonics must be very precisely like a square wave. The brick wall filter messes that up. We don't get a pure square wave. There is no ringing to hear. What we do hear is the straight-forward effect of the brickwall filter.
You're specifically referring to the Gibbs phenomenon here - phase shifts have nothing to do with this. The square wave just had its harmonics lopped off.


Right, and chopping off the higher harmonics disturbs the required precise phases and amplitudes. Not due to phase shift like I erroneously said, but rather due to massive screwing with the amplitudes. They're gone!

QUOTE
If the temporal smearing or ringing or dampenened sinusoids (or whatever you want to call them) exceed the temporal masking threshold, no matter what you call the phenomenon, it will be audible.


Are you aware of any modern brickwall filter in an >= 44KHz ADC or DAC that actually rings or smears outside of the multi-millisecond temporal masking interval?

QUOTE
That should not be that hard to reproduce for filters exceeding 4000 samples for khz-range filtering.


Evidence?

QUOTE
FIR filtering of rumble is usually a fool's errand, but I could imagine a suitably high-Q filter operating at 30hz to be plausibly audible with a suitable choice of listening equipment. I have no evidence to back that statement up though


A brick-wall rolloff below 30 Hz is going to be audible with many examples of musical program material simply because it is going to remove low frequency information that we hope is audible, or else it isn't worth getting rid of, or if it is music, preserving.

It's my belief that people who worry about equalizers having audible effects are missing the point of having equalizers! ;-)
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Axon
post Dec 4 2008, 01:17
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QUOTE (Arnold B. Krueger @ Dec 3 2008, 17:26) *
QUOTE (Axon) *

Absolutely not. If there were a peak, you would have resonance. Resonance is not ringing. Ringing is temporal spreading - and a dampened sinusoid is quite obviously an instance of that.
Now you are making a semantic argument, and established authories go against you: <a href="http://en.wikipedia.org/wiki/Ringing" target="_blank">http://en.wikipedia.org/wiki/Ringing</a>

"In electrical circuits, ringing is an unwanted oscillation of a voltage or current." When bells ring, it is due to resonances, but now you want to say that bells don't ring? I'm not buying any! ;-)
OK, OK, maybe I did take a semantic turn there. I was speaking of resonance in terms of a second-order system, where a definite peak exists in the response. At the same time though, these damped sinusoids we are speaking of seem quite unwanted to me smile.gif

QUOTE
A long FFT of a complex wave, when processed by the inverse FFT transform, gives back a good reproduction of the original wave Ringing, and all. Making FFTs long does not make them inaccurate.
And you're the one complaining about semantics here? tongue.gif I'm talking about magnitude power spectra, not FFTs. Their graphical interpretation is entirely subject to debate. In particular, short-time power spectrum (shorter in time than the length of ringing) will clearly show frequency peaks in the transition band before/after the square wave transitions in your example; a long time power spectrum would not show it.

QUOTE
Unfortunately I was unable to download the referenced files. They appear to have been deleted. However, the presence of an audible difference does not guarantee that the difference exists for the reason hypothesized.
Quite true, but given knowledge of the implementation, it appears to be the most plausible explanation to me.

QUOTE
Are you aware of any modern brickwall filter in an >= 44KHz ADC or DAC that actually rings or smears outside of the multi-millisecond temporal masking interval?
No - I forgot to mention that I am aware of the research about phase shifts at brickwall frequencies etc being basically inaudible. I get that.

But the original discussion was about rumble frequencies, right? The 10-30hz regime is a different ballgame entirely when it comes to audibility.

QUOTE
QUOTE
That should not be that hard to reproduce for filters exceeding 4000 samples for khz-range filtering.
Evidence?

<whips it out>



Seriously, would you like for me to try to make a test of this? Perhaps implementing the same class of filter at two different numbers of taps, and demonstrating an ABX based solely on the ringing?

QUOTE
QUOTE
FIR filtering of rumble is usually a fool's errand, but I could imagine a suitably high-Q filter operating at 30hz to be plausibly audible with a suitable choice of listening equipment. I have no evidence to back that statement up though


A brick-wall rolloff below 30 Hz is going to be audible with many examples of musical program material simply because it is going to remove low frequency information that we hope is audible, or else it isn't worth getting rid of, or if it is music, preserving. It's my belief that people who worry about equalizers having audible effects are missing the point of having equalizers! ;-)
I tended to agree, until I figuratively threw the foobar2000 eq against a wall a few years ago. There are real disadvantages to excessively high-Q filters.

For the record, when I derumble, I use a 6th order Butterworth at 25hz for L+R, and an 8th order Butterworth at 35hz for L-R.

This post has been edited by Axon: Dec 4 2008, 01:18
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2Bdecided
post Dec 4 2008, 12:58
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I've attached a .zip archive containing 5 files

ringing_original.wav - a series of impulses to filter.
(I could have used almost any broadband signal, but impulses make the effect really obvious.)

ringing_4kHz_LPF_brickwall.wav - the effect of filtering the signal with a brick wall filter. I used the FFT filter in CEP - the corresponding .gif file shows the settings.

ringing_4kHz_LPF_gentle.wav - the effect of filtering the signal with a more gentle filter. Again, the corresponding .gif file shows the settings.


I didn't take much care with this, but it proves my point: brick wall filtering rings, and if the transition band is within the audible range, this ringing is audible.

Take a listen for yourself - I can clearly hear the ringing at 4kHz in the brick wall filtered version. And yes, it does sound a bit like a bell!

The version with the gentle filter avoids this (though it's not a very careful attempt to do so - just a quick application of the "spline curves" feature in CEP).



I wrote a response to some of the points raise, but my PC crashed. No bad thing. There's no real answer to some of the points, because they're so...!

E.g. the suggestion that ringing should be examined via a 65k FFT! Even the worst temporal smearing any audio codec has ever created would be invisible under such examination - you don't examine temporal effects using long FFTs!

As for "what is ringing" - "that's not ringing, it's Gibbs" - ?!?!?!!!!!

Cheers,
David.

P.S. Please, no more talk of phase shifts - the CEP FFT filter isn't doing any. It's linear phase.

This post has been edited by 2Bdecided: Dec 4 2008, 13:01
Attached File(s)
Attached File  ringing.zip ( 67.03K ) Number of downloads: 292
 
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krabapple
post Dec 4 2008, 18:10
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QUOTE (2Bdecided @ Dec 4 2008, 06:58) *
I've attached a .zip archive containing 5 files

ringing_original.wav - a series of impulses to filter.
(I could have used almost any broadband signal, but impulses make the effect really obvious.)

ringing_4kHz_LPF_brickwall.wav - the effect of filtering the signal with a brick wall filter. I used the FFT filter in CEP - the corresponding .gif file shows the settings.

ringing_4kHz_LPF_gentle.wav - the effect of filtering the signal with a more gentle filter. Again, the corresponding .gif file shows the settings.


I didn't take much care with this, but it proves my point: brick wall filtering rings, and if the transition band is within the audible range, this ringing is audible.

Take a listen for yourself - I can clearly hear the ringing at 4kHz in the brick wall filtered version. And yes, it does sound a bit like a bell!


I don't hear a bell sound at all -- I hear the two filtered versions both sounding considerably more 'muffled' than the original. And both sounding a little different from each other too.

(using foobar2k v 9.5.3, output at 16 bit Direct Sound , no dither, to onboard SigmaTel Audio on a Dell PC, into Koss TD61 headphones)
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lvqcl
post Dec 4 2008, 19:00
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QUOTE (krabapple @ Dec 4 2008, 20:10) *
I don't hear a bell sound at all -- I hear the two filtered versions both sounding considerably more 'muffled' than the original. And both sounding a little different from each other too.

They are 'muffled' just because they are lowpassed. But "brickwall" version also contains some hiss (around 4 kHz).
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Arnold B. Kruege...
post Dec 4 2008, 21:35
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QUOTE (2Bdecided @ Dec 4 2008, 06:58) *
I've attached a .zip archive containing 5 files

ringing_original.wav - a series of impulses to filter.
(I could have used almost any broadband signal, but impulses make the effect really obvious.)

ringing_4kHz_LPF_brickwall.wav - the effect of filtering the signal with a brick wall filter. I used the FFT filter in CEP - the corresponding .gif file shows the settings.

ringing_4kHz_LPF_gentle.wav - the effect of filtering the signal with a more gentle filter. Again, the corresponding .gif file shows the settings.


I wonder how many controls have been involved in any listening tests comparing these files, as they are supplied with 3 significantly different amplitudes, the closes still about 0.5 dB apart.

I normalized them, checked average RMS to make sure that they didn't different much in that direction, doubled their length by cutting and pasting, and then ABXd them. I could easily hear a difference and immediately scored 16/16, but the difference was not night and day.

I also checked the frequency response of the two filtered ways and found that they varied by upwards of 6 dB in the transition band. The transition band for the gentle filter is about 200 Hz wide for a center frequency of 4 KHz, which may be enough to explain much of why they sound different. They are very different filters and may sound different simply because of the obvious difference.

Using impulses for a test like this seems like an extreme worst case. Music with impusive percussion would be more like a reasonable worst case.
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Dynamic
post Dec 4 2008, 22:36
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QUOTE (Arnold B. Krueger @ Dec 4 2008, 21:35) *
I wonder how many controls have been involved in any listening tests comparing these files, as they are supplied with 3 significantly different amplitudes, the closes still about 0.5 dB apart.


That's surely going to happen with impulses (e.g. approximations to a Dirac delta function).

The energy of a delta function is evenly distributed in frequency (just like white noise). If you filter out everything above 4 kHz you're going to lose a lot of energy (leaving 4000/22050 = 18% of the original total energy if it's sampled at 44100 Hz), and lose a good deal of peak amplitude, having spread the impulse temporally. That's what it's supposed to do. The two filters probably pass a similar proportion of the total energy, but one has a gentle cut-off that should cause little temporal spread and the other is a brick wall that should create a lot of temporal spreading.

The main thing we're trying to do is listen for any ringing and temporal spreading effects, not really to ABX and calibrate volume levels.
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Arnold B. Kruege...
post Dec 4 2008, 23:51
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QUOTE (Dynamic @ Dec 4 2008, 16:36) *
QUOTE (Arnold B. Krueger @ Dec 4 2008, 21:35) *

I wonder how many controls have been involved in any listening tests comparing these files, as they are supplied with 3 significantly different amplitudes, the closes still about 0.5 dB apart.


That's surely going to happen with impulses (e.g. approximations to a Dirac delta function).


Of course.

QUOTE
The main thing we're trying to do is listen for any ringing and temporal spreading effects, not really to ABX and calibrate volume levels.


I'm very surprised to hear people on Hydorgen Audio dismissing the idea of introducing scientific controls into their listening tests.

I guess this is just a game, and without any serious intent?
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krabapple
post Dec 5 2008, 00:34
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QUOTE (lvqcl @ Dec 4 2008, 13:00) *
QUOTE (krabapple @ Dec 4 2008, 20:10) *
I don't hear a bell sound at all -- I hear the two filtered versions both sounding considerably more 'muffled' than the original. And both sounding a little different from each other too.

They are 'muffled' just because they are lowpassed.



Ah, also the sound of me being dumb. laugh.gif


QUOTE
But "brickwall" version also contains some hiss (around 4 kHz).



I heard that as a tiny bit of extra 'detail' compared to the gentle filter, rather than as a bell.
The two filtered versions were easy to ABX, whether 'raw' or level-matched with Replaygain.
(With replyagain's boosting, --~ +25 dB -- the 'detail' became a whistling)

foo_abx 1.3.3 report
foobar2000 v0.9.5.3
2008/12/04 18:37:48

File A: ringing_4kHz_LPF_brickwall.flac
File B: ringing_4kHz_LPF_gentle.flac


without replaygain

18:42:27 : Test started.
18:42:41 : 01/01 50.0%
18:43:07 : 02/02 25.0%
18:43:12 : 03/03 12.5%
18:43:16 : 04/04 6.3%
18:43:19 : 05/05 3.1%
18:43:23 : 06/06 1.6%
18:43:26 : 07/07 0.8%
18:43:41 : 07/08 3.5%
18:43:49 : 08/09 2.0%
18:43:56 : 09/10 1.1%
18:44:00 : 10/11 0.6%
18:44:09 : 11/12 0.3%
18:44:19 : 12/13 0.2%
18:44:23 : 13/14 0.1%
18:44:26 : 14/15 0.0%
18:44:30 : 15/16 0.0%
18:44:36 : 16/17 0.0%
18:44:39 : Test finished.

----------
Total: 16/17 (0.0%)

with replaygain:

18:37:48 : Test started.
18:38:25 : 01/01 50.0%
18:38:32 : 02/02 25.0%
18:38:37 : 03/03 12.5%
18:38:43 : 04/04 6.3%
18:38:46 : 05/05 3.1%
18:38:50 : 06/06 1.6%
18:38:53 : 07/07 0.8%
18:38:57 : 08/08 0.4%
18:39:01 : 09/09 0.2%
18:39:07 : 10/10 0.1%
18:39:11 : 11/11 0.0%
18:39:14 : 12/12 0.0%
18:39:18 : 13/13 0.0%
18:39:22 : 14/14 0.0%
18:39:27 : 15/15 0.0%
18:39:30 : 16/16 0.0%
18:39:33 : 17/17 0.0%
18:39:42 : Test finished.

----------
Total: 17/17 (0.0%)

This post has been edited by krabapple: Dec 5 2008, 00:48
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lvqcl
post Dec 5 2008, 01:14
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QUOTE (krabapple @ Dec 5 2008, 02:34) *
I heard that as a tiny bit of extra 'detail' compared to the gentle filter, rather than as a bell.
The two filtered versions were easy to ABX, whether 'raw' or level-matched with Replaygain.
(With replyagain's boosting, --~ +25 dB -- the 'detail' became a whistling)

Well, it's that whistling I called 'hiss'... It is really simple to ABX; and this effect of steep filter is obvious at spectrograms (left is the signal after gentle filter, right - after brickwall)

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Arnold B. Kruege...
post Dec 5 2008, 12:55
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QUOTE (krabapple @ Dec 4 2008, 18:34) *
QUOTE (lvqcl @ Dec 4 2008, 13:00) *

QUOTE (krabapple @ Dec 4 2008, 20:10) *
I don't hear a bell sound at all -- I hear the two filtered versions both sounding considerably more 'muffled' than the original. And both sounding a little different from each other too.

They are 'muffled' just because they are lowpassed.



QUOTE
But "brickwall" version also contains some hiss (around 4 kHz).



I heard that as a tiny bit of extra 'detail' compared to the gentle filter, rather than as a bell.
The two filtered versions were easy to ABX, whether 'raw' or level-matched with Replaygain.
(With replyagain's boosting, --~ +25 dB -- the 'detail' became a whistling)


A whistling?

I hear the difference between my two level-matched samples as the brickwall version having a little more snap, or detail.

The two samples still don't have the same frequency response +/- 0.1 dB. Therefore, of course they are going to sound different.

Occam's razor suggests that the obvious frequency response difference is the cause of the audible difference, not the differences in ringing.

If we're going to attribute the difference to ringing and not frequency response, then the frequency response of the two samples needs to be matched, with the only difference being the amount of ringing.
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Slipstreem
post Dec 5 2008, 13:19
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QUOTE (Arnold B. Krueger @ Dec 5 2008, 11:55) *
The two samples still don't have the same frequency response +/- 0.1 dB. Therefore, of course they are going to sound different.
Are you seriously saying that the human ear can detect amplitude errors within the passband as small as 0.2dB, or am I misinterpreting this? I thought it was closer to 2dB. Feel free to poke me in the eye with a sharp stick if I'm being an eejit. blink.gif

Cheers, Slipstreem. cool.gif
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krabapple
post Dec 5 2008, 17:53
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QUOTE (Arnold B. Krueger @ Dec 5 2008, 06:55) *
A whistling?

I hear the difference between my two level-matched samples as the brickwall version having a little more snap, or detail.


I'd expect that's because your level matching didn't boost them by ~+25dB, as replaygain does.

QUOTE (Slipstreem @ Dec 5 2008, 07:19) *
QUOTE (Arnold B. Krueger @ Dec 5 2008, 11:55) *
The two samples still don't have the same frequency response +/- 0.1 dB. Therefore, of course they are going to sound different.
Are you seriously saying that the human ear can detect amplitude errors within the passband as small as 0.2dB, or am I misinterpreting this? I thought it was closer to 2dB. Feel free to poke me in the eye with a sharp stick if I'm being an eejit. blink.gif

Cheers, Slipstreem. cool.gif



Nope, it's 0.2dB , in the part of the passband that humans are most sensitive to (midrange). Here's a web tutorial that appears to corroborate the figure

http://www.avatar.com.au/courses/PPofM/loud/Loud1.html

QUOTE
The minimum change in SPL required to give a detectable change in the loudness sensation (JND in sound level) is roughly constant and of the order of 0.2 - 0.4 dB in the musically relevant range of pitch and loudness.


I suspect the lower figure (0.2 dB) is from experiments using pure tones, but don't have a reference for that.

This post has been edited by krabapple: Dec 5 2008, 18:08
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Axon
post Dec 5 2008, 18:13
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Aren't the samples supposed to have differing peaks? The total energy in each pulse should be very nearly equal. Normalizing them is an incorrect procedure here - in a very real sense, they're already normalized.
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