I have been playing with DSP linear phase FIR filters for audio for 7 years and I don't think I have reliably heard preringing, even with gross amounts applied, while listening to music.
Whether digital XO, with any XO type or slope, driver linearization, room correction, excess phase correction, and correction filters for DAC's, whether 100 taps or a million taps, it seems to fit in the same category as the dreaded "jitter". Everyone talks about it, but no-one can hear it... Btw, here is a jitter test to listen to: jitter_1
I have listened to hundreds of variations of linear phase FIR filters and not really convinced I have ever heard preringing, even when not using preringing compensation, which are part of Acourate https://www.audiovero.de/pdf/AcouratePRCen.pdf and Audiolense https://cdn.computeraudiophile.com/article-images/2017/1205/mitch/Audiolense-help.pdf both commercial products I have reviewed extensively on ComputerAudiophile.com.
You can find several "preringing" articles on Archimago's site, including numerous listening tests that show inconclusive results. Here is one example of Arch's articles: Archimago's Musings: Audiophile Myth #260: The Detestable Digital Filter Ringing and Real Music...
Folks should understand when viewing an impulse response "displays" in FFT audio tools, that the display itself is heavily weighted towards high frequencies, making the visual display way worse than it is. Switching the display to a step response instead of an impulse response, gives a more weighted view across the frequency range. A quick review of that here: Measuring Loudspeakers, Part Two | Stereophile.com
Mark, if you really want to pursue this, I would design a listing test so that you can switch back and forth between linear phase FIR filters in a convolver while listening to music. I would focus on a track that is sparse with a bass/kick drum, as if there were to be audible preringing it would be most noticeable on a bass drum. It would come in the form of a pre-echo or almost that the bass drum sounds reversed. There would be a low frequency ramp up sound before the hit of the beater on the bass drum. This would be your best bet to hear it. Once there is more than two, three or more instruments, forget it. Which is why for music, like the dreaded jitter, it is simply a non issue.
Hope that helps and enjoy the music!
Whether digital XO, with any XO type or slope, driver linearization, room correction, excess phase correction, and correction filters for DAC's, whether 100 taps or a million taps, it seems to fit in the same category as the dreaded "jitter". Everyone talks about it, but no-one can hear it... Btw, here is a jitter test to listen to: jitter_1
I have listened to hundreds of variations of linear phase FIR filters and not really convinced I have ever heard preringing, even when not using preringing compensation, which are part of Acourate https://www.audiovero.de/pdf/AcouratePRCen.pdf and Audiolense https://cdn.computeraudiophile.com/article-images/2017/1205/mitch/Audiolense-help.pdf both commercial products I have reviewed extensively on ComputerAudiophile.com.
You can find several "preringing" articles on Archimago's site, including numerous listening tests that show inconclusive results. Here is one example of Arch's articles: Archimago's Musings: Audiophile Myth #260: The Detestable Digital Filter Ringing and Real Music...
Folks should understand when viewing an impulse response "displays" in FFT audio tools, that the display itself is heavily weighted towards high frequencies, making the visual display way worse than it is. Switching the display to a step response instead of an impulse response, gives a more weighted view across the frequency range. A quick review of that here: Measuring Loudspeakers, Part Two | Stereophile.com
Mark, if you really want to pursue this, I would design a listing test so that you can switch back and forth between linear phase FIR filters in a convolver while listening to music. I would focus on a track that is sparse with a bass/kick drum, as if there were to be audible preringing it would be most noticeable on a bass drum. It would come in the form of a pre-echo or almost that the bass drum sounds reversed. There would be a low frequency ramp up sound before the hit of the beater on the bass drum. This would be your best bet to hear it. Once there is more than two, three or more instruments, forget it. Which is why for music, like the dreaded jitter, it is simply a non issue.
Hope that helps and enjoy the music!
My 30 years or so of experience listening to digital filters is that there are audible differences between minimum, linear and maximum phase implementations. The audibility is most evident on percussive sounds that necessarily involve higher frequency components. Possibly lower frequency components are just as audible, but remain obscured by other low frequency roll-offs in the chain, such as due to the loudspeakers, as mentioned previously, or by the one or more coupling capacitors in the signal path, for example.
As I remarked previously, however, generalisations are hard to come by: Each application of minimum or linear phase filtering should be considered in respect of where and how it will be used. Nevertheless, the listening experiment I described previously will show the limits of audibility in a maximum phase example - please try it: A linear phase filter will then by definition fall within that limit.
So I state again, there are applications where linear phase filters are audibly advantageous; There are other applications where they remain detrimental and where minimum phase (or thereabouts) will be preferred. In an artistic scenario, anything goes and, as noted before, has probably gone already and not been noticed. But to add some perspective...
...The ear remains largely insensitive to phase and we are talking here of a linear phenomenon. Even on percussive sounds, where there is an audible change, the character of the sound can hardly be described as offensive to the ears. This is very different to jitter, which is offensive to the ears, and for good reason, decades of engineering efforts have brought about its attenuation to (in most cases) below the level of audibility.
As I remarked previously, however, generalisations are hard to come by: Each application of minimum or linear phase filtering should be considered in respect of where and how it will be used. Nevertheless, the listening experiment I described previously will show the limits of audibility in a maximum phase example - please try it: A linear phase filter will then by definition fall within that limit.
So I state again, there are applications where linear phase filters are audibly advantageous; There are other applications where they remain detrimental and where minimum phase (or thereabouts) will be preferred. In an artistic scenario, anything goes and, as noted before, has probably gone already and not been noticed. But to add some perspective...
...The ear remains largely insensitive to phase and we are talking here of a linear phenomenon. Even on percussive sounds, where there is an audible change, the character of the sound can hardly be described as offensive to the ears. This is very different to jitter, which is offensive to the ears, and for good reason, decades of engineering efforts have brought about its attenuation to (in most cases) below the level of audibility.
Thanks to everyone for the good discussion.
@BYRTT, appreciate the kind words. Yeah, the on-axis stuff has gotten very easy to get right. My time is mostly spent now with the painstaking process of trying to optimize on and off axis.
But i also decided I need to find ways to truly hear whatever problems might exist that are maybe being masked....hence this thread.
I guess really, it's getting harder and harder to find the subtleties that improve sound.
@mitchba, great stuff, thx. Yes, I've built the ability to switch back and forth between min-phase and linear phase crossovers. I completely agree with your advice on test music.
It's funny how I can hear clear improvement with lin-phase on some tracks, but certainly not all.
I'm guessing this reflects on the mastering speakers used...that maybe where speakers with minimal group delay were used, it allows lin-phase crossovers to shine.
I can't say I've heard any pre-ringing yet...another, hence this thread
And yes, impulse responses are so heavily weighted towards high-freq response, at least when viewed on a linear scale, that it's hard to glean low freq info. Wesayso showed me some time back how to look at band-filtered impulses for that info.
@soundbloke, thx for your continued comments. I'll try the maximum phase comparison...it's a suggestion I've never thought to try.
Time to embed a set of max-phase xovers into a FIR file...
@BYRTT, appreciate the kind words. Yeah, the on-axis stuff has gotten very easy to get right. My time is mostly spent now with the painstaking process of trying to optimize on and off axis.
But i also decided I need to find ways to truly hear whatever problems might exist that are maybe being masked....hence this thread.
I guess really, it's getting harder and harder to find the subtleties that improve sound.
@mitchba, great stuff, thx. Yes, I've built the ability to switch back and forth between min-phase and linear phase crossovers. I completely agree with your advice on test music.
It's funny how I can hear clear improvement with lin-phase on some tracks, but certainly not all.
I'm guessing this reflects on the mastering speakers used...that maybe where speakers with minimal group delay were used, it allows lin-phase crossovers to shine.
I can't say I've heard any pre-ringing yet...another, hence this thread
And yes, impulse responses are so heavily weighted towards high-freq response, at least when viewed on a linear scale, that it's hard to glean low freq info. Wesayso showed me some time back how to look at band-filtered impulses for that info.
@soundbloke, thx for your continued comments. I'll try the maximum phase comparison...it's a suggestion I've never thought to try.
Time to embed a set of max-phase xovers into a FIR file...
Again, this will not help you to necessarily hear the difference in pre- and post-filter responses per se. Two maximum phase crossovers are as complimentary as their minimum phase counterparts. What you will here are the effects (if any) of the errors in the summation off-axis.
Build yourself an equalizer instead, which I appreciate is not exactly "hi-fi", but it will be more useful in hearing the characteristic attributable solely to a pre-response. And then you can also experiment with frequency and cut-off rates too.
When I was mixing (music) everyday there was a period where I wanted to see what all the fuss was regarding linear phase EQ. For a few months, I went through entire projects with say 12-30+ individual tracks and would change the Equalizer on each track to be linear phase or minimum phase. I also varied sessions from 48khz -> 96khz. At either sample rate, the cumulative nature of the filter topology happening across all the source materials was audible. It was not necessary to ABX the difference (which I had done many times before on other processes.) A printed mix using linear phase sounded cold, hard and more spacious at times. OTOH, the minimum phase mix sounded normal temperature and smoother (more "gooey" in comparison.)
The audibility was most obvious in the transient details of the percussion (with some shorter events losing some of their weight & impact and becoming more brittle & clipped sounding.) This was especially true for snare and short white noise events.
In nature objects don't vibrate before they are struck, so linear phase processes are somewhat orthogonal to how physics and our ears work. The mixing and mastering engineers I knew and followed seemed to go back to minimum phase after a brief "oooh look at that shiny pony" moment.
5 years on, i've been mostly working with minimum phase (when I have time for the studio) and don't miss or even really want to work with LP processing. Although, this last year I have been curiously playing around with Transverse Equalizers. TEQ seems to produce a linear phase like result, which again I've observed to have harder edges and a less bodied feeling.
The audibility was most obvious in the transient details of the percussion (with some shorter events losing some of their weight & impact and becoming more brittle & clipped sounding.) This was especially true for snare and short white noise events.
In nature objects don't vibrate before they are struck, so linear phase processes are somewhat orthogonal to how physics and our ears work. The mixing and mastering engineers I knew and followed seemed to go back to minimum phase after a brief "oooh look at that shiny pony" moment.
5 years on, i've been mostly working with minimum phase (when I have time for the studio) and don't miss or even really want to work with LP processing. Although, this last year I have been curiously playing around with Transverse Equalizers. TEQ seems to produce a linear phase like result, which again I've observed to have harder edges and a less bodied feeling.
Thx
but I'm not connecting the dots yet , on how the equalizer will help me discern pre-response..I know high Q peaking gain sucks under any implementation, so I'm assuming you are proposing some other EQ test ...?
Pls let me describe my current setup, to help paint picture....
4-way crossed at 100, 650, and 6300Hz. I said linear phase LR8's in an earlier post; I should have said LR16's (I likes steep !)
Oh,I'm not at all adverse to EQ. But I like a different way to approach tonality. Each of the 4-bands band has it's own volume control, all collectively controlled by a DCA. Using an X-32 mixer.
I find this the best way to adjust tonality and still maintain linear phase, because all on-the-fly shelving filters are IIR.
Mark, perhaps you can clarify your question about preringing. I am presuming that you are referring to linear phase FIR correction filters for loudspeakers (and possibly room correction) correct? Meaning music reproduction. I.e. using linear phase FIR filters for speaker digital XO, speaker driver linearization, with perhaps some overall amplitude correction and maybe even some excess phase correction... all to do with loudspeakers (and room) correction...
Some of the posts above deal with music "production" and not music reproduction using a convolver in the music reproduction chain. As an ex recording/mixing engineer, I would tend to agree with some of the comments above, but again, I think it should be clarified by you exactly what you are using the linear phase FIR filters for...
Also note if using digital loudspeaker (and room) correction with FIR filtering, hosted in a convolution engine, you are also correcting for the entire playback chain: measurement software on PC-->DAC-->(preamp if any)-->power amp(s if using digital XO) -->speaker(s if using digital XO). I left out the mic, mic preamp and ADC, but that is also being corrected... This is of course entirely a different application than music production.
Some of the posts above deal with music "production" and not music reproduction using a convolver in the music reproduction chain. As an ex recording/mixing engineer, I would tend to agree with some of the comments above, but again, I think it should be clarified by you exactly what you are using the linear phase FIR filters for...
Also note if using digital loudspeaker (and room) correction with FIR filtering, hosted in a convolution engine, you are also correcting for the entire playback chain: measurement software on PC-->DAC-->(preamp if any)-->power amp(s if using digital XO) -->speaker(s if using digital XO). I left out the mic, mic preamp and ADC, but that is also being corrected... This is of course entirely a different application than music production.
Thanks a lot for the reply !
I'm not sure what you mean by change the Equalizer on each track to be linear phase or minimum phase. ???
If it were simply changing EQ's that were being applied to tracks, I don't think that gets close to a linear-phase system.
It's kinda a big deal to change the entire speaker's system processing from minimum phase to linear phase.
Were you doing that? If so, pls tell how. Thx
I am not advocating EQ for anything in particular, just trying to provide a means where people can isolate the audible character of filters with pre-responses that are different from their counterparts with only post-responses.
Using crossover filters will not likely help you much in that quest because (as I mentioned in my first post), the intention of any crossover would be to give a perfect sum of the crossed-over channels. You will then not have much of the pre-response left to hear (barring any relatively high level early reflections in which the summation is no longer correct).
Also as a point of clarity (sorry!), LR crossovers are not linear phase: Q cannot be defined for linear phase filters. What we have are linear phase filters with a magnitude response that uses an LR magnitude response as a target.
And whilst I welcome the introduction of rooms and EQ implementations, as it is one of my specialist subjects, it will probably only serve in complicating what should be a simple thread still further
Using crossover filters will not likely help you much in that quest because (as I mentioned in my first post), the intention of any crossover would be to give a perfect sum of the crossed-over channels. You will then not have much of the pre-response left to hear (barring any relatively high level early reflections in which the summation is no longer correct).
Also as a point of clarity (sorry!), LR crossovers are not linear phase: Q cannot be defined for linear phase filters. What we have are linear phase filters with a magnitude response that uses an LR magnitude response as a target.
And whilst I welcome the introduction of rooms and EQ implementations, as it is one of my specialist subjects, it will probably only serve in complicating what should be a simple thread still further
Good questions. Let me clarify how these are equatable scenarios and qualify my example a bit more to reduce confusion.
When mixing a multi-channel recording it is customary to have HPF on vocal tracks and LPF on sources that don't contain (or require) high frequency information as well as normal cuts and boosts using band-pass or shelving filters across the spectrum. Sometimes up to 3 or more filters will appear on a single track of audio, for example a bass guitar. An entire recording session of 12 tracks might have ~16, or more (6, 12, 18, 24, 36 or even 48dB/Oct) filters engaged to produce a mix that audiophiles then fuss about for years
. I also fuss, so consider me audio kin. Anyhow, all of these filters[eq] (in a session mixdown process) can be run as minimum phase or linear phase. Mastering and mixing engineers choose which filter topology to use (often per track or even filter) in order to make the best sounding mix. A lot of us studio people experienced the linear phase fad of 2012 and moved back to minimum phase because it was more natural sounding (as confirmed by minimum phases sensible physics.)
How is this the same to your situation with speaker crossover architecture?
In my view, equalization and crossovers are based on the same principle of summed phase shift. When I see the word Equalizer or the word Filter or the word Crossover, I see the same math with different co-efficient's or feedback topologies. At whatever frequency and steepness a filter is employed (in the mixing of the material) or (in a digital crossover for the speaker), it will have the same effects on the reproduced signal. In the mixing example we have many filters applied to the material (far more than your crossover) and thus the cumulative effect becomes very apparent. In the crossover scenario you are working with, we only have a few filters, thus it is *much* less audible. For example, If your crossover is not centered in a particularly sensitive region like the 400Hz-1.2Khz transition between thwack & noise portion of a snare, then *I* probably wouldn't notice the pre-ringing. Some people might be more sensitive to it.
I'm not saying that LP is flawed, I'm just saying that it causes an unusual artifact in that it blurs the leading edge of transients which is uncommon in nature and on some material with the filter at the right/wrong frequency, I can hear it.
In your setup, can you ABX the difference between Linear and Minimum phase crossovers?, because the only real difference I'm aware of is the filter's ringing (position, amplitude & duration.)
When you asked who has heard pre-ringing, I described circumstances under which linear phase processing makes pre-ringing very audible and also cited an example where it may be obvious individually (mostly snares and percussive noise bursts.)
Hope that clarifies. I'm not taking sides here as I enjoy pre-ringing on occasion. I would love a filter without any ringing but it's probably an np-complete problem that none of us have time to solve
Hi Mitch, sorry for any lack of clarity.
Yes, I have been referring to the use of FIR for loudspeaker correction and alignment, for all the tasks you mention. Reproduction only.
Any comments about recording/mastering were just conversational asides.
(I do not use any type of room correction, other than careful speaker placement.)
For measurement, I use dual channel FFT which at least helps to isolate some of the chain components you mentioned.
My gut says drivers represent 98+% of the non linearity's I'd like to fix.
Maybe measurement mic is part of that... hope the calibration curve is worth a damn lol
For me, a linear phase FIR file for a driver needs to:
a. first include min-phase driver correction for both in-band, and out-of-band response through summation region
b. include driver phase-only adjustment, if needed and prudent
c. include lin-phase crossover filters, and HPF and low LPF on outside bands
Levels, polarities, and delays, can of course go into the FIR file as you know, but i generally adjust those separately.
So when I'm asking about pre-ringing, it's in the context of trying to hear it in the above speaker setup. (4-way)
How is this the same to your situation with speaker crossover architecture?
In my view, equalization and crossovers are based on the same principle of summed phase shift. When I see the word Equalizer or the word Filter or the word Crossover, I see the same math with different co-efficient's or feedback topologies. At whatever frequency and steepness a filter is employed (in the mixing of the material) or (in a digital crossover for the speaker), it will have the same effects on the reproduced signal.
I'm not saying that LP is flawed, I'm just saying that it causes an unusual artifact in that it blurs the leading edge of transients which is uncommon in nature and on some material with the filter at the right/wrong frequency, I can hear it.
In your setup, can you ABX the difference between Linear and Minimum phase crossovers?, because the only real difference I'm aware of is the filter's ringing (position, amplitude & duration.)
When you asked who has heard pre-ringing, I described circumstances under which linear phase processing makes pre-ringing very audible and also cited an example where it may be obvious individually (mostly snares and percussive noise bursts.)
Hope that clarifies. I'm not taking sides here as I enjoy pre-ringing on occasion. I would love a filter without any ringing but it's probably an np-complete problem that none of us have time to solve
Thanks for such a nice detailed reply.
I guess I see a fundamental difference between parametric filters and crossovers, as far as phase goes. In my mind, parametrics do a phase warp around center frequency, that screws with group delay rather narrowly; whereas xovers have a much more spectrum wide effect on group delay. I guess shelving and high-pass/low pass fall into the crossover camp too.
Anyway, I have no experience at all with trying to EQ source material with linear-phase. All my efforts have been on the speaker system alone.
As I read your experiments in 2012, I was left thinking the only real way to have compared the lin vs min track EQs, would be if you were listening on a linear-phase speaker system. It's like comparing on a crooked ruler until the speaker is linear phase (as long as pre-ringing isn't really an issue Lol )
My best method of comparison so far has been to set one speaker up lin-phase, side-by-side to another one min-phase, using the same crossover topology and order. I get measurably higher, fast-transient SPL's with the lin-phase, but that doesn't always translate into more pleasing sound. I think smear can be our friend when recordings aren't top notch. Bass most always seems a little tighter with linear phase.
If I can find a way to hear pre-ringing soon, I'm just gonna shut up and enjoy what i have
PS I laugh like you too, when i hear folks shunning EQ, knowing what all goes on in the studio...
My Eq is I have a gain control on every driver section !...it can fix almost any tonality problem !
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My lack of clarity too (sorry !!) Yes, LR is just a topology.
FYI, have you had a look at FirDesigner?
It has a great assortment of filters that can be used min, lin, or max.
Love this program...
I don't know that application. I come from a world of hand-cranked machine code and designing from first principles on MATLAB. But LR magnitude responses have no particular merit when used for complimentary linear phase crossovers.
Also just to make it clear, I did not mean to imply that EQ was a bad thing in an artistic scenario. Even in the replay chain it has its uses. The issue is finding a non-crossover based filter with which the effects of pre-responses can be heard.
I think you are describing the same thing I hear. But you might also like to try compensation of the system low frequency roll-off. My impression here is definitely of tighter bass, but oddly too also of greater bass extension.
But as I hinted at earlier, once we compensate for the system roll-off, we then open the Pandora's Box of all the possible low frequency system roll-offs the signal has been subjected to in the recording process....
Yes, I do compensate for natural low-freq rolloff when I have the available taps at hand, and latency is not an issue. Sounds like we are on the same page.
Totally agree about Pandora's Box.
I think many people already realize that recordings reflect the tonal balance of the mastering studio.
For example, if we listen to a recording on a flat system that sounds bass deficient, it means the mastering studio was probably listening to excess bass.
I think a lot of the difficulty now in evaluating the effect of making our systems linear-phase down low, is the same tonal balance problem just mentioned.
If the studio optimized the recording while listening to significant group delay, our linear-phase low end is likely to sound worse. (I think this is what you were saying, I'm just paraphrasing again for extra clarity to any following along.)
When both mastering and playback is linear-phase ...moolah
Yes, I do compensate for natural low-freq rolloff when I have the available taps at hand, and latency is not an issue. Sounds like we are on the same page.
Totally agree about Pandora's Box.
I think many people already realize that recordings reflect the tonal balance of the mastering studio.
For example, if we listen to a recording on a flat system that sounds bass deficient, it means the mastering studio was probably listening to excess bass.
I think a lot of the difficulty now in evaluating the effect of making our systems linear-phase down low, is the same tonal balance problem just mentioned.
If the studio optimized the recording while listening to significant group delay, our linear-phase low end is likely to sound worse. (I think this is what you were saying, I'm just paraphrasing again for extra clarity to any following along.)
When both mastering and playback is linear-phase ...moolah
What I was saying is that you can actually render the recording linear phase, just you would need know the the roll-off, of which there may well be more than one.
As for the mastering engineers listening to excess bass, I would suggest they might just be listening at higher levels where the equal loudness contours flatten out somewhat.
Digging this thread up because I heard something odd today that might be preringing.
I've been using linear phase lr4 crossovers 950hz with -8db peq's at about 350 and 3000hz to flatten my horns for about six months or so now. I wrote my own impulse file generation tool in Python with a c# gui that let's me adjust filter order, freq, and manage up to three peqs., Crank out an eapo config file and dump it in real-time into eapo. Never heard an issue and this was a huge improvement over lr2 passive setup. I did a custom gui because eapo isn't exactly great for a 10' gui.
So the past couple of days I decided to add sinc "brick wall" option and it didn't take long to hear some weird artifacts. Here's a very tangible example. Second track on eno "taking tiger mountain by strategy" opens with just some snare hits and I could hear a "whistle" fainty before each snare hit that I know isn't on the recording. Switched back to the lr4 filter and it's inaudible. My filters use 65535 taps and compare extremely well with those generated by rephase in both magnitude and phase. Just to confirm the same filters built in rephase also exhibited this.
Is this whistling the sound of preringing or some other artifacts?
I've been using linear phase lr4 crossovers 950hz with -8db peq's at about 350 and 3000hz to flatten my horns for about six months or so now. I wrote my own impulse file generation tool in Python with a c# gui that let's me adjust filter order, freq, and manage up to three peqs., Crank out an eapo config file and dump it in real-time into eapo. Never heard an issue and this was a huge improvement over lr2 passive setup. I did a custom gui because eapo isn't exactly great for a 10' gui.
So the past couple of days I decided to add sinc "brick wall" option and it didn't take long to hear some weird artifacts. Here's a very tangible example. Second track on eno "taking tiger mountain by strategy" opens with just some snare hits and I could hear a "whistle" fainty before each snare hit that I know isn't on the recording. Switched back to the lr4 filter and it's inaudible. My filters use 65535 taps and compare extremely well with those generated by rephase in both magnitude and phase. Just to confirm the same filters built in rephase also exhibited this.
Is this whistling the sound of preringing or some other artifacts?
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Interesting....
Were you using a complimentary brick wall x-over? I think I remember Pos recently saying brick walls need to be fully complimentary.
I sometimes think I can hear faint percussive sounds ahead of the real deal, when the track is completely silent before the percussion. But that's been when experimenting on the edge.
(I don't view 96dB / oct as on the edge because I can't hear anything then...)
If you make a decision on whether what you're hearing is pre-ringing or some other artifact, please post back, thx
Were you using a complimentary brick wall x-over? I think I remember Pos recently saying brick walls need to be fully complimentary.
I sometimes think I can hear faint percussive sounds ahead of the real deal, when the track is completely silent before the percussion. But that's been when experimenting on the edge.
(I don't view 96dB / oct as on the edge because I can't hear anything then...)
If you make a decision on whether what you're hearing is pre-ringing or some other artifact, please post back, thx
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