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In Reply to: Some answers posted by Frank.. on June 14, 2004 at 13:39:57:
Thanks Frank for supplying some of the answers. They are correct.Here are rest of answers:
*** Why are the maximum and minimum sample values different for each of the formats? ***
Because they are different recordings - can't guarantee hitting the record button at exactly the same time etc and samples not quite at same positions in the music.
*** I see the clipping on the charts, but not in the tables (actually, the normalized DSD gets clipped in one sample?) ***
The tables just count how many samples hit 0dB FS (that's why it says "Possible clipping"). since i set the gain such that the entire recording is under 0dB FS, there should be no instances of peaks hitting 0dB FS (except for normalized DSD where it's been normalized so that 1 sample is at 0dB FS).
*** When doing the statistics, had you already filtered the sample as described in the end, or was it before? I'm not sure this makes a difference, but... again, curious. ***
Before. By the way, the link is still to the old article - these are not new results. I didn't bother posting the new results as they are so similar.
*** You also show the clipping in the DVD-Audio, directly related to the original clipping on the CD. ***
the clipping is not related to the CD. the CD clips in hundreds of places, the DVD-A only in a few.
*** So if you have no clipping in the DSD, it can mean many things: the transfer from analog tapes has been done more carefully for the DSD , or DSD handles dynamics better than PCM. It may also mean that clipping has been avoided by using a very good limiter... ***
correct. DSD also handles overload differently than PCM - i don't understand why but there are a few AES papers about it. i'm sure ted smith can explain. bottom line is it doesn't "clip" as badly as PCM - although it's best never to overload it as it does deteriorate the sound.
the AES paper i refered to would suggest it's best not to exceed -3dB FS for PCM to avoid any possibility of distortion when processing the analog reconstructed waveform. actually, to be conservative, -6dB FS, to allow for 6dB of "headroom" - but it would be very rare for the analog reconstructed signal to get this high.
*** Maybe I didn't understand a few things, but: in my little mind, if the difference between minima and maxima is greater over a sample, you have "more" dynamics, and RMS power measures that. ***
the difference between minimum and maximum RMS power is an indication of dynamic range. the SACD will always do poorly compared to the DVD-A/CD in this measurement because of ultrasonic noise.
Actually, in my new recordings (done on the DVD-2200) the CD recording has the highest dynamic range (around 89dB!!!) - this is because of lack of ultrasonic noise in the CD. it doesn't mean that CD is better than SACD or DVD-A.
What i refer to as "relative dynamics" can be roughly measured by difference between maximum and average RMS values. as you can see, SACD is higher than DVD-A which in turn is higher than CD on this measurement. it can also be measured by comparing the difference between any two arbitrary points on the waveform across the recordings.
*** It takes many operations in your paper to demonstrate higher dynamics: first normalizing, then computing an average, and then comparing the maximum to that average... when the original difference is actually higher for the DVD-A and CD than for SACD? ***
No, the relative dynamics differences can be observed simply by eye on the waveform - no computation is required. The computation is to measure the extent of the difference, but the differences have always been large enough to be visible to the eye even at screen resolution.
*** A contrario, if you have higher and faster peaks in the DSD version, and if this is consistent throughout the sample, then shouldn't you have a greater difference between minima and maxima in the statistics? ***
No, because of DSD ultrasonic noise, as I've mentioned. that's why it's better to look at difference between maximum and average RMS power, not maximum and minimum.
*** BTW: I still wonder about whether I should get rid of all that noise when I record SACDs, did you notice any difference on your system before and after you used the filter? ***
If you don't intend any post processing on the recording, then you don't need to filter. If you doing some heavy crunching on it (ie. you are doing a "remix") then filtering is a good idea.
DSD's weakness is that it is really only suitable for end delivery (i.e. listening to music), the ultrasonic noise complicates the editing process. To me, DSD is the equivalent of a vinyl "direct to disc" recording. Get it right the first time, and try and avoid any post processing. Or use it in the final mastering stage only - if you edit extensively, do it in PCM (heresy to the SACD fanatics, i know!).
Follow Ups:
I understand better now. Last night I did check with my version of Look of Love, and I think the SACD is displaying instances of limiting (track 4 for sure, and I think track 7, but other instances are suspicious). Nevertheless, it's interesting if it is confirmed that DSD can handle overload better than PCM.I didn't have time to do it today, but I'll try to post my figures tomorrow, they look very similar, I think.
(I'm off to see the Dutch beat the Germans :)
Best
Well, not really, but hot pics of the SACD waveforms at least :)Christine,
I can't post things in a very professional way on a web page, but here's part of the data I got with the Diana Krall album. I tried to record the entire album at 32/192 but my PC/software refuses to let me write files bigger than 4Gb, so I could only record the first nine tracks in one file (about 45mn), track 10 is in a separate file.
I recorded using my Lynx soundcard, using CEP and checked with Sound Forge. The results were identical but I preferred doing everything in CE to use the same statistics, and not have to do file saving (saving a 4Gb file took 10mn!!) or any copy/paste.
Anyway, as you know the volume level is very low on the SACD as on all SACDs, so I normalized the entire file to 99.8dB. I won't go into too many aspects, but here's my impression:
- the SACD isn't clipped when you focus on the highest peaks (not like the bad cases on the DVD-A that you showed), but it has a very distinct signature similar to using some form of limiting. This is very clear in track 4 and maybe track 7, but in other instances as well. The only type of music where I get this type of pattern is either music that has badly clipped at the recording or at the mastering (eg Pink Floyd) or electronic music when the synths are set to a constant volume, but never on acoustic music when the levels and dynamics are never exactly the same. In short, it looks like some sort of limiting has been used, but I can't say how and when.
This is easily checked on the most obvious track #4, I drew a red line on the limit that has been imposed on the track (again, this could be at the recording or at the mixing stage). The same thing can be found on other tracks, including track 7 and 9
- regarding frequency distribution and analysis, I do not find the missing frequencies as described, but I do see a small peak at -80 or -82dB, but I think that's a problem with my soundcard since I installed additional hard disks, i think the noise floor was lower before (not sure, but I'll check that). When I applied the Bessel filter as per your description, that little "bump" didn't disappear, but it was lower, towards -90dB, so maybe the filter helps in some way.
- To conclude, when I did try to use a 6th bessel filter, first at 25760Hz, then at 15760, and it didn't change the frequency distribution, but it changed (slightly) the statistics, which are as follows
32192 no filters (norm 99.8dB) Same, filtered Bessel @ 25670HzLeft Right Left Right
Min Sample Value: -31076.61 -32168.22 -30810.97 -31084.36
Max Sample Value: 31428.37 32702.85 30772.38 31140.45
Peak Amplitude: -.36 dB -.02 dB -.53 dB -.44 dB
Possibly Clipped: 0 0 0 0
DC Offset: 0 0 0 0
Minimum RMS Power: -132.77 dB -127.44 dB -138.04 dB -128.78 dB
Maximum RMS Power: -5.16 dB -5.97 dB -5.16 dB -5.97 dB
Average RMS Power: -19.14 dB -18.76 dB -19.15 dB -18.76 dB
Total RMS Power: -17.93 dB -17.65 dB -17.93 dB -17.66 dB
Actual Bit Depth: 32 Bits 32 Bits 32 Bits 32 Bits
(Sorry about the layout).
There are some notable differences with your own figures, but this may be due to the fact that the statistics are computed on the whole sample of the first 9 tracks, not just one song, so maybe there are much louder tracks in the sample.
I think the only significant factor is that the filter lowered the minimum RMS value (I'm not sure how that makes sense, since it is supposed to delete samples above certain threshold, in this case 50kHz). The difference between minima/maxima as per our previous discussion (which to me means "dynamics") would then increase. But it is unclear to me whether this is an artifact due to the filter or whether the "real" dynamics appear after the filter has been engaged, maybe you and Frank have a better idea. Average and total RMS power is unchanged, which is strange.Special bonus:
The following charts are really more ornamental than anything, but they show how much ultrasonic noise is really floating in the SACD tracks. On the "linear" view, I highlighted the frequencies where I have peaks, because it seems that on all my recordings I get the same peaks on those frequencies. The largest peak at around 60kHz seems higher than usual, but I'm not sure.
Obviously, this was done before the Bessel filter was applied.
They also show some interesting differences between left and right channel, but I have no idea where that comes from, it looks like the two channels didn't receive exactly the same treatment. I'll double check with other recordings whether that is the case or if that could come from my soundcard.I hope this helps in your analysis
Best
Nice work Eric,Regarding the hf peaks in the spectrum plot:
You should check if the peak at 60kHz correlates with your monitor frequency. Simply adjust the videocards scanrate. If the peaks shifts to another frequency you can be certain that it's interference from your videocard.***I think the only significant factor is that the filter lowered the minimum RMS value (I'm not sure how that makes sense, since it is supposed to delete samples above certain threshold, in this case 50kHz). The difference between minima/maxima as per our previous discussion (which to me means "dynamics") would then increase. But it is unclear to me whether this is an artifact due to the filter or whether the "real" dynamics appear after the filter has been engaged, maybe you and Frank have a better idea. Average and total RMS power is unchanged, which is strange.***
The filters lowers the wideband noise contribution. It's normal that the minimum RMS value will be lower after applying a filter.
This is true because the noise energy and the signal energy ratio is small.It's also logical that the avarage and total RMS power changes very little. The signals energy is mostly concentrated below the filters cutoff frequency. Also the signals energy is much stronger than the noise contribution.
Thanks Frank,Very smart idea! I'll check my monitor and video board, because I think something's not right (I'm sure they don't share any IRQ though, but I'll try to see how I can check that.
(more work!)
Cheers
yes - you've definitely found signs of peak limiting - no doubt about it.i'll try and record the whole album this weekend on both DVD-A and SACD and see if i can replicate your results.
*** I'm off to see the Dutch beat the Germans ***Is World War 3 about to start? :-)
Actually, i do want to point out a misconception in Frank's reply:
***Without the bessel filter these sample sizes are missing indicating a low low level resolution. ***
This is not true, and is a misunderstanding on Frank's part. The low level resolution is still there, it is "masked" by the DSD ultrasonic noise.
To take an extreme example:
Let's say you have two sine waves, one at 1kHz but at -50 dB. another at 48 kHz but at -25dB (assuming PCM at 96/24).
If you do a histogram analysis it will likely show no samples at -50dB. Does this mean the 1kHz signal is not there? no, if you play it you will clearly hear it. in contrast, i doubt you can hear the 48kHz signal recorded at a much higher volume.
now if you filter away the 48kHz, you will find the 1kHz magically reappearing in the histogram at -25dB. it's not magic - it was always there, overlaid with the 30kHz signal.
this is the same phenomenon happening with LPs. many people point out LP's poor dynamic range but others will say they hear more low level resolution on LP than CD.
again, LP's "noise" - primarily surface noise in a good set up - is primarily below 1kHz. The dynamic range above 1kHz is very high - at least on par with CD. the surface noise is an artefact our ears can easily filter out, allowing us to hear most of the low level detail that is there.
so it is with DSD. all the "noise" is above 20kHz, where we can't really hear it. but we can hear the low level resolution in the audible band.
***If you do a histogram analysis it will likely show no samples at -50dB. Does this mean the 1kHz signal is not there? no, if you play it you will clearly hear it. in contrast, i doubt you can hear the 48kHz signal recorded at a much higher volume.***This isn't accurate.
There should be samples visible in any case. Every audio signal and even noise is crossing zero so statistically there have to be low level samples included in the count.(That there are none could be a fault in the software creating the histogram.)
"again, LP's "noise" - primarily surface noise in a good set up - is primarily below 1kHz. The dynamic range above 1kHz is very high - at least on par with CD. the surface noise is an artefact our ears can easily filter out, allowing us to hear most of the low level detail that is there."
The dynamic range of LP isn't anything near that of 16 bit pcm.
There is no cutting head in the world that is able to handle high frequency at full scale output. In fact high frequencies are filtered at 16kHz and there is a slight boost around 12kHz to sort out cutting head anomalies.
Noise levels from sacd's I sampled turned out to be fairly high.
Only with filtering you get lower noise levels. ( and improved accuraccy! ) But it's a trade of against bandwidth.There are already 'natural' filters in the listening chain, our speakers and ultimately our ears.
The audiosoftware's tools like the equalizer can show very low noise levels for steady state test signals just by averaging across hundreds or thousents of perdiods of the steady stade signal.
However this doesn't mimic the behaviour of our hearing.
The hf noise is naturally filtered out because it's too fast for our inner ear to be able to even detect it.
By using a bessel filter set to mimic this natural behaviour of our hearing brings the result you get from the statistical tools in your audio software more in line with what humans hear.
*** (That there are none could be a fault in the software creating the histogram.) ***So if PCM doesn't show low level samples it's a fault of the software, but when the DSD doesn't show low level samples it's a fault of DSD? :-)
*** The dynamic range of LP isn't anything near that of 16 bit pcm. ***
I regularly get a dynamic range of 80-90dB on recordings of my LP for frequencies above 500 Hz.
*** Noise levels from sacd's I sampled turned out to be fairly high. ***
Weren't you going to post MP3s? Maybe i missed that posting.
***So if PCM doesn't show low level samples it's a fault of the software, but when the DSD doesn't show low level samples it's a fault of DSD? :-)***But your graph did show low level samples from DVDA from your Denons output.
Only the dsd source has a missing range of sample sizes missing in the histogram. (Although there are counts near -85dB!)
This really needs further investigation.
***I regularly get a dynamic range of 80-90dB on recordings of my LP for frequencies above 500 Hz.***
I do too, but that turn out to be impulses from scratches mostly. :-(
*** Noise levels from sacd's I sampled turned out to be fairly high. ***
Notice the very strange noise only occuring when massed strings play at a very low level.
Frank
*** Only the dsd source has a missing range of sample sizes missing in the histogram. (Although there are counts near -85dB!)This really needs further investigation. ***
It has been investigated, and I've since posted about it (can't remember when, though). The "missing samples" phenomenon is quite common for recordings that are "noisy". i've replicated it in recordings of both LPs and DVD-As so it's not unique to SACD.
*** I do too, but that turn out to be impulses from scratches mostly. :-( ***
Well, maybe you do need to upgrade your turntable. as i've said, i am getting full dynamic range of conservatively 80-90dB all the way to 20kHz and beyond. i'll post some graphs in my next article.
Thanks for the MP3. can you tell me the exact timings of your recordings - i would like to compare against my system on the weekend.
***Well, maybe you do need to upgrade your turntable. as i've said, i am getting full dynamic range of conservatively 80-90dB all the way to 20kHz and beyond. i'll post some graphs in my next article.***80..90 dB is practically 'impossible'. Dynamic range is reduced with a compressor for the disc cutter. It's also a possibility that your element arm combination/setting overshoots on transients.
You need a test disc to check if that's the case.***Thanks for the MP3. can you tell me the exact timings of your recordings - i would like to compare against my system on the weekend.
***I'll see what I can do. I have placed markers at the 'candidate silence' locatations in the track but not at the exact spots that are used in the mp3 sample.
The first and the last locations are fairly easy to spot though. :)
Note that I sampled the left and right front channels of the multichannel mix. I did this intentionally because usually there is ambiance added in a stereo mix picked up with an additional set of mics further out in the venue. The stereo mix could also be made from a pair of spaced omni's.(AB)
Ambient surround recording often use more 'discrete' mic setups with cardiods pointing in 5.1 directions.
There is a good chance the front mics pick up less hall ambiance and deliver better defined sonics from the direct sound field.
This way it's easier to seperate the noise from the ambient sounds.
*** 80..90 dB is practically 'impossible'. Dynamic range is reduced with a compressor for the disc cutter. It's also a possibility that your element arm combination/setting overshoots on transients. ***I must admit - I admire your ability to "spin" just about anything to suit a conclusion that you want to make. I particularly like the way that if you measure a limitation then it's proof that the medium is at fault, but if someone offers a counter example then their equipment is at fault :-) Have you considered a career in PR? The iraq situation needs people like you right now. :-)
"Impossible"? Evidence, please. Your repeated mention of disc cutter limitations just doesn't "cut" it - given that Discrete Quad LPs produced in the 70s require frequency response to 50kHz, so the technology was available even then.
And can you elaborate on exactly how a "passive" stylus tracking grooves can somehow cause "overshoots on transients"?
Frank
***I must admit - I admire your ability to "spin" just about anything to suit a conclusion that you want to make. I particularly like the way that if you measure a limitation then it's proof that the medium is at fault, but if someone offers a counter example then their equipment is at fault :-) Have you considered a career in PR? The iraq situation needs people like you right now. :-)***I was only pointing out a *possible* flaw in your arm/element setup.
There are testrecords with a test signal approaching a 'squarewave' as best as it could be reproduced by a mechanical analog system.
With an oscilloscope the overshoot/resonance can be checked.You have sampled vinyl. Just view the effect of a loud scratch by lookin in your wave editor to see how arm/cantilever resonance come into play.
As in the case of the missing samples. You didn't present evidence of missing samples with pcm or cd at the time.
It's clearly because of noise and the method used by the software to calculate the graph.
It isn't counting sample size occurances it's averaging them.
Due to the noise the the 'missing samples region' shows little because it averages out to near zero or below a treshold set in the software's algorithm.However it's still a valid indication that low level resolution might be poor.
***"Impossible"? Evidence, please. Your repeated mention of disc cutter limitations just doesn't "cut" it - given that Discrete Quad LPs produced in the 70s require frequency response to 50kHz, so the technology was available even then.***
I have had a guided tour in a vinyl pressing plant with it's own cutting facility.
It was clearly explained why this dynamic range compression and hf frequency roll off is needed.It was also explained at an informal lecture I attended by veterans in this business who have cut thousends of disc.
Do some research and you'll find out that it's the simple truth.
I simply refuse to buy into the current audiophile trend to blowup vinyl's capabilties into mythical proportions.
It's just a nice sounding format that served it's purpose in it's day.This dynamic range limiting isn't such a bad thing as it's contributing a lot to the particular charm of vinyl playback.
***"Impossible"? Evidence, please. Your repeated mention of disc cutter limitations just doesn't "cut" it - given that Discrete Quad LPs produced in the 70s require frequency response to 50kHz, so the technology was available even then.***
Now you are confusing extended bandwidth with dynamic range capabilities.
'Discrete' Quad LP's. A joke I presume. :) Quad LP's used some kind of matrix processing or carrier wave technology to encode the mc in the analog domain.
The technology was available but in the end hardly practical and difficult to implement.***And can you elaborate on exactly how a "passive" stylus tracking grooves can somehow cause "overshoots on transients"? ***
It's a mechanical device. The mechanics are comparable with that of a car's suspension hitting a pothole or bump in the road.
A more sporty trim and you feel every dent in the road (and ruin your back) or a more limo type setting where you can become seasick.Sample a loud scratch and look how your setup reacts.
*** However it's still a valid indication that low level resolution might be poor. ***If that is so, it is poor on ALL formats equally: LP, CD, DVD-A and SACD, since the behaviour can be replicated on recordings from all three formats.
*** Do some research and you'll find out that it's the simple truth. ***
Actually, i have done some research and what you say doesn't appear to be the simple truth. it is true that there are many contraints and limitations in pressing vinyl and it is possible to make a bad LP - there are many examples in the 70s and 80s due to commercial constraints of putting as much music as possible onto a side that filtering and compression was necessary.
However, there are also ways to avoid or mitigate those limitations. And given that LP is a "dead" format these days, LPs pressed for audiophiles are done by who do take the extra care.
*** Now you are confusing extended bandwidth with dynamic range capabilities. ***
No I wasn't. You were the one who said it was "impossible" to reach 20kHz at full dynamic range. but clearly the technology to do so was there in the 70s.
*** The technology was available but in the end hardly practical and difficult to implement. ***
So then it's not impossible to have extended frequency response and dynamic range, then, as you originally claim?
As for practical/difficult, my father had a Quad capable system, supporting both matrixed and discrete Quad LPs. you do need a special stylus for the discrete quad lps, but they were no more expensive than a reasonable stereo only stylus. we played quad for many years. it was quite practical and no more difficult than stereo. And even when quad died, a lot of the technology (such as the stylus shape and half speed mastering) was carried over to improve stereo LPs.
*** Sample a loud scratch and look how your setup reacts. ***
I have. It looks exactly like a loud scratch.
***No I wasn't. You were the one who said it was "impossible" to reach 20kHz at full dynamic range. but clearly the technology to do so was there in the 70s.***No you didn't get my point or are twisting my words.
I said it is impossible to reach 20kHz at the full dynamic range *you* claimed. (80..90dB)Fancy audiophile pressings are limited by the cutting heads limitations just as much as normal pressings.
If hf content is pushed to hard then the cutting head gets into trouble.***So then it's not impossible to have extended frequency response and dynamic range, then, as you originally claim?
***Yep, flat response up to 50kHz? Simply not possible.
The encoded quad info in the disc beyond 20Khz had a lower level.
*** I said it is impossible to reach 20kHz at the full dynamic range *you* claimed. (80..90dB) ***And i supplied a counter example to your "impossibility"
you may not realise this, but not all cutting heads are equal. there are different types, and they have different limitations. also, 20kHz at full dynamic range is easy to obtain if you slow down to half speed mastering. widening the spacing between grooves will also help.
just because you have visited one pressing plant and attended one seminar means you know everything there is to know.
by the way, DVD-A has limitations that require filtering as well. 96/24 5.1 channels require MLP encoding to fit into maximum bitrate of 9.6 Mb/s. however, MLP assumes it is able to achieve a certain level of compression.
Some program material cannot be compressed easily and generate "illegal" files. For example, by definition white noise cannot be compressed, so 96/24 5.1 of white noise will generate an illegal MLP file. The solution is - guess what: filtering the high frequencies.
SACD avoids this issue by specifying a higher maximum transfer rate for the player: 16 Mb/s. so if the material cannot be compressed by DST, it is still legal and able to be transferred to SACD without filtering. you do reduce the maximum playing time though.
***by the way, DVD-A has limitations that require filtering as well. 96/24 5.1 channels require MLP encoding to fit into maximum bitrate of 9.6 Mb/s. however, MLP assumes it is able to achieve a certain level of compression.***'require filtering'? Apart from very rare occurring events in normal music content other filtering requirements than that to fulfill the nyguest criteria are not neccesary.
***Some program material cannot be compressed easily and generate "illegal" files. For example, by definition white noise cannot be compressed, so 96/24 5.1 of white noise will generate an illegal MLP file. The solution is - guess what: filtering the high frequencies.***
White noise at full level is very usefull in a music format.
Filtering at 30kHz instead of 40kHz probably solves the issue.
No big deal.***SACD avoids this issue by specifying a higher maximum transfer rate for the player: 16 Mb/s. so if the material cannot be compressed by DST, it is still legal and able to be transferred to SACD without filtering. you do reduce the maximum playing time though.
Obviously dst is less efficient in reducing the bitrate so a higher bitrate from the player is needed.
The maximum playing time is less than that of a dvda with mlp.
DVDA with mlp is a more flexible format. If the bitrate is too high than mild filtering can be used to remedy the problem.
Filtering hf is only needed during the difficult passage.Maximum playing time isn't compromised by a fancy ability to handle vary rare occurring musical signals or non musical signals like wideband noise at high levels.
Frank
you were very quick to criticize limitations in vinyl technology that require certain signals to be filtered or compressed in rare circumstances (and even then the problem can be solved by using half speed mastering or widening the groove spacing).but a very similar limitation is in DVD-A where certain signals require filtering before they can be mastered onto DVD-A.
***I must admit - I admire your ability to "spin" just about anything to suit a conclusion that you want to make. I particularly like the way that if you measure a limitation then it's proof that the medium is at fault, but if someone offers a counter example then their equipment is at fault :-) Have you considered a career in PR? The iraq situation needs people like you right now. :-)***I was only pointing out a *possible* flaw in your arm/element setup.
There are testrecords with a test signal approaching a 'squarewave' as best as it could be reproduced by a mechanical analog system.
With an oscilloscope the overshoot/resonance can be checked.You have sampled vinyl. Just view the effect of a loud scratch by lookin in your wave editor to see how arm/cantilever resonance come into play.
As in the case of the missing samples. You didn't present evidence of missing samples with pcm or cd at the time.
It's clearly because of noise and the method used by the software to calculate the graph.
It isn't counting sample size occurances it's averaging them.
Due to the noise the the 'missing samples region' shows little because it averages out to near zero or below a treshold set in the software's algorithm.However it's still a valid indication that low level resolution might be poor.
***"Impossible"? Evidence, please. Your repeated mention of disc cutter limitations just doesn't "cut" it - given that Discrete Quad LPs produced in the 70s require frequency response to 50kHz, so the technology was available even then.***
I have had a guided tour in a vinyl pressing plant with it's own cutting facility.
It was clearly explained why this dynamic range compression and hf frequency roll off is needed.It was also explained at an informal lecture I attended by veterans in this business who have cut thousends of disc.
Do some research and you'll find out that it's the simple truth.
I simply refuse to buy into the current audiophile trend to blowup vinyl's capabilties into mythical proportions.
It's just a nice sounding format that served it's purpose in it's day.This dynamic range limiting isn't such a bad thing as it's contributing a lot to the particular charm of vinyl playback.
***"Impossible"? Evidence, please. Your repeated mention of disc cutter limitations just doesn't "cut" it - given that Discrete Quad LPs produced in the 70s require frequency response to 50kHz, so the technology was available even then.***
Now you are confusing extended bandwidth with dynamic range capabilities.
'Discrete' Quad LP's. A joke I presume. :) Quad LP's used some kind of matrix processing or carrier wave technology to encode the mc in the analog domain.
The technology was available but in the end hardly practical and difficult to implement.***And can you elaborate on exactly how a "passive" stylus tracking grooves can somehow cause "overshoots on transients"? ***
It's a mechanical device. The mechanics are comparable with that of a car's suspension hitting a pothole or bump in the road.
A more sporty trim and you feel every dent in the road (and ruin your back) or a more limo type setting where you can become seasick.Sample a loud scratch and look how your setup reacts.
Pretty good game, though, I think the Dutch came close to a second goal towards the end, and they should have been awarded at least one penalty.I was really impressed by the new German players, like this young devil Bastian Schweinsteiger, I think we will hear more of that guy !
Cheers
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