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Kenny Gioia
09-29-1999, 12:54 AM
I'm noticing a lot of people complaining about the Pro tools Quality so I asked a manufacturer of a $350,000 digital board why I should spend so much when I already have a digital board.

He basically said that pro tools resolution is not high enuff once you combine all those tracks. For every track you add, you need another bit. I think that means a 24bit session with 32 tracks would require 56 bits on the path.

He said it's less of a problem if you keep the volumes down on each track so your not combining too much signal.

I don't know if I understand him correctly but I definitely notice the more I add the worse it sounds. And that never happened on my Neve board.

So I think my plan is to ruff mix everything in Pro tools. Then rent 4 AD8000's and book an SSL room and Mix right in pro Tools but having all the tracks individually going out all the outputs to the SSL with the Faders all at 0Vu (so as to keep my mixing moves and save time).

And then of course use all ouboard EQ & compressors because I know those plug In's are ruining my sound. Everytime I bypass them the mix sounds better.

Well if we could all afford the Neve Capricorn we wouldn't be at the Digi Site. Peace :•)

Digi Engineering
09-29-1999, 08:06 PM
His statement about 1-bit per track is incorrect. To combine 32-full code 24-bit phase aligned signals you need 6 additional bits (2 to the 6th power is 32). Also, you almost never have phase aligned signals so in the typical case you rarely need more than 3 additional bits. The 24-bit optimized mixer has 48-bits of internal resolution.

What resolution was the digital console that you were looking at? If they were using 32-bit floating point DSP's than the mantissa is 24-bit's and since audio is usually normalized to between -1 and 1 in a floating point system has the same dynamic range as 24-bit fixed point.
-Mike Rockwell

tdmuser
09-29-1999, 09:10 PM
Mike Rockwell writes:
>The 24-bit optimized mixer has 48-bits of >internal resolution.

Yes but those 48 bits are TRUNCATED back to 24 bit fixed point. And as far as I can tell this is the only reason why Pro Tools mixes sound thin. WE WANT DITHER. I don't care if it needs twice the DSP. I choose AUDIO QUALITY over savings. I and many other pro audio engineers (who care about the sound) BEG you to write a dithered mixer.
And to end this note, Mike THANK YOU for your contributions to the list. I hope that these posts will help Dave Lebolt and other Digi managers to consider spending time improving Pro Tools sound quality.

Mini Me
09-29-1999, 11:16 PM
To the success of Ricky Martin's CD which was recorded and mixed almost conpletely with Pro-Tools, doesn't anyone believe that the systme is capable of stellar mixes?

Mini Me
09-29-1999, 11:25 PM
I read all you comments regarding Pro Tools sound quality and I'm very much interested in purchasing the entire system, Protools 5.0, (3)-888/24 and ProControl.
Any opinions?

Thanks for your suggestions.

millimix
09-30-1999, 12:00 AM
hear ye, hear ye, im dying to see what one of the die-hard pro-tools users has to comment on the last posting,i personally have some mixes ive done on a pt24 with 1 dsp farm ,2 888-24's,2 yamaha 02r's and the world of high end out board gear, that blows away mixes done in new yorks most expensive studios, on the other hand ive got mixes done on a mix plus with two extra mix farms(mixed internally)audio was recorded through ad-8000's,and the overall sound sucks big time,one user told me that the pt24 sounds better than the mix plus, i find this far fetched since they are basically the same product,through my tests the only direction i can point the blame is to the internal mixer, can someone shed some light on this!!(Also 16 Channel pro-control for sale 12.5k)

[This message has been edited by millimix (edited 09-29-99).]

09-30-1999, 01:03 AM
Why do Mackie DB8 desks also claim to have had a hand in the Ricky Martin Hit in thier promo literrature?

blairl
09-30-1999, 07:30 AM
Mike Rockwell,

Thanks for your contributions! I have a question regarding 48 bit processing. You mentioned that the Pro Tools mixer uses 48 bit internal processing and yet people are saying that the signal is truncated back to 24 bits. I was reading some information provided by Sonic Solutions which states the following:

"High-resolution, 48-bit processing delivers the highest signal-to-noise ratio needed for working with High-Density Audio™. While common 24-bit and 32-bit floating point processors require that digital audio signals be truncated to 24 bits following all DSP operations, thus adding distortion to the signal, 48-bit processing removes this limitation."

Can you comment on this? Is Sonic Solutions using a different technology to acheive this? I am interested in your response. Thanks!

Pete45
09-30-1999, 04:29 PM
To Mike Rockwell:

Can we assume that there is some kind of interaction of mixer elements in Pro Tools and can we consider this as digital crosstalk,related to TIME DIVISION MULTIPLEXING.



Thanks.

Digi Engineering
09-30-1999, 04:52 PM
You should absolutely not assume that. There is zero crosstalk on TDM and within the mixer. Time division multi-plexing either works perfectly or it doesn't. There would be very obvious catastrophic results if TDM were not working such as full code digital noise.
-Mike Rockwell

benderbro
10-01-1999, 07:45 AM
To Mike Rockwell:

You answered Pete45's question but what about blairl? You know, the question right before that about Sonic Solution's statement. I too would be interested in your response to that.

------------------
benderbro@aol.com

El Guapo
10-01-1999, 09:21 AM
I tried to avoid spending the loot for a digi system that seemed profesionally usable. I waited 3 years for the other promised DAW products which would be better, faster, and less expensive than digi to the extent that I tried several solutions (Akai, Paris, etc.) and finally, (unfortunately) got a Mackie d8b and MOTU 2408's last year. It entirely bombed.

It's interesting that a digi system is compared to (what I would call) expensive digigal boards. Gee, I also wish my 40K digi system sounded like a 400K system. What is more interesting is that so many people believe it should be able to.

Get off your duff Digi and make this product sound like a 400K recording system now!

I expect the complementary upgrade by the end of the 4th quarter, 1999, before the power goes out.

MICHAEL JACKSON
10-01-1999, 01:59 PM
Pro Tools works perfectly fine and sounds realy good.I've done lots of mixes on it,engineered 10 albums(most in 16 bits!).I think there is too much crap talk on the subject of the sound quality.I've also had experience on all sorts of Neve boards and SSL and nothing gives me such a good sound,flexibility as Pro Tools does and DIGI is not paying me to say this!!!I've recently done A/B test between Apogee AD-8000 and 888 I/O on a pop song in front of several clients and believe me or not-nobody could tell the difference in sound quality!
I believe that it all comes down music beeing recorded.When you recording jazz or classical material you would propably want to go 24 bits and 96 khz with Prism or Apogee in order to get the edge.But with pop,rock,dance,rap...believe me it doesn't matter!!!

tnie
10-01-1999, 04:33 PM
Mike Rockwell,

I also would like to hear the answer to blairl's question. The one dealing with Sonic Solutions........

10-01-1999, 04:40 PM
I get acts nagging me about the type of desk they want me to use, (mostly Neve but they dont know the difference betwween the 80's ok ones and the real deal 60's & 70's ones)
Digi's commitment seems giant, lots of cool stuff coming out for PT.
Options versus internal sound? Options can be take em or leave em,. Sound? Now that's something you can't bull**** about, word will get around.
It would be coolest for my buisness if Pro Tools became known as the "Neve" of the DAW's, rather than the "Dr Gadget".
Very exciting times ahead for us all.....
Thank you Digi!
http://www.digidesign.com/ubb/images/icons/smile.gif
Jules

Digi Engineering
10-01-1999, 04:55 PM
Ok. To my knowledge, Sonic Solutions still uses 56k series DSP's in their workstations. The are the same DSP's that are used in PT systems. 56k's are capable of processing with double precision which is 48-bits wide. It takes more cycles to process 48-bits wide than to process 24-bits in many cases. Most of the plug-ins that run on TDM do carry the results in the 56-bit accumulator (8-bits are for overflow). This includes the 24-bit optimized mixer. The only potential difference between TDM and Sonic is that they may maintain the 48-bits when they transfer the data between multiple DSP's where with TDM it needs to be converted to 24-bit by truncating, rounding or dithering. At some point the Sonic system has to convert to 24-bit to get out AES or go to a file.

I'm not sure what the term "High Density Audio" means. It sounds like a bunch of marketing junk to me. I have to say that I personally don't think anyone could hear the difference between the 48-bit accumulated signal being truncated, rounded, or dithered. We are talking about differences that are down -138 dB. Your source material only contains noise down there since no converter in the world can actually deliver more than around 21-bits worth of information. I suppose that some plug-ins (like reverbs) could generate something useful down there but you can't hear it because no D/A converter can resolve it and no analog monitoring system could reproduce it.

All that being said, in a few weeks we are going to do an experiment with all you folks being the subjects. We are going to post a number of audio files that differ in only one respect. One set will have been gain changed through the PT mixer with dither noise added after the fact. Once set will have been gain changed with 48-bit precision dithered to 24-bits. A few of them will be just two different passes with the PT mixer as controls. The questions that will be asked are:
Can you hear the difference?
Which one sounds better?
This will be a truly double blind test. I encourage everyone to download these files and join in on the experiment. Again this won't happen for a few weeks as we are in the final throws of getting PT 5.0 out the door. It'll be really interesting to see if anyone can actually hear the difference.
-Mike Rockwell

Disco_Doctor
10-01-1999, 07:42 PM
Mike, now THAT is a really cool idea!

In the interest of making sure that the test is not manipulated by either Digidesign or its detractors, I'd like to suggest that you strongly consider giving an objective third party the "answers" as to which file is which before you post the files to the public...just so there's no question at all that Digidesign is being completely honest about which file was which when the results of the test are made public.

One other thing - I wouldn't be surprised if very few people could hear the difference under the conditions you are describing - that is, one piece of source material being gain changed one time through the PT mixer vs. one piece of source material being gain changed one time with 48 bit precision and dithered to 24. That may be, as you often claim, too subtle to detect. What would be a more realistic comparison would be if, in addition to those single-pass test files, you took some mix stems - say eight to twelve tracks worth - and gain changed and summed all of them through the PT mixer and this 48 bit precision/24 bit dithered process - which would be a more real world example of the sort of accumulating "thinness" and loss of clarity that people are claiming to hear when mixing large projects in Pro Tools.

I hope you'll consider these suggestions...I look forward to the test!

http://www.digidesign.com/ubb/images/icons/smile.gif

PS - Please don't use Spice Girls or Kid Rock for the source material!

Steve MacMillan
10-01-1999, 09:08 PM
might I suggest...

Beethoven's Moonlight Sonata

J Harry
10-02-1999, 01:30 AM
This is a fun test you can do at home.

Take a split stereo file- vocals, acoustic guitar, piano- something with a bit of detail and dynamic range

Create split stereo pairs via three methods

1. Bounce to Disc
2. Analog Bounce (Direct Out into 888/24 inputs)
3. Recording to tracks via the aux busses

Import the bounce to disk and compare the three.

Hope some other folks will try this and post their results

JH

Joris Vincken
10-03-1999, 10:56 AM
Michael Jackson (?!) it's thru that almost no non pro audio person can hear the difference between bit rates, other convertors or even cheaper microphones. From experience with my clients I do know however that once you combine all those little improvements, especially after using a lot of plug-ins etc that people do hear the difference. As for the internal 24/48 bit question, for me it doesn't matter which sounds better. It would just be easier if I had more headroom when mixing and don't have to worry what sounds better. The speed I gain is what matters. When using multitrack 16 bit can sound just as good than 24 (when using a lot of tracks). It's just that with 24 bits you don't have to worry about levels that much and still have it sound good.

Rail Jon Rogut
10-03-1999, 11:41 AM
Mike

You don't think it's time to replace the old 24 bit TDM buss with a newer 64 bit TDM buss with your next major hardware upgrade? This would certainly take care of the problem.

As to the blind test -- I doubt that any results you garner from a blind test performed under varying conditions by varying users of whom we know nothing about their qualifications, will give any valid results. You may want to do what they did with the watermarking technology -- Warner Bros. held A-B-X listening tests at different areas of the US with professionals in the audio field including members of MPGA (Ed Cherney, Massenberg, etc) and mastering engineers (Doug Sax, Bob Ludwig, bernie Grundman) -- and based upon those results they could make a qualified assessment.

I also think another valid test would be to perform these tests in a professional recording studio where you can A-B between a full mix where the outputs of Pro Tools are taken individually out of Pro Tools into faders on an SSL 9k and summed using the 9k's summing buss vs. the stereo mix summed inside Pro Tools.

Best regards.

Rail

------------------
Recording Engineer

Rail Jon Rogut
10-03-1999, 04:19 PM
Hi Monte

I agree that the dither in the current TDM buss will probably solve most of the ills - but would consume CPU time in processing... so I thought the logical step would be to make the wider TDM buss, which wouldn't require any additional processing overhead (well perhaps some to handle the larger bit width numbers) - thus allowing Digi to maintain their current track count.

I also agree that the floating point math in the 001 may indeed sound superior, but as George Massenburg told me, you certainly can get acceptible sound quality using fixed point if it's done correctly. He felt the main culprit was the truncation -- without that the quality should improve.

Best regards.

Rail

------------------
Recording Engineer

Monte McGuire
10-04-1999, 12:02 AM
Mr. Rockwell,

Thanks for taking this subject seriously. I really look forward to such a test, but I'd like to make a few suggestions as to how to approach the whole thing.

The problem I have with the sound of the PT mixer happens when I have a large number of inputs summed to mix. It's not that the PT code is misbehaving in some sort of bug like way with large mixes, I think it's simply a case that emphasises the problems with the current mixer. So, any test files made with the two mixers should be made from a typical large scale mix, and not just a few tracks run through the mixer.

The tracks themselves should be sourced through a reasonably clean chain too. If a number of plug ins that round or truncate back to 24 bits are used as inserts on each channel of the test mix, the importance of the final truncation at the end of the stage will be minimized and any differences won't be as audible. We'd simply be comparing twice damaged audio to thrice damaged audio and yes, the differences won't be that significant. So, restrict plug in usage to those plug ins that do 48 -> dither to 24 processing only.

I'd also like to suggest that you use artificial reverb in the mixes. Reverb gets altered by truncation pretty readily, and it's therefore a good test of processing quality for both the aux send mixers and the main stereo mix. Truncation at either end will shrink and harden the sound of the reverb.

Finally, I would prefer it if you could post two unidentified mixers instead of the mixes themselves. You know, mixer A and mixer B and let us decide whether we can hear differences and decide which is which. I guess it would be readily apparent which mixer is which if you looked at allocator, but maybe there's some way to make the non dithered mixer out of the dithered mixer by patching out the dither code. In this way, the two mixers would use up the same DSP resources and be indistinguishable except for their processing quality.


Again, thanks a lot for taking this seriously!! I look forward to the tests...

Monte McGuire
mcguire@world.std.com

Monte McGuire
10-04-1999, 12:43 AM
Rail,

I don't think a super wide TDM bus is needed if every processing step dithers the accumulator before going to the TDM bus. In that case, the only difference would be the small resolution loss from adding dither at the 24 bit level, and that is a very small increase in noise. Remember, true uncorrelated noise is relatively benign, but the cross modulation distortion products caused by truncation are not.

To put some numbers on it, assume you have three inserts on each track of a 32 channel mix. The added noise of the dither between each plug in amounts to a noise floor of around -140dBFS per track. Summing 32 of these tracks at unity gain yields a noise floor of -125dBFS.

Currently, that's about as quiet as a single channel of conversion, so I don't think it's really going to have a huge impact. Plus, if the mixer worked properly and you could use the faders to attenuate individual tracks, then the resulting noise floor could be a good bit lower - there is no need to run all the faders at unity gain and the noise floor could drop quite a bit.

However, if dither is not going to be used, then yes, a wide TDM bus would help a lot. It seems to me that this is a great argument for dither in that we could get the same sound quality in a 24 bit chain as we could from a true 48 bit path, plus a little extra noise.

One final comment about a wide TDM bus. Native floating point based systems are usually undithered between processing steps and they have an internal chain that is essentially the same width as their 24+8 bit accumulator. Given that the new Digi 001 system uses only floating point processing and has session file and plug in compatibility with PT TDM, it would be a great test to run the same mix on an 001 system as well as the current TDM mixer and a new dithered TDM mixer.

Since floating point processing will clean up the plug ins as well as the mixer, I wouldn't be surprised if a mix sounds best on the 001.
I know I have been pleasantly surprised by the quality of work some of my friends are doing on low end floating point systems and frankly, if the editing and latency on those systems were better, I'd have jumped ship!


Monte McGuire
mcguire@world.std.com

10-04-1999, 12:43 AM
A 64 track session would be an ideal test example!
Jules



[This message has been edited by Julian Standen (edited 10-03-99).]

Digi Engineering
10-04-1999, 01:17 AM
Thanks for all of your feedback. We'll take it into consideration when we create the test files. A couple of comments:

Monte: In your post you state that 32-bit floating point is 24+8 bits worth of precision. This is not actually the case. With 32-bit floating point there are actually 24-bits worth of mantissa. The other 8-bits are for exponent. As far as I know all of the 32-bit host systems out there normalize their values to between -1 and 1. Anything greater than 1 is considered clipping. This is actually less resolution than the 48-bit fixed point that is used in the PT 24-bit optimized mixer.

Disco: It's unfortunate that you think that we would manipulate the results. We will post the honest results and descriptions of the techniques used to create the sound files at the end. I hope that you will trust us to be honest. This test is really to help us decide the best way to make PT sound better not to be able to make some marketing claims.

Rail:
Since we can probably never make our mixer sound like the SSL (analog vs. digital) it doesn't seem like a very useful test. If there is a digital console that you think sounds a lot better than PT that would be useful info. Extending the width of TDM would actually increase the processing overhead substantially. 56k DSP's have a 24-bit external data bus. It would double the overhead to write 48-bits to a wider TDM bus. Again I think that it's easy to get caught up in the more is better syndrom. 64-bit fixed point would have a 384dB dynamic range which would basically be mostly a waste of bandwidth. Let's see how the tests come out before we jump to that conclusion.
Thanks,
Mike Rockwell

J Harry
10-04-1999, 02:07 AM
Mike,

It would be great if you used an SSL Axiom

JH

Disco_Doctor
10-04-1999, 10:17 AM
It's unfortunate that you think that we would manipulate the results.

Mike, why do twist around everything I say? I said "in the interest of making sure that the test is not manipulated by either Digidesign OR ITS DETRACTORS". That way, no one at Digidesign can ever misrepresent the results, and no one who disagrees with or challenges the results can accuse Digidesign of manipulating the results.

I did not say, nor do I actually believe, that you or any other particular person would use the test for marketing purposes. But just because YOU are an honest guy with a good intentions, that doesn't mean everyone in Avid's boardroom and marketing department is. We're talking about a huge corporation here Mike - of which you are not the CEO or CFO - and you can't, with a straight face, speak to the motives of everyone who works there. My lack of unequivocal faith in Avid Inc. is not "unfortunate" - it's pragmatic.

If you want this test to have long term integrity, the "answers" should be given to a third party. I stand by what I said.

I doubt that any results you garner from a blind test performed under varying conditions by varying users of whom we know nothing about their qualifications, will give any valid results.

Maybe so. But I also think that with a public test like this, the listener results may average out and provide some useful information. There's plenty of qualified ears out there in Digi's customer base. Your suggestion is good though - Digi should also subject itself to the opinions of MPGA members and top mastering engineers as well as its broad audience of amateur and professional customers.

I'm sure there will be plenty of situations where producers and engineers will get together with their peers to do this testing in professional studios. That will lend some additional validity to this public test.

One other suggestion Mike. Set up a temporary forum on the DUC specifically for the test results. Post instructions that outline exactly how users should perform the testing and a questionaire that lists specific questions for the respondents to answer. That will make it a lot easier for Digidesign to extract useful data out of all the feedback.

http://www.digidesign.com/ubb/images/icons/smile.gif

Rail Jon Rogut
10-04-1999, 11:00 AM
My thought was that every time the signal has to enter a TDM node, it also has to be dithered down to 24 bits, and at that stage it's done by the CPU - dithering on returning out of the plug-in or TDM mix node would be done by a DSP chip... of course I have no real knowledge of this and it would be up to Mike to let us know exactly where and how the dithering was being performed. You may be correct... I just don't know (it would be nice to have a block diagram one day).

Regards.

Rail

Digi Engineering
10-04-1999, 02:33 PM
It is true that you need to have head room to mix together phase synchronous audio. This applies to analog as well as to digital. Analog consoles let you go to around +28dBU before you clip the stereo bus. The 24-bit optimized mixer has around 30 dB of headroom above zero before it clips internally. This was not the case in the past with the 16-bit optimized mixer which would clip pretty easily and you had to bring down the source faders. With the 24-bit optimized mixer you can bring down your master fader if you are clipping the output, just like you would do on an analog console.
-Mike Rockwell

Kenny Gioia
10-05-1999, 10:02 PM
Boy, look what I started!

Well I'm glad to see that mostly pros are using the system that I am. We are bitching about a pretty good sounding system.

Let's suppose Digi does post an A:B test against an SSL analog console mix. Let say the SSL kicks the Digi Mix's butt. Ya

Is everyone on this thread gonna go out and sell there Pro Controls and buy an SSL???

Ya gotta believe it's pretty cool to buy a system that costs about $30,000 (all in) that people are complaining about it not sounding as good as an SSL.

Is anybody calling up the $30,000 console makers (Amek) and asking why there consoles don't sound like the SSL.

They also don't include Graphic Editing!!!

Hey Amek!! Where's the graphic editing???

Kenny Gioia
10-05-1999, 10:04 PM
Also does anyone know if the trunication still happens if you insert a piece of outboard gear in a channel instead of a plug in???

RMD
10-05-1999, 11:42 PM
I went to the digi 5.0 tour tonight at a local dealer. I mentioned some of the comments I've been reading about the sound of the mixer. He said he is totally unaware of any problem. There are 300 000 digi users worldwide who are completely satisfied with the sound of protools. Was unaware of the upcoming "test" but did say digi has always been good at improving their product.

This leads me to wonder whether people on this conference have "enlightened" auditory abilities or are in a deep minority amongst digi users.

Whatever the case it does make for some great reading.

cheers to all
Robert M. DiVito

johnwhynot
10-06-1999, 01:33 AM
Well, I say to hell with the statistics. I'd be interested just for my own edification/confusion. I love listening tests.

But comparing two mixes done on different systems is not terribly instructive in my experience. Too much chaos in the process. Brings to mind a demo I saw for BASF tape, in which a live band was recorded to two 24-track machines and the mixes put up on two sides of a SSL9000J. Of course the BASF sounded fantastic! Until I noticed that the group master on the BASF side was about .5 db louder. An error, yes, but just one element in a vary large collection of potential cumulative errors. Bye the way, I switched to BASF after doing a comparison of 2-track recordings done on the same machine. It did indeed sound better than 499.

Anyway, the more rigorous the test the more fun the examination of the results, I reckon. Keep the source the same for each run of tests. I'll definitely listen hard!

nipple
10-06-1999, 11:19 AM
Speer,

Judging from my own experience I would say trimming back the source faders sounds better to me than slamming the master bus. Just did an A-B between the two the other day. I had built a mix that the client was loving but it was killing the master and it was kind of bugging me so after they left I tried it with pulling the master down and with trimming the automation on the source faders. The difference for me on this particular tune was that when I pulled the source faders down the mix seemed to clean up a bit, seemed to have more depth and stereo spread albeit at the expense of some punchiness. Your mileage may vary.

Digi Engineering
10-06-1999, 11:23 AM
Speer,
The clip meter on the Master Fader does actually tell you when the internal bus is clipped even when you bring the Master Fader down. There is around 30 dB of internal headroom on the 24-bit optimized mixer so you should be able to hit it pretty hard and bring down the Master Fader with NO audio degradation. This was not the case on the 16-bit optimized mixer. Were your problems with the 16 or 24 bit Mixer?
-Mike Rockwell

Monte McGuire
10-07-1999, 12:50 AM
Mr. Rockwell,

Sorry I was careless about the terminology - by referring to 32 bit floating point as having 24+8 bits of resolution, I meant 24 bit mantissa and 8 bit exponent, as you stated. If you talk about resolution strictly, then only 24 of those bits should be counted as resolution. Sorry for the sloppiness! I guess the point I wanted to make is that 32 bit float doesn't really have any more instantaneous resolution than straight 24 bit integer, it merely has an exponent which gives it greater dynamic range. I shall be more careful with "reserved words" in the future!!! ;-)

As for Produceher's question about hardware IO inserts, they are 1:1, unprocessed paths between the 24 bit TDM bus and your IO interface. If your IO interface handles 24 bit digital IO (as do all interfaces attached to a PCI card), then there is no truncation or any other sort of math involved - it's just a data copying operation.

Of course, if the external box you're feeding can't handle full 24 bit IO, then you'll get truncation, but simply looping through a hardware IO loop only adds delay - it is a perfect clone otherwise. Systems using NuBus cards do not have this property. Digital IO is only 16 bit wide on PT3 NuBus hardware.

Speer wrote:
<<Monte, interesting idea. I suppose Digi could make it mandatory for all plug-ins to dither back to 24. Of course, they would have to also do that for the Digi-Rack stuff. This would lessen the amount of power they can claim for a core system, so of course they won't and here we go again...>>

I didn't mean that only dithered plug ins should be available on the platform as a whole, only that the ones used to prepare the test mixes that Mike is talking about should use 48->24 bit dither. This is simply to provide the least damaged signals to the mixer and make the test more optimistic.

And, finally, RMD said:
<<I went to the digi 5.0 tour tonight at a local dealer. I mentioned some of the comments I've been reading about the sound of the mixer. He said he is totally unaware of any problem. There are 300 000 digi users worldwide who are completely satisfied with the sound of protools. Was unaware of the upcoming "test" but did say digi has always been good at improving their product. This leads me to wonder whether people on this conference have "enlightened" auditory abilities or are in a deep minority amongst digi users.>>

The guy giving the demo must have loved you ;-)

Seriously, I don't think the whole DSP quality issue is a _huge_ problem and PT DSP quality as a platform is higher now than ever. But, it's still something that if it were improved, it could greatly enhance the usefulness and value of the system to more than a few people. And wouldn't it be a great thing to have PT be the DSP quality leader for a change...

I know for a fact that a lot of people operate DAWs in very foolish and signal damaging ways, they don't hear the difference and the work still gets done. But you know what? A lot of stuff sounds pretty poor out there.

There will always be a bunch of people out there who worry about the details and a bunch who don't even know what they don't know. All I care about is advancing the quality of my productions and right now, DSP quality ranks in the top 10 list of tech things to fix in my system. Of course the current level of quality is usable and on an absolute sense it's better than ever, but I don't think I'd still be doing this if the quality of my work wasn't improving.

From my experience with high resolution plug ins, I think it would make a worthwhile difference, and that's why I continue to ask for it.

Monte McGuire

RMD
10-07-1999, 11:04 PM
Monte,
I wasn't vicious or anything. I just voiced some concerns and he answered to the best of his ability. Nothing to do with whether he likes me or not. Just trying to get useful information which is what this is all about. (and I must say your posts are always well informed and very useful!)

Regards,

Robert DiVito

Digi Engineering
10-08-1999, 12:14 AM
Thanks for you reply Monte. As ever you are the level headed guy. Let's see what these tests turn up. The plan is to only allow 48 to 24-bit dithered changes in the test files so we should really be able to see if folks can hear the difference.
-Mike Rockwell

Joris Vincken
10-08-1999, 01:13 AM
Maybe the dithering to 24 bit in every plugin is a good thought. Why not make it switchable, this way if you want it, you lose some DSP and otherwise nothing changes.

10-08-1999, 05:06 AM
Yes I agree with Speer, what you have here is a bunch of contientious engineers used to "old time gainstaging".
We are a bit in the dark (and feeling mushroom like) about the best way to procede with the various, potentialy "source sound changing" signal path options in the whole TDM system and crave some guildlines for it.
Jules

bluemt
10-08-1999, 07:58 AM
I'm in agreement with the need for better documentation and re-engineering (at least of the GUI at this point)of metering. I'm Protools mastering my first project and am dealing with all these issues. I'm not too hip on the "pull up the faders and guess method" for avoiding overloading the bus.

Rail Jon Rogut
11-10-1999, 05:04 PM
Mike Rockwell wrote:

Since we can probably never make our mixer sound like the SSL (analog vs. digital) it doesn't seem like a very useful test. If there is a digital console that you think sounds a lot better than PT that would be useful info.

Okay, today I finally had some free time, so I went over to Ocean Way and set up a system in their Oxford room. To clarify, I did not do double blind testing, but I did do single blind tests.

Setup: 24 bit session with a stereo 24 bit track. Duplicate the same stereo tracks on tracks 1, 2; 3, 4; 5, 6; and 7, 8 - output tracks 1, 2, 3 and 4 out of outputs 1 & 2 - output tracks 5, 6, 7 and 8 out of outputs 5, 6, 7 & 8. Return outputs 1 & 2 into a monitor return position and return outputs 5, 6, 7 & 8 into channel line inputs. Group the faders on the Oxford where the returns for 5 through 8 return and set them up at 0dB. Each pair of tracks in Pro Tools was panned hard left and right. Using the 24-bit optimized mixer. The Oxford was the word clock source, fed through a USD and all inputs to the Oxford were AES/EBU.

Mute tracks 3, 4, 7 & 8 in Pro Tools. I had a 1 kHz tone region on all tracks and played that to verify levels were correct at the stereo buss and also at the 2 track monitor return position.

First test: Brought the levels of tracks 1 & 2 down 3 dB in Pro Tools - brought the grouper down 3dB on the Oxford. Checked with the tone that the levels matched, then blindly compared the input to the monitor return on the Oxford. It wasn't obvious, but there was a xylophone solo in the track, and at that point it was obvious that the image of the Oxford's stereo buss was wider.

Second test: Unmuted tracks 3 through 8 in Pro Tools and brought the level of the faders in Pro Tools for tracks 1-4 down to -5.1dB. Tracks 5 through 8 remained at 0dB in Pro Tools. On the Oxford, lowered the gain of the group fader to -5.1dB. Checked levels with the tone and then did the blind A-B test again. This was much easier to discern the difference between the input and the monitor return.

The test track which we were using was pretty full, it was a big band style song - so we thought we'd try using another piece of material - so we transferred the Alanis Morisette song off the City of Angels soundtrack into Pro Tools and used that for the next tests.

I set up the tracks as before - just replacing the audio regions.

Third test: Muted Pro Tools tracks 3, 4, 7 & 8 and brought the faders of tracks 1 and 2 in Pro Tools down 1dB. Tracks 5 and 6 in Pro Tools were kept at 0dB. The Oxford group was brought down to -1dB and the levels were checked with the tone. Imaging difference was not as apparent as before in test #2, but there was definitely a smearing of the high frequency information in the Pro Tools mix vs. that of the Oxford.

Fourth (and last) test: Kept everything as set up in test #3 but brought the Pro Tools tracks 1 & 2 down to -5.1dB and the Oxford grouper down to -5.1dB. Checked levels with the tone. Here the difference was obvious and not at all subtle. The intro of the song has a vocal with dense reverb and the image of the Pro Tools mix just collapsed compared to the Oxford which still sounded dense.

I would suggest that you try these tests yourself.

Best regards.

Rail


------------------
Recording Engineer

[This message has been edited by Rail Jon Rogut (edited 11-10-99).]

[This message has been edited by Rail Jon Rogut (edited 11-10-99).]

[This message has been edited by Rail Jon Rogut (edited 11-16-99).]

Steve MacMillan
11-11-1999, 09:22 AM
That does it, I'm selling my house and buying an Oxford! Maybe I should wait until I can charge $10,000.00 a mix, hmmmm.

It would be great if the next major hardware rework of ProTools (the one with 96k, 128 tracks, and 2048 TDM nodes) addressed these internal resolution questions. I would pay more to be as good sounding as an Oxford or Sonic. Until then I'm still making surprisingly good sounding records completely in ProTools.

sm

Rail Jon Rogut
11-11-1999, 10:49 AM
I wouldn't run out and sell the house Steve... But I was asked about a comparison to a digital console, and that's what I did. If Digi can improve their mix buss to better approach the Oxford's -- then I'd like them to. Personally, I think improvements in sound quality and trying to improve the responsivness of the Pro Tools program when you have sessions with thousands of edits with crossfades should be made a high priority. Adding more features may attract more new users, but fixing the old problems should also be addressed.

Best regards.

Rail

------------------
Recording Engineer

11-11-1999, 04:26 PM
Here here!
I'd go for that over some squelching direct connect synths! There will be a stampede back to analog if we aren't carefull!
Jules

P.S. RAIL surely after a while with teeny drum crossfades and the like you can make a decision that you have the master tracks and bounce em down? or not?

[This message has been edited by Julian Standen (edited 11-11-99).]

tnie
11-11-1999, 06:54 PM
Digi is not listening !!! Its funny... Digi always speaks about upgrades directed to people who suggest: Midi, new scrubbing tools, things that are cosmetic, new marketing ploys, better adds in Mix mag.... but never address the sound problem. This thread and others with same topic keep showing up but the issue is never addressed. Why?


Is tape really dead ? With Digi's project Mix system I think not !!!!

Rail Jon Rogut
11-11-1999, 08:47 PM
Julian wrote:

Rail, surely after a while with teeny drum crossfades and the like you can make a decision that you have the master tracks and bounce em down? or not?

If I were the producer and knew for certain that the edits were final, then I could use the AudioSuite Duplicate function every 30 seconds and remove unused regions from the session - thus reducing the number of edits and crossfades. However, when hired as an independent Pro Tools editor - I prefer to edit all the drums and afterwards build the crossfades and then play it for the producer to get his okay -- if there's a place where he'd like me to possibly massage the edits, I'd prefer to not be locked into duplicated/edited regions. Having to sometimes perform close to 500 edits across 11 tracks, this can add up to a very large number of edits and crossfades which brings Pro Tools down to its knees.

Regards.

Rail

------------------
Recording Engineer

evanb
11-12-1999, 08:52 AM
I am not sure I really understand all of this but if the problem is truncation at the master fader for the stereo bus...

Does the bus signal get truncated before or after the master fader? Could you use a flat insert EQ that Dithers to 24 bits over the stereo bus master fader to take care of the problem before truncation?

Just an idea.

Steve MacMillan
11-12-1999, 09:20 AM
Rail,

I very much enjoyed reading your ProTools comparison with the Oxford, and I don't doubt you findings at all. I suspect there is more to this than mixer dithering, but we will never know until Mike Rockwell writes his high quality mastering mixer plug-in with internal dither at every stage. I would love to have the option of using it depending on DSP resources.

Thanks.

[This message has been edited by Steve MacMillan (edited 11-12-99).]

Kenny Gioia
11-13-1999, 04:39 PM
Rail,

What I found most amazing about your tests, were that you were doing them with full mixed tracks. (Not individual)

The fact that you even notice a difference scares the hell out of me.

It would be frightening to hear the difference if you had a full blown 32 track session to campare it to. It'll probably be night and day.

More tracks would destroy the Stereo Buss.

Thanks for the test!!!!! Peace :•)

Monte McGuire
11-13-1999, 08:19 PM
evanb wrote:
>> I am not sure I really understand all of this but if the problem is truncation at the master fader for the stereo bus...Does the bus signal get truncated before or after the master fader? <<

It's before the inserts on the master fader. The mixer operates internally at some higher resolution, reportedly 48 bits, and any mixed signal that is fed to the TDM bus will get truncated or rounded (I don't know which) to 24 bits first.

>> Could you use a flat insert EQ that Dithers to 24 bits over the stereo bus master fader to take care of the problem before truncation? <<

It wouldn't help because the truncation happens inside of the mixer plug in, before there's a chance to do anything about it.
Dithering after truncation does no good... it has to be done to the wide signal inside of the mixer plug in before it gets sent back to TDM. This one's for Digi to fix... there's no way to do this at home.

Monte McGuire

Disco_Doctor
11-14-1999, 03:48 PM
I've done thousands of hours of drum edits over the years, and probably had to crossfade a total of ten times, and then only on the overheads...Also, you can usually leave little gaps of silence, even in the overheads, and no-one will hear it.

?!!!?? I'm surprised to hear you say that Speer! I agree that crossfades aren't always necessary on the kick and snare tracks - but "gaps of silence in the overheads"!? I've never been able to do that without it being very audible. I don't want to derail the original topic of this thread...I was just a little surprised by that comment. I guess my answer is yes, my drum edits have pretty much always needed meticulously applied crossfades.

Thanks for the test results Rail! Very interesting stuff! I imagine if the same test was done with say, 24 tracks of mix stems - that would really begin to tell the tale.

One other thing that would be interesting to try (hint hint!): The same test, but use a Filterbank input gain fader instead of the Pro Tools mixer faders...

http://www.digidesign.com/ubb/images/icons/smile.gif

Rail Jon Rogut
11-15-1999, 02:00 PM
Speer

I have also done drum edits without crossfades, however that takes a lot longer - and sometimes the budget or the time constraints of the session don't allow that, so at times I've resorted to creating batch crossfades. This may not be the optimal method, but when I do have to use it - the software should be able to handle the workload.

DiscoD, I'll have to try and get back there to do that test -- shoulda done it when I was there (I know, I know).... I'll see what I can do.

Mr. Mediator, you seem to agree with our conclusions... I personally do still use a 9k for mixing (given the budget), but I would like to see Pro Tools actually aproach its full potential.

Best regards to all.

Rail

------------------
Recording Engineer

Jonny Atack
11-17-1999, 03:55 AM
Sorry to be off-topic, but I have to agree with Speer on the drum editing. If you cut the drums just before the attacks, cross-fading is almost never necessary. And as far as I know, cutting a 1-sample hole of emptiness between two overhead tracks is the fastest way to get rid of pops if you're using the Auto-Fade function in the PT prefs. I don't believe anyone can hear a one-sample hole in the overheads. I can't, and it's far faster than doing crossfades that don't always solve the problem the first try.

Rail, I assume that you manage to work more quickly by avoiding the many zoom changes involved in precision editing i.e. by rough cutting and then fixing the pops afterwards. But do the batch cross-fades get all the pops out? Does that work well?

Thanks again for the tests you made -- great stuff.

Best Regards to all.

blairl
11-17-1999, 09:34 AM
Back to the issue of Pro Tools sound quality. I'm very interested to see the "test" posted so we can do some listening tests. In the meantime, I have done some research on the topic. It seems that all of the major digital consoles use 32 bit floating point processing which is maintained at 32 bit until the final output where it is dithered down to 24 bit. These major consoles include the Capricorn, Axiom, and Oxford. Sonic Solutions maintains a 48 bit path throughout the entire process until final output where the signal can then be dithered to 24, 20, 16 or whatever you choose. Pro Tools rounds the signal to 24 bit each and every time the signal passes through the TDM bus. I am aware that the math says that the noise from rounding or truncation is much too low for humans to notice, but somehow it seems that people are noticing some changes in the sound quality. This question is for digi engineering. Is the 24 bit TDM bus a hardware issue or can this be changed? Sonic Solutions has long been considered an excelent system when it comes to sound quality. Why not shoot for the best and maintain that 48 bit path throughout? There must be something to this when digidesign is the only major manufacturer that does not use 32 bit floating or 48 bit fixed processing. I am aware that Pro Tools is considered to be 48 bit fixed, but the TDM 24 bit bus issue seems to cancel this out. Can these things be fixed with software solutions or is all of this a hardware limitation?

[This message has been edited by blairl (edited 11-17-99).]

11-17-1999, 01:09 PM
Those who say they use no fades for drum edits, do you sync you PT back to to 2"? (I think not!). That is where click n pops really show up, isn't it?
Jules

LGiordano
11-18-1999, 06:38 AM
blairl wrote:

all of the major digital consoles use 32 bit floating point processing which is maintained at 32 bit until the final output where it is dithered down to 24 bit.

Even the 02R has 32-bit processing. The manual is vague about whether its fixed or floating, but either way, its better than 24-bit fixed.

the math says that the noise from rounding or truncation is much too low for humans to notice, but somehow it seems that people are noticing some changes in the sound quality.

That's because they're using their EARS, not math. I think its fair to say that most of us do not have an agenda to pick holes in PT. Its exactly the opposite, we want it to be better than anything else, not worse. The editing model is really good, and the sound quality has to catch up.

This question is for digi engineering. Is the 24 bit TDM bus a hardware issue or can this be changed?

I think the real decision that Digi engineering has to make is not CAN it be changed but SHOULD it be changed. There's probably a conservative faction lobbying for keeping it the way it is, and a progressive group gunning for change. In any case, they can ignore us, passively listen, or actively solicit our feedback. For the survival of the company, the latter seems the obvious choice. I hope that's the way they go.

creaturestudios
11-19-1999, 01:37 AM
Just a half a thought...

People, let's not forget that Pro Tools can do 4 things. Recording, editing, mixing, mastering. Doesn't mean you have to.

Is it the best recorder out there?
No. If you want to use it for recording go ahead. Just make sure you have all the extras. Awesome mics, mic pre's, compressors, eq's etc.

Is it the best editing software out there?
There are others but it's up there.

Is it the best for mixing?
From this thread you can tell that it's not.

Is it the best for mastering?
No. But many have used it for that purpose with great results.

Whether you use Pro Tools for preproduction or as an editor or for your two track master let's not forget that it is not the best thing out there nor does Digi claim it to be. If you've heard a better mixer. Kudos!!!
Go use it. Doesn't mean you can't use your system for something else. Remember it can do 4 things. Sure I'd like it to sound better. I'd also like it to wash my car walk my dog and pay my bills. Let's be reasonable.
For under $30,000 you can get alot done on your Pro Tools system. But let's not forget those high quality beatles recordings. Those were oh so much better than what people are producing today using Pro Tools. Let's face it guys. We can talk technical till were all blue in the face. If the performance wasn't there to begin with, it didn't matter what equipment we're using. Try telling Ricky Martin or Nine Inch Nails (both ends of the music spectrum) not to use Pro Tools because it sounds thin. They'll laugh. They're making millions. If you think Pro Tools sounds like crap then that's some of the best sounding crap that I've heard in along time. Except for Ricky Martin. I'm not really into that.

Peter Cosmic
11-19-1999, 04:28 AM
I guess the problem is not that most people think ProTools is crap. It's just that
ProTools doesn't seem to aproach it's
full potential, like Rail Jon Rogut stated before.

The hardware technologie available today is getting faster and faster so the integration of a higher resolution algorithm for the Mixer seems to be not so much a problem of missing computing power any more. Especially if you think of the newer dsps and the power of G3/G4s.

nipple
11-19-1999, 10:23 AM
>>I guess the problem is not that most people think ProTools is crap. It's just that
ProTools doesn't seem to aproach it's
full potential, like Rail Jon Rogut stated before.<<

Doesn't reach it's full potential? What exactly is it's full potential and how will anyone ever know when it is reached? We all have these lofty, wispy musings about what "Pro tools could be". If only this, if only that - blah blah blah. Name one other DAW that you would rather work with. Name one other DAW that is as close to a complete audio solution as Pro Tools. Try doing a complete album in Cubase, Studio Vision, Logic, DP, Soundscape, Sadie, Sonic etc... Just go ahead and try. I promise that you will come screaming back to Pro Tools. Now I'm not talking midi here although I've use PT 5.0 (beta) and it's a nice start.I don't seem to have problems with PT being thin, it's just a different animal. Learn how to work your machines people or go back to analog boat anchors or better yet your ADATs with a Mackie board - now there was a great sounding combo(sarcasm). Oh yeah, how much is that Oxford in the window? Can I edit my drum tracks with it? Can I record anything on it without an external tape machine? How much did your Pro Tools rigs cost?

Steve MacMillan
11-19-1999, 11:50 AM
nipple,

You are totally right, but maybe you should switch to decaf.

http://www.digidesign.com/ubb/images/icons/smile.gif

Jonny Atack
11-19-1999, 03:29 PM
Dear Mike and friends at Digi,

Any news on that DUG-wide sound quality test you folks were preparing last August/September?

I'm sure you're more than a little busy with the many current prod releases, but after getting us all excited with your sound test plans it would be great to actually see (hear) it.

Thank,

Jon

Jonny Atack
11-19-1999, 03:42 PM
On second thought, if preparing the test would mean shifting resources away from development of the top-secret, soon-to-be-released higher-res TDM mixer plug...

...then keep up the good work! All the best.

[Benjamin]
11-21-1999, 03:17 PM
My worries are of a different kind, I think. Basically, I just wanna work out how to get optimal results from my gear, whatever it is, and avoid pitfalls. With Protools this means, for instance, How do I get the best possible 2-track mix? Bouncing and letting protools process from 24-bit to 16-bit? Or do I dither straight to 16-bit as I bounce, and which dithering should I use? Or, if there's an issue with protools bounces, should I record in a background application? or on an external machine? And so on.. I don't really wanna complain too much about protools, but I want the knowledge to do the best I can do with it.

------------------
[HYPERSONIC]

Disco_Doctor
11-22-1999, 02:40 PM
I don't really wanna complain too much about protools, but I want the knowledge to do the best I can do with it.

Definitely! Hence the need for Digidesign to publish detailed specs on the internal workings of the Pro Tools mixer.

http://www.digidesign.com/ubb/images/icons/smile.gif

Bernie Frantsits
11-22-1999, 04:37 PM
Definitely! Hence the need for Digidesign to publish detailed specs on the internal workings of the Pro Tools mixer.

Definitely! Hence the need for Digidesign to publish detailed specs on the internal workings of the Pro Tools mixer.

Definitely! Hence the need for Digidesign to publish detailed specs on the internal workings of the Pro Tools mixer.

yes yes yes, protools mixing is no fun!

I worked for about a year in 5 different Studios here in Germany with the same result as above. every user should be happy!

bernie

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

Rail Jon Rogut
11-23-1999, 07:18 PM
nipple, I think you're missing the point. Pro Tools is the best editing tool available and is at the top of the totem pole for its editing features, no argument... however when it comes to sound quality it's near the bottom of the pole compared to every other DAW on the market. Sure it's better than mixing on a Mackie analog board, but that's not where Digi have targeted their audience (in their advertising) -- they state that "a Pro Tools/ProControl system runs circles around consoles and recorders costing 5 or 10 times more". Functionally maybe, not from a sonic standpoint. As for its potential, we know that it has the potential to sound better by not truncating the signal anywhere throughout the signal path, so what we're asking for is not beyond the realm of reality. The problem is that Digi has to be able to deliver this without losing their track count or using up too much DSP.

I respect your attitude, and if you're happy with the Pro Tools product as it currently exists - then fine. However for those of us who actually wish to be able to use the product to its full potential - I would suggest you sit back and accept any improvements that are finally made to the product brought on by our diligence.

------
Jonny Atack wrote:

Rail, I assume that you manage to work more quickly by avoiding the many zoom changes involved in precision editing i.e. by rough cutting and then fixing the pops afterwards. But do the batch cross-fades get all the pops out? Does that work well?

Jonny, I've actually programmed some QuicKeys which I use and I can do the edits really fast -- if I then go and do a batch fade across all the drum tracks with the xfade set to centered and between 10 and 20 msec it works fairly well. I then listen down to the drum tracks and if I hear anything wrong - I go back and adjust the xfade or edit to fix the problem. This shaves off at least 2 hours of zooming in and adjusting each edit so that no xfade is required.

Best regards.

Rail

------------------
Recording Engineer

mariusdevries
11-30-1999, 05:06 PM
nipple: "Try doing a complete album in ....... Logic ...... Just go ahead and try. I promise that you will come screaming back to Pro Tools........"

hmmm....I did, and I didn't.....

Steve MacMillan: " nipple, you are totally right, but maybe you should switch to decaf..."

i did this too. no significant improvement.

Regards. mdv

11-30-1999, 05:44 PM
I'm sold on PT & I bought a Pro Control. how can we get Digi to improve the sound quality?? http://www.digidesign.com/ubb/images/icons/grin.gif http://www.digidesign.com/ubb/images/icons/grin.gif
we need to move forward here, most threads don't come up with new angles now
http://www.digidesign.com/ubb/images/icons/frown.gif

Jules

tnie
12-02-1999, 10:03 PM
I hope they improve the sound quality by the time Mac comes out with the G5 and system 10, because many companies are doing the same business. Pro-tools Mix is just a project type DAW it has tons of bells and buttons, but when it comes down to sound quility it always falls short compared to let say Sonic Solutions. :P http://www.digidesign.com/ubb/images/icons/blush.gif

KymataDigital
12-03-1999, 09:17 AM
96K PT to be seeded to developers in 3rd quarter.

You didn't hear this from me.

Bryan
12-04-1999, 06:41 PM
Assman, is that you?

DaveG
12-04-1999, 11:59 PM
I'm a musician, not an engineer, but at least in a very broad sense I do understand the issues being raised here. I just have a couple of things to toss into the discussion.

First, obviously everyone is talking about the 24-bit optimized mixer in the full-blown Pro Tools/24 system. Now that the Pro Tools 5.0LE software is going to support 24 bits without Pro Tools hardware, when operating in 24-bit mode what kind of mixing algorithm will LE employ? Will it be similar to the one used for 24-bit sessions in previous versions of Pro Tools, or does the 24-bit optimized mixer (with 48 bit internal resolution) need the processing "oomph" from Pro Tools/24's dedicated DSP chips?

Second (this isn't actually a question), for what it's worth I'd be glad to lose some of the convenience of totally realtime operation if it meant a gain in fidelity. It seems that in a completely software-based environment like Pro Tools LE you could have one digital mixing algorithm for realtime track playback, which one could use as a reference while mixing. And then have an offline, non-realtime mixing algorithm for doing test and final mixes that makes no compromises and maintains big wordlengths as it computes the result of all your digital tracks summed along with EQ and plug-ins. I for one would be willing to let the computer sit there and number-crunch away for half an hour mixing a 5-minute song if it meant a substantial increase in the smoothness and transparency of my final mixes. (I know that might not fly in pro studios under pressure of time, but sometimes good things come to those who wait...)

Kenny Gioia
12-05-1999, 09:57 PM
The biggest problem I'm having with all of this is that I already own one Pro control.

I'm buyin another for studio B and expanding A with two fader paks for 24 faders. I just don't want to spend another $20,000 if I should just buy a couble of 02R's

Dave Lebolt
12-09-1999, 11:06 AM
Hi folks,
Sorry it's been a while since you've heard from us on this issue. Things have been very hectic around here getting Pro Tools v5.0 out the door, and many other things. For those of you who don't know me, I'm Dave Lebolt, Director of Product Strategy at Digi. I work with the engineers and product marketing teams to design our products (including the feature details), and look out for our future plans as well.

First, I wanted to pass on to you all that Mike Rockwell, who's been involved in this (and many other) discussions on the DUC has moved on to work at Avid headquarters. Mike was a key contributor to Pro Tools' success, and I've worked closely with him for close to 6 years. Now he'll be working over at Avid helping them with everything Avid does (which includes everything we do!). Luckily, we've got a lot of great engineers here at Digi who have been with us for a long time, and they'll be on the DUC now and again, as will product managers (who are in charge of their specific products). Also, you should know that while Mike had contributed to this discussion thread, he did not work on the DSP code for our mixer, so we'll have no problem dealing with ongoing development in this area.

This will be a long note, so let me summarize for those of who have been following this thread. People who are interested in the details can read the whole thing…

- We've developed a new mixer Plug-In that does the usual 48-bit accumulator thing, but dithers any output stream to 24-bit. This includes any output pair or post-fader send. A price is paid to support the dither, as you'd expect: the largest mixer that can be created on a single DSP with this mixer is 47x2, as compared to 59x2 with the undithered version that we send out today. This is a decrease in DSP efficiency of approx. 20%.
- The Customer Service group will post 2 versions of TDM mixer for you guys to play with (they'll give you a link to an FTP download, or let you know the details via this thread in the next day or so). One will be the new dithered mixer (let's it "Mixer A"). The other one will be a special version of the same one we ship (let's call this "Mixer B"). But this special version of the shipping "24-bit Optimized" mixer will be set so that it is limited artificially to a maximum of 47x2 inputs. This will allow you guys to do basic blindfold testing by swapping the mixers in and out of the DAE/Plug-Ins folder in the Finder. (Of course, we won't be telling you which is the dithered mixer. Some of you clever folks out there will find out ways to see which one is which, but let's not ruin this for others who just want to see if they hear a difference without knowing!)
- I've done some blindfold tests here, but I figure we should just give this mixer out to you to see if you hear a difference yourselves… I think this better than our doing any tests here and just posting files to you… We'll probably ship this mixer out in our next release (shortly), but we wanted you guys to have a chance to do some blindfold testing, and tell us if you hear a difference.

Then, hopefully, our Web group here will set up a way for you folks to e-mail in some of your test evaluation files for us and others to listen to. If and when we do this, please list all the steps you performed to create these snippets. I'm sure we'll all be interested in the results.

Here's a tip about how to set up a good A/B test:
1. Create multi-track mixes with different program material. (I say multi-track, because this is where people said they heard a difference most… with multiple tracks and gain changes.) To be as objective as possible, don't use any DSP Plug-Ins.
2. Bounce the mixes to 24-bit stereo files using the two different mixers.
3. Import the results (the two pairs of dual-mono files from the Mixer A and Mixer B bounces) into a new session. Line them up on two grouped track pairs.
4. Go to Preferences/Operations, and uncheck the option that is normally enabled for Solo buttons to "latch." This allows you to instantly switch between two mix stems using the solo buttons.
5. Listen back, and switch soloing between the pairs to try to I.D. any differences.
6. With your new "comparison" sessions, since you're not changing gain (setting faders at unity), there should be no difference between the two mixers… use either one to check the results.
7. Perform a blind-fold test: have someone switch the results without your seeing (or hearing) them switch. If you want it to be really honest, the operator will not even switch all the time, and certainly not when "cued" by you as the listener. This forces the listener to actually identify when and if the sound has changed. If you're not accurately identifying a difference greater than 50% of the time, you're statistically not hearing any difference. (I've been tripped up on this a few times… try it!)

Now, On to the Details…
There have been a lot of posts here that vary from the subjective to the objective. It would take too long to reply to each and every one of the points brought up, but I'll try to answer a few. I'm putting this information out to be helpful, and have made it as factually correct as I know how, so bear with me…

32-Bit Floating Point vs. 24-bit Fixed Point
Some of you asked whether or not 32-bit floating point is just plain superior to 24-bit fixed point (however it's implemented). I know that many of you may know about all this stuff, but there are many readers of this forum, so I feel I should give a bit of an overview on this subject for those who don't.

From what I've been able to learn, a well-designed 24-bit fixed point digital audio system is *equal to (or better than)* a well-designed 32-bit floating point system. This issue is difficult to educate people about, because it's a bit like trying to differentiate between Intel CISC processors (used in PCs) and PowerPC RISC processors (used in Macs). Some people tend to look at the number "32" and say it's bigger than "24" and they're done. But it's not so simple...

32 is Better Than 24, Right?
Not really... Remember, we're not strictly talking about 32 bits representing the audio signal here *(the bit depth or word length).* The signal recorded to disk and sent in and out of the I/Os on all of these DAW or mixing systems is still 16- or 24-bit (not 32-bit). Instead, we're talking about how the programmer works with the math results when they're performing mixing and processing operations.

What Does "Floating Point" Mean?
Floating point arithmetic allows a programmer to represent the result of complex equations so that the number of digits representing the exponent and mantissa can "float" or "move." This translates to a more flexible scheme for representing very small numeric values (values between +1,0, and -1) which are the remainders from their calculations. It does not necessarily represent greater numerical precision, however (for example, we hold on to 48 bits of precision with our fixed point math until we truncate it on output).

Floating Point vs. Fixed with Digital Audio
In a digital audio system, most every time mixing or processing occurs the result of the calculation will be a long integer (another way to say "big number"). When working with 24-bit audio, those numbers get pretty large, because the results of the calculations (if listed in standard base 10) would have many places after the decimal point. A properly designed audio system will try to maintain as much accuracy as possible when manipulating the bits, to avoid losing resolution. Loss of resolution can result in poor audio quality. (Some changes would not be audible to most people, but could be audible to "golden ears" people.)

The programmer attempts to "hold onto" the remainders from these equations as accurately as possible as more and more them take place. At some point, however, they've got to do one of two things:
- Truncate or round the result (cut off the numbers that they can't "hold onto" anymore)
- Apply dither to the result and then round to the lower number of bits. Dither injects a low-level non-correlated noise into the signal to avoid distortion artifacts caused by quantization, by reducing those artifacts to white noise. Repeated applications of dither will raise the noise level (as discussed by earlier notes on this thread), but proper use of dither can offer better results.

This process happens in all systems, whether fixed or floating point. Remember, the programmer has to eventually round, truncate or dither the numeric precision they've been "holding on to" when they send a file to disk or out of the system. This is because the word length is fixed at 24 (or 16) bits. And, keep in mind that audio that comes in to converters on a Pro Tools system and gets sent to disk, is sent there unaltered…

Our DSPs vs. Their DSPs
In our system, we use the 56-bit accumulator that is in the Motorola DSPs we use to store values as we do the math. This means that you can get a very high level of precision. The "24-bit Optimized" mixer that we have shipped since Pro Tools|24 was released stores 48-bits of precision in our mixing operations. This allows for high headroom (close to 28 dB). At the point when the data is placed on a physical output or into a submixed aux input or send, the output is truncated to 24 bits of precision. This new mixer that we're giving out for "golden ears" evaluation *dithers* the output streams instead.

In many competitive systems, there can actually be less precision available when performing certain operations. The register system used in, say, a SHARC DSP is very different from the Motorola system. It is difficult to directly compare the differences between the two (different chip architectures and fixed point vs. floating point). In a lot of ways it's up to the skill of the programmers putting the system together to get the best result.

I've seen some pretty outrageous claims on the internet (including our DUC, but not here in this thread), about the Motorola DSPs we use being really old technology or not having real capability. This is patently untrue. And, fixed point Motorola DSPs are used in many excellent products including ours, Sonic Solutions, Eventide, Z-Systems, Soundscape, Lexicon, and many others. Sonic Solutions has made their mark in the mastering market, which is most critical about audio quality, and Motorola DSPs (and fixed point math) are used. And many products on the market that use SHARC DSPs and advertise using large numbers of them, are in fact, using older, cheaper versions of that DSP family. We are using a "state of the art" DSP family design.

Clear as mud, eh? The bottom line will always be (I hope): how does it sound!??

And the bottom line is, we care about getting the highest audio quality possible. We've created this mixer at your request, and hope that there is an audible benefit (or that it will help set your mind at ease). You'll be interested to know that I spoke to a well-known music mixer who is the owner of an Oxford (and a lot of other stuff, including Pro Tools). He tells me that, to his knowledge, the Oxford is 32 bit fixed point, and truncates the result to 24-bits. There are dithering options on the outputs only (perhaps they're dithering internal bussing; he didn't know for sure).

What's Audible?
This is pretty subjective area, but I think it's worthwhile mentioning (to the non-tweakheads that are reading all this stuff): the effects of truncation (as described above) is "way down there," and is theoretically beneath audibility. 24-bit audio yields a dynamic range of 144 dB. If we agree normal human listening levels don't normally exceed 120 dB (the threshold of pain), we are leaving 24 dB of headroom to play with. This means that any effects of truncation, (and any sidebands that may be generated), should be below the threshold of audibility.

Try this simple test. Take a good pair of headphones and a very clean amplifier source (or use some hi-quality monitors). Turn up some program material until your ears say "stop… loud enough." This is the loudest level you want to expose your ears to (safely). Now, loop over an area of the material, and lower down your fader until you can't hear anything at all (or it's dropped into the noise floor of your excellent monitor system). (Pro Tools lets you see this level by Command-clicking on the fader to read HR peaks. You can reset this peak by clicking on the indicator area.) It's interesting to note that most people can't really hear levels below -90 dBfs or so. This is 6 dB less than the theoretical dynamic range of a CD. As Mike Rockwell noted earlier, today's 24-bit converter technology, (even on the most expensive custom converters), doesn't really produce any more than about the equivalent of a 21-bit result (the lowest 3-bits are "dancing in the noise"). Now, consider that the truncation results we're talking about are all hovering beneath this level, and you'll understand why I'm mentioning it.

Now please don't take this to mean that I believe that you guys are not hearing anything… I'm going to do the Oxford test myself with a big mix as soon as I possibly can (probably past the New Year). We've all heard about situations in "audio history" where specs disagreed with what people could hear. And if there's a difference we *want* to know… I'm just pointing out that on a practical basis, the levels at which truncation (and any generated artifacts) occur *should* be inaudible.

One Other Tip for Comparing Audio Files…
When comparing systems, try this simple test… If you can't do a "bit for bit" compare using Sonic or Sound Designer II or something, just follow these steps to see if there are differences:
1. Create final mixes produced on any two systems (with all other things being *exactly* equal, such as fader levels matched to within .1 dB, etc.).
2. Import them into your DAW editor of choice. Line up the files in pairs against each other so that they're lined up *exactly* to the sample.
3. Flop the phase on one of the pairs. If the sound sources are identical, you'll hear nothing at all (because if they're the same, and of opposite polarity, they'll cancel each other out). This is a more "scientific" way to tell if you're hearing anything… it's an old audio trick from way back… try it. You can do this in Pro Tools using any Plug-In that offers phase buttons, including TimeAdjuster. I like to do this by bussing to a stereo aux. input (each pair to its own aux.). Be sure that you place the same PI on the tracks (to match delay!) Or, if you don't want to run through any Plug-Ins, you can use the Invert A/S Plug-In.

Metering
On 10/8 Speer mentioned that he wished that O/L indicators came "on" for him on master faders at -3 or -5. Speer, this is because the meters on master faders read both the signal on "input" (to the mix node) and output (of the master). If you pull down the master fader so that the clipping indicators extinguish, you'll get exact and accurate input clip indication. Which brings me to…

Pull Down the Master to Trim or Adjust the Faders?
This was mentioned several times. This is one area that is *not* the same as an analog mixer. Believe it or not, it will not make a difference whether or not you lower the level of all channels strip faders feeding the 2-bus or lower the master fader. In fact, we recommend running your faders at normal levels (the sweet spot), and trimming down the master.

If you perform the exact same gain changes, the results (sample to sample) are *numerically identical.* (I don't know about you, but numbers being equal to other numbers means "the same" to me.) Why is this true? If you're adjusting gain, and storing a 48-bit result (as we are), then you can do this with absolute precision and with no loss of data integrity. The truncation to 24-bits in the shipping 24-bit Optimized Mixer occurs *after* the gain adjustment on the fader. We've set up the Master Faders so that the meter monitors clips on the input *or* output side (as mentioned above).

Hope this answers some of your concerns, and looking forward to your impressions about the dithered mixer.

blairl
12-09-1999, 01:30 PM
Thanks Dave! I'm exited to try the new mixers.

I have a question about what you said. You mentioned that Pro Tools maintains a 48 bit signal until it hits a physical output or submixed aux input. I believe your exact words were,

"At the point when the data is placed on a physical output or into a submixed aux input or send, the output is truncated to 24 bits of precision."

Then you said,

"We've developed a new mixer Plug-In that does the usual 48-bit accumulator thing, but dithers any output stream to 24-bit. This includes any output pair or post-fader send."

By the way I'm only quoting so people that read this post can get a quick understanding.

My understanding has always been that the signal is not only truncated at the points that you mention, but that it is also trucated each and every time it passes through the TDM bus by using inserts or plug-ins as well. So if I have a vocal track that contains say Q10, C1, De-Esser etc..., each time it passes to one of these plug-ins and comes back, the signal is being truncated (or rounded). Is my understanding incorrect? I hope so. The reason I mentioned these plug-ins is that I know there are several plug-ins that automatically dither back to 24 bits, but most of them don't.

If it is true that the signal is truncated each time it passes through a plug-in, is the new mixer you have developed going to address this issue? Or is it only going to dither at the final output, aux input and send stages?

Thanks for your time and consideration!

Dave Lebolt
12-09-1999, 02:59 PM
Hi Blair,
Thanks for you note. Your impressions about what happens are correct. When I made that statement, I was referring only to the mixer plug-in itself. The mixer plug-in can't change the data "width" of the TDM bus, but it *can* dither down the data (from the 48-bit result that is stored by the mixer). That dithered result is then sent to a send or physical output.

As has been brought up previously, we have a 24-bit TDM bus, and when passing audio from one Plug-in on an insert to another Plug-in on an insert, the data path is 24-bits. It's then up to the plug-in to do any dithering (if desired). Keep in mind that this is issue is not tied to "double-precision" arithmetic that some plug-in developers use (where they say that they get 48 bits of precision). They still have to truncate, round or dither to get to 24-bits when the signal leaves the plug-in itself.

The signal *is* reduced to 24 bits, but as I've indicated, most people do not feel that this is a problem. I've pointed out why that might be a valid point of view in my post above, when discussing our ability to hear any results of the truncation (as any truncation results and sidebands generated are technically "below the level of audibility," i.e., accounting for headroom, greater than -100 dB down).

As you know, it's true that you can design DSP Plug-ins that dither on the output, but this will definitely cost quite a bit of DSP power, which translates into less instances of that plug-in per DSP. And, it will also be increasing the noise floor (although by a small amount. However this does accumulate through the mixing process or use of many mix points or inserts). By the same token, one can use higher sample rates (like 96 or 192 kHz), which will also greatly diminish the available pool of DSP power available for use in a given system design.

I would suggest performing some of the tests I've described above with no plug-ins (just the mixer) and see if you're hearing a difference with a blindfold test. If you (and others) honestly are, output dithering in the realm of plug-in design may become a more important, both to our in-house plug-in developers, and to third-party developers. Right now, many people (including many hi-end customers) are happy with the results they're getting with the present system.

We all know that audio performance is always evolving and improving (we hope!). Let's see what some listening tells us. I'm sure you'll agree… our ears, (unprejudiced by knowledge of which source we're hearing or influenced by other's opinions in a room), are our best sources of judgement.

Jules
12-09-1999, 05:07 PM
If we like it, can we keep it? Or will it time out? Is it a fully functional mixer with midi too? PT 5?

Jules

Auditioning the results inside PT seems pointless Surely a better test would be 2 mixes to an MO device,. Some folks have 2 PT systems, side by side would be another good test.
Does the source material need to be "fresh" not, ruined http://www.digidesign.com/ubb/images/icons/wink.gif already by Pro Tools TDM bus? http://www.digidesign.com/ubb/images/icons/grin.gif

Jules


[This message has been edited by Jules (edited 12-09-99).]

Dave Lebolt
12-09-1999, 05:49 PM
Hi Jules,
Yes, it's fully functional. No hit on any other area like MIDI, and I don't believe it times out. We'll probably be releasing it with a newer maintenance release for Pro Tools v5.0 in the first half of next year.

Re your comments about having to audition outside of Pro Tools, feel free, but auditioning outside of Pro Tools should not be necessary, since you'll be listening to files that are dual mono 24-bit. This means that if you place the faders at unity, you'll be performing no gain change to 24-bit files, and thus no truncation or bit modification will be taking place. So either mixer (not knowing which is which) should be o.k. here.

Also, files that you've recorded into Pro Tools via a disk track input are not messed with in any way... there's no truncation taking place when recording straight to disk (as mentioned earlier on this thread). You can use any files that have passed directly to disk from an input without any concern that they may affect your mix evaluation testing. Only 24-bit sources that are being summed together are affected (e.g., summing several mics via a stereo aux to 2 disk tracks).

[This message has been edited by Dave Lebolt (edited 12-09-99).]

[This message has been edited by Dave Lebolt (edited 12-09-99).]

Disco_Doctor
12-10-1999, 01:05 AM
Hi Dave!

First off - thank you! I figured this was put on the back burner forever with Mike leaving and the hectic PT5 schedule and all that...it sure is great to hear you guys have been working on this!

We've developed a new mixer Plug-In that does the usual 48-bit accumulator thing, but dithers any output stream to 24-bit. This includes any output pair or post-fader send.

I want to be 100% clear about a how things work on this new dither-mixer in the context of a real world session so I know for sure what testing will be valid and what won't be valid. I know you're busy, so if you could just answer YES/NO to these questions when you have a chance, I'd really appreciate it!

I have a stereo pair of audio channels, and I group the faders and pull them down -1.5db, and send the output of those channels to TDM buss 1-2.

Question #1) Is dither applied to the output of the audio channels prior to the signal reaching TDM?

I create a stereo aux, and set the aux input to buss 1-2 to receive the signal. The output of the aux is 888/24 Output 1-2. I lower the output of the aux -1.5db.

Question #2) Is dither applied to the output of the aux feeding the master buss?

I create a master fader for 888/24 Output 1-2. I lower the output of the master fader -1.5ddb.

Question #3) Is dither applied to output of the master fader prior to the signal going back onto TDM to pass through the final master fader inserts?

Question #4) The last point in the signal path - the last insert on the master fader: Presumably, no dither is applied here, right? I apply my own final stage dither as usual?

I add a stereo POST fader send to the above mentioned aux and set the level to -1.5dB. I route the output of the send to TDM buss 3-4.

Question #5) Is dither applied to the output of the POST fader send prior to reaching the TDM buss?

I add a stereo PRE fader send to the above mentioned aux and set the level to -1.5dB. I route the output of the send to TDM buss 5-6.

Question #6) Is dither applied to the output of the PRE fader send prior to reaching the TDM buss?

Question #7) Assuming that the answers to questions 1 through 5 are true (never mind #6): If I open a session with the new dither-mixer, and all the plugins in the session are plugins that dither their output coming off the TDM buss, it would be fair to attribute any audible differences to the mixer...?

Question #8) VERY IMPORTANT: Does the "dither preferences" setting in the prefs panel affect the dither depth that the new dither-mixer is using?

Question #9) Lastly, will the new dither-mixer work with PT5?

Thanks again Dave!

http://www.digidesign.com/ubb/images/icons/smile.gif

blairl
12-10-1999, 08:51 AM
Hi Dave:

I've been thinking all night about this new mixer plug-in and I was going to ask some additional questions when I got in this morning but it looks like Disco covered it.

I do however have one additional question. Right now the data "width" of the TDM bus is 24 bits. Is there any chance of making a TDM bus that maintains 48 bits in and out? Would this be a hardware or software issue?

Thanks!

Barnaby Bristol
12-10-1999, 03:04 PM
It would be both a hardware and software issue to create a 48bit TDM bus. One design advantage in expensive digital consoles is maintaining wide data paths throughout and handling any bit reduction properly.

I look forward to the dithered mixer test. It isn't going to be an easy test because I don't think it's going to be a clear, "oh, this sounds better than that" issue. Over the course of production the benefits add up. Of course you can't A/B entire productions because you're making decisions throughout the production's entirety based on the tools you're using. You'd make different decisions based on what you hear.

I do believe it will be an improvement. The products of undithered truncation are about as alien to our auditory systems as you can get. And I'm continually impressed by how discriminating they are.

Oh. And a dithered mixer plugin will probably force me to buy another MixFarm. Let's burn some DSP!

Barnaby

[This message has been edited by Barnaby Bristol (edited 12-10-99).]

Rail Jon Rogut
12-10-1999, 03:27 PM
Hi Dave

Thanks for addressing this subject.

I too have a question. With the new dithering mixer, if we have a simple session with a single stereo pair of tracks panned hard left and right without any fader moves - is the algorithm inteligent enough to know that the bit depth hasn't exceeded 24 bits, and therefore no dither is required? IOW, can we go through the system (as we can now) without changing the data stream?

Have you asked Mick to do any tests to compare this new mixer with his Oxford, and if so could you share the results?

Best regards.

Rail

------------------
Recording Engineer

Disco_Doctor
12-11-1999, 03:47 AM
is the algorithm inteligent enough to know that the bit depth hasn't exceeded 24 bits, and therefore no dither is required?

Excellent question...thanks Rail!

http://www.digidesign.com/ubb/images/icons/smile.gif

Dave Lebolt
12-11-1999, 06:54 AM
Here's a reply to Disco Doctor's questions:

I have a stereo pair of audio channels, and I group the faders and pull them down -1.5db, and send the output of those channels to TDM buss 1-2.

Question #1) Is dither applied to the output of the audio channels prior to the signal reaching TDM?

Yes. It is applied after the master fader that adjusts gain on that buss (if any).

I create a stereo aux, and set the aux input to buss 1-2 to receive the signal. The output of the aux is 888/24 Output 1-2. I lower the output of the aux -1.5db.

Question #2) Is dither applied to the output of the aux feeding the master buss?

Yes. Same as #1

I create a master fader for 888/24 Output 1-2. I lower the output of the master fader -1.5ddb.

Question #3) Is dither applied to output of the master fader prior to the signal going back onto TDM to pass through the final master fader inserts?

Yes. The fader adjusts gain at the summing point of the mixer. Then it's dithered.

Question #4) The last point in the signal path - the last insert on the master fader: Presumably, no dither is applied here, right? I apply my own final stage dither as usual?

Yes.

I add a stereo POST fader send to the above mentioned aux and set the level to -1.5dB. I route the output of the send to TDM buss 3-4.

Question #5) Is dither applied to the output of the POST fader send prior to reaching the TDM buss?

Yes. The mixer services any "mixed summing point".

I add a stereo PRE fader send to the above mentioned aux and set the level to -1.5dB. I route the output of the send to TDM buss 5-6.

Question #6) Is dither applied to the output of the PRE fader send prior to reaching the TDM buss?

Yes.

Question #7) Assuming that the answers to questions 1 through 5 are true (never mind #6): If I open a session with the new dither-mixer, and all the plugins in the session are plugins that dither their output coming off the TDM buss, it would be fair to attribute any audible differences to the mixer.

Yes, very much so. BTW, you say, "dither the output off the TDM buss." Perhaps it would be more correct to say "dither the output of the whatever insert they reside on." This is because the signal is passed as 24-bit directly from TDM (the insert "output" to the plug-in). The output of the plug-in is where the dither would be taking place.

The point at which signals are mixed together and sent to physical outputs or sends is where dither is applied with this prototype evaluation mixer. For example, if you have several signals bussed into a stereo aux. input, the signals are dithered by the mixer when they are summed and placed up onto the TDM buss to be routed. If you then apply a plug-in, (if the plug-in uses dithering), then the plug-in's dither is applied at the output of the insert that the plug-in resides on. Since the audio stream has to hit the *input* of the plug-in first, the dithering from the mixer "does the right thing," (if one believes that these all of these things are making an audible difference).

Question #8) VERY IMPORTANT: Does the "dither preferences" setting in the prefs panel affect the dither depth that the new dither-mixer is using?

No. The mixer is not affected by that preference. That preference refers to AudioSuite processing only, and is there because people working in 16-bit sessions and file-processing their material with AudioSuite plug-ins would want any processing that involves gain changing to dither to the current bit depth of their source audio files.

Question #9) Lastly, will the new dither-mixer work with PT5?

Yes.

Dave Lebolt
12-11-1999, 07:08 AM
To BlairL: both HW and SW are required to make a TDM bus that is wider than 24 bits. I'm sorry, but I can't comment to you right now about when and if that would appear in a future product.

Rail, you ask if you get dither even if you don't touch the level. The answer is yes, on this puppy, the dither occurs all the time. I'll see if it's possible to get the change you described added.

As far as my listening tests go with consoles like the Oxford, I have nothing new to report to you yet… but I'll be glad to share stuff with you when I have. Besides, I'm interested in what *you* folks are actually hearing (as compared to the more theoretical aspect of our conversation here).

BTW, guys, I may not be able to spend as much time as I just have in the coming week on this issue, so please bear with me if it takes a little while to get back to you. I'll try to keep up.

Dave Lebolt
12-11-1999, 07:21 AM
Hey folks,
Just load this URL to get your download of these mixers to evaluate: ftp://ftp.digidesign.com/pub/support/misc/test/mxrs.hqx

In the next few days, we should be able to give you an e-mail address where you could upload snippets of "bounce pieces" that you have created from both of these mixers that may (or may not) illustrate any audible differences. To keep this as objective as possible, use the tips for blind listening that I mentioned in my note above.

If we get this upload thing together, keeping file lengths to minimum will be useful. Looking forward to your impressions.

Jules
12-11-1999, 07:33 AM
Thanks Dave,

While your on here, the fixed playback head scroll feature on PT 5 is unbearably jittery on my G3 266 (192 meg). Any tips or is that just the way it is? (unusable)

Rail Jon Rogut
12-11-1999, 10:32 AM
Thanks Dave

To make it easier, here's a link to the download:

http://download.digidesign.com/support/misc/test/mxrs.hqx

or go to the bottom of the Pro Tools section in Downloads and download it from there.

Rail

Rail Jon Rogut
12-11-1999, 11:16 AM
Dave, in your original post you wrote a description of how to perform the AB test:

Here's a tip about how to set up a good A/B test:
1. Create multi-track mixes with different program material. (I say multi-track, because this is where people said they heard a difference most… with multiple tracks and gain changes.) To be as objective as possible, don't use any DSP Plug-Ins.
2. Bounce the mixes to 24-bit stereo files using the two different mixers.
3. Import the results (the two pairs of dual-mono files from the Mixer A and Mixer B bounces) into a new session. Line them up on two grouped track pairs.
4. Go to Preferences/Operations, and uncheck the option that is normally enabled for Solo buttons to "latch." This allows you to instantly switch between two mix stems using the solo buttons.
5. Listen back, and switch soloing between the pairs to try to I.D. any differences.
6. With your new "comparison" sessions, since you're not changing gain (setting faders at unity), there should be no difference between the two mixers… use either one to check the results.
7. Perform a blind-fold test: have someone switch the results without your seeing (or hearing) them switch. If you want it to be really honest, the operator will not even switch all the time, and certainly not when "cued" by you as the listener. This forces the listener to actually identify when and if the sound has changed. If you're not accurately identifying a difference greater than 50% of the time, you're statistically not hearing any difference. (I've been tripped up on this a few times… try it!)

However, because the new mixer always dithers, I think that before step 5 - the mixer should be changed back to the old original shipping 24 bit optimized mixer for a true valid comparison.

Thanks.

Rail

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Recording Engineer

Juicy
12-11-1999, 11:21 AM
Unfortunately, the bulletin board software doesn't appear to support creating links to FTP sites. UBB Code appears to prepend "http://" to the beginning of the URL, no matter what. Take a look at the status bar in your browser when you mouse over the FTP link.

-B...

Jules
12-11-1999, 04:45 PM
The epic instructions & then debates as to how to set up to judge the results are a little worrying. http://www.digidesign.com/ubb/images/icons/frown.gif
Good luck to the remaining few propellerheads! Let us "ears only" folks know how you got on when you have a break from "the lab"! http://www.digidesign.com/ubb/images/icons/grin.gif

Jules

bcobin
12-13-1999, 10:26 PM
Just found out about this thread... Suffice it to say that I've done the mixer comparison on a full mix and the results are very revealing indeed...

There is no doubt in my mind which mixer I'll be using from now on... If you're mixing in PT, you want to try this...

-b

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[This message has been edited by bcobin (edited December 13, 1999).]

ProdInfo
12-14-1999, 01:06 PM

12-14-1999, 01:26 PM
Greetings!!!!

This thread has become rather large and our system has difficulty indexing it. So, I'm closing this thread here, but re-opening one called "Pro Tools Sound Quality (continued)???"

http://duc.digidesign.com/showflat.php?Number=29413

Please post under that new section. Thanks.

Webmaster

Dave Lebolt
12-15-1999, 12:07 AM
Folks,
If you're doing these tests, I have one more piece of information for you: I have just found out from the developing engineer that the dithering on the special mixer will only work on MIX cards. The amount of DSP required to do this on the older "vintage" DSP Farms makes it impractical to support. Just wanted to make sure you were all aware of this.