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I heard some audio technician say that dvd-audio is largely superior to sacd, and that sacd is someway "out of standard". Is it true? In what sense is dvd-audio superior in some technical sense?
Follow Ups:
and searching the archives on this topic, there has been massive amounts of discussion on this point over the past few years. Both "hi-rez" formats seem to do a little better than CD in 2 channels for most people and they both have multichannel capability. In the end, if you are choosing one or the other, I would strongly suggest looking at available releases, the catalogs are quite different and this is probably far more important than minor differences that may or may not exist in intrinsic sound quality.
it is music that matters most and quite sadly the two formats catalogs are different. unless one looks at a player that plays both.
In terms of digital formats you have DVD-A vs CD, DAD (DVD without video content), MP3 and variants, HDCD, etc...except for DVD-A all of the others are either lossy formats (not all of the info IN is used to reconstitute the original signal due to bitrate transfer limitations bits/sec/channel) or have lesser resolution (sampling freq. and bit resolution)...if the recording and equipment used to record (ex the mic) can take advantage of the higher resolution than the numbers are clearly in favor of DVD-A for *equal and competent* implementation of the playback gear (ex. the player)...not much you can argue there in terms of higher fidelity (other pratical matters such as music availability is another matter but assuming thats not what "superior" refers to here).In terms of high resolution formats there are 2 main formats currently 1) DVD-A, 2) SACD/DSD both are lossless (ie, use all the info on the disk to reproduce the signal). SACD however has, in the main, a lower bit resolution of 20bit (equivalent) vs 24bit (actual) for DVD-A which is at the edge (beyond in fact) of what can currently be done with current DAC (digital to analog converters) considering inherent noise of circuits (in fact your speakers will never reach that kind of resolution...but they are analog...however the use of higher resolution source affords better processing thus improved sound).
THe noise for DVD-A is also comparatively lower since it goes inversely as the bit resolution. On the other hand, SACD uses 1 bit conversions at very high sampling freq. (@2.8 mega hertz, it integrates many sample points to compensate for errors..but still not enough as an eng. design compromise), these conversions produce a lot of noise that is pushed out of the "audible" range (commonly accepted as 20-20kHz) by a process called "noise shaping". That noise shaping, while deemed innocuous (from its marketing), still remains significant in the audible range and can be detrimental to your amps (unless you have a player with a default 50 kHz filter, to take away that energy from your amps).
So for SACD the noise increases with freq. (not for DVD-A) to the point where the dynamic range (ie, the difference between the lowest and the highest resolved signal) of SACD becomes even less than CD! You can even argue whether SACD is better than CD! on listening and from technical consideration SACD is NOT a high res. format (it being inferior to even CD in some "aspects"). It has been marketed as such by Sony and Philips however (looking for licensing revenue to offset current loss from the expiring CD licensing) and has really hampered the adoption of the only true current commercial High resolution format in DVD-A.
There are bigger problems yet for SACD. Most of the technology to date is based on PCM (digital signals of DVD-A/CD/DVD, ect.) whether on the recording side or playback side. There are currenlty NO effective way to perform DSP in the digital domain (mixing and processing in studio, room correction, barely on bass management, etc.) based on SACD/DSD digital data..unless you convert to PCM...or you start with PCM (which defeats the purpose of ever going to DSD/SACD in the first place)...Years of perfecting PCM technology would go down the drain if DSD/SACD were to become the standard...Sony et al. does not care about improving the process, rather, they care about developing a new source of licensing revenue (ie, Sony came out with SACD *exaclty* 20 yrs after their licensing rights on CD expired).
Lets not forget that theres is no SACD with video content currently available (as opposed to DVD-A which currently carries some video content)...Sony is working on SACD II to do that which will render all current players/processors for SACD obsolete...
in short, SACD is a compromised technology that is not even on par with CD, that disregard pratical and entertaining aspects of playback and recording...why it has ever seen the light of day as a hi-res. format is beyond me (well not really considering the financial push)...and i do consider detrimental to achieve the best possible music reprodcution out of our systems.
This being said, some people like the softened, edgeless, forgiving, compressed, sand-brushed, fake cymbal, sound out of SACD. So you may want to try both over many different SACD and DVD-As to see what is to your liking (which is the bottom line)..some good ex. DVD-A is Goodwin "swinging for the fences", Marvin Gaye (raved about around here), Beethoven 9th Abravado, Queen night at the opera, etc.
If you look at the DVD Audio Standard there can be bit depths of 12, 16, 20, and 24. Sample rates can be 44.1, 48, 88.2, 96, 172.4, and 192.
If you research DVD audio discs that are out, many are 24/44.1 or 24/48 which isn't a whole lot different than standard CD. It is not until you get into the high sample rates 96 and up that DVD audio really distances itself from redbook CD and there is not a lot of recordings available with these resolutions. Studios who have been making 16/44.1 CD's for years do not have the equipment to do 24/192 any more than they have the equipment to do DSD. To do either of the new formats correctly takes a significant investment in new hardware.Unfortunatly many SACD's are just DSD conversion of the same low res PCM recordings and since you cannot create data that was not there in the first place. These are no better than the PCM that was started with. Pure DSD recordings are by definition high sample rate and are more widely available than 24/192 DVD Audio Discs. Yes it is true, DSD signals need to be sampled to multibit for DSP, but a DSD capable workstation does this at a very high sample rate. Again, this is also required for high sample rate LPCM, but not by definition DVD Audio which does not need to be higher bit or sample rate than Redbook.
The comment about SACD noise causing damage to amps is just a ruse to spread Fear Uncertainty and Doubt. Reality is that the max noise level I have seen in any of Stereophile's test's is -40db. While when testing Amps, Stereophile has been measuring amplifier frequency response out to 200khz with a high level signal for years. Stereophile is not having amps blowing up left and right in this test are they. It is just not an issue.
From a marketing standpoint both DVD Audio and SACD are not about bringing hi res to the masses, it is about bringing out multichannel and yes it is about royality revenue. The big corporations do not care about about a few audiophiles or they would not keep turning up the level of CD's until there is no dynamic range left. Do we want to bring up the fact that DVD Audio was delayed for YEARS as these corporations argued about what Watermarking Technology to use? Talk about screwing something up!
The only area where from a Technology standpoint that SACD does not meet CD is about 10db higher noise floor at 20khz. There is not a system out there where you can tell the differnence between a -90db and a -100db noise floor so this point is irrelevant.
From a listening stand point, I can't tell the difference between the SACD which started from a DSD recording and PCM which started from a 24/96 or higher recording. I can tell the difference between the above and 24/44.1 and 24/48 which at this time make up a lot of the DVD Audio Titles available.
Cheers
is that the "limitations" of DVD-A you mention with regard to actual recording resolution (Fs and bit) are not *inherent* to the format. I also pointed out in my initial post what you indicated in yours: you do need to have the right equipment (I gave the example of the mic) to take advantage of the higher resolution afforded by DVD-A...Thus DVD-A has the potential of establishing a superior standard.On the otherhand I see *inherent* limitations in current SACD technology that make it sub-standard (with respect to what a high res. format should be) no matter what the recording technique...
and thats where my argument lies in the inherent capabilities and shortcomings of the 2 formats...in light of this, ill add that even with current recording techniques, a well recorded session (delivered in SACD) may come out superior to CD if that one is not done as carefully. In large part, the recording art can be determinant beyond the format capability (no point of resolving a lesser recording)...this, indeed, applies to both formats in questions.
To address another of your point...Ive seen a mention a couple of time of players having builtin (and defeatable maybe) high freq. cutoff filters to protect potentially unstable amps from the presence of high freq. noise inherent in SACD <100 kHz..the level are apparently high enough to cause concern and have someone implement a protection for it at cost. Why would the manufacturer include this feature if not for engineering concerns, where would the "ruse" be in that? I believe this (these) players were Sonys but cant be sure, if someone knows please let us know.
whilst NonA's post is so laughably biased/inaccurate i didn't even bother to respond, i just thought i'll add a few words to yours.*** 24/44.1 or 24/48 which isn't a whole lot different than standard CD ***
i probably think 24 bits is a bigger advantage over higher sampling rates. on my system, i can hear more of a difference between 44.1/16 and 44.1/24 than 48/24 and 96/24. higher bit depth - at leaast during the recording process - gives you more headroom which is vital for PCM recording, which does not tolerate saturation very well.
*** To do either of the new formats correctly takes a significant investment in new hardware ***
Agreed but i think moving to hi-rez PCM is less of a stretch than investing in DSD. although 48/24 is still quite common in the older equipment, many studios have already standardised on 96/24 or better even for engineering CDs. DSD requires new toolsets and is not as easy to edit as PCM.
*** many SACD's are just DSD conversion of the same low res PCM recordings ***
'many'? in my opinion, it's just a very small minority - Stephen would have a better idea.
*** where from a Technology standpoint that SACD does not meet CD is about 10db higher noise floor at 20khz ***
i'm not sure this is true. i think the SACD spec has a lower theoretical noise floor than CD throughout the entire audible spectrum. the stereophile graphs that everyone keeps pointing to are more to do with individual player performance than about the format.
*** 24/44.1 and 24/48 which at this time make up a lot of the DVD Audio Titles available ***
again, not sure i would agree with that. i have about 50 DVD-As, and only 1 title has a 44.1/16 2ch track (miss e ... so addictive) and only about 2 titles have 48/24.
since you pickedup a spectrum analyzer...but before your head explodes reads this."whilst NonA's post is so laughably biased/inaccurate i didn't even bother to respond.."
im sorry i cant be good enough for you to indulge in a response (aside of the arrogant one above)...
This is a discussion forum isnt it (lets start slow an' easy so that you understand, shall we?:))?
if you see "inaccuracies" do post and elaborate...and we'll address back and forth 'til we reach a logical "concensus"...thats the whole point of forums...(errratum humane est?) I'm not immune to it...and welcome any "counter-point"..
Your arrogance makes you think we'll accept baseless critic such as this: "...NonA's post is so laughably biased/inaccurate..." without further input and that we'll take it as a statement of fact because of what? You're a Guru and need not justify because of what you fancy yourself to be?...This reeks of delusional arrogance...
and, by the way, I have nothing against arrogance but only from those who can back it up...and here's a piece of constructive criticism that will save you some hurt when you venture in other circles: you can't back it up and you're not a Guru :-)
this entire thread is absolutely astonishing.please, all of you, go back to your systems, put on your favorite recording, and RELAX. why are you wasting so much time fighting about which format is better, when there is so much to listen to and so little time?
i really enjoy the threads that help me get more out of what i have, or inform me about recordings i haven't heard or don't even know about.
this forum often reduces itself to a playground brawl, with the same short tempers and arrogance as displayed daily in a kindergarten sandbox. y'all need to lighten up, stop feeling guilty about what format you bought into, and enjoy the music.
... and i don't own a spectrum analyzer. But i do apologize for the swipe about "laughably biased/inaccurate".i didn't want to respond to your post because i did think it was biased and inaccurate. you've listed all the advantages of PCM/DVD-A and all the disadvantages of DSD/SACD, but neglected to point out the disadvantages of the former and the advantages of the latter. Since in the past you've shown a disinclination to have an "open mind" i didn't think it was likely that i will be able to persuade you otherwise so i didn't bothered.
in reality both formats are good, and both formats have their compromises/limitations. i certainly enjoy both, and i've listened critically to both on a variety of players and setups (i've personally reviewed half a dozen players on my system). i own more than 150 SACDs and 50 DVD-As.
as an example of your bias/inaccuracy, you mention something about SACD have an "equivalent" bit depth of 20 compared to DVD-A at 24.
well, first of all, i don't think that DSD is necessarily "equivalent" to 20 bits resolution. in fact, at least one paper has quoted that you need about 384/32 to be able to represent DSD accurately for editing purposes.
but let's assume for the sake of discussion SACD actually is equivalent to 20 bits of resolution. you are still comparing apples with oranges because 24 bits is an upper theoretical limit for PCM 96/24 or 192/24 resolution. the actual resolution is likely to be less because:
(1) PCM resolution is directly proportional to signal amplitude. if you have a sine wave at maximum amplitude to that the peaks are encoded at maximum values, then yes, that sine wave is recorded at 24 bits resolution. if your sine wave was half that amplitude, you are recording at effectively 23 bits resolution (because the most significant bit is never used therefore it is "wasted"). very low level signals are effectively being recorded using only a few bits of resolution. In most well recorded music, the average peaks of the signal is far below the maximum, so the effective resolution is less than 24 bits. Low level signals are probably being captured at less than 10 bits resolution.
DSD on the other hand has an interesting property in that resolution is not linearly proportional to signal amplitude. yes, it is still somewhat dependent on signal strength, but DSD is probably better at capturing low level detail.
to take an extreme example - a zero signal is recorded as a stream of zeros in PCM - effectively with only 1 bit resolution. a zero signal is recorded as a string of alternating 1s and 0s - no bits are "wasted".
(2) PCM hard clips when signal exceeds the maximum level. therefore when recording PCM, it is absolutely important to preserve some "headroom" to avoid the possibility of clipping. good engineers will leave around 2-4 bits of headroom when recording. that's one of the reasons why early CD recordings didn't sound too good - they were effectively being recorded at only 12-14 bits resolution instead of 16. Modern 24-bit recordings are likely to only be effectively recorded at 20-22 bits. In fact, the original reason why studio equipment is at 20 or 24 bits when the delivery format is only 16 bits is to allow engineers to have that headroom without sacrificing the quality of the final product (CD).
DSD on the other hand is more tolerant of clipping. it behaves more like analog tape in this respect.
(3) As we know, current technology makes it very hard to realise more than about 20 bits of resolution in A/D and D/A converters. even if it were possible to build accurate converters, most analog circuitry has a noise floor that will prevent full 24 bits being passed through. so the 24 bits are "marketing" bits - they are not really realisable practically.
Once you take those into account, PCM does not seem to have an inherent advantage in resolution. i would say hi-rez PCM and DSD have "similar but different" resolution - DSD with an advantage in low level accuracy and impulse response, PCM with a higher theoretical dynamic range and better editability.
i could make similar points to just about every statement you've made in your post so hopefully you'll see why i chose not to initially respond otherwise i would still be typing!
Christine Tham wrote:
"1) PCM resolution is directly proportional to signal amplitude. if you have a sine wave at maximum amplitude to that the peaks are encoded at maximum values, then yes, that sine wave is recorded at 24 bits resolution. if your sine wave was half that amplitude, you are recording at effectively 23 bits resolution (because the most significant bit is never used therefore it is "wasted"). very low level signals are effectively being recorded using only a few bits of resolution. In most well recorded music, the average peaks of the signal is far below the maximum, so the effective resolution is less than 24 bits. Low level signals are probably being captured at less than 10 bits resolution.DSD on the other hand has an interesting property in that resolution is not linearly proportional to signal amplitude. yes, it is still somewhat dependent on signal strength, but DSD is probably better at capturing low level detail."
I'm not sure where you got this information, but it's incorrect. For a linear A/D or D/A converter, resolution in bits is the base 2 log of the number of discrete states available. So it is a constant and has nothing whatever to do with the signal level. If we take an example of an analog system with input v, where the relationship of the output to the input is given by f(v), if it's linear it must display the property that:
f(a * v) = a * f(v) where a is a constant. Basically, if I scale the input by a, the output must scale by a for linearity. If this relationship doesn't hold, distortion will (usually) be generated (unless this is caused by a simple DC offset).
Now think of what would happen if the resolution were a function of signal level in an A/D or D/A. This says the quantization step size would need to be variable with signal level. So if I scaled the input v by a, the amount the output would scale would depend on the value of v. For an ideal DSD converter to have improved "resolution" at low signal levels compared to a standard converter, its quantization step size would need to be smaller at low signal levels. But then if we looked at a graph of output count vs input voltage, it wouldn't be a straight line and would thus be a distortion generator.
look at it this way. using a 24 bit A/D converter record a musical signal with a low gain setting, such that the entire signal is completely encoded only in the lower 12 bits.the results would be no different if you had recorded the musical signal with a 12-bit A/D converter but at optimal gain setting.
actually, practically the results will be worse off than using a 12 bit A/D converter in the first place, since the lower bits of an A/D are typically less accurate and less linear than the upper bits.
you may ask - well, why record at such a low gain level? why not set the gain such that the signal is encoded using the full 24 bits?
the reason is if you are recording live, you have no way of knowing in advance what the maximum peak of the music will be, so you want to be conservative and set the gain at a level where you think is optimal, then back down a bit to leave some headroom just in case the music turned out to be louder than you think. even if you have a luxury of asking the orchestra to play the loudest passage so that you can calibrate gain beforehand, you still want to leave some headroom "just in case".
a symphony orchestra is the worst case scenario. a symphony can be at an average level of 70-80 dB SPL but suddenly peak all the way to 120 dB or more for the finale. the quandary is: what do you set the gain at? if you set the gain for 70-80dB the signal will surely clip at the end (big no no). if you set the gain low enough to handle the peak at the end then most of the time the music is being recorded at -50dB below max which means you are not using the full 24 bits of the A/D converter, you are using maybe the lower 16.
i find that DSD, being a 1-bit PDM, is better at capturing low level signals. i've heard some DSD recordings done at low gain that sounded much better than the equivalent PCM recording done at low gain. the goal is still to record with the gain as high as possible, but no higher.
i still stand by my assertion that good 24 bit PCM would probably have been done with 2-4 dB headroom, which means the "effective" resolution of the recording is only 20-22 bits, assuming a "perfect" A/D converter. normalization, followed by effects processing at 24 bits of course increase the apparent resolution, but doesn't completely hid the fact that the original tracks only have 20-22 bits used.
Christine,The example you give of using only 12 bits of a 24-bit converter, do you realize that this signal is 20*log(2^12) or 72.25 dB below full scale? That's pretty conservative recording procedure if you ask me :-). The quantization errors you're referring to (neglecting noise) are the same regardless of whether it's DSD or PCM conversion. There's absolutely nothing magical about DSD that improves low-level signal handling behavior, except possibly in the minds of some marketeers. In fact, it's been shown that the noise is higher for DSD than PCM at the high end of the frequency band in the ideal case.
You may have found subjectively that the DSD recordings and/or equipment sound better at low levels, but I suspect that's a quality of implementation issue, either in the recordings or the equipment. Someone once said, "In theory, theory and practice are the same. In practice they are different" :-). All I'm saying is that there's nothing in the theory that says this should happen. In practice, I've found that good examples of both DVD-A and SACD both sound excellent and I have no preference for either one.
i was purely using 12 bits as an extreme example.however, i have had on occasion required up to 8 bits of headroom (recording at an average peak level corresponding to 16 bits using a 24 bit A/D converter) to record a very dynamic symphonic work.
in terms of theory vs practice, i agree - my observation were empirical observations, never suggested that there may or may not be a theoretical basis to it (although i have read a paper that suggested a theoretical basis for better low level performance of DSD - can't remember the reference though)
my point to NonA (which he didn't understand, or didn't want to understand) was he was comparing practical DSD performance with theoretical PCM performance and that is an apples vs oranges comparison.
in practice, as we both agree, both PCM and DSD can deliver excellent results, although each of them has some quirks.
you should have no problem amplifying and filtering your recording signals to match to your ADCs adjustable input gains and, thereby, achieve the predither 20-23 bit accuracy whether "practice or theory"...you do need to know your equipment specs as i pointed out earlier, and you can even find commercial apps that do that for you (signal cond.), or simply built the circuit yourself on a board (easily)...you also need to do a bit of trivial calculations, calibration (to verify specs, very important! since the manufacturer specs are not always accurate), and drawup a block diagram of the whole process to clarify things...this "trial and error", "rule of thumb" approach you advocate is not professional, and youd do well to get on board (literally :-) if you do this for a living...
Your point 1) "PCM resolution is directly proportional to signal amplitude.." no, the resolution (the Voltage resolution) remains the same for the converter whether at full or zero amplitude...you are confusing this with nonlinearities of the converters and/or SNR.2) "PCM hard clips when signal exceeds the maximum level..." the output with DSD has the same structure as in PCM when cliping, DSD scheme just reads.."same, same, same, same, etc". from the delta sigma modulator....there's no "soft" clipping...you're confusing this issue with filter/analog circuit transient response to a "discontinous" signal input (the clipping).
3) "As we know, current technology makes it very hard to realise more than about 20 bits of resolution in A/D and D/A converters...". did you even read the post you dismissed? This is what i wrote: "..24bit (actual) for DVD-A which is at the edge (beyond in fact) of what can currently be done with current DAC (digital to analog converters) considering inherent noise of circuits.."
So these are the points you were so keen on looking down upon...well thanks for the input.
.
nt
"(1) PCM resolution is directly proportional to signal amplitude. if you have a sine wave at maximum amplitude to that the peaks are encoded at maximum values, then yes, that sine wave is recorded at 24 bits resolution. if your sine wave was half that amplitude, you are recording at effectively 23 bits resolution (because the most significant bit is never used therefore it is "wasted"). very low level signals are effectively being recorded using only a few bits of resolution. In most well recorded music, the average peaks of the signal is far below the maximum, so the effective resolution is less than 24 bits. Low level signals are probably being captured at less than 10 bits resolution.DSD on the other hand has an interesting property in that resolution is not linearly proportional to signal amplitude. yes, it is still somewhat dependent on signal strength, but DSD is probably better at capturing low level detail.
to take an extreme example - a zero signal is recorded as a stream of zeros in PCM - effectively with only 1 bit resolution. a zero signal is recorded as a string of alternating 1s and 0s - no bits are "wasted".
"Nonsense, no bits are "waisted". All 24 bits are used to express a sample value.
In digital math zero's and one's are equally important.Another factor is that if you record with a proper gain setting the low level signals are far below the threshold of being able to hear them.
Usually the input gain of the recorder channel is adjusted just below clipping. This is just 'good practise' of the record engineer.
That dsd is somehowe preserving low level detail better is a myth that resulted from clever marketing and pseudo scientific white papers. Hires pcm with dither and noise shaping has far better low level sound quality.
That the lower pcm bits end up burried in the analog stages noise floor is perhaps a beneficial side effect as this masks any errors left over from processing and/or improper dithering after bit reduction from 32 bit processing into 24 bit audio files.
as usual, you have missed the pointas an extreme example lets say that the gain is set so that the signal never exceed 12 bits. then effectively the 24-bit PCM recording is indistinguishable from a 12-bit PCM recording. the other 12 bits are never used - therefore "wasted."
*** Usually the input gain of the recorder channel is adjusted just below clipping ***
that is fine if you can predict what the loudest part of the signal will be. in a real life situation, you often can't - that's why you need to leave about 2-4 bits of "headroom" just in case.
if you have ever recorded a live performance and successfully avoided clipping you will know what i'm talking about. i've had too many recordings spoiled by an inadvertent clip caused by too aggressive a gain setting that i stick to my 2-4 bits of headroom philosophy.
You're point misses the mark.
It's simply not an issue.Any half decent record engineer adjust his levels for a maximum input to capture the best possible dynamic range. That was true for tape and is still true for digital recording. Tape saturation was, and still is, used for effect.
Simply ask a drummer, singer or any other participant to make a loud noise and adjust your levels accordingly and add some headroom to compensate for the effect that live performances tend to get louder during the session.
Add to that the knowledge and experience and you know how to compensate for sufficient headroom.
This is just good practice and hold true for any format.
You still refer to 'waisted bits'. Which shows a lack of understanding the basic pcm theory and why it works so well for audio recording.
If you adjust your equipent for lets say a maximum loudness level of 100dB with 6db headroom. You use 23 bits dynamic range but you still encode the sample value with 24 bit precision so no bits are waisted!
Each 6dB headroom 'costs' only 1 bit.Now imagine a soft sound at 40dB during the performance. This sounds dynamic range is captured within 13 bits. (Sample values are still encoded with 24 bit precision!) Any distortion errors due to a lower signal level within the dynamic range window of your recording setup fall far below the hearing threshold.
You're point isn't valid because the end users volume level is usually adjusted in accordance with the highest loudless levels.
Suppose the users listens to this recording at 90dB loudness.
Any artifacts already below hearing threshold will get quiter another 16 dB.
*** Simply ask a drummer, singer or any other participant to make a loud noise and adjust your levels accordingly ***if only it was that simple!
if you've done a reasonable amount of live recording, you will realise it does not work like that in real life.
and never underestimate the dynamic potential of live music. particularly symphony orchestras!
i have been burned many many times.
just what do you do with a symphony that plays at 70-80dB for 95% of the time and then go to 120-130dB right at the end?
if you set the gain to peak at 130dB you will be recording 95% of the symphony at -50dB below peak. if you set the gain any higher the material will clip at the end.
"just what do you do with a symphony that plays at 70-80dB for 95% of the time and then go to 120-130dB right at the end?"Adjust for maximum input if you prefer the purist approach or use (mild) dynamic range compression to prevent trouble.
But again it isn't really a problem because the volume levels in a domestic listening environment are scaled down by the volume setting.
And as a consequence any artifacts with it.With 24 bit artifacts are already very low. With proper dither this is not an issue.
PS 120..130dB levels are hardly reached at the usual mic position.
*** Adjust for maximum input if you prefer the purist approach ***you do realise then you are effectively recording 95% of the symphony at effectively 16 bits resolution (or less), and the full 24 bits are only used for the last 5%?
*** or use (mild) dynamic range compression to prevent trouble. ***
and just exactly how do you do that? if you do it post A/D, don't bother - you've already lost the resolution and "waisted" (sic) the bits. if you do it pre A/D, you'll have to do it in the analog domain.
tell me Frank, just how many times have you actually done a recording, in live conditions? using what equipment?
***you do realise then you are effectively recording 95% of the symphony at effectively 16 bits resolution (or less), and the full 24 bits are only used for the last 5%?***That's not an issue. What is important is that the samples are 24bit accuracy. (actually 21..22bits with the best of the equipment available today)
You fail to see that distortion and artifacts are far below the signals level. If 16 bits are 'used' at a 70 dB signal then distortion is at least -90dB below that relative level.
***and just exactly how do you do that? if you do it post A/D, don't bother - you've already lost the resolution and "waisted" (sic) the bits. if you do it pre A/D, you'll have to do it in the analog domain.***
You need a compressor/limiter in the analog domain.
***tell me Frank, just how many times have you actually done a recording, in live conditions? using what equipment?
***Running out of valid arguments? :)
I have no experience with recording full orchestra's under live conditions (yet). However I don know how it's supposed to be done.
It's important to setup during a rehearsal to get a starting point for the setup. If you are surprised by a simple loudness issue during a performance of a Mahler or Wagner you didn't came prepared very well.
Frank
please don't use an analog dynamic compressor/peak limiter as you suggest.these things do serious damage. you can pretty much forget ever achieving the full potential of a 16 bit recording, let alone a 24 bit recording, if you use them.
*** You fail to see that distortion and artifacts are far below the signals level. ***no, i never said that recording at effectively 16 bits will not yield anything other than 16 bit accuracy.
you seem to be finally acknowledging *my* point, which is that 24-bit PCM recordings do not always have 24 bit resolution - it depends on the signal strength. due to the need to have headroom during the recording process, most 24-bit PCM recordings are effectively utilising less than 24 bits.
*** I have no experience with recording full orchestra's under live conditions (yet). ***
maybe you should actually try it one day. then perhaps you will appreciate what i am talking about.
i'm not saying i am an expert. i have made more botched recordings than acceptable ones. at least i understand how difficult it is to record under live conditions that make me appreciate why it is an art rather than a science, and salute those who do it well.
*** If you are surprised by a simple loudness issue during a performance of a Mahler or Wagner you didn't came prepared very well.
***perhaps, if you've actually experienced recording under live conditions you will realise it is not about not knowing the material, asking the musicians to play their loudest bits first so you can set gain levels etc etc..
no human set of musicians will play the same piece of music at exactly the same loudness levels twice.
i repeat: never underestimate the dynamic potential of live music. those who do not understand this statement have not done live recording before.
just plugging along what shes read in a pamphlet...hope she doesn't do that for a living...or that shes doing the work "pro bono" for the customers' sake...:-)..even then i dont think theyre getting their moneys worth :)
NT
you getting old man, your posts on this thread have contributed nothing but wasted space, step outside and get some fresh air.I can see you wagging your tail and meek bark of yours everytime you see fit to come in defense of your pals that need none. Your pack mentality works elsewhere...not here...move on...
NT
nt
well, wrong again...you should know the sensitivity, dynamic range, and gain applied in the chain...with this you can easily calculate what you should expect as your maximum amplitude...all these things you can find easily in the manufacturer specs (Yep, even for the mic)...the 6 dB rule is for lazy people...
"DSD on the other hand is more tolerant of clipping. it behaves more like analog tape in this respect."No it doesn't. There is no saturation effect. If the signal clips you end up with large sequences of zero's (or ones).
You're even advised to keep record levels below -6dB.
With pcm you can max out if subsequent processing is done with 32bit or higher accurracy.
there was a discussion about this several weeks ago, which you probably missed out on. speak to ted smith about it.of course, just because it saturates better, doesn't mean one shouldn't try and avoid saturation. some engineer posted in reply to ted that it's a good idea to keep to -6dB below saturation as there are some artefacts close to saturation point.
Tape saturation is entirly different than DSD 'saturation'.
It's better to call it overload in that case.The fact that it's already discussed changes nothing.
Frank
Frank, have you done any DSD recording?it is very easy to deny or dismiss something you have no first hand experience of.
What has my experience got to do with this?It doesn't change the fact that overloading a dsd dac is very different from pushing a tape into saturation.
The 'official' advice is to stay below -6dB with dsd to allow headroom for processing and prefent artifacts.With pcm you can use all the available headroom for recording without bothering about processing if you move into a DAW with more headroom if it's processing in at least 32 bits.
PS.
I gather from previous postings that you have a lot of experience with overloading converters. :)
*** The 'official' advice is to stay below -6dB with dsd to allow headroom for processing and prefent artifacts. ***you got it backwards. See Ted Smith's post and graemme's reply.
- http://db.audioasylum.com/cgi/m.mpl?forum=hirez&n=174106&highlight=DSD+saturation+Ted+Smith&r=&session= (Open in New Window)
Staying -6dB below clipping and calling it 'build in headroom' is turning things around.DSD clips gradually because it runs out of bits to represent the signals wave shape. To suggest that this behaviour mimics tape saturation look more like an attempt to keep the dsd myth alive.
With any digital medium the amount of headroom is at the discretion of the record engineer.
regardless of whether you are calling the glass half empty or half full, your original statement is still factually incorrect, and that is all i am saying.*** The 'official' advice is to stay below -6dB with dsd ***
it's not 'official', it's not 'advice', and it's not '-6dB'
you love jumping in and saying statements that are incorrect. it really lowers your credibility.
accusations of arrogance in a post dripping with same (e.g. "lets start slow an' easy so that you understand"). There is no basis for debate. In your experience, SACD sounds bad, apparently verging on terrible. In the experience of others, it sounds fine. Unless you sit in the same room listening to the same thing, how is debate going to resolve this? Even then, there can be disagreement, I've seen it happen. To each their own.
put that in there on purpose...You have a point about the comparison issue...eventhough i think personal bias is more of an issue there...in my listening comparisons I ask "Ok, does it sound closer to the real thing?" more often than not the DVD-A comes closer (and even CD) compared to SACD if well recorded...
As for "terrible"...no, those are more of "audiophile relative differences" and the disproportionate importance "we" attach to them...add on top of that the practical restrictions SACD imposes on processing (at the recording or customer level), hence my position...
Good God, man! Post your system, and get some help soon! :-)
nt
You might recall, DG did listening comparisons where they couldn't reliably distinguish the DSD recording from 192/24 PCM. For you to claim that the final product is so utterly inferior to DVD-A...well, something just doesn't add up. :-)
DG was just being 'kind' to SACD by saying that. (It could have been much, much, worse!) :-)
...we'd probably get all sorts of entertaining hyberbole from the two of you, a bit like the IAR80 website. Then, if they made the tests blind...you'd be scratching your heads like monkeys (again). You might even prefer the DSD encoding! :-)
even how bad it can sound...but then again i wouldnt expect you to dig into it :-)
Or even tested?
I have recorded the SFSO Mahler 1st Sacd with 24/96 pcm.Noise floor sits at -75dB according to adobe audition analyzer scale.
Low level detail sounds grainy and noisy.The opening part of this recording is very suitable to crank up the gain and listen for sacd short commings.
it was on whole wheat bread. Microscopic particles of wheat were expelled. The grain you hear was actually in the air of the hall.
it's a live performance - if you crank up the volume you can clearly hear it's audience noise and reverb. i can tell it's reverb even at normal listening levels - that's what makes it a good recording.as for the low level detail sounding grainy and noisy, what can i say, your PCM recording is obviously not very good :-)
I did the same for a quite pcm recording.
Dead quit and far cleaner low level sounds.I even got -82dB from a very quite vinyl track with the same setup.
Of coarse you can hear audience noise and reverb. I even get the sounds of turning papers. The point is that these very low level sounds where grainy and noisy.
Better AD conversion is on it's way. (Motu 896HD)
it's very easy to get dead quiet for a studio recording - much harder for a live recording. and i wouldn't trust a live recording that sounds dead quiet - all the life has been processed away!i'm picking up -60 to -70 in my own living room, and drops down to -50 when the fridge turns on.
my point was the low level sounds on the mahler 1st were not grainy and noisy on my system (in fact, they were highly detailed), so the fault does not lie with the recording or the format.
I'm talking about how the quality of these very low level sounds heard through a headphone.
The gain was cranked op about 40dB. (That is gain before line level/headphone amplification.)Another disc where you can listen for low level detail is the Sony SBM demonstration disc. (Redbook listen to the water and chimes demo with and without SBM)
The sound quality of the demo track with the gain cranked up sounds far more natural than the low level detail on the Mahler sacd.
it's very difficult to make a "grainy and noisy" recording sound clean and detailed.on the other hand, it's very easy to corrupt a good recording into sounding noisy and grainy.
so tell me again why you think it's "included in the format"?
well, you could say perhaps your ears are more "golden" than mine, but let's not go there, shall we?
I listened through a magnifying glass. :)Through most domestic systems you are not capable to hear that kind of low level detail.
I hope you're not falling for the sacd has superiour resolution marketing crap. Because it really isn't true in it's 64*fs incarnation.
.
Christine,What Frank is reporting, ie amplifying the quiet section by 40dB could be a very long shot for a lot of software and converters. I wonder about some kind of post processing needed (eg whether any electrical artifacts or DC offset appear in the process). I would also check for the results at various bit rates.
But still, it's an interesting experiment, especially if you can compare it with an equivalent PCM version of the same material.
I plan to try the same thing to see if I get the same results, hopefully I can find a quiet section in one of my titles (don't feel like buying a Mahler title just to do a test :)Best
Eric
to keep the ad converters noise and non linearity out of the equasion as much as possible.
So what did you do exactly? You set your player to play a specific section, amplify it by 40dB (or just below clipping) and record the output from your receiver or preamp into your soundcad?Are you not recording / measuring distortion from the receiver if you do that?
Yes, you are including all the analog noise and distortion too if you capture from the player.With the Lynx you should be able to capture low sound levels with a good microphone and get raw pcm audio directly.
I used a 24/96 external usb interface M-Audio I had on loan.
I'm thinking of buying a Motu 896HD.
When I do I will put together a DVD Audio sampler with various tracks as example for various technologies. (SBM HDCD DVDA CD sacd)I also found a way to demonstrate the effect of dither.
I downrezzed a PCM recording with a bit crusher plugin to the point where bit resolution is 4 bits. By mixing in white noise through another channel before it is bitcrushed you can easily demonstrate the effect of adding dither noise.
Hey Frank,No wonder you've been so quiet :)
If you publish your tests, you should post it on this board...
With the Lynx, I was thinking of capturing through balanced inputs directly from my preamp, why would a microphone be better? (don't have mic preamps, either).
About the Motu 896HD: it looks cool, with so many inputs, I guess you can do multichannel recording and all kinds of cool stuff... but I heard their technical service is not always reliable so be careful where you buy it.
BTW, did you double-check your USB IRQs? I had some problems with that issue, and it clearly affected CE Pro (Adobe Audition) recording and playing at high resolution rates. I found that reserving one IRQ to the recording device helps a lot.
Best
Eric
You can use a microphone to capture low level sounds to test the low level sound quality of your dac. You can record familiar sounds to check if it sounds natural if you record with low gain setting.If you capture the output of a sacd or DVD Audio player you need a device with sufficient gain.
I didn't experienced problems with the usb(2) device.
do you mean: taking a well recorded SACD and recording onto 96/24 PCM (thus negating any benefit of DSD in the process), complain about the ambience and use it as evidence of DSD's "shortcomings", then compare it with a studio PCM recording as evidence of PCM superiority, then complain about the ambience being "noisy and grainy"?no thanks :-) it's already been done once, and that's enough.
Hold your horses, Milady...I think the experiment is interesting, especially if it's conducted with several digital recording devices. Of course the experiment is not scientific, but I have no pre-conception at all about DSD and will not draw any cheap conclusion from the experiment.
In fact, the experiment will tell very little about DSD, only about PCM. Since we don't know what happens with equivalent PCM data, the fact that it's DSD is irrelevant, and in fact, all the experiment says so far is that 24/96 PCM sounds "grainy" when amplified by 40dB on specific hardware :)
It could very well be that someone else, with different equipment and settings, would not find the result to be "grainy", or that the equivalent PCM material will also sound "grainy". Or it may be that with specific equipment and settings neither DSD nor PCM data captured at 24/96 sounds grainy.
What's the point? Well, from a practical point of view, since PCM is the only option available to me for recording analog data in high resolution, I find it interesting, because it may impact the way I record my SACDs at 24/96.
My first (very unscientific :) attempts at recording SACDs at 24/96 yielded similar results, I thought the sound was better at 24/48... but I also noticed that I had better results when recording a mono signal (less data to process) so the problem was on the processing side. So I upgraded my Motherboard to benefit from a faster bus, bought new hard disks, and changed the IRQ settings to avoid any conflict with my soundcard, and the results have greatly improved. But I didn't go as far as Frank, amplifying some sections by 40dB to check the sound...
I'll try that over the weekend, hopefully it will rain :)
Best
Eric
Frank has an ax to grind and can always be counted on to find problems with SACD. He is like some of our Supreme Court justices in the U.S., before the case is presented, everyone already knows what their ruling will be. It is always more interesting to hear tests from someone who is objective.
You are a bad judge of character.And already down to the point where you need to snipe at a person instead of contributing to a discussion.
There are no DSD benefits...If you do this with 24bit 192kHz the only thing you are listening for is the low level sound quality from each format. As long as you observe a difference in this comparison the results are valid.
Frank
isn't this a bit of circular logic?you don't believe there are any DSD benefits.
to prove your point, you record DSD onto PCM. so that even if there were any benefits, they would be lost.
then you hear what you think is a problem.
therefore, it proves your hypothesis that there is no benefit to DSD.
So you amplify the most quiet moments to hear artifacts or defects in the DSD version?Have you made a similar comparison with a dual format title (Diana Krall, Marvin Gaye, Swing Live, etc) ?
Best
It's not easy to get a quite part on these discs.
Mahlers 1st is well suited for this.NB it peaked at -6dB halfway through the first part. I adjusted the recording level just below digital clip level on that peak to get the best possible dynamic range.
Frank
I'll try to find some quiet or "silent" moments in my titles to see if I get the same thing you mention.Will post the results
Best
NonA,To be fair, most of the PCM A/Ds and D/As are multi-bit Delta-Sigma convertors, and as such their noise does rise as a function of frequency, however it isn't as dramatic as the rise with DSD.
nt
When was that?
Duilawyer is correct - NonA can't see his own error.
Regards,
Geoff
thought it was obvious...thx.
even though delta sigma has been around for a while, and DSD archiving also from Sony...only when they figured that they needed a replacement, did they use a Direct Stream Dividend scheme for customer playback :).i just did a quick search on Goggle on "DSD" this is an interesting bit i got from audioholic website where it came from:
"....Let's take a look at what would happen if we decided to work in DSD format tomorrow:
25 years of knowledge of pcm---dump it
High end pcm workstation---dump it
800 GB of hard drive---don't dump it but buy 800 more
Expensive multiple A/D and D/A's --dump 'em
New digital studio wiring---rip it open again and add 4 times as much
20 years of engineering thinking about new digital microphones---forget it---dump that too
Expensive outboard gear--wrong format---dump it and re-quip (with what?)
Review fax from major labels that says just send us your pcm work and we'll convert it to DSD. ---Weep!It's a total joke and we refuse to let the dullards in hi end hi-fi drive our investment. Those who push us to do this re-equipping will not guarantee our bank loan will they? This is even before one reviews the research of Lipshitz and Vanderkooy into the drawbacks of DSD.
Sony didn't tell you that you could record DSD on a regular PCM multi-track did they? No, they said we have some new gear to show and install for you.
We have an increasing load of work for DVD-A. Now someone tell me (a tough customer who has done the listening) why we should change to DSD.
Ealing mobile recording, Ltd., Chicago
Note: The above comments show us that in order for a studio to fully produce SACD discs they will need to fully re-equip their studios with a lot of new hardware and software. This is a very expensive proposition. Many recording studios are in the middle of an economic recession, and with the possibility that the SACD format will have limited distribution, many studios have serious reservations about buying into the SACD format....
"
Geez, I'm glad my daughter doesn't come here... I can't believe the math level on this boardNo wonder we don't get more high-resolution releases, the publishers must be thinking we can't read big numbers...
("Joe, don't bother with the 24/192 version, I just read the DVD-Audiobahn, those guys can't even count their fingers" )
nt
.
nt
CD rights EXPIRED.
. . . i.e. the expiration of Sony's CD patents around the time of the new millennium WAS a factor which strongly influenced the timescale of SACD's introduction (give or take a couple of years).It’s rather like the handover phase of the baton in a 400 metre relay sprint race (i.e. there is no sudden discrete changeover since the two runners are running together for about 10-15 metres on the racetrack.).
Ironically, when we apply that analogy to SACD, the only thing we can see happening now, is that after some four years already ( !! ), it still doesn’t look like Redbook CD is going to hand its baton over to SACD, however hard the latter seems to be trying to grab it! :-)
Your point is different, "why SACD was "invented".
I am not denying that.I am disputing the 20 year addition. Simple. It is a math problem. Sony's patent ran out of 2000, 20 years later is 2020.
Microsoft is running with the Intel Coach, Toshiba et al. promess an easier passing of the "baton", Sony/Philips are using stereoids but cant afford to get caught again, the file sharer are undercutting the whole scene and breaking race rules, gear manufacturer are sponsoring the race along with the music and movie industry...we, niche audiophiles, cheer left and right
please share, best.PS: CDs came out at the begining of the 80s...patent for CD lasts 20 yrs (w/o further modification to keep it going)..so 1980+20= yr 2000 (end of CD licensing rights). SACD came out commercially in 1999 or 2000 not sure...now see above...am i missing something??
Do the math.
my smileys usually mean "kidding around"...thanks for pointing out the discrepancy in that sentence...I thought what i meant to say was obvious (1980, 20 yrs, CD license expiring, etc.)...I didnt think i needed to correct explicitly.btw, it seems *I* got the good stuff...and im sharing too -> :-)...how did you like the Eric Clapton "on the road" DVD?
I will get it, but there are strange things happening at my local tower, from fully DVD stocked, to lot of empty bins. Thanks for the reminder.
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