ROUTING, THE MYSTERY UNVEILED!

A space for learning and studying the Scope environment and music-making in general.

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Nestor
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Post by Nestor »

We have talked about everything in this forum, and about our boards. But I guess… little in regard to routing possibilities, examples and procedures… perhaps because every user is a “unique” user in the way he connects the devices. It is for sure true that every virtual studio of ours is different from one another.

It is perhaps time already, to give it a real in depth go, so we can investigate HOW others work, and start doing a more complex use of routing. It would be also an extremely efficient way for beginners to Creamware Boards, to start their journey playing save! We all have realized that it is rather difficult to get started with Pulsar, and the main reason for it has always been ROUTING.

Seriously, PLEASE get involved and put your knowledge here:
Please, post your understanding of the topic, and get VERY CLEAR about what you say, thinking you are talking to people who many be totally new to the concept. So it is good to stay away from complicated acronyms or half-way descriptions, it would be wise to make full descriptions, step by step, so people can follow these, as if they were tutorials, or they could eventually be used by the CWA next PDF, as suggested already by users, so to build a better PDF manual for new and old users.
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Post by Nestor »

INTRODUCTION:
You are an experienced musician, but you are new to the computer-music world. You know that the most affordable and flexible way of having a recording studio at home, is through a computer system, everybody knows it today.

You went around for a few months, looking for the best and cheapest hardware to be used in your new system, the one you have just bought to accomplish your purpose. Perhaps a friend of yours, or a few good reviews in the net, perhaps some specialized magazines or this forum you are in right now, have got your attention to Creamware Audio solutions, and you finally got one of these all-powerful recording studio-cards home. Well, first of all, let me tell you: you did the right thing!

You have just installed “THE” musician’s dream in your PC or MAC… You’ve got one of the Creamware Audio recording studio-cards available in your system: Luna or PowerSampler with 3 DSP, Pulsar I with 4 DSP, Pulsar II or XTC with 6 DSP, PowerPulsar or SCOPE with 15 DSP (Digital Signal Processor) units onboard. You can have up to three of any of these cards, in any combination, in a single system. The more DSP units you have, the more “virtual space” you own to place your gear around. Every time you add DSP power to your system, you add new rooms, so more musicians can perform there for you, to achieve your musical ideas, more instruments will have room and more connections between them all will be handled. You can easily have a philharmonic orchestra at home if you like, at any time! What I like most about this is that these kinds of musicians never complain :lol: and they seem to be always waiting the director, “me”, in the right mood.

Ok so, now that you have it, you would like to know how to use it; it doesn’t seem to be a straightforward process at first sight nevertheless, doest it? You start scratching your head in perplexity and ask yourself: what it is all about? You know it must be difficult, there are too many options and you don’t understand even a single one of those strange words in front of you, written inside a bunch of blue little rectangles placed left and right of your computer screen, everywhere… and there are plenty of them to load!

I know… you’ve got particularly lost when you tried to BUILD your first project symbolizing your virtual studio-recording setup.

Perhaps the first thing you need to understand is that Creamware Audio cards are not “sound cards”. Yea, of course they offer you “sound” in the traditional way as other PC and MAC sound cards do, but they are rather FULL RECORDING STUDIOS, like those used by music groups and orchestras, to record their songs and get them published, but everything handled in virtual spaces. That is to say: mixers, synthesizers, amplifiers, samplers, box-drums, and even some instruments like a guitar for instance, instate of being there in front of you as hardware, are there as software through the screen. Think of a Creamware Audio Card as a number of computers inside a computer. Every DSP unit is in fact, similar to a little computer, so you actually have several computers inside your computer system that is why these cards are able to process an enormous amount of digital audio in real time, without latency.

Because your card is a full virtual recording studio inside a single computer box, it does a myriad of things at the same time. These things need to relate each other to be able to function, like in real physical recording studios. That is why it is difficult to manage Creamware Audio software at first, because there are too many concepts and you have to be able to understand it as a whole before going to build your projects or you may get seriously frustrated.

There it is the big name: ROUTING!

What the hell routing means anyway? Well, when you plug a guitar to an amplifier, you are in fact routing the audio signals played in the guitar, to the amplifier so the sound can be heard. You may have for instance, a Chorus pedal as well; you know you have to build a connection bridge between the guitar, the chorus pedal and the amplifier. Then you’ll need two cables to achieve the connection, one from the guitar to the chorus pedal, and a second one from the chorus pedal to the amplifier. Let’s go on… the amplifier itself may be routed to a mixing table; the mixing table could be routed to an FX processor, and the FX processor to a second mixer table, and finally this second mixer table to a recording device. All of this cables going here and there, so “sound” can go through them is what we call routing. In a hardware, physical studio, you’ll do it with real cables, with Creamware Audio cards, you do it virtually, with imaginary cables that, via software, connect virtual devices to each other so you can build a recording studio project, or as many as you need, for every specific musical need.

You may blame me of being over simplistic, well, I don’t mind this accusation, then I blame you :smile: for being unaware of beginner difficulties and foreigners that don’t speak English, French or German, which are the only three languages available for Creamware Audio systems.

More to come... :smile:

<font size=-1>[ This Message was edited by: Nestor on 2004-02-17 10:28 ]</font>

<font size=-1>[ This Message was edited by: Nestor on 2004-02-24 22:12 ]</font>
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Post by Nestor »

FIRST STEPS:
You can open and close the Routing Window using the F9 key of your computer keyboard; this is very handy for the sake of more virtual visual space to work. Remember that you do everything in this screen, so it is easy to have 20 devices open in front of you, and they can bother a little, that is why it is handy to close them sometimes. You may for example be programming modular synths patches; you would like to concentrate only there, in modules surfaces, you can actually do everything without closing, but well, it is just more pleasant to your eyes if you close the Routing Window and even the File Browser with all the mess they can produce with too many cables and devices around, so they disappear from your sight till the end of a task. You can close the File Browser pressing the F10 key of your computer keyboard.

Tip: press the F9 and F10 keys together; it is a fast and comfortable way to switching on and off the whole thing for better sight.

More to come...
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<font size=-1>[ This Message was edited by: Nestor on 2004-02-17 11:12 ]</font>
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Post by Nestor »

PROJECTS:
What is it a project? (file extension *pro) A project is a finished specific setup of a virtual recording studio ready to load and use, it is a template of interconnected devices you can use and modify as you need. Sometimes you would need to do some Rock music, and so there will be a project with a good distortion, a big hard drum, a solid bass, etc., but you may also need doing some Arabic kind of music, there you’ll have a project with some Ethnic Arabic instruments like the Ud, the Qanoon, the Nay and the Mihbaj, ready to load! Isn’t it amazing? You may even be working with 10 different projects like a Rock project, an Arabic project, a Jazz project, a Classical project, a Techno project an so forth., and you’ll be able to load them instantly, in a matter of seconds. Of course, you can handle only one project at a time.

When you start, a number of projects are given to you as example, but they are of little or no use if you don’t understand the concepts to construct them yourself.

Here you have a summary of most important points from the Pulsar Manual (a little bit updated by myself) worth reading attentively:

SCOP Fusion Platform is more versatile and flexible than any other system. Depending on the modules available-both those that came with the package and those you’ve acquired individually-you can configure your system as a modular synthesis platform, as a full featured digital mixer with effects, as an I/O (in and out) system for your HDR (Hard Disk Recording) system, as a sample player, a full featured sampler, or all of this at the same time – simply load the modules you need, as long as DSP computing capacity permits.

SCOPE Fusion Platform is completely modular. All modules – synthesisers, mixers, effects, sample players, full featured samplers, and even audio drivers and hardware I/Os – can be loaded from the list of available modules. This means you can always adapt SCOPE Fusion Platform 100% to fit whatever your needs happen to be at the moment. Modular in turns means economical, since only those modules which are actually loaded consume DSP power.

With SCOPE Fusion Platform, “realtime” really means real time. As the DSPs handle all the work of audio processing without help from the host CPU, we’re able to avoid the typical latency that goes along with dependency upon the operation system of the computer.

The SCOPE Fusion Platform needs never become boring. You’ll always be able to come up with new ways to use it, which in turn will provide new inspiration for your work. This is a system without limits! (I can swear this is true, and not just some marketing rubbish)

More to come...
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Post by cannonball »

brav0!!!!!


ale
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Post by wayne »

wow, Nestor, the nay!!!

played in a band with one of these - tricky to play and keep the reed conditioned, but an amazing axe, to be sure!

looks like a recorder, sounds like a bendy arabic low oboe, or something :smile:

The f9/f10 thing is indispensible!

sorry for ot :grin:
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Post by scary808 »

I have created a thread in the projects forum concerning a subject I've talked about a couple of times. It's about logic effects as inserts in SFP & vice versa. It's complete with screenshots, a short tutorial, .pro files, & .lso files. I think it could also help those who use other DAWs get some more depth to their system.
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Post by Nestor »

That's great Scary, thanks! Could you please link it here? :smile: Cheers.
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Post by scary808 »

Oops! I forgot to paste the link. Here it is:



http://www.planetz.com/forums/viewtopic ... forum=14&0
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Post by Nestor »

Pulsar ANALOG SOURCE - ANALOG DEST Modules:

Going to Software I/Os, you’ll find the place where all the modues are ready to load into the screen of the SFP:

Image

In the SFP (Scope Fusion Platform) software, you have basically, two kinds of module representations: The first one is a virtual kind of module representation, which represents virtual hardware (none existent but as software). The second one is a virtual representation of REAL hardware that can be within or outside your computer box. Within your computer box could be a second board like a synths sound card, for example; outside your system box could be a hardware keyboard, an FX processor, a hardware mixer, or all of them together.

Some connections so, actually NEED more than just connecting your software modules between them. You’ll eventually need to connect external gear; this is the case with these modules we are studying right now. Remember this well, so you don’t get confused later. It seems improbable, but it is awfully common to find people that have gone a long run of troubles, because they didn’t figured out that MIDI cables, for instance, do not transmit any sound, but pure rudimentary data, and that without the proper Analog outs, there is no possible sound coming out of a keyboard. I have always thought that a good analogy about this: think about a word processor, in respect to a printer, the words are just data, but not printed ink. In the same way, MIDI information is similar to those words, but it is not sound. These kinds of confusions are common, so it is good to sort out which modules represent “hardware” and which represent “virtual” devices, you can do it easily reading the Pulsar manual.

Pulsar Analog Source and Dest drivers represent your card’s Analog inputs and outputs respectively.

Image

As you may guess, the little rectangle called “Pulsar Analog Source” is the representation of your line IN, through which the sound gets to your system to be processed into digits, and that is why the processed Analog sound is called “digital audio”.

The second little rectangle here, called “Pulsar Analog Dest”, represents the actual OUT through which the sound gets out of your system, i.e., to your speakers so you can hear it back, transformed itself into Analog audio.

I strongly recommend you to read this, take your time and go through it slowly, understanding clearly the points, it will give you a good and simple background to understand how the conversion takes place: analog to digital and back from digital to analog, inside your Pulsar board, go here:

http://www.blazeaudio.com/howto/bg-digital.html

Here you have some more information regarding to how analog and digital recordings work, in case you want to get deeper, with a very clear explanation:

http://entertainment.howstuffworks.com/ ... gital3.htm

Get into it, understand it fully and come back :smile:

More to come…


<font size=-1>[ This Message was edited by: Nestor on 2004-02-24 22:07 ]</font>
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Post by Nestor »

S/P-DIF:

You can do a lot with S/P-DIF, but you've got to watch the clock.

Go here to get to the Hardware IOs modules:

Image

The first thing one must learn about S/P-DIF is how to pronounce it. I say it "ess pee diff (as in Jiff)." You may imagine other ways of saying it. Perhaps you really do want to go around saying "spee-diff." Sort of like people who think it's cool to say, "boxen :smile: ."

S/P-DIF comes in two flavors, copper and plastic. These flavors may be used alone, or in combination. Typically:

S/P-DIF copper. RCA connector. e.g. Pulsar Classic I/O whip - yellow Dig In & Dig Out.

S/P-DIF's cousin AES/EBU - XLR connector. e.g. Scope Classic Plus whip - yellow Dig In & Dig Out.

Plastic or "optical" S/P-DIF uses the same connector as ADAT. The Samplerate menu (under Set) lets you choose whether a Pulsar II or newer card will treat one of its ADAT ports as 2 channel S/P-DIF or as 8 channel light pipe.

These are the modules:
Image

I will refrain from any participation in the plastic vs. copper debate except to make you aware (in case you were not) that there is a debate. Your search techniques should avail you should you feel this matter of plastic vs. copper is worth investigating.

In theory, S/P-DIF and AES/EBU aren't supposed to communicate, but... If you are curious, try a search using both terms plugged-in to your friendly local search engine (yes, I use Google too, but what about giving the other guys a chance now and then - that'll leave more bandwidth for me back at google :8 ).

There are ways of getting S/P-DIF and AES/EBU to pass information to each other. The simplest method I've heard uses a Drawmer M-Clock to re-clock a CD-Player's S/P-DIF output to the AES/EBU input on a Scope Plus. The M-Clock also controls the timing of the CreamWare environment. The ability to reclock allows more than one S/P-DIF input to be active so the streams can be cleanly mixed. No nasty crackles or pops.

But even absent something like a master clock that can re-clock digital inputs while working with CreamWare's clock board, the SynCPlate; digital output from an S/P-DIF device into the CreamWare environment should still be crackle and pop free. As long as you've chosen S/P-DIF as clock source in the Samplerate submenu under Set.

If you've never changed the clock in sfp before, note that you have a major selection of two buttons at the top and minor selections below. Both major and minor buttons have to be pushed - and even then it can take awhile. Just wait, the clock rate should eventually show in the samplerate window. Until it does, don't waste your time trying to do anything else in sfp.

Oh, and mute anything sensitive downstream from your sfp environment before you change the clock. Give your Creamware device some privacy at this intimate time. Think of it as if your Pulsar or Scope were molting :smile: .

Be aware that the quality of the S/P-DIF device's clock is entirely up to the S/P-DIF device's manufacturer. Myself, I really appreciate having a CreamWare environment with a SyncPlate - and VDAT. Before I got an external clock, I tended not to use the S/P-DIF as clock very often. But you can.

Even if one somehow manages to rig an AES/EBU to S/P-DIF adaptor or even an S/P-DIF to AES/EBU adaptor using readily available parts, there's still that old master-slave thing to deal with.

If there's a choice, typically, the most expensive device supplies the clock. You could try listening, but it does make sense that if there's a big price difference, the more expensive should have better parts - at least you hope so. Of course that is only valid for items made about the same time.

S/P-DIF sources could be a CD player, Mini-Disc player, DAT player (recorders) or another (computer with) Pulsar I or II Classic.

The S/P-DIF source device should be master and CreamWare the slave. (Clock is possible exception.)


S/P-DIF dests could be a DAT recorders or another (computer with) Pulsar I or II Classic.

It's a lot easier going out, by the way. In that case, your clock should come from the Creamware device unless another clocking arrangement makes more sense.

The tricky part in going out to an S/P-DIF device is making sure you've chosen the right clock speed (for example 44.1 for CDs or 48k for DATs). Believe it or not, there's a debate on that, too. As always, good search techniques will get you through - if that's where you want to go :smile: .

by john (Known in the forum as Jabney)

Thank you very much Jabney, for your contribution!

More to come...


<font size=-1>[ This Message was edited by: Nestor on 2004-03-05 15:04 ]</font>

<font size=-1>[ This Message was edited by: Nestor on 2004-03-05 15:15 ]</font>
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Post by Nestor »

S/P-DIF Continued:

Once you have all your clocking issues dealt with, then use S/P-DIF in the same way you would Analog In or Analog Out.

The most direct way is to drag the S/P-DIF In and S/P-DIF Out icons from the Hardware I/Os section of the menu onto the surface of the CreamWare sfp RoutingWindow. This provides a two channel (typically stereo) input from, and a two channel output to some digital device located outside the computer box.

An Example:

Let's say you want to listen to the digital output of a Tascam DA 30 Mk II DAT recorder. That model DAT machine has both S/P-DIF and AES/EBU inputs and outputs. If you have both a Scope or Pulsar Classic card and a Scope or Pulsar Plus card in your computer, you could connect using either S/P-DIF or AES/EBU. Which? I'd say use the one for which you have the better cable.

Please be aware that expensive digital cables do exist and not all of them are snake-oil. Still, I'd hesitate before spending thousands (or even hundreds) on a digital cable. At least make sure that you are using a digital - as opposed to audio, mic' or line - cable. (If you want to get technical, the difference between digital and audio cables has to do with impedance, and here it's a sort of 'sideways' measurement. Not at all like a simple resistance measured from one end to the other wherein a 1000 meter cable would have more resistance than a 1 meter cable of the same gauge. A search engine can provide more answers for the curious.)

If you insist, you could listen and try to pick whether AES/EBU or S/P-DIF sounds best. Good luck trying to hear a difference, though. A primary reason for using any digital connector is to assure that the signal is passed along without any alteration. So S/P-DIF, AES/EBU, ADAT lightpipe over a few feet should all sound exactly the same - in theory. For this example we'll stick with S/P-DIF.

info added 2/28/04

I'd like to thank Tom (known on PlanetZ as Astroman) for pointing out a 'fly' in the ointment of digital perfection. The error correction applied in a CD player for example, is not as robust as the error correction applied in a computer to a file. If you need to get as close to the original as possible, then a grabber program (e.g. cdparanoia) to convert the CDDA file to a .wav file is the way to go. Then, you probably wouldn't be using S/P-DIF, instead you'd just use your computer's data CD reader.

One other thing: it is not always a bad thing to go through a digital to analog/analog to digital cycle. In fact the conversion hardware may add or subtract just enough to give you the sound you are looking for - in a mix, for example. Conversion may be (in fact, has to be) less faithful than pure digital, but a little color can be useful now and then. For now, we'll persevere with the S/P-DIF, though.

Once you've made the physical connection between the DAT machine's S/P-DIF output and the CreamWare's Digital In, turn on the DAT machine. Load the DAT tape you want to hear and hit Play on the DAT machine. When you hit Play, an already recorded DAT tape will transmit its clock rate to the DAT machine. You should be able to tell the clock rate on the DAT by seeing either 44.1 or 48k on the DAT machine's display. Once the DAT's clock is displayed, you should be able to rewind the tape and hit Pause. The clock info should still be there. (A CD player gets its clock information from a valid CD and except for the parts about loading the DAT tape _ use a CD instead _ and 48k _CDs only do 44.1, the above information is also applicable.)

Start the CreamWare sfp environment, and adjust the SampleRate (under Set) so that CreamWare is a slave to the appropriate S/P-DIF input stream. Then just connect the output pads of the S/P-DIF In icon to a couple of channels on your VDAT or STM-2448 or what-have-you.

An Alias:

Scope and Pulsar (II) Plus users with AES/EBU digital connectors will still use the S/P-DIF icons (at least as of sfp 3.1c) since there is not a dedicated AES/EBU input or output icon. There is a workaround if you think this could cause a problem, say you have a clueless, yet stubborn, techno-snob client who might think the AES/EBU signal path was somehow polluted by having even the label "S/P-DIF" in the path. You can create your own AES/EBU-labeled icon. (This has more practical applications, too. So read on.)

Perhaps you say, "I have so many things connected to my CreamWare environment, the back of my computer looks like a porcupine at the beauty parlor. How can I possibly remember what goes where?"

Not to worry. CreamWare has the Tools to solve your problem. In fact, the Tools menu is where the "Source S" icon is found. For our DAT example, we will drag the Source S icon onto the CreamWare RoutingSurface and double click to open the icon's surface.

The first thing to do is to change the icon's name from "Source S" (which stands for Source Stereo) to something more meaningful, perhaps "DAT Play." It's as easy as changing any other text field in CreamWareLand, but this one actually displays as the name on the icon.

Below the name you will see two fields to the right of some potentially confusing labeling: Output then Left and Right. "But aren't we working with inputs?" you ask. Yes, but only into the DAT Play (formerly known as Source S) device. The output of the DAT Play icon is the stream it gets from the hardware (S/P-DIF) Input which in turn becomes ... well, the good thing is, you can't mess it up. When you click in the Left or Right field with the Right Mouse Button, a list of Sources is displayed. Choose an unused S/P-DIF Source then choose L Out. Repeat for the Right channel. Then you are almost done.

I recommend setting-up a separate directory for the icons you create using Source M, Source S, Destination M and Destination S tools (as well as the similar but more complex External Effects of various flavors under Effects/Stereo (or Mono)/Others). Then, you can save your new icon (using the save sub-icon on the device's upper-right-corner) so you don't have to do it all over again the next time.

There is one "gotcha" lurking behind these alias icons: the icon for the hardware I/O you want to connect must be physically present on the CreamWare RoutingSurface. I like to hide the physical icon below the bottom of the screen, because it does not have to be connected to anything. The connection is made with the alias icon.

Oh, and sometimes the alias icon dialogue may show that a physical connection is "occupied." The best way to deal with that is to experiment. A hint, before you experiment with stealing connections for alias icons, make sure that you can see the connections that are "occupied" then watch what happens :smile:

What We've Done:

Let's review. You decided that you want to use the S/P-DIF Out of a DAT player. You made a physical connection from the DAT player to your CreamWare card's Digital In and you made sure to use digital cable. You started playing the DAT tape so that it would furnish the clock.

On the CreamWare RoutingSurface (or using your favorite method) you connected the right-facing pads of the S/P-DIF source icon to two left-facing audio pads of some other icon. You made sure that CreamWare was slave to the appropriate S/P-DIF stream using the SampleRate dialogue. You set the levels in software, knowing that the DAT player's output levels have no control over the S/P-DIF stream. If you were comfortable with the idea of S/P-DIF, perhaps you set-up an alias icon. Now what are you waiting for? You're in CreamWareLand: make some music!

A Messy Post Script:

At some point, you may want to use an external device with S/P-DIF In and S/P-DIF Out. "Nifty," you think, "Now that device can stay entirely in the digital realm." Perhaps. The reason it gets messy is that sometimes it works, sometimes it doesn't. In general, later generation devices (CreamWare and external) tend to be more cooperative. And strict compliance to S/P-DIF standards may make a highly compliant device appear to be less capable than a more loosely-spec'ed device. Experimentation done at your leisure is the answer. A live show (or with an anxious client breathing down your neck) is not where you want to experiment with double (or even single) sided S/P-DIF.

Watch the Clock (or, Messy Post Script Part II):

A single active S/P-DIF stream appears to provide a stable clock to the CreamWare environment. Why not let CreamWare default to be slave to a DAT or CD player? Here's why not (in case you missed it above): the S/P-DIF clock that you might think of as coming from the DAT player or CD player is actually coming from the DAT tape or CD disc. If you don't have a DAT tape or a CD loaded, then you don't have clock.

john

Thank you Jabney for your contribution!

More to come...

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<font size=-1>[ This Message was edited by: Nestor on 2004-03-05 15:05 ]</font>
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Post by Nestor »

tnx John for that huge effort, but unfortunately it almost ends when it starts to become interesting :smile:
Imho the pdf doc is pretty good at explaining the basics of digital routing as long as it concerns a single device and only in one direction.

But I can't even remember the case reading from and writing to a device on the same port is covered in the manual.

It's stated though that digital fx 'roundtrips' are impossible on generation one cards (4 DSP Pulsar), but do work in all 2nd generation boards (from Luna on) by seeing the SPDIF specs a bit non-specific (as you mention).

So what finally happens is specific to the manufacturers implementation and there seem to be many different viewpoints - you cover this topic by the suggestion to try it out in non-pressure situations, which seems the only way to go.

It's important for the beginner to know there are specs, but they are of little use when it comes to practical applications - and it's not always one's own stupidity when a setup doesn't operate.

Maybe I didn't search extensively enough, but I do miss all technical details (in the original documentation) about Pulsar's clock specs, how this and that is related, what has priority and the pure signal specs.

Then come the more complicated setups with at least 2 external digital units, like an ADAT IO-box and an fx unit.
A digital mixer, ZLink and fx unit to make it even worse.
What gear works good together and what is to avoid ?
When does an external clock source provide a noticable improvement and what units have a good reputation (that stuff is expensive).

Imho we need to collect some hands-on facts from existing setups, preferably from the more experienced users - who have probably the least time :smile:

One import fact I'd like to contribute is error correction, which isn't mentioned at all yet.
Everyone tends to interpret the digital audio streams the way data is read from a harddisk, but it's not like that - it's more like modem data sent down the phone line.

It's important to realize that an identical copy[/] of such a stream does not mean the data was read identically (and reliably) from disk or tape - it just means that the error correction has a consistent way to deal with it.

cheers, Tom

By Tom (Known in the forum as Astroman)

ps: I found the last sentence somewhere on this site http://www.tnt-audio.com/int.html
some understandable (!) facts about digital signals, clocks, jitter and aliasing
http://www.tnt-audio.com/clinica/convertus1_e.html

Thank you Astroman for your contribution!

More to come...


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Music is the most Powerful Language in the world! *INDEED*


<font size=-1>[ This Message was edited by: Nestor on 2004-03-12 20:24 ]</font>
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skwawks
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Post by skwawks »

Very interesting and very informative thank you very much .:grin: Maybe you could highlight a little section saying it's the "SAMPLERATE WINDOW" dummy, thats where you can sort it.
If the helpfull guy in the shop Hadn't said make the dbx386 the master clock I would have wondered for a while about those clicks and pops .:???: As it was I still had to think for a while concerning the whereabouts of the above mentioned window . Is that dumb or what ? It's a Culture this creamware stuff ,but thanks for trying Nestor . Could you try really hard about the sts thingo's
now they are obtuse :oops:
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hesnotthemessiah
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Post by hesnotthemessiah »

Hi Nestor. Thanks for the information. I just joined this forum and am due to become a Creamware convert very soon (looking at the Creamware Scope Pro options). Just a couple of questions:-

(1)"the S/P-DIF clock that you might think of as coming from the DAT player or CD player is actually coming from the DAT tape or CD disc. If you don't have a DAT tape or a CD loaded, then you don't have clock."

Not too sure what you mean by this? The clock comes from the unit (CD player/DAT machine etc) regardless of whether a CD/tape is loaded doesn't it?

(2) I have a naff Philips CD player connected to my current audio card's S/PDIF in one PC. I therefore have to use this CD player's clock rather than the audio card's actual clock. This PC is connected via ADAT to my other PC which uses this ADAT signal as it's digital clock source. So all digital clock sources start from the naff Philips CD player. I guess I am stuck with using the Philip's CD player's clock as the source. But if I was to install more than 1 Creamware Scope card in one PC and use more than 1 S/PDIF into this PC, wouldn't this cause problems? Each of these Creamware cards in this PC would have to use the card's digital in source as clock but each card is connected within this PC so wouldn't their clocks have to sync to the same clock source. Hope that made some sense!

Next topic "The Big Bang - How it all started!".
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at0m
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Post by at0m »

All CW cards in a single machine are always at the same clock rate. Only one device, PCI card or external hardware, can be master. The CD player can't be slaved. If you have more of such devices, you can connect all of them to the available digital inputs of the cards. But you can slave the cards to only one of them at a time. If these external devices have digital inputs, slave them to the cards, which in their turn are slaved to the CD player - this shouldn't give any problems.
The only problem arises when you need multiple devices that don't have digital input at the same moment. Then you'll have to choose which you use. Or use the analog output instead of the digital connection, then no sync is required.

Hope this helps...
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Nestor
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Post by Nestor »

I really hope to finish this one with the help of people like atOm here. Unfortunately, time has become a bad enemy for me nowaday. We'll see...
dawman
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Post by dawman »

Nestor,
As you once gave Agamemnon of Mycinea great advice, so you share your great advice with us here at the Z. I just finally took time to read this great thread as I am away from my DAW 4 a while. You should be commended 4 this quality of information. I hope you grace us with more of your work and knowledge. If you program modular with the same zeal you applied to the routing info., I want to see some of your patches!!

To Us, And Those Like Us,
fraz
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Post by fraz »

Nestor,

I have some questions regarding this thread but I'm not sure if it's the correct place to post them so please let me know. You may wish I post in the general discussion forum, I'm not sure.

Thanks.
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Nestor
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Post by Nestor »

Perhaps is good to keep everything in a single place, so it is then easier for other people looking for the same info after you.

Keep them coming.
*MUSIC* The most Powerful Language in the world! *INDEED*
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