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anyone recording at 192 on the regular? Dynamics Plugins
Old 26th February 2013
  #31
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Quote:
Originally Posted by Daniel Comport View Post
Hi famous yard,

It really isn't about Nyquist, in digital, it's about intermodulation distortion - forget rates and think about harmonics - Why 192?

plain and simple - it has won in blind tests over and over again -

prime example is an arpeggiated synth with fluctuating filter and freq offset from tempo - fullness and clarity is essential.
Can you post a link to those blind tests?
Thanks in advance.
Old 26th February 2013
  #32
Quote:
Originally Posted by Meltias View Post
48 all the time. Fact is no band or consumer or producer i have worked with has ever been able to tell the difference once its output at 44.1 for CD or as an mp3. And no film producers i have worked with have ever been able to tell the deifference between 48 and 96. The only time I use 44.1 is when I am doing dance tunes and need to go get sweeps and other fx from sample libraries that seem to usually be at 44.1. In the end it is not what WE hear, its about what the client, producer and consumer hears and the vast majority of the time they never hear the difference. Higher than 48 just causes huge file sizes and processor load. And actually more important to me is 32 bit over 24 bit cos i do notice a difference in both dynamic range and granularity.

Sent from my PG06100
Do you record at 32 bit? What program do you use?
Old 26th February 2013
  #33
Lives for gear
i find the whole resolution debate pretty funny. as i do the itb versus analogue debate.

but anyways.

i can clearly remember not that long ago. maybe 12 months ago, people laughing at those using higher sample rates. tabling science that shows people can't hear above 20khz, referencing shootouts from professionals that clearly provide evidence that no-one can hear a difference between 41khz and 96khz. even dan lavry got miss quoted and that was used as evidence. there are pages upon pages online that prove catagorically that 44/1 khz was enough.

then somone, dunno who, maybe god himself actually realised that the science wasn;t being accuratley portrayed or maybe there is something else going on that isn't being measured. but regardless they did some more listening and low and behold a eureka moment occurred when they realised that actually, 96khz sounds a **** load better. most of those experts that staked their name on the 44.1 culture are pretty much ignored these days or gone quiet. even mr lavry himself said, umm... you know what guys. actually 60khz is where it's at.

wow, now everyone is trending to 96khz. so those that have been using higher sample rates for a long time just stay quiet until occasionally someone asks. "so what sample rate do you use" then it starts all over again.

all of the experts table all of the data that proves beyond a doubt that there is no reason to now go beyond 96khz. all of the people using 192khz sit back watching. thinking, wow i've heard all of this before while they eye off technology that will get them even higher sample rates so their music can sound better agian.
Old 26th February 2013
  #34
Lives for gear
but still people go on about how higher sample rates are not needed. but mr massenburg and mr neve possibly think differently.

so does the chief tech at philips. he seems to have some vision as to higher sample rates.

DSD Vs PCM from the Head Engineer at Phillips in 2005 taken from an online article

(Thanks to Mr. Rich Mays for this Article)


And, yes, he is still the Chief Engineer at Philips Digital Systems Labs. He was very forthcoming and I hereby thank him for his time.


TH:
Studio people who have compared the live misc feed to DSD and PCM say that DSD is much better and they cannot tell the two apart. If, as you say, it is flawed, why are the studio engineers all for it. As someone on the forum said, it is analog without all the problems.

BP:
1) Sound of DSD:

The HF noise and the low-level nonlinearities of DSD do not get in the way of the sound as we hear it. However, it is impossible to build a production chain using the format.

DSD is in its place in one (1) application: Mastering from analogue. When the recording chain is completely analogue, you can feed the audio from the analogue mastering into a DSD A/D converter and cut that signal straight onto a disc without any further processing. It is in this application that DSD can be viewed as pretty transparent. When you convert the signal twice, however (such as when using a DSD recorder as the tracking medium), the second conversion is no longer transparent, due to the HF noise present in the source signal hitting a second analogue deltasigma modulator.

In this I have a serious gripe with AudioQuest. Before DSD, they tracked onto a 15ips 1/2" two track. These tapes were first used to master to ordinary CD, later again to SACD for a reissue, which indeed was a sonic forward step. However, they then switched to using a DSD recorder for tracking as well. Grundman then mastered it as usual (ie through an analogue mastering studio) onto the DSD master recorder.
The recording guys may well have found the DSD recorder better than their analogue two-track in a dry shoot-out, so you can't blame them for having made that choice. The resulting SACD releases however, are below anything. What's worse, there's no way of salvaging the recordings for a better-sounding reissue (unless they had the analogue two-track running as a backup). As an example, get hold of the classic "BluesQuest" sacd, which was made from analogue tapes. Then compare Doug McLeod's "Whose truth whose lies", where a DSD machine was used. If you're a bit of a sensitive person you'll run away screaming.

I think we could say that DSD is analog with a few extra problems. Serious ones.

2) Total transparency:

Apart from that I consider any claim of "not being able to discern between a live feed and DSD" as something of a hyperbole. While flicking a switch during the music will indeed not reveal any serious deficiencies, more controlled listening (e.g. listen - rewind - switch - listen) will certainly get you hearing a lot more. I can't even insert a unity gain stage (low-noise low-distortion etc) in the signal path without hearing degradation, let alone a pair of converters.

3) Jitter:

When a signal is fed straight from an ADC into a DAC, they share the clock. When you sample a signal with some jitter, and you reproduce it with the *same* jitter, the jitter has NO impact on the sound (jitter which impacts the sound most, incidentally is of the LF kind, which does not affect noise specs, and which is typically not attenuated by PLLs). With DSD this works - the delay between ADC and DAC is nearly zero. With PCM it doesn't, because there is a delay of several milliseconds between the two, meaning sampling and reconstruction see different jitter. This means that a live vs DSD will always sound more transparent than when you take the digital signal to tape and replay it.

4) PCM implementation issues:

On Nanophon - Home page you can read a number of the late Julian Dunn's excellent papers on how compromised implementation of digital filters account for many of the deficiencies noted with PCM. These can be readily solved with due care for details and at the expense of only mildly increased computational burden.


TH:
How do you feel about the DSD workstations? The existing ones, I believe, are all using DSD-wide, or PCM-narrow.

BP:
The workstation from Merging is in itself OK in that the data is kept in 352.8kHz/32bit floating until it is written to an sacd master file. Currently the incoming data is DSD, causing a "two DSD conversions" problem (although apparently in the digital domain the sonic degradation is less than when it happens in the analogue domain). If you can somehow get the data in through a non-DSD converter (preferably at 352.8kHz/24bit or so), it would become a digital equivalent of the "ideal analogue chain" in that only one deltasigma coder is in the signal path (when writing the master). That would be fine.

If you do have to edit a recording which is already in DSD, it can be done in a sonically transparent way by using the "transport" mode of the workstation. In this mode, the DSD data is preserved exactly except during the edits (crossfades). The two conversions problem would be restricted to those edits only. Of course, any form of processing, such as level change, EQ or mixing, is not possible in this mode. It's really just cutting and splicing.

The workstations from the Sony camp (I don't know names) use 2.8224MHz, 8 bits, still noise shaped. This still has the same problem as reconverting to DSD every time, but reduced by 48dB. Still I wouldn't recommend it. The Merging approach is the most suitable one.


TH:
You mention the use of a non-DSD converter (preferably at 352.8kHz/24bit or so). EE Dan Lavry argues that the bigger the numbers the less accurate the conversion is. He says that there really is no need to go beyond 24/96 (48k bandwidth) and anything more will not result in a more accurate signal but rather more noise and distortion.

BP:
Dan is correct in that as the sampling frequency is increased, the available signal to noise ratio inside the nyquist band decreases. However, when you keep the bandwidth across which you are measuring SNR constant (e.g. you measure noise across 20kHz) and when noise shaping is used, this trend is often reversed. DSD is precisely a case in point. I have one converter here (homegrown discrete circuit) that puts out 1 bit at 2.8224MHz. Measured across its nyquist bandwidth (1.4112MHz), its SNR is useless, well below 6dB. However, taken across 20kHz, it delivers the full 120dB. When you push the sampling rate too high, performance will again resume a downward trend.

Dan's converters are multibit, non-noise shaping. Such converters will not even hold their precision at constant bandwidth when sampling rate is increased. Since his converters have ultra-low noise as their hallmark, he's quite right to maintain a reasonable sampling rate.

When I propose to use a 352.8kHz converter, in practice that would be a noise shaped converter designed to offer maximum SNR performance over as wide as possible a band, but not up to 176.4kHz (not feasible). It would still have a "noise shaping tail" in the nyquist band, but much less so than DSD. At the final DSD conversion stage, the noise of the ADC would be negligible compared to the DSD noise, so the output spectrum would be pretty much the same as that of an analog signal converted to DSD in one go.

A design which is in the work at home uses 64fs, 16level PWM with 7th order noise shaping. This would offer up to 129dB SNR (limited by amplifier noise) over 80kHz, which is 4 times as wide as DSD. This would insure all the flexibility of PCM while a later conversion to DSD is not compromised.


TH:
How do you feel about DSD as an archiving format? Is the transparency good enough for Sony's precious analog master tapes? And what makes DSD ideal for archiving as opposed to PCM? Or does it matter? I read Neil Young chose hi-res PCM format for his masters.

BP:
When sony came up with DSD as an archiving system there was hardly even 96kHz PCM around. If at that time they had some old tapes to archive before they fell apart, DSD was the best available. However, since DSD is a liability in terms of processability, archiving to DSD now is no longer a good idea and use of 192/24 is warranted instead. Since SACD is probably here to stay we should view DSD as strictly a release format, in the same way as we didn't produce on vinyl, but music was brought to the home on it.


TH:
What do you say when you hear audiophiles make comments that DSD has all the "air" and "smoothness" of analog?

BP:
In my own experience, high speed PCM also produces the air DSD has, while the "digital glare" of some PCM can even be solved at low speeds. It is caused by the narrow alias-band that is present between 20 and 24.1kHz. Removing this band prior to playback restores naturalness and focus.

I don't suspect DSD of any "euphonic" effects, although, who knows, the HF noise?

Admittedly, the DSD camp has been able to mobilise more audiophile designers (folks like Ed Meitner), resulting in the analog circuitry of the best DSD converters sounding better than that of most available PCM converters. Doing a straight shoot-out is actually quite challenging technically, as the two formats normally use different converters, necessarily producing a different sound. This is another reason for me to do this 352.8kHz converter, because its output can be converted to either DSD or 192/24 while compromising the performance of neither. This would finally allow a direct comparison.


Gardo:
Are the production chain problems insoluble, or not yet solved?

BP:
The "production chain problem" ie. the fact that signal quality quickly and irreversibly deteriorates as it passes through subsequent processing stages, is in itself not solvable. The root cause lies in the high HF noise level. This noise is indistinguishable from the wanted signal, so it cannot be stopped from accumulating every time a signal is converted to 1-bit. This is not to say there are no workarounds. However, the mere fact that such workarounds are necessary shows that the DSD format itself was misconceived.
1. "Production DSD". To use a 1-bit signal at 128fs or 256fs instead of 64fs. Especially in the case of 256fs, signal and noise no longer overlap. The signal bandwidth is specified, as before, at 70...80kHz. The quantisation noise only comes out of the analogue noise floor at 80kHz as opposed to 20kHz in the case of 64fs. Now, signal and noise can be separated, quite simply by filtering. It follows that every processing step still involves filtering to remove the HF noise from previous conversions and deltasigma modulation to recode the processed signal into 1-bit. This constitutes a considerable processing overhead.
2. PCM-narrow. To use 352.8kHz, 32bit floating point as the intermediate format. This is PCM by all means, but according to the listening tests carried out by Philips and Merging, the decimation and upsampling filters needed to convert to and from 2.8224MHz do produce audible artefacts. The advantage here is that this format is practical enough to maintain throughout production, store on hard disk etc. Converting from DSD to PCM-narrow is done only once, namely as the data comes into thr production chain. Conversion to DSD is only done once, namely at mastering. Moreover, the 352.9kHz data can be derived straight from a better-than-DSD AD converter (which means most of all present day converters), burdening the production chain with no or greatly reduced HF noise.
3. DSD-wide. 2.8224MHz, 8 bit. This format still requires all processed data to be noise-shaped back, but to 8 bits instead of 1. This can be done many times before noticeable headroom reduction or other adverse effects set in.


Gardo:
If the problems with PCM are easily solved, why haven't they been? That's a real question. If I understand DP, he's saying that removing aliasing effects in the 20khz-24.1khz band takes care of digital "glare." How would this be accomplished? Better brick-wall filters? Aren't those aggressive digital filters part of the problem?

BP:
In order to "solve" the PCM problems, it is required that a certain amount of care is given to all steps in the production chain. It is currently no problem to build a set of AD/DA converters that avoid all of the pitfalls in PCM. However, until everyone takes the same actions to get their digital filters right, these issues can reappear at any time.
Here's a concise list of the problems:
1. Half-band filtering: The upsampling (interpolation) and downsampling (decimation) filters are often specified such that the frequency response is symmetric around fsh/4;0.5 (on a linear amplitude and frequency scale. fsh=2fs). The effect of this symmetry is that every second coefficient is 0, except for the middle one, which is unity. This halves the number of computations necessary to implement a filter of the same steepness. However, the symmetry means that the response at 0.5fs is -6dB, a far cry from the total rejection called for by the nyquist theorem. Most if not all of PCM implementations actually do not conform to the nyquist criterium.
Now here's some funny psychology. Looking at the data sheets of converters you'll find that the passband goes to 0.4535fs, so the stopband starts at (1-passband) ie. 0.5465fs. At 44.1kHz, this translates into twentythousand point zero Hz. Apparently the belief must have been that as long as everything is well until 20.0k, all is cool and nobody will care about aliasing between 20.0k and 24.1k.
2. Equiripple filtering: These days chips are fast enough to run longer filters, so the half-band filter has gone somewhat out of fashion. One thing in which half-band filters are seen to be something of an overkill is in passband ripple. Due to the symmetry, a HB filter that has -100dB stop-band rejection has +/-0.0001 dB flatness in the passband. Not requiring a halfband characteristic but a looser passband spec (such as 0.01dB) will make the filter shorter in terms of time. All coefficients are now nonzero, but the total number of coefficients can be shrunk considerably, resulting in a reduction of group delay. Group delay is a serious problem in multitracking where a musician needs to hear herself play in the mix while tracking.
The result is that the filters are now commonly spec'ed as 0.01dB (or worse) flat until 20.000kHz (re 44.1kHz), stopband from 24.100k, because that was the accepted practice till now, no? Requiring a nonaliasing response would either mean shrinking back the passband to about 18kHz (a commercial no-no) or increasing the filter length (where the exercise was all about shortening it). The alias band thus remains status quo at 0.4535fs to 0.5465fs.
Why am I making such an issue of this flatness affair? Well, this periodic ripple corresponds to two small secondary spikes (echos) in the impulse response. One at the start, one at the end. For a filter that's flat to +/-0.01dB, these spikes are -66dB, and are a serious threat to stereo imaging and produce pretty obvious time smear (well, obvious to us perfectionists - aren't we a flea in the fur of the industry?)
3. Sharp filtering
A sharp filter has the effect of psychoacoustically enhancing the corner frequency, resulting in a kind of unnatural brightness or roughness. This effect is to my knowledge the most innocuous of effects associated with PCM. Also, making the filter steeper from what it is to begin with has no additional "brightening" effect.

How to attack the problems:
1. The aliasing problem can be solved at once, anywhere in the audio chain, using a single lowpass filter that enters stop-band before the alias band ie before 0.4535fs. A good place to do this is at reproduction or before final dithering. This means that halfband filters as in 1 may be used throughout without deleterious effects. I find that running a CD through an ultrasteep filter (pb to 18.5kHz, sb from 20kHz, eliminating all aliases that were created anywhere in production) results in an improvement in contrast, depth and precision of the stereo image.
2. The echos due to equiripple filters cannot be removed except by employing surgical precision on a case-by-case basis (like I'm doing to solve this problem on a TI SRC4192 - putting a DSP before and after the chip to compensate for the echos in the int and dec filters. In short, equiripple filters must be avoided at all cost. Simply specifying a higher sampling rate and a looser filter has the same effect of shortening group delay, but this implies that chips specified in this way are no longer suitable for the low sampling rates of 44.1k and 48k.
3. As we're already increasing sampling rate in view of the group delay problem, we can push it further to obtain a more natural roll-off after 20kHz.

Recipe for perfect PCM:
Specify a sample rate well above twice the audio band, e.g. 192kHz (lower is arguably acceptable too but since we're slowly standardising on 192 who cares)
Specify all interpolation/decimation filters as halfband, 0.4fs to 0.6fs transition.
Put exactly one non-halfband lowpass filter in the chain (e.g. at replay or before final dithering) that enters stopband at 0.4fs. Specifying its passband at 20kHz will allow for a very smooth roll-off and hence very short and practically ringing-free impulse response.

A more detailed paper concerning the aliasing and equiripple problems can be found at Nanophon - Home page (read them all - excellent stuff).
A description of a method to calculate the "final lowpass filter" is AES preprint 5822, by Peter Craven.


Gardo:
Every recording medium has fundamental flaws. CD tried to address some of the fundamental flaws of vinyl and introduced flaws of its own. I'm sure this will be true no matter how much net improvement is realized in recording and playback. The question for me is whether SACD (generally speaking) sounds better than CD (generally speaking). On my modest system, it certainly does.

BP:
On my less than modest system it does too. I'm only saying that of all the possible solutions that could be formulated for the deficiencies of CD, DSD is the least intelligent one.


Gardo:
BP says that "SACD is here to stay." Do we think that's a step forward or a step back?

BP:
Sonically it's a step forward. In terms of practical usefulness it's more like putting the world's population on a spaceship to colonise a new planet without first checking if there's water and oxygen on it, and if the conditions on earth were really so bad we needed to leave it.


Gardo:
If I'm following your recommendations correctly, it seems that the second sentence should end "do not produce audible artefacts." Is that right?

BP:
Indeed: "do not". I've made the same typo in a different discussion not so long ago and it made things a bit confusing.


Gardo:
Also, I'm gathering from your remarks that you do not believe there is any reason to have a passband greater than 20khz, so I take it you do not agree with those who say there's extra information up there that we need to be able to reproduce. Is that correct?

BP:
I was a bit unclear there...
The passband only needs to be flat up to 20kHz, but it shouldn't drop off after that - there's quite a lot of sonically relevant information above 20k. In the case of the 192kHz system I proposed, the stopband (-80dB or better) would start at about 77kHz. The -3dB point would lie at 36kHz, -6dB at 43kHz, -10 at 49kHz. This means a lot gets through and it's a nearly perfect compromise between impulse response and bandwidth.
Also keep in mind that this filter should be applied only once over the entire signal chain. If it's done in the customer's player, he can even chose between this and a flatter (but "ringier") response. My hunch is that the slow rolloff option is the most sonically transparent.


I have also included a couple of other questions that some of you might find of interest:

TH:
With regards to SACD playback, I notice that some multiformat players convert DSD to PCM before output. Does this conversion degrade the SACD sound in any audible way?

BP:
The conversion to PCM in these players is done by chips by NPC (2 versions available). They can downsample to 8, 4, 2 and even 1fs. How badly the sound is affected depends very much on which chip/setting is used. Someone here has ordered samples so we can measure the filters. The spec says nothing about them. A lot depends on how well the filters are implemented (see last mail). Probably the 8fs version is relatively innocuous. Funnily enough this is done in order to allow the DSD signal to be reproduced by Burr-Brown multibit DAC chips.

SACD players exist in an enormous variety, up to and including devices (usually the cheapest ones) that downsample to 44.1kHz. Also their DVD signals are reduced to 48kHz first. This is why indeed you have to be careful about buying a cheaper player.

The audiophile brands each (or at least many of them) have their own converter philosophy, which warrants attention to detail but not necessarily knowledgeability :-)

I personally use a first generation SACD player (Marantz SA-1) which happened to be lying around here. I modified it to deliver the DSD at three BNCs at the back and have my own DAC to convert it to analogue. The same DAC will take 192/24 (but it does not yet have all the fancy filters present - next version) so by the time I can knock off a DVDA player somewhere I can use that too.


TH:
You seem to be an audiophile as well as an engineer. What is you opinion of vinyl as a delivery format for analog master tapes? The re-issuing labels like the late DCC, Classic Records, Sundance, etc. seems to put out some really good sounding releases that, unlike the old days, do not seem to be compromised (compressed or limited) in any way?

BP:
The good thing about current vinyl is that you know the customers will have good styluses and arms. The compression in early days was necessary to prevent some styli from jumping out of the groove. The first to set such qualms aside was Telarc in the early 80s. Indeed the record carried a warning that some turntables would not play it properly.
I'm not a vinyl fan, mostly because the pinch distortion puts me off. On the other hand, I am aware that better alignment can reduce that problem to an acceptable point.

The ideal release medium is transparent and neutral. I find that good vinyl can offer very good transparency but not neutrality. Depending on their tastes, audiophiles will put up with a deficiency on one aspect or the other. Some will like it for the transparency and will hardly notice the colouration. Others will just be put off by the colouration and not notice it is offering something regular CDs aren't.

In terms of sound quality I believe both SACD and DVDA to be good formats for releasing remasters on. As said before, my problem with SACD lies in the practical side of affairs.
Old 26th February 2013
  #35
At the current point in time, computers aren't powerful enough to work at 192k on a reasonably sized project, unless you have very small plugin requirements or track counts.

That's why no-one works at 192k. For me, the minute (if any) sonic improvement would be offset by an operational ballache. No thanks! I'll work at 96k for smaller projects, otherwise 44.1/48k it is.
Old 26th February 2013
  #36
Quote:
Originally Posted by gouge View Post
i find the whole resolution debate pretty funny. as i do the itb versus analogue debate.

but anyways.

i can clearly remember not that long ago. maybe 12 months ago, people laughing at those using higher sample rates. tabling science that shows people can't hear above 20khz, referencing shootouts from professionals that clearly provide evidence that no-one can hear a difference between 41khz and 96khz. even dan lavry got miss quoted and that was used as evidence. there are pages upon pages online that prove catagorically that 44/1 khz was enough.

then somone, dunno who, maybe god himself actually realised that the science wasn;t being accuratley portrayed or maybe there is something else going on that isn't being measured. but regardless they did some more listening and low and behold a eureka moment occurred when they realised that actually, 96khz sounds a **** load better. most of those experts that staked their name on the 44.1 culture are pretty much ignored these days or gone quiet. even mr lavry himself said, umm... you know what guys. actually 60khz is where it's at.

wow, now everyone is trending to 96khz. so those that have been using higher sample rates for a long time just stay quiet until occasionally someone asks. "so what sample rate do you use" then it starts all over again.

all of the experts table all of the data that proves beyond a doubt that there is no reason to now go beyond 96khz. all of the people using 192khz sit back watching. thinking, wow i've heard all of this before while they eye off technology that will get them even higher sample rates so their music can sound better agian.
You need to actually read why Dan Lavry thinks 60kHz is about right. This is to allow enough leeway for the filters on the converters to not affect the audible high end. He says 192 actually sounds worse because of the misunderstanding about how digital audio works. It's not a case of "higher is better".

It's a very good read.
Old 26th February 2013
  #37
Lives for gear
yep i've read what dan said.
Old 26th February 2013
  #38
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Arksun's Avatar
Quote:
Originally Posted by gouge View Post

3) Jitter:

When a signal is fed straight from an ADC into a DAC, they share the clock. When you sample a signal with some jitter, and you reproduce it with the *same* jitter, the jitter has NO impact on the sound (jitter which impacts the sound most, incidentally is of the LF kind, which does not affect noise specs, and which is typically not attenuated by PLLs). With DSD this works - the delay between ADC and DAC is nearly zero. With PCM it doesn't, because there is a delay of several milliseconds between the two, meaning sampling and reconstruction see different jitter. This means that a live vs DSD will always sound more transparent than when you take the digital signal to tape and replay it.
This part of the interview I don't quite get. He's arguing a jitter advantage, but citing the example as ADC to DAC which is real time. But as I understand it jitter behaviour is random. So if you record to DSD and playback later, any jitter from the clock will be randomly different to how it was when recording, thus will not be exactly the same. Better then to have as stable and low jitter a clock as possible in the design. If its the ultimate in real time monitoring, a simple cable would be the best option.
Old 26th February 2013
  #39
Lives for gear
I record at 192 just to make my dog happy! He barks whenever i record at anything lower!
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