Griffinator, et al who participated in the "Harsh CD" Thread...

[Deang]

I don't want to hear anything better -- I have to keep my house.

[Mallette]

My welcome to Lynn as well. It is good to hear almost precisely my own experience from independent ears.

Dean: Let's see. LP's=free to a few bucks for most of them. CD's, 10-20.00 for most of them. Give it some thought...

From the excellent input of Griff and others, the mystery has only deepened for me as to just WHY the higher sample rates sound better. However, the fact that they do, and it is significant, has only been further supported. In my mind at the moment, the most likely candidate is simply more "dots" in the picture and the unknown depths of the human brain that perhaps are able to percieve more than we think. I am less certain about ultrasonics and the other issues. The impact of jitter is still debated in audio circles.

The good news is that, unlike 30 ips tape and SACD, very high sample rates are available to us way down the food chain.

Dave

[Glimmertwin]

this has been a very interesting and instructive thread ...thank you!

It may have answered one of my lingering questions: why do some CD's sound so good to me, while others sound really bad?

[Mallette]

Glimmertwin: Rather than individual CD's, are there catagories that, in general, sound better to you than others?

Dave

[Marvel]

Dave,

I'll answer that one, at least from my perspective.

I have some classical recordings that sound very good to me. One is a '93 RCA Red Label DDD at Abbey Road, of a Schuuman Symphony.

A Berlioz work, that I don't remember the label (I'm at work).

Maura O'Connell -- Celtic/Nashville, w/Bela Fleck, Mark O'Connor, Jerry Douglas -- lots of acoustic instruments as well as electric.

Of course I have a cheap CD player, JVC of some vintage, so the source electronics could be masking the CD a lot. I admit it.

Moving from the JBL 4311 pair to the Heresy IIs seems to have made an improvement, but mostly in dynamics. I haven't hooked the JBLs back up to make another comparison.

I don't really listen to much real rock music. The "it" I had mentioned in my earlier post, was acoustic bass, acoustic guitar, sax, trumpet, percussion. The ProTools mix was good. I could send you the CD if you would want to hear it.

Marvel

[Griffinator]

I have no doubt in my mind that there's a genuine improvement in sound quality going from Redbook to 24/192 - but the rest I'm a little iffy about.

Lynn - you mentioned that they SRC'ed from 24/192 down to the other formats? To me, this is already a problem. SRC is not transparent on any system short of the absolute top-shelf stuff (and no, Apogee is not top-shelf)

I had a chat with George Massenburg a while back about this. He and some friends of his did some blind testing with 24/96 source SRC'ed with $20,000 converters down to 24/48 and none of them could tell the difference. That said, the same converters tracking at 24/48 or 24/96 did show a slight difference in overall tonal quality.

SRC is not the simple math we all assume it to be. It's a lot more complex than "removing every other sample" to cut samplerate in half - there's all kinds of algorhithms available to make this process work, and you can rest assured that the very best do not incorporate anything resembling "removing every other sample" methodology.

[Griffinator]

One more thing - I should correct a statement I made earlier.

The general consensus amongst engineers is that 24/48 is a sufficient recording format for CD production - all the potential errors created in the studio by effect tails, fades, etc. at this bit depth and samplerate will not be audible in the final product.

Of course, CD being the inadequate product that it is, recording 24/96 and finalizing to a 24/48 product would be the preferable choice.

The key for the engineer is, how realistic can I make the final product given the parameters I have to work with. Until the record companies start demanding DVD-A or SACD instead of CD, we have to work in CD format.

Oh - and in response, Marvel, the dBx 376 channel strips are equipped with digital audio outs. If I chose, I could go straight S/PDIF out of the 376's into my DAW at 24/96. I just don't see the point of such an exercise when the final product I'm turning out is 16/44.1.

[Mallette]

I want to know more. My experience in downsampling doesn't jive so far. My downsamples of live material still sound better than all but a handful of commercial CD's, and I assume they are using these big bucks machines.

Why shouldn't you simply truncate a 24 to 16?

And why shouldn't you throw away every other bit going from even divisions? Simple math (the kind I understand) suggests you are doing exactly what you would expect in halving resolution.

Even Sound Forge has a "dither off" mode.

I'd also comment that it is difficult to understand why big bucks would yield any improvement in an operation that is all computer.

OTOH, I am here to learn.

Dave

[LynnOlson]

Hi folks, thanks for the great discussion!!! Lots of good input from folks who do this for a living, people who ought to know.

The 2001 VSAC demo was a long way from a scientific double-blind test, but I've heard Keith Johnson's classical recordings at the CES before and this demo considerably exceeded what I heard then. The recording and playback gear was pro-grade stuff from Ed Meitner and Keith Johnson, actual data sources were removable hard disks, and Johnson was playing recordings of Eastern European symphony orchestras that he made for reference purposes (although some are commercially available as downmixed HDCD recordings).

Although the sound was very good - certainly a lot better than the dreadful sound you usually hear at the CES, or worse, the S'Pile shows - it was not the best sound I've ever heard. That was visiting the BBC in 1975 and hearing a discrete quadraphonic recording of Beethoven's 9th Symphony, and listening to a first-generation mastertape made on a customized Studer running at 30IPS with no Dolby processing. Speakers were professional BBC monitors with internal bi-amplification and the microphone was a Clarec Soundfield prototype. I have never before or since heard any recording with the startling sense of realism and audience presence that one had - at one point, the audience got excited and started clapping along with the music, and you could hear every handclap with crystal clarity, in every direction (this was 360 degree quadraphonic mastertape), along with the 100-strong chorus going full shout - all with NO distortion whatever. Usually a dense piece like the 9th is just a roar of hash and distortion at the climaxes - not this recording. It certainly would be fantastic if the BBC ever released this on multichannel SACR or DVD-A - it would be hands down the best recording of the 9th ever made.

To descend from the sublime to the geekiest, there's bunch of MLSSA graphs and stuff over at the Odds and Mods forum.

[LynnOlson]

"Lynn - you mentioned that they SRC'ed from 24/192 down to the other formats? To me, this is already a problem. SRC is not transparent on any system short of the absolute top-shelf stuff (and no, Apogee is not top-shelf)"

In all honesty, I don't know what Meitner et all were using for SRC - although K. Johnson's gear would be a safe guess. I'm not a recording guru, so I don't know the status of Meitner/KJohnson stuff in the pro world. Might be good, might be crummy, dunno.

I have to confess I'm mostly an analog guy, but a lot of that is sour grapes from the early Eighties, when the recording industry made the rapid switch from analog to bare-bones 44.1/16 without dithering and the worst A/D converters imaginable. I still think of the Eighties as a lost decade, when recorded sound quality actually went backwards for the first time since the invention of the phonograph.

It took PCM a long long time to get up to 15IPS mastertape quality, something like 15 to 20 years realistically. I'm old enough to remember the conversion from mono to stereo - that was a LOT more effort all around, and took less than 5 years, and involved media as diverse as pre-recorded tapes, movie-sound, LP's, and FM radio! So the "hard-to-do" excuse doesn't cut a lot of ice with me. Stereo sound was a MUCH more difficult and costly conversion process, and happened three times faster.

Here we are, a full twenty years after the introduction of the CD, and high-rez finally equals the quality of an all-analog mastertape made in 1975. 20 years of effort - for what? So people can walk around with MP3 players that sound just a little better than 1970's cassettes?

[Marvel]

Here's a question for all, and since it has been mentioned here before, I'll ask again. I have an almost mint LP of Dire Straits Brothers in Arms album. It sounds simply fantastic to me, and although it is a rock album, has acoustic instruments on it. But it was done in '82 on a Sony digital multitrack. Anyone else have the LP and care to comment?

Griff,

There's the rub. We still have to do the best we can with the tools and final release media we can. Maybe record the highest quality (24/192 or higher) we can, downsample and relese it. When better playback systems are here, re-release the product. That is what all the digital remastering craze is all about, but we know that the product isn't necessarily better -- just more convenient.

I have heard of many technical folks who say that although there can be differences, today it is not hard to design a good software SRC. That could be used as an excuse to accept what you have, or it could be true. Programming is over my head, there's too much math. I have worked with engineers (EE), who would sit and stare for hours at a whiteboard full of calculus, and finally go "Oh, this needs to be..." reach up and change something. I appreciate the hard work that goes into it. Nowadays, there is no need for totally hardware based SRCs. It's far better to have it in software, because you can always update with ease (relative to updating hardware). Our company had hardware products also, and now they collect dust because it was too hard to update.

I agree with Dave that more money doesn't always mean better quality.

Marvel

[Mallette]

A week from today we should have some input on the subject. I will be hosting the DFW Hornheads to listen to the live recording. It's a great group of diverse ears of all tastes and ages. I don't plan anything overly scientific, though I'll make some attempt at trying to keep an even playing field.

Also, great mention of the London quad recording. My own faith is in four mics, 360 coverage for the majority of acoustic music events. The center and subs make no aesthetic sense to me except for a very few subjects. Before I got into surround, one of my mottos was "Why only two mics? Well, I've only got two ears!"

Given all the discussion, I feel my location successes are probably a combination of simplicity and my mic plans. Knowing from the beginning I didn't have "big boy" toys to work with, I resolved to attempt to maintain the most direct signal path possible. No mixers or anything not absolutely necessary. I've labored to produce the shortest source to mic distance to the point of studying the pickup patterns and sighting the angles.

Beyong that, as Marvel said, I have a lot of faith in software processes. I believe good software is always better than another signal path. The simple revelation that click and pop reduction via software does not nor cannot impact anything but the target noise made a believe out of me. Even having worked with computers since 1979, I am basically of the analog generation and such "nuances" tend to get lost there.

The age of digital excellence is probably being born now. As we gain control of these tools, I believe the "little guys" will be freed to woo and pursue the thousands of worthy acts and performers that formerly had no voice.

It's exciting. Fun, too. Try it, you'll like it.

Dave

[Marvel]

Dave

You are making, for want of a better term, simple recordings. Four channel surround. Given that, you can afford to use 24/192. I know studios using DAWs, and they end up with 50-80+ tracks. They have been able to stream 24/96 data off the hard drives, but (if I wanted to do the math) I doubt they could do the 24/192 data with that many tracks. The new Pro Tools can, but with great hardware assist and even more money. I don't mean to say that your recordings are easy to make BTW, but compared to all the work that goes into what Griffinator is doing in a studio environment, the setup is easier.

Steinberg isn't at this point going to code Cubase to do above 96k, but it is a part of Nuendo ($1200). Sonar only does to 96K as well. So other than your inexpensive N-Track, most software is over $1k to be able to handle the 192k material.

Can't wait to hear feedback (no pun intended) on your demonstrations at your place.

Marvel

[LynnOlson]

I'd like to throw in my two cents worth in favor of 96/24 DAD's. They play on any DVD player, including the fancy SACD/DVD machines, and most recording systems these days, even for entry-level DAW systems, support the 96/24 format.

What's annoying about 192/24 DVD-A is that the medium defaults to a miserable 384kps Dolby Digital if the player isn't a late-model DVD-A player, and worse, consumer-grade DVD-A players leave a lot to be desired sonically unless you pay kooky prices for Krell or suchlike. As we know, the matter of transporting high-rez digital out of a transport to a high-rez external DAC is mired in Hollywood copy-protection schemes and legal shenanigans.

SACD/DSD is hostile to home recordists, with almost nothing available for DAW systems. The front-end stuff is very thin on the ground compared to 96/24 equipment, which seems to be everywhere.

Which leaves 96/24 as a format that plays on ALL DVD players - from the cheapest COSTCO special to super-high-end - and is well supported on the recording end. My question, though, is 96/24 DAD mastering software hard to get or use?

[Mallette]

I fully agree that what I am doing is simplicity itself. However, I believe the skill of developing a mic plan for such simple recordings requires significant comprehension. To me, it is very simple. I put the mics where my ears want to be, and use mics that respond very much like ears. Very simple. I am, OTOH, astounded at how few supposed professionals seem to either understand this concept or how to implement it. I attended the dedication of a very large pipe organ at a very rich church a while back. They had hired a very expensive audio service to record. The mics were all hung (and there were several) high in the air in front of the various divisions. I thought, Crikey, if that is where this organ sounds best, somebody must have goofed up.

As the old Quaker hymn sez, "Tis a gift to be simple..."

I have yet to hear a multi-miced acoustic event that sounded right. Not one.

One more experience comes to mind. The Meyerson Symphony Hall in Dallas is considered to be one of the acoustical triumphs of the 20th century. Having subscribed for years, I can confirm this. When the local tv station announced this years Christmas concert, they said "You are in for an acoustic FEAST! 36 microphones will be used to provide 5.1 surround!" I went, OH MY GOD, what can you do with 36 microphones except really screw up...

I couldn't listen for more than about 10:00 before giving it up. It was horrible. Complete imaging chaos, with a hum to boot. I'd have used 4, two on the choir and organ, two on the orchestra. The hall does the rest.

Just my simple opinion, of course.

Dave

[Marvel]

Dave,

My mind is simplicity itself. There is an art to correct mic placement, and the fact that you basically use four mics doesn't mean it is easy. Many years ago I recorded a small junior college choir, using two mics. It was long before tv was stereo, but I recorded in stereo. I didn't really know what I was doing, and made a mono mix for use at the station where I worked. A younger friend of mine, still a college student, heard the playback and immediately said "you mic'ed in stereo didn't you?" I've learned a bit more since then. Most people didn't notice, but he caught the phase cancellations, and funny eq that it all caused. If I had done an M/S or other correct stereo recoding with the two mics I could have fixed it, or it would have worked out okay, but alas, I hadn't done it that way.

Lynn,

I think a lot of software is available to master at 24/96, but there still needs to be a lot of talent. I could screw it up in a heartbeat. I don't know about acoustic, live recordings such as Dave does, but I think the majority of studios send their two track masters out to a mastering lab for final multiband compression/EQing/tweaking, and like the mix on CD or tape (reel to reel), with the tape not cut right at the beginning and end of the music, but with the residual tape noise on the front and tail. The CD could be as a wav with higher bit/sample rates. They spend big bucks on their equipment and can make an incredible difference on the end product (could be good or back, depending on your perspective).

Marvel

[Griffinator]

FWIW...

I do tend to mix in as high a resolution as I can extract from my system (currently 24/96) for one simple reason:

When my system gets done processing effect tails, fades, crossfades, etc, having the ability to apply a solid dithering algo and a solid software SRC (it ain't hard to do a good SRC algo, but it damned sure ain't cheap, whether hardware or software) rather than leave all those LSB errors audible is worth the trouble.

Mallett - I'm going to try to explain this to you again. You are totally mixing up your terms.

- dithering is a randomization function used on the bit rate (word length) in a PCM signal. It's sole purpose is to approximate a noise floor where none exists, because truncation creates all manner of ugly, audible rounding errors.

- bit rate refers only to the PCM description of the amplitude of a signal. 1 bit = @ 6dB of dynamic range.

- SRC is an acronym for sample rate conversion - self explanatory.

- sample rate is the number of snapshots per second taken of the waveform's frequency. It has nothing to do with amplitude. When an SRC "resamples" a waveform, that's exactly what it's doing - it's recreating the waveform then taking new snapshots of the recreated waveform at a faster or slower rate (upsampling or downsampling)

This is why truncating a signal has absolutely nothing to do with the signal's sample rate, and why you cannot remove "every other bit" in order to change 24/192 to 24/96.

I'm not trying to be sarcastic, I'm just trying to help you understand the way the technology works, and hopefully better understand the arguments I'm presenting.

[Griffinator]

----------------

On 8/9/2003 3:59:55 PM LynnOlson wrote:

SACD/DSD is hostile to home recordists, with almost nothing available for DAW systems. The front-end stuff is very thin on the ground compared to 96/24 equipment, which seems to be everywhere.

----------------

Actually, that's really not true. You don't have big names (yet) producing DSD front-end ADCs, but it's coming. There are low(er) cost front-end solutions coming out on a regular basis. Genex just released their 8-channel DSD/192Khz PCM ADC unit for just over $2,000 list. Yeah, it's a big bite, but it's a hella lot cheaper than most of what's out there.

Software for SACD? Check out Pyramix - http://www.merging.com/2002/html/pyramix.htm - they've released addons for this system to handle DSD signals and author in surround SACD and DVD-A protocols. It ain't cheap, but it's a damned good option for PC-based SACD/DVD-A production. The feature list, IMO, embarasses ProTools considering the cost/performance ratio. Does everything PT does, and more, for a lot less.

[AndyKubicki]

I'm glad you guys are discussing this here so the rest of us could learn a little about this. I would love to hear your reactions to my second post in the Experiencing new level of detail post regarding vinyl vs Red Book CD. My interest is for making a decision as to whether to ultimately get into SACD as well as to learn to make better recordings using both MIDI and audio tracks.

BTW, is the main use for a SRC mostly to make DATs brought into a studio with unusual sampling rates (like 48K) compatible with the more standard sampling rates used in studio equipment (like 44.1 or multiples thereof)?

From the excellent input of Griff and others, the mystery has only deepened for me as to just WHY the higher sample rates sound better. However, the fact that they do, and it is significant, has only been further supported. In my mind at the moment, the most likely candidate is simply more "dots" in the picture and the unknown depths of the human brain that perhaps are able to percieve more than we think.

Dave, I think you're on to something there. We don't really perceive more dots in a higher resolution picture unless we look through a microscope, yet we experience it as "sharper" or more detailed. We are not looking at waveforms on an oscilloscope, we are hearing the whole thing.

[Mallette]

I do not think you are being sarcastic, Griff, though perhaps you are a bit impatient with my loose use of technical terms.

I didn't go back to see where I'd mixed my metaphors, but I wouldn't be surprised. I am exploring here, not trying to write an A.E.S. paper.

I have a pretty good understanding of the quantizing process, but that doesn't not explain to me what sounds good and what doesn't.

Though I did not go back to see what I'd messed up, a clue was in your statement

>This is why truncating a signal has absolutely nothing to do with the signal's sample rate, and why you cannot remove "every other bit" in order to change 24/192 to 24/96.

When I refer to truncating, I am referring to dropping the least significant 8 bits of a 24 bit signal.

When I am talking about disposing of every other bit, I am refering to sample rate. I'd still like a clear explanation of why this is not possible. Much of the digital audio world thinks it is and I see no flaws in the logic.

That doesn't mean the aren't any.

Dave