Hey folks, I am new to the board here, and I had a question regarding DSD for anyone out there with the chops.
Basically I have gone through as many previous posts on all the issues from filtering, downsampling, to sigma-delta; however I still am curious about two particular things:
1-Both DSD and 24-Bit audio supposedly have approximately the same amount of dynamic range at around 120 dB; however, is there a difference in the quantization levels within the two systems. Obviously how it is done is slightly different, but I am talking about how large these levels are that the signal most be approximated to. (basically a micro-decibel type figure).
2-While we all know the Nyquist Theorem guides our bandwidth in typical PCM, what guides the bandwidth of a DSD system. Obviously it filters out high frequencies (or higher frequencies for that matter), but what is really the ratio used to figure out the bandwidth of a DSD system with such a strange sampling method.
I hope some folks out there know more than I do, cause this has been bugging the hell out of me and I just can't seem to find any info relating directly to these two issues.
Thanks,
Schex
To find the dynamic range at a particular frequency for either format (using dithering in PCM), take the number of bits used to represent a quarter of the wavelength, then take 20*log(2^number of bits). For the nyquist frequency at 24 bits PCM, that's 20*log(2^24) ~= 144 dB. Lower frequencies have a higher dynamic range thanks to dithering, since the greater number of samples per wavelength tend to "average out," allowing the perception of sample values in between the actual sample values.
PCM is quantized to the nearest sample value, which is 1/2^bits per sample of full scale. Because dB isn't linear, each sample does not have the same difference in dB. The difference between two samples near silence can be 6 dB, while the difference between samples near full scale is .000001 dB.
DSD is a bit more complicated, you can't really call what it does "quantization," since technically everything is quantized to either positive or negative full scale. All you can really say is that it has a certain dynamic range at a certain frequency.
I guess quantization may not be the best term; however, if understand it correctly that is, there is a finite amount of dynamics in each of those positive or negative values. Since these values each represent a change, the system has to have some way of knowing how large of a change that is going to be, right?
I guess what I am searching for there is some type of way to compare this value of how much resolution DSD can produce within it's own dynamic range, as to how much resolution (1/2 quantization level size) PCM is able to.
Maybe this is off the mark, I am not sure. But then again that's why I am the kid asking the question, and you guys that actually understand it are helping explain it
The possible resolution is exactly the same for the same bitrate (i.e. number of bits used to represent the same amount of time). If you can encode a quarter of a 96 khz sine wave using 24 bits, there will be 16,777,216 different possible amplitudes for that sine wave, regardless of the format you use.
EDIT: This is, of course, assuming you're not using any form of compression on the data.
Hi,
See DSD Spectrum (http://people.zeelandnet.nl/eschoester/jaccos_place/PropertiesCD_DVD/DSDOutput.pdf) for more info regarding the Dynamic Range. Please note that the electronics after such a DAC are worse than this. The performance is thus dominated by the electronics after the DAC!
the bandwidth of a DSD system
That would be 1.4 MHz, one half of the sample frequency.
Regards,
Jacco
Wouldn't a signal at half the sampling frequency in DSD actually represent a DC signal?
Yes and no. Remember it is just a sigma-delta modulator with two level output. The "yes" means that the value is a DC value, of coarse but that is always the case. In conventional multi-bit sigma-delta's the output is also DC value, but they chance every sample time. The more frequent they change in value as function of time, the higher the frequency is sampled.
Regards,
Jacco
Hey folks, I am new to the board here, and I had a question regarding DSD for anyone out there with the chops.
Basically I have gone through as many previous posts on all the issues from filtering, downsampling, to sigma-delta; however I still am curious about two particular things:
1-Both DSD and 24-Bit audio supposedly have approximately the same amount of dynamic range at around 120 dB; however, is there a difference in the quantization levels within the two systems. Obviously how it is done is slightly different, but I am talking about how large these levels are that the signal most be approximated to. (basically a micro-decibel type figure).
2-While we all know the Nyquist Theorem guides our bandwidth in typical PCM, what guides the bandwidth of a DSD system. Obviously it filters out high frequencies (or higher frequencies for that matter), but what is really the ratio used to figure out the bandwidth of a DSD system with such a strange sampling method.
I hope some folks out there know more than I do, cause this has been bugging the hell out of me and I just can't seem to find any info relating directly to these two issues.
Thanks,
Schex
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This paper is always a good read:
[a href="http://sjeng.org/ftp/SACD.pdf]Why 1-bit Sigma-Delta conversion is unsuitable for high-quality applications[/url]
This one is even better:
Why Direct Stream Digital is the best choice as a digital audio format (http://www.extra.research.philips.com/mscs/publications2001/dr_dsd.pdf)
Nooooooo!!! (http://www.hydrogenaudio.org/forums/index.php?showtopic=20252&hl=)
Oh, good, another religious flame war is building.
I've yet to be able to ABX one of my multichannel/dual channel SACD's from the remastered CD version (there's a few titles that overlap my CD's), so I (personally) doubt that there is much difference between DSD and PCM in real life. Which means that unless you are interested in picking up a couple of surround recordings (as I've done), stick to CD
stick to CD
Yes, CD can be very good. The problem is that we still don't know how to do it in practise. After all these years we are still learning.
I've yet to be able to ABX one of my multichannel/dual channel SACD's from the remastered CD version (there's a few titles that overlap my CD's), so I (personally) doubt that there is much difference between DSD and PCM in real life. Which means that unless you are interested in picking up a couple of surround recordings (as I've done), stick to CD
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I have to agree with you there, Cygnus X1.
I cannot hear the difference between the SACD and CD layers on any of my SACD's.
I have been sampling SACD output into 2496 wave files and comparing it with 44.1k CD-layer rips, and my ABX scores are appalling. Here is an example:
-------------------------------------
WinABX v0.42 test report
09/26/2004 20:00:30
A file: M:\audio_experiments\01_Quality_of_silence_track1.wav
B file: M:\audio_experiments\01_quality_of_silence_track1_16_96.wav
20:02:18 1/1 p=50.0%
20:02:36 2/2 p=25.0%
20:02:51 2/3 p=50.0%
20:03:08 2/4 p=68.8%
20:03:29 3/5 p=50.0%
20:03:51 3/6 p=65.6%
20:04:05 4/7 p=50.0%
20:04:19 4/8 p=63.7%
20:05:07 4/9 p=74.6%
20:05:36 4/10 p=82.8%
20:05:55 4/11 p=88.7%
20:16:56 4/12 p=92.7%
20:17:23 reset
Note: I can't seem to get WinABX to play 24-bit 96khz files on my system, so I dithered the 2496 SACD sample down to 16 bits for my ABX test.
When doing FFT's of sampled SACDs, I have noticed that some SACDs (NOT all) still have the same 20khz lowpass cutoff as CDs- LOL. I guess I really wasted my money on those disks. The characteristic "rising noise floor" of SACD is also clearly visible in the FFTs, but it is still only about -70dB at 45khz.
stick to CD
Yes, CD can be very good. The problem is that we still don't know how to do it in practise. After all these years we are still learning.
[a href="index.php?act=findpost&pid=244827"][{POST_SNAPBACK}][/a]
I'm not sure what you mean by this statement; there's a wealth of excellent, properly mastered CD titles on the market. The problem is that recent recordings are poorly mastered, which is the fault of the industry and recording engineers, not the CDDA format itself. CD's are capable of over 100dB with dithering, and a frequency response of 22KHz. I've yet to see real, conclusive research that a) a greater bit depth makes a big difference over properly dithered 16-bit audio, and b) that the presence of ultrasonics in music is detectable in a blind test.
And if we indeed are still learning how to coax the best sound out of a CD (with things like dither, etc), why is that a compelling reason to abandon the format for something new and (yet) unproven? Smells like marketing to me....I'm sure that the recording industry is drooling over the prosepct of making people dish out $18 for new albums that aren't rippable and don't cost them anything more to produce.
Hi,
I cannot hear the difference between the SACD and CD layers on any of my SACD's.
Well, I can and especially in our listening room where there is much better equipment than I have.
a) a greater bit depth makes a big difference over properly dithered 16-bit audio
This is indeed a good question. Properly dithered is also misleading since the time domain itself is not better than before dithering, but the mean of several sine waves will resemble closely the original. I am not interested in sinewaves that continue forever. However, I think 24 bits is preferable.
b) that the presence of ultrasonics in music is detectable in a blind test
I did some experiments and there is nothing to hear. At least I didn't, may be my ears are not good enough. Perhaps correlated ultrasonic sounds can be heard, I am not sure yet. But the effect of low-pass filtering can be heard, ie. musicians were able to hear the effect of 50 kHz low-pass filtering. So 96 kHz is not the way to go in my humble opinion. 192 kHz is ok.
Well, I can and especially in our listening room where there is much better equipment than I have.
Are they still using the Marantz SM-2 with (27 kHz) B&W 801? Hope they installed coupling caps after blowing 6 of them.
Properly dithered is also misleading since the time domain itself is not better than before dithering
You do realise that dithering is all there is to DSD?
But the effect of low-pass filtering can be heard, ie. musicians were able to hear the effect of 50 kHz low-pass filtering. So 96 kHz is not the way to go in my humble opinion. 192 kHz is ok.
Please accompany your statements with some proof.
I was in one of the tests you refer to, and did not hear the 50kHz difference until it was pointed out. After that it no longer was a blind ABX test. And even so, with a 27 kHz system, I wonder what portion of 'hearing' was tweeter distortion, instead of 'more information'. If there's really 'music' in supersonics, the system should be blameless to at least that bandwidth (with thanks to Douglas S for the adjective ).
In later reports (http://www.extra.research.philips.com/mscs/publications2001/dr_edit.pdf), a conclusion was drawn upon, amongst others, this test. The staggering number of 8 listeners was called upon as significant. The AES should have read Hydrogen's TOS...
Being one of the 8 and remembering my poor performance, I find yours an ambiguous statement.
Sorry about fueling the fire here, but I found it necessary to include some inside info about 'research'.
I cannot hear the difference between the SACD and CD layers on any of my SACD's.
Well, I can and especially in our listening room where there is much better equipment than I have.
Most likely they are differently mastered. There was a thread around here recently, where an email from some official guy was posted who acknowledged the fact that the CD layer was mastered differently. In particular they aplied more dynamic compression to be "competitive" in the CD changer.
This is indeed a good question. Properly dithered is also misleading since the time domain itself is not better than before dithering, but the mean of several sine waves will resemble closely the original. I am not interested in sinewaves that continue forever. However, I think 24 bits is preferable.
Dithering improves the perceived dynamic range, not less, not more. Whoever claimed anything else? A PCM sampled signal can be perfectly reproduced within the limits of bandwidth and dynamic range. If you have a signal with infinite precision then you can deconstruct that into an infinite number of sine waves without loosing any information at all. It's just a different represantation of the same thing.
But the effect of low-pass filtering can be heard, ie. musicians were able to hear the effect of 50 kHz low-pass filtering. So 96 kHz is not the way to go in my humble opinion. 192 kHz is ok.
The effect of the lowpass may be that the intermodulation within the speaker is changed. Speakers are nonlinear. Input 40kHz and you may get an output of 40kHz. Input 60kHz and if you are lucky you will also get 60kHz output. But play them both at the same time and a signal of perhaps 8kHz will be output which is audible. In this case a lowpass which removes content above 50kHz would also remove the 8kHz tone.
I would go so far as to recommend band limiting the signal fed into the speakers to reduce the effects of intermodulation distortion, resulting in a more faithful reproduction.
1-Both DSD and 24-Bit audio supposedly have approximately the same amount of dynamic range at around 120 dB; however, is there a difference in the quantization levels within the two systems. Obviously how it is done is slightly different, but I am talking about how large these levels are that the signal most be approximated to. (basically a micro-decibel type figure).
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What is your question ? (Sorry, I did not understand. But hopefully my comments below are of help for you)
2-While we all know the Nyquist Theorem guides our bandwidth in typical PCM, what guides the bandwidth of a DSD system. Obviously it filters out high frequencies (or higher frequencies for that matter), but what is really the ratio used to figure out the bandwidth of a DSD system with such a strange sampling method.
[a href="index.php?act=findpost&pid=244689"][{POST_SNAPBACK}][/a]
DSD is not that different to PCM. just with a very high sampling rate but only one bit per sample. In both systems you make a per-sample quantization error. Since DSD has one-bit samples the power of the quantization noise is very high but it's possible (more or less) to shift most of the power into the ultrasonic range and out of the audible range because of the very high sampling rate. So you've about 120 dB within 0-20 kHz and
much less above.
Theoretically the quality of a PCM/DSD stream mostly depends on the bitrate since you can compensate for fewer quantization levels by using high sampling rates and noise-shaping techniques
to some extend - so DSD may seem like a logical choice to eliminate complicated circuits for upsampling/filtering in a player. An SACD player only sends the "bittrain" to a 1BIT DAC and applies an analogue lowpass filter to remove the ultrasonic quantization noise (pretty damn simple compared to DAC circuits which accept PCM data)
On the other hand there are currently some problems with DSD:
- You can't apply full dithering which leads to non-linearities and possibly percible limit-cycles (birdies) in some situations
- It's hard do design good and stable noise shaping filters for 1BIT quantizers.
Also - according to the paper disgustipated mentioned - DSD's quality per bit ratio is less effective compared to PCM. (The 2-level quantizer cannot fully be compensated by noiseshaping). It's quite interesting that 4*44.1=176.4 kHz at 8 bit/sample results in higher quality using
half the bitrate compared to DSD, 64*44.1 kHz at 1 bit/sample.
HTH,
Sebastian
DSD is a solution for the wrong problem. If the recording engineers and labels actually did their jobs, we would all be satisfied with Red Book digital for the rest of our mortal lives (for 2-channel audio at least).
DSD is a solution for the wrong problem. If the recording engineers and labels actually did their jobs, we would all be satisfied with Red Book digital for the rest of our mortal lives (for 2-channel audio at least).
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Amen.
Perhaps correlated ultrasonic sounds can be heard, I am not sure yet. But the effect of low-pass filtering can be heard, ie. musicians were able to hear the effect of 50 kHz low-pass filtering. So 96 kHz is not the way to go in my humble opinion. 192 kHz is ok.
[a href="index.php?act=findpost&pid=244838"][{POST_SNAPBACK}][/a]
How?
1) Most recordings don't contain any musical information past 20kHz. Recordings from 1890-1945 probably only go up to 10kHz at best. Recordings done in the analog tape era (late 40's-80's) might have information past 20kHz, but not much. Digital recordings done in 44/16 or 48/16 will only have information up to the low 20's. It's only
very recently that we've been able to capture higher frequencies in recordings, making the extra bandwidth a moot point for 95% of recordings out there. Moreover,
2) Most mics don't capture a ton of ultrasonic information past our hearing limits. Most speakers in people's homes don't go past 20kHz (if even that ), and most amps don't either. So, the frequencies won't be there, even if the medium is capable of storing them.
So, even if
we could hear a 50kHz tone in isolation, would it matter for music? Would people be able to reproduce it on most recordings, on the equipment usually found in homes? No. Therefore, 44.1 or 48kHz is probably all we'll ever need.
PS- I'm a classically trained musician and can't even detect a 17kHz lowpass on most music, let alone 50kHz.
Bringing this back to DSD....
I think the reason for DSD being interesting to audiophiles has more to do with marketing than technical superiority. Open any SACD and you'll find a little insert that explains how "DSD closely resembles an analog waveform," resulting in a warm, analog-like sound quality. It's the old "16/44 makes a waveform look like a pile of bricks" argument all over again. The problem is, has anybody ever proven the existence of a correlation between a "smooth" waveform and "warmth?" I've come to believe that what many people call "warmth' is a codename for distortion. And with DSD, there's lots of that to go around beyond 20kHz.
I guess I just don't see why using PCM is "wrong." 24/96 audio, which is overkill, results in a frequency response of 48kHz and a dynamic range of 144dB over the entire range, unlike DSD, which has to shove all the noise from its 1-bit process up in the ultrasonic bands, where the dynamic range falls off sharply. Multi-bit PCM just seems like a simpler way of accomplishing the same thing, without having to dither all that noise. Am I right in thinking this way?
Edit: typo
If the optimal recording/processing/playback method really
is 1-bit noise-shaped, then DSD would drastically reduce the complexity of the playback stage - basically a very simple decoder and a class D amplifier. But
- it is definitely not the best recording format - everybody uses multi-bit DS nowadays, for what I understand are overwhelmingly good technical reasons, requiring a (drumroll) PCM to DSD conversion.
- it is definitely not the best processing format - PCM is far, far, far easier to work with. Is there even a full-featured DAW which supports DSD natively? Pro Tools sure doesn't (its SACD plugin (http://www.highfidelityreview.com/news/news.asp?newsnumber=11278979) is a PCM to DSD converter. So to use most modern hardware for processing you need (drumroll) PCM to DSD conversion.
- its economy for playback is made questionable by the rapidly shrinking cost of good 24/192 playback devices, as well as "universal" players that play SACD and DVD-A from a single chip, IIRC.
This is an interesting article regarding >20kHz hearing.
The world above 20kHz (http://www.earthworksaudio.com/f_wpapers/beyond20khz.html)
Basically it suggestes that response in the time domian determines the transparency/naturalness of a microphone or loudspeaker. Perhaps this relates to high resolution digital recording as well.
Also, as a recording engineer, I don't like to take the blame for all bad recordings. The issue is more complicted than one person or group of people not doing their job(s).
Also, as a recording engineer, I don't like to take the blame for all bad recordings. The issue is more complicted than one person or group of people not doing their job(s).
[a href="index.php?act=findpost&pid=244950"][{POST_SNAPBACK}][/a]
It's amazing that the issue is in fact "complicated". If only the industry would treat clipping and overdriving with the same respect that one would with a lit match and a gasoline pump.
Also, as a recording engineer, I don't like to take the blame for all bad recordings. The issue is more complicted than one person or group of people not doing their job(s).
[a href="index.php?act=findpost&pid=244950"][{POST_SNAPBACK}][/a]
It's amazing that the issue is in fact "complicated". If only the industry would treat clipping and overdriving with the same respect that one would with a lit match and a gasoline pump.
[a href="index.php?act=findpost&pid=245173"][{POST_SNAPBACK}][/a]
Well said.
Red Book CD is more than enough, when properly mastered. But at the rate things are going nowadays it's just pathetic. It's obvious to anyone that they just want everyone to buy into SACD or DVD-A (depending on which company you are) by them claiming their decently mastered SACD or DVD-A is better than a crappily mastered CD. Well of course, they made it that way.
SACD and DVD-A = solutions to a problem that shouldn't exist anyway. Industry likes them because they can throw more DRM at the consumer
CD = perfectly fine product when put together greatly. Industry doesn't like them because they are too easy to copy and rip.
In the end, the industry wants to screw the consumers over. It's all in their interest while the uninformed think they are getting a superior product when in fact they really are not.
This is an interesting article regarding >20kHz hearing.
The world above 20kHz (http://www.earthworksaudio.com/f_wpapers/beyond20khz.html)
Basically it suggestes that response in the time domian determines the transparency/naturalness of a microphone or loudspeaker. Perhaps this relates to high resolution digital recording as well.
Also, as a recording engineer, I don't like to take the blame for all bad recordings. The issue is more complicted than one person or group of people not doing their job(s).
[a href="index.php?act=findpost&pid=244950"][{POST_SNAPBACK}][/a]
Well, I did not read the linked paper, as my patience wears thin at this point in that nearly every reference I have been given in such subject has been primarily speculation(S) without proper evalution/listening tests to back up the assertions.
But let's address the 'issue' that is 'complicated'. Please be more specific. Techncially, it's a rather simple problem from my perspective: overcompression and excessive hard limiting. Techncially, just reduce the overall amplitude so that hard limiting is not an issue. Then stop compressing the signals into a primarily 15-20dB average envelope. Ordinary CD has well over 90dB available! But this seems to go way past technical problems and enter into the fray of the cliche' coined 'loudness wars'. I don't blame you or other recordig engineers for this problem. It seems that most of the time this is a problem that is forced upon the mastering engineer(or mix engineer if they are also dong the mastering) by the artist(s) and/or record companies that are giving you the paycheck. But how to teach the ignorant artist(s) or record companies that they are actually reducing sound quality?
-Chris
Also, as a recording engineer, I don't like to take the blame for all bad recordings. The issue is more complicted than one person or group of people not doing their job(s).
[a href="index.php?act=findpost&pid=244950"][{POST_SNAPBACK}][/a]
It's amazing that the issue is in fact "complicated". If only the industry would treat clipping and overdriving with the same respect that one would with a lit match and a gasoline pump.
[a href="index.php?act=findpost&pid=245173"][{POST_SNAPBACK}][/a]
Well said.
Red Book CD is more than enough, when properly mastered. But at the rate things are going nowadays it's just pathetic. It's obvious to anyone that they just want everyone to buy into SACD or DVD-A (depending on which company you are) by them claiming their decently mastered SACD or DVD-A is better than a crappily mastered CD. Well of course, they made it that way.
SACD and DVD-A = solutions to a problem that shouldn't exist anyway. Industry likes them because they can throw more DRM at the consumer
CD = perfectly fine product when put together greatly. Industry doesn't like them because they are too easy to copy and rip.
In the end, the industry wants to screw the consumers over. It's all in their interest while the uninformed think they are getting a superior product when in fact they really are not.
[a href="index.php?act=findpost&pid=245181"][{POST_SNAPBACK}][/a]
Actually, quite a number of musical professionals/engineers sincerely believe that CD is not sufficient in it's bandwidth and dynamic range for hi-fi playback. Of course, I have not seen one of these people support their concerns with valid research, or even participate in proper DBT tesing using proper equipment/methods to produce a fair test. Could it be that some engineers just give better attention to the SACD/DVD-A mix just because they beleive CD to be inferior(thus less worthy)? Their is also the issue of 'radio play'. Bob Katz on his site claims that CDs may be made loud in the hope it will 'be louder' on air. But I've seen various articles to this from experts in radio compression systems that state a highly compressed CD is more likely to be distroted worse, and not be actually louder due to the functional paramaters of the broadcast compressions systems. The accuracy of any of this; I do no know.
-Chris
This is an interesting article regarding >20kHz hearing.
The world above 20kHz (http://www.earthworksaudio.com/f_wpapers/beyond20khz.html)
Basically it suggestes that response in the time domian determines the transparency/naturalness of a microphone or loudspeaker. Perhaps this relates to high resolution digital recording as well.
Also, as a recording engineer, I don't like to take the blame for all bad recordings. The issue is more complicted than one person or group of people not doing their job(s).
[a href="index.php?act=findpost&pid=244950"][{POST_SNAPBACK}][/a]
Very interesting company, and website.
Personally I now have no doubt that 'time-domain' or phase accuracy is the key to 'life-like' recording and reproduction of sound/music. The evidence is mounting inexorably.
It's the reason some equipment 'works' which shouldn't (and vice-versa) when asessed by most other measured parameters.
RF
edit; this is 'IMO', of course (at least for the time being!)
I've heard U2's new track "Vertigo" on the radio many times and I can just tell, based on the radio broadcasts, that it is a horrendous recording. On the radio it sounds very distorted, hard-limited and compressed and I listen to a non-commercial station that generally applies a lighter touch to dynamics compression than your typical corporate Top 40 station. Unforunately, the song does have that lively, in-your-face sound that grabs your attention. It rather reminds me of the production attitude of Oasis's Be Here Now and it wears your patience very thinly.
SACD and DVD-A = solutions to a problem that shouldn't exist anyway. Industry likes them because they can throw more DRM at the consumer
CD = perfectly fine product when put together greatly. Industry doesn't like them because they are too easy to copy and rip.
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IMHO the problem starts with having sound sampled at 44.1khz. I think 44.1khz is not enough, not because we can hear beyond 20khz (at least I can't), but because the transient phenomenon of the strings in our ears when they start resonating will be different when the ultrasonic-component of some instruments attack is there - as compared to when it's not there because of our friend mr. brickwall filter.
If the last statement is incorrect feel free to hit me :-)
Oh yes - and I do think that industry doesn't like CDs because they are just to easy to copy (including ripping to WAVs/MP3s). But see, when SACD/DVD-A become really really popular someone will find a way... and it'll all start over again with BlueRay-Audio or whatever.
I think the general consensus is that if ultrasonic content generates harmonics in the audible range, those harmonics will end up in the recording anyway. Also, 192 khz sampling won't do you any good if your mics or your speakers only go up to 20 khz. You'll fry a tweeter every other song.
Heh, nobody 's gonna fry any tweeter. If the tweeter won't play it it just won't play it. Good tweeters play up to 40khz+ without problems (some scan-speak, morel, most AMTs, ...). Microphones at least won't cut off with a 100 or 200dB/oct and there are microphones that go higher - no big deal.
Also I'm not sure I understand what you mean by "creating harmonics" - what I tried to say has absolutely nothing to do with harmonics or overtones. Just with mechanical resonators with finite Q and the "moment when the music starts playing". Just view it from a time-domain point of view and you see that the term frequenzy isn't even necessary to see why heavy filtered 44.1khz just can't sound like the original.
Of course all this is again just my opinion ...
IMHO... I think...
If the last statement is incorrect feel free to hit me :-)
That will not be necessary. Let's turn it around, and say if your statement is correct provide some proof.
Just view it from a time-domain point of view and you see that the term frequenzy isn't even necessary to see why heavy filtered 44.1khz just can't sound like the original.
Of course all this is again just my opinion ...
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With all due respect, and without trying to be a "TOS trumpeter"...
If I wanted to, I could make a new thread and post about how transcoded mp3s sound better than mpcs, and then qualify that statement by saying it's "just my opinion."
Providing objective support for claims of sound quality is the very basis of HA. There's a very good reason for it. If we all just posted unsupported opinions all day, there would never be any progress in our accumulated knowledge. We would be running around in circles.
That said, your hypothesis about being able to detect ultrasonics by means other than hearing is interesting. I believe you're saying that even though we can't actively hear them, our body reacts to them in a perceptable way?
It's an interesting idea, and one that should be pursued with research and testing. But not posted about as if fact while it's still in the hypothesis/opinion stage.
SACD and DVD-A = solutions to a problem that shouldn't exist anyway. Industry likes them because they can throw more DRM at the consumer
CD = perfectly fine product when put together greatly. Industry doesn't like them because they are too easy to copy and rip.
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Multichannel is an improvment. A real one. For that reason, CD is outdated.
umm.... this person assumes that frequency and waveform are different things, when they are merely alternate representations of the same thing. this stuff about the outer hair cells i don't buy at all. if these were really "waveform analysers", then people would be able tell the difference if they wired their speakers the opposite way (minus-to-plus and vice versa).
the truth is that it is cheaper to make DSD DACs, so those with vested interests are trying to convince us with smoke and mirrors that their method is best.
i DARE you to ABX a properly lowpassed dirac pulse in DSD and LPCM. please, i will eat my words if you can provide me with compelling evidence based on more than just this awful pseudoscience.
i mistrust anybody who thinks using big words equates to intelligence. it is simply a sign of pretension and arrogance. like those goddamn "scientists" that deny that global warming is down to human activity, and in the same breath deny that being on the payroll of an oil company could hold any possible conflict of interest.
Well.
A Frequency is not a waveform, that much is clear. You can decompose/transform a waveform into frequency information - but that's another thing.
Also I never even tried to state that the human ear can act as a "waveform-detector". So I figure I can't even express my thoughts here without being misinterpreted just to prove that I don't have a clue what I'm talking about. Nice.
However. What I stated I can't proove because I can't ABX things I don't have access to, and I lack the math to provide you with a formal proove. I was though hoping that my statement was clear enough so that people who are reading this, some of which that seem to know a lot more math than I do, would understand what I was trying to say, and maybe could express it in more technical terms. Seems like I failed. I apologise, my native language is german, and so is my thinking, and many german expressions do not translate to english well. But I'll try again.
I was suggesting to imagine how one hair-cell in the human ear will react when fed with a dirac-pulse filtered only "by air" (it won't carry infinite high frequenzies) as opposed to the reaction when fed with a dirac-pulse thats "properly lowpass-filtered".
The hair can be thought of a resonator. It's able to resonate at a specific frequency and it's damped. The perfect dirac-pulse will be a infinitely short pressure-burst with a certain amount of energy. When it hits the hair (lets assume there was silence before that and the hair is not moving) it will excite (accelerate) the hair proportional to it's energy. Each hair will be excited no matter what it's resonance frequency is (each hair is set into motion and then resonate - each one with it's own frequency - for a short while).
When you now start filtering that pulse you will cut away frequencies and thereby cut away energy. And more, if you filter the pulse you will shift part of that energy in time - i.e. the dirac-pulse will be "stretched" in time. The effect is that the hairs will have time to start moving bevore the end of that pulse is reached, so the amplitude will build up slower and the amount of energy that will be absorbed by the hair will further decrease. I can't calculate by how much since I don't know how fast those hairs move when resonating. I can only guess (not even estimate) that the effect should be small.
So. The point of all this is, if you cut away energy from the pulse by lowpass-filtering it, the amplitude of the resonance of the hairs will be decreased, and it will take longer to build up completely (to max. amplitude), although only some microseconds.
That's what I was trying to say in my first posting. I admit that's absolutely no proof that humans can hear the difference, but I think it proofes that the mechanics of hearing would allow for the detection of the difference.
That all if the above is correct - and I repeat, I can't prove it because I just don't know enough maths & physics to prove it formally. However if there is a flaw in my logic or any assumtions that I made do not hold, that flaw should be easily found by someone who e.g. has a degree in physics and/or math. So please just let me know if this is correct or not, and if not, where I made a mistake.
umm.... this person assumes that frequency and waveform are different things, when they are merely alternate representations of the same thing. this stuff about the outer hair cells i don't buy at all. if these were really "waveform analysers", then people would be able tell the difference if they wired their speakers the opposite way (minus-to-plus and vice versa).
BTW, polarity is ABXable, there are some AES articles to this effect, and there are excellent physiological reasons why it is audible.
i mistrust anybody who thinks using big words equates to intelligence. it is simply a sign of pretension and arrogance. like those goddamn "scientists" that deny that global warming is down to human activity, and in the same breath deny that being on the payroll of an oil company could hold any possible conflict of interest.
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Got a chip on your shoulder, eh?
There's a reasonably clear difference between arguing from a cogent but incompatible paradigm, and throwing around big words to confuse people. What we are discussing is definitely the former.
So. The point of all this is, if you cut away energy from the pulse by lowpass-filtering it, the amplitude of the resonance of the hairs will be decreased, and it will take longer to build up completely (to max. amplitude), although only some microseconds.
That's what I was trying to say in my first posting. I admit that's absolutely no proof that humans can hear the difference, but I think it proofes that the mechanics of hearing would allow for the detection of the difference.
That all if the above is correct - and I repeat, I can't prove it because I just don't know enough maths & physics to prove it formally. However if there is a flaw in my logic or any assumtions that I made do not hold, that flaw should be easily found by someone who e.g. has a degree in physics and/or math. So please just let me know if this is correct or not, and if not, where I made a mistake.
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There are less of mathematical rather than physiological reasons for why this would not be true. After the transducer stage of human hearing (hairs + neural circuitry turn sounds into pulses), there is a ton of additional processing that goes on to generate what we call sound. Even if ultrasonic content does affect the response timings of each frequency, it's highly possible that later neural processing could filter it out.
If each hair behaves halfway linearly, it not only will respond to a pulse in terms of a driven (ultrasonic) component and a resonant (sonic) component, but that behavior
should be equivalent to having the signal filtered through a transfer function, just like how an RLC circuit can be described both in terms of a differential equation and in terms of a transfer funcion. I don't think it's quite like "1 frequency = 1 hair" - a whole bunch of hairs fire, and there may be a partial response for hairs further away from the real frequency. It's the brain's job to make sense of all the generated pulses, and it's a distinct possibility that it knows how to get rid of ultrasonics as part of the process.
BTW, polarity is ABXable, there are some AES articles to this effect, and there are excellent physiological reasons why it is audible.
The only detection of polarity by human hearing that I am aware was achieved using assymetrical waveform test signals. Not actual music. Do you have references demonstrating such discrimination with valid DBT listening tests in actual music program material?
-Chris
The only detection of polarity by human hearing that I am aware was achieved using high amplitude assymetrical waveforms in a hearing test. Not actual music. Do you have references demonstrating such discrimination with valid DBT listening tests in actual music program material?
-Chris
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I haven't bought these articles yet ($80! for them! ow) so I only have the abstracts on hand. Admittedly all I've done is search for "polarity" on AES, but....
- Observations on the Audibility of Acoustic Polarity. JAES Volume 42 Number 4 pp. 245-253; April 1994. Greiner, R. A.; Melton, Douglas E. A number of experiments are described which show that absolute acoustic polarity is clearly audible in certain select cases of reproduced sound from acoustical instruments. The nature of the audible differences and the characteristics of the temporal signals which lend themselves to audibility are described. A large double blind listening experiment using varied musical program material is described as well.
- Comments on -Observations on the Audibility of Acoustic Polarity- and Author's Reply. JAES Volume 43 Number 3 pp. 147-149; March 1995.
- Proofs of an Absolute Polarity. 91st AES Convention Proceedings (September 1991), preprint #3169. Johnsen, Clark. Absolute polarity is an interesting phenomenon (wherein) those who don't hear the effect mostly doubt the opinion of those who do. (John Roberts, AES) A newly-devised test (herein called triple-blind) once-and-for-all assesses polarity audibility variously among audio engineers, hobbyists and musicians. Results decisively affirm the sensation; many trial subjects moreover testify that Absolute Polarity's palpable reality constitutes an essential addition to the audio engineering armament.
- Observations on the Audibility of Acoustic Polarity. 91st AES Convention Proceedings (September 1991), preprint #3170. Greiner, R. A.; Melton, Douglas E. A number of experiments are described which show that absolute acoustical polarity is clearly audible in certain select cases of reproduced sound from acoustical instruments. The nature of the audible differences and the characteristics of the temporal signals which lend themselves to audibility are described. A large double blind listening experiment using varied musical program material is described as well.
- Another Look at the Importance of Transducer Polarity in the Recording Studio. 91st AES Convention (September 1991), preprint #3172. Kaiser, James A.; Hedden, Gary H. Assuming identical polarity of the left and right loudspeakers in a reproducing system, earlier studies have investigated the audibility of absolute versus reversed polarity. Little is known about the importance of absolute polarity in headphones. This paper presents experiments which reveal the need to determine and maintain absolute acoustical polarity whenever headphone foldbackis utilized in the studio. Simple tests and practical solutions are described.
I also heard that the cochlea has something of a rectifying effect on the sound, but it'll take me more time to dig up that info.
Obviously, examiation of the actual papers is required to know the specific conditions(and how this could relate to actual music-not just extraordinary samples) and any flaws in the tests(if apparent). I dont have any of these specific papers, but I'll ask a friend who has access to some of this material if he can get me some of these papers. I'll email you copies if I get them(and if you want them).
-Chris
The only detection of polarity by human hearing that I am aware was achieved using high amplitude assymetrical waveforms in a hearing test. Not actual music. Do you have references demonstrating such discrimination with valid DBT listening tests in actual music program material?
-Chris
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I haven't bought these articles yet ($80! for them! ow) so I only have the abstracts on hand. Admittedly all I've done is search for "polarity" on AES, but....- Observations on the Audibility of Acoustic Polarity. JAES Volume 42 Number 4 pp. 245-253; April 1994. Greiner, R. A.; Melton, Douglas E. A number of experiments are described which show that absolute acoustic polarity is clearly audible in certain select cases of reproduced sound from acoustical instruments. The nature of the audible differences and the characteristics of the temporal signals which lend themselves to audibility are described. A large double blind listening experiment using varied musical program material is described as well.
- Comments on -Observations on the Audibility of Acoustic Polarity- and Author's Reply. JAES Volume 43 Number 3 pp. 147-149; March 1995.
- Proofs of an Absolute Polarity. 91st AES Convention Proceedings (September 1991), preprint #3169. Johnsen, Clark. Absolute polarity is an interesting phenomenon (wherein) those who don't hear the effect mostly doubt the opinion of those who do. (John Roberts, AES) A newly-devised test (herein called triple-blind) once-and-for-all assesses polarity audibility variously among audio engineers, hobbyists and musicians. Results decisively affirm the sensation; many trial subjects moreover testify that Absolute Polarity's palpable reality constitutes an essential addition to the audio engineering armament.
- Observations on the Audibility of Acoustic Polarity. 91st AES Convention Proceedings (September 1991), preprint #3170. Greiner, R. A.; Melton, Douglas E. A number of experiments are described which show that absolute acoustical polarity is clearly audible in certain select cases of reproduced sound from acoustical instruments. The nature of the audible differences and the characteristics of the temporal signals which lend themselves to audibility are described. A large double blind listening experiment using varied musical program material is described as well.
- Another Look at the Importance of Transducer Polarity in the Recording Studio. 91st AES Convention (September 1991), preprint #3172. Kaiser, James A.; Hedden, Gary H. Assuming identical polarity of the left and right loudspeakers in a reproducing system, earlier studies have investigated the audibility of absolute versus reversed polarity. Little is known about the importance of absolute polarity in headphones. This paper presents experiments which reveal the need to determine and maintain absolute acoustical polarity whenever headphone foldbackis utilized in the studio. Simple tests and practical solutions are described.
I also heard that the cochlea has something of a rectifying effect on the sound, but it'll take me more time to dig up that info.
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Obviously, examiation of the actual papers is required to know the specific conditions(and how this could relate to actual music-not just extraordinary samples) and any flaws in the tests(if apparent). I dont have any of these specific papers, but I'll ask a friend who has access to some of this material if he can get me some of these papers. I'll email you copies if I get them(and if you want them).
-Chris
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Thanks, I'd definitely be interested.
Multichannel is an improvment. A real one. For that reason, CD is outdated.
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So you agree with Jean-Michel Jarre when he says that CD is bad (he finds vinyls better than CD) and can't do multichannel?
Some artists did Surround CDs (like Isao Tomita) isn't it multichannel? (Limited I agree)
But the SACD and DVD-A are for 5.1 sound which are to me useless for portable players.
There are less of mathematical rather than physiological reasons for why this would not be true. After the transducer stage of human hearing (hairs + neural circuitry turn sounds into pulses), there is a ton of additional processing that goes on to generate what we call sound. Even if ultrasonic content does affect the response timings of each frequency, it's highly possible that later neural processing could filter it out.
If each hair behaves halfway linearly, it not only will respond to a pulse in terms of a driven (ultrasonic) component and a resonant (sonic) component, but that behavior should be equivalent to having the signal filtered through a transfer function, just like how an RLC circuit can be described both in terms of a differential equation and in terms of a transfer funcion. I don't think it's quite like "1 frequency = 1 hair" - a whole bunch of hairs fire, and there may be a partial response for hairs further away from the real frequency. It's the brain's job to make sense of all the generated pulses, and it's a distinct possibility that it knows how to get rid of ultrasonics as part of the process.
Yes, it's a possibility, but I see no reason why it should be like that. Why should the brain filter out ultrasonic - why should the brain throw away useful information? I mean "the brain" has no idea what ultrasonic is - and to me it just seems pretty far fetched that it should filter out something when there is no point in "learning" (in an evolutionary sense) such a thing.
But again, I really don't wanted to claim that we can hear the difference - because I just don't know. I just wanted to say that the human ear will react differently to lowpass-filtered sound, and that I don't see why so many people seem to be so sure that we just can't detect it.
If the response in the inner ear looks different than why shouldn't it sound different? If the response is just very slightly different - then maybe it will sound just very slightly different. But different anyway :-).
So. Now I've written in this posting "I just don't know" where I wrote in my first one that I think that 44.1khz is insufficient. So. I think it's insufficient because there is a _good chance_ that we (at least some of us) can hear the difference. CD is o.k. as a cheap medium for the masses, as a replacement for tapes etc. It's just not "everything we need".
You'd be surprised to find out how much the brain throws away. It's its basic working mechanism to prevent information overload.
A good understanding of this mechanism (without 'maybe' and 'could') will help us achieve the same in the processing and storage on our machines. Storing and processing everything that 'could be' out there is and endless process.
Ultranalog, storing 4-bit delta-PCM data with 1mhz should not be a problem at all. And that would cover pretty much. Maybe not everything, but much much more than 44.1khz/16bit. However.
Does the brain throw away so much? I always thought that it more often just "packs" data - and that's a huge difference. And even if the brain throws away much information in the hearing process - the question I'm asking is why should it throw away "every bit of information about the ultrasonic component"?
Of course you're right when you say a good understanding of that mechanism would help, but I'm afraid until now it's just a on of mother nature's secrets.
Standards are set upon knowns, not guesses. The very word ultrasonic implies proper research has yielded solid results. So until this established knowledge (http://www.audioholics.com/techtips/roomacoustics/physicsofhearing.php) is overthrown, this is what we have to go by.
Ah... this isn't leading anywhere. And it's gone pretty OT. I'm out :-)
FWIW, i wish i had the equipment, resources and enough good samples to do my own tests on these things... at the very least my test result would be an excellent system to listen to
as far as the polarity thing goes, i'm sure it would be ABXable in quite a few cases, however not for the reason that the ear is percieving the "shape" of the waveform, or else polarity would be a dead giveaway (which surely it is not, or it would be much easier to prove the difference). the reason would be more related to the mechanics of the ear (after all, the ear drum's excursion would undoubtedly be different in cases of reversed polarity, and with the right sounds it should be a cynch for the brain to tell a difference). i suspect that the polarity difference is only really audible for low frequencies at relatively high amplitudes. but i can't say that for sure (and i haven't read the articles).
as far as an unfiltered dirac pulse causing the hairs in the ear to behave differently to a lowpassed dirac pulse, that sounds intriguing. certainly there's room for a physical difference, and possibly even a perceptible one. you can't argue with conservation or energy after all. all i can say is that this needs some fairly extensive testing done.
right now, the problem i see is that the playback chain is too complex and variable (amps, preamps, different speakers, different crossover networks, etc, etc, etc), and we really can't say for certain whether a difference that is heard due to ultrasonic content is caused by the physics of the ear, or by artefacts produced by the speakers (distortions caused by the ultrasonics). in these cases, one could argue you get a more true-to-life sound by intentionally low-passing the input signal - after all, the instruments themselves would not have produced these distortions in a live performance. perhaps the extreme steepness of the 44.1K brickwall isn't an ideal lowpass for these purposes, but certainly 1-bit DSD may also not be optimal. 48Khz gives a fair amount of room for lowpassing, i think.
i remember reading that multi-bit A/D converters get diminishing returns when bitrate and sampling frequency are increased together (was it a Dan Lavry article?). for example, physical limitations of the electronics cause sampling rates above about 50Khz to resolve less than 24 bits. rather like the uncertainty principle, if you increase temporal resolution, you will decrease dynamic resolution.
here's the link:
http://www.hydrogenaudio.org/forums/index....50&hl=dan+lavry (http://www.hydrogenaudio.org/forums/index.php?showtopic=20550&hl=dan+lavry)
and also:
http://www.google.com/groups?hl=en&lr=&ie=...lr%3D%26hl%3Den (http://www.google.com/groups?hl=en&lr=&ie=UTF-8&oe=UTF-8&threadm=atldigi-D2CB2D.00431915112003%40news1.news.adelphia.net&rnum=1&prev=/groups%3Fsafe%3Dimages%26ie%3DUTF-8%26oe%3DUTF-8%26as_usubject%3D*384kHz%2520PCM%2520%253F%253F%253F*%26lr%3D%26hl%3Den)
AFAIK the hairs inside your cochlea are arranged in groups. The sensory cells which detect the movement feed their output into a small number of nerve cells, which then generate the actual impulse which is sent to the brain.
So in the end there is only one signal sent to the brain which is derived from the movement of a larger number of hairs. Those hairs cover a certain amount of area and now here is my point to counter the speculation about sharpness of impulses.
My guess is that the impulse may be as broad as the group of hairs for one to start noticing a difference. I'm not sure, maybe even a number of hair groups are linked to one signal generating cell. That should decrease temporal resolution significantly.
What I'm trying to say: with our (the people on this board) limited knowledge of the workings of the inner ear, one can make reasonable speculations in almost any direction one wants without ever finding something we could call "the truth".
I was suggesting to imagine how one hair-cell in the human ear will react when fed with a dirac-pulse filtered only "by air" (it won't carry infinite high frequenzies) as opposed to the reaction when fed with a dirac-pulse thats "properly lowpass-filtered".
The hair can be thought of a resonator. It's able to resonate at a specific frequency and it's damped. The perfect dirac-pulse will be a infinitely short pressure-burst with a certain amount of energy. When it hits the hair (lets assume there was silence before that and the hair is not moving) it will excite (accelerate) the hair proportional to it's energy. Each hair will be excited no matter what it's resonance frequency is (each hair is set into motion and then resonate - each one with it's own frequency - for a short while).
When you now start filtering that pulse you will cut away frequencies and thereby cut away energy. And more, if you filter the pulse you will shift part of that energy in time - i.e. the dirac-pulse will be "stretched" in time. The effect is that the hairs will have time to start moving bevore the end of that pulse is reached, so the amplitude will build up slower and the amount of energy that will be absorbed by the hair will further decrease. I can't calculate by how much since I don't know how fast those hairs move when resonating. I can only guess (not even estimate) that the effect should be small.
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...but a "resonator" is a filter by another name. In other words, the "hair" "filters" the signal (and that's not how it works, but still...) and just won't respond to the ultrasonic stuff that it's not made to respond to.
Imagine a long heavy pendulum. If you push it slowly, it'll move. Low frequency resonance + low frequency input = movement. However, imagine tapping it sharply with your knuckles - it'll hurt your knuckles and barely move the pendulum. low frequency resonance + high frequency input = no response.
That assumes the ear is a linear device, which of course it isn't. There may be something we don't understand in there which responds to ultrasonics (there
is, already proven, if you count bone conduction) - but it's not linear resonators responding quickly to a time domain signal which they would ignore in the frequency domain - the two are linked, and you can't have one without the other.
If there's one mechanism for detecting pitch, and another (with different resolution) for detecting onset, that might make sense. It's a nice idea, but I haven't seen any proof.
Cheers,
David.
@2Bdecided:
yeah, i know what a resonator is and that one can also express it as a bandpass filter. but every filter has a limited Q and simple mechanical filters tend to have a Q < 1.
I was not trying to say that those hairs will react much on ultrasonic in general, but I wanted to show that there's at least a possibility that the time of the first signal sent to the brain is different if depending on the filter used - i.e. how much it cuts off and how exactly the IR of the filter looks.
Imagine your long heavy pendulum. Now form a signal that's silence at first and then somewhere add the resonance frequency of the pendulum. Let the transition (volume change) from silence to the resonance frequency be sharp. That rapid "amplitude modulation" (or volume-change) "generates" higher frequency components in the range of the transition. Now filter the whole waveform with a sharp lowpass filter that's passband starts just about the resonance frequency. What you'll get is a signal with a very less shart transition from silence to the full level of the resonance-frequency. If you now have that pendulum excited by the original waveform/signal once and another time by the filtered waveform/signal you'll see the difference. It's not much, but it's there. Depending on what filter you use you'll get a response that starts when it should but not as strong as it should (e.g. filter without pre-ringing) or even a response that starts before it should (e.g. filter with pre-ringing). And the amplitude-change will always take longer with the filtered signal. Got me now :-) ?
@Gecko:
What I'm trying to say: with our (the people on this board) limited knowledge of the workings of the inner ear, one can make reasonable speculations in almost any direction one wants without ever finding something we could call "the truth".
Correct. Indeed I didn't want to proove anything (about the ability of humans) by my postings in this thread since my knowledge is far too limited. It's just that I've read the absolute claim (or rather dogma) "it's impossible" (regarding the questionable ability to detect differences > 20khz) far too often ... and I think it's not impossible with what we know. Nothing more, nothing less.
---
I've answerd because I've been quoted and personally addressed and I don't intend to be inpolite or go and "hide" after I've made a statement. But I still think this has gone "very OT" (not that I'm so inoccent of pushing this further OT) ... :-)
I found this when searching through the hyperphysics site mentioned earlier on this board. Some food for thought about the inner ear:
http://hyperphysics.phy-astr.gsu.edu/hbase/sound/corti.html (http://hyperphysics.phy-astr.gsu.edu/hbase/sound/corti.html)
At least I haven't been a total liar.
I cannot hear the difference between the SACD and CD layers on any of my SACD's.
Well, I can and especially in our listening room where there is much better equipment than I have.
Most likely they are differently mastered. There was a thread around here recently, where an email from some official guy was posted who acknowledged the fact that the CD layer was mastered differently. In particular they aplied more dynamic compression to be "competitive" in the CD changer.
This is the case with the two stereo layers of the Pink Floyd 'Dark Side of the Moon' SACD, for example, as demonstrated by measurements published in the 'high end' journal Stereophile.
I have also somewhat less rigorously found difference between the remastered CD version of the first few Peter Gabriel albums, versus their SACD (two-channel nonhybrid). SPL meter readings showed the CD versions to be a consistently louder than the SACD ones, when fed throught he same outputs (e.g., the front l/r channels of my SACD player's mutichannel outs, or the 'CD out' channels of same, both of which pass analog only)
When mastering is probably different, all bets as to *format* being the cause of sonic difference, are off.
Btw, people on this thread might be interested to know that Dan Lavry, master engineer of A/D converters , now has his own forum where high-level objective discussion of digital audio technology (included DSD) is taking place --
http://recforums.prosoundweb.com/ (http://recforums.prosoundweb.com/)
Bringing this back to DSD....
I think the reason for DSD being interesting to audiophiles has more to do with marketing than technical superiority. Open any SACD and you'll find a little insert that explains how "DSD closely resembles an analog waveform," resulting in a warm, analog-like sound quality. It's the old "16/44 makes a waveform look like a pile of bricks" argument all over again.
No no no, not *bricks*...*stairsteps*.
But let's address the 'issue' that is 'complicated'. Please be more specific. Techncially, it's a rather simple problem from my perspective: overcompression and excessive hard limiting. Techncially, just reduce the overall amplitude so that hard limiting is not an issue. Then stop compressing the signals into a primarily 15-20dB average envelope. Ordinary CD has well over 90dB available! But this seems to go way past technical problems and enter into the fray of the cliche' coined 'loudness wars'.
The sick thing is, if people *want* to dynamically compress their music, then mfrs could simply add
a button to playback devices to allow that *choice* (DVD players already have a form of it -- usually called 'night listening mode' or somesuch). Radio stations *already* have outboard compressors. To have the DR limitation irreparably 'hard coded' into the product, is just ridiculous and sad.
- Proofs of an Absolute Polarity. 91st AES Convention Proceedings (September 1991), preprint #3169. Johnsen, Clark. Absolute polarity is an interesting phenomenon (wherein) those who don't hear the effect mostly doubt the opinion of those who do. (John Roberts, AES) A newly-devised test (herein called triple-blind) once-and-for-all assesses polarity audibility variously among audio engineers, hobbyists and musicians. Results decisively affirm the sensation; many trial subjects moreover testify that Absolute Polarity's palpable reality constitutes an essential addition to the audio engineering armament.
Re: Clark Johnson and the claims of this AES preprint, see this story related on rec.audio.high-end by Tom Nousaine (http://www.google.com/groups?q=g:thl1365314558d&dq=&hl=en&lr=&ie=UTF-8&selm=56U0c.162097%24uV3.712702%40attbi_s51)