So here's a question for you brainiacs...

record/edit with same soundcard a must?


If yer using a particular soundcard (PCI, FW/USB interface, ISA :laugh: ) is it really important to be using that card while editing/mastering if it was the one used to track?

I’m thinking wordclock master here. If the soundcard was the wordclock master when tracking, you’d want to have that as the master when editing/mastering as well right? I have NO scientific explaination for my assumption…that’s where you come in…I’m just going off of instinct.

What’s the deal? ???

I disagree with you on this assumption. Once your material is recorded using an arbitrary soundcard or interface, your recordings are in the digital domain. Period. As long as you reproduce (for mixing, monitoring, overdubbing etc.) at the same bit depth/sampling frequency as you recorded at, no significant alterations to the sounds recoded should occur. Any editing done to the material should affect the material independently of the soundcard.

That said, the quality of the soundcard plays a role when recording, overdubbing and mixing down, i.e. whenever the data representing the sounds is converted back to or from audio. Mastering quality is only as good as the mastering plug-in allows.

I have taken material recorded on one (portable) machine and transferred it digitally onto a stationary machine a lot of times without any trouble, wordclock or not. (BTW, what are the assumed defects introduced in this way supposed to be?! ??? )

regards, Nils

Nils is right.
You’re not editing nor mastering with your soundcard. You only use it to listen back.
Generally speaking, if you have a choice, you would always try to have the device with the best (=lowest jitter) clock to provide the wordclock signal.
Most important when recording, but also when playing back.

I agree with Nils and Hansje.

Here’s where you can get into trouble: recording via one card while monitoring via another. Unless you’ve slaved one’s clock to the other, they won’t be synchronized, and eventually they’ll drift off (if the recorded wave file is long enough). The solution to that is simple, just slave one to the other.

Hi Guys:
I’m not in total aggreement with the replies here to this topic… or, I need some explaination to what might be happening to the tracks I have received on past collaborations I have been involved with…

Here’s what I’ve encountered here, regarding some issues I was faced with on tracks that I’ve received and worked with on “Time-Lines” … here…

For example…

I have worked on “Guide/Ghost” Tracks here where as many as say 8-16 tracks were placed on a time-line and then “Rendered” to a Two-Track Composite Mix.

The “Two-Track” rendered mix was then Compressed with .OGG or Monkey’s Audio. for uploading, where someone, would import the Two-Track to a Time-Line for adding additional tracks/track… Then, a Composite mix of the added tracks… along with the individual tracks would be compressed and up-loaded for “Importing back the the original Time-Line”, for insertion to the “Original” Time-Line… that all the “Guide/Ghost” Tracks were first created, on…

To Find…

1/ The added track begins, Out-of-Sync with the Original Time-Line

2/ Sample/Accurocey of the added track to/of the original Time-Line has some added “latency” that is different from the added tracks on the Composite Two-Track versus the added tracks to the “Original” Time-Line…

I stand corrected… But mabey, there is an explaination to what is occuring here… other than what I’m refering to…

What I think happens… is…

The “Orignal” Tracks were created /A-toD with one set of converters/clock… The Added track/tracks were created with another set of converters/clock as-well-as a different “Latency” is introduced the the “Added” Tracks that causes “Sample In-accrocey”… (sorry about the Spellings)… I can’t spell, today… either…

A Latency Coeffentcey is different or has been inserted into the tracks/track between the two “Tracking” DAWs…

Jeff… anyone? Got another idea? as to what has happened?



When editing/mastering there is no practical disadvantage to using a different soundcard.
However, as Bill rightly points out, when collaborating, different soundcards (and hence different clocks) introduce varying degrees of offset. The worst scenario is a clock that is not stable and wanders around during recording.
For my money, it seems most soundcard clocks are the absolute pits. Drift and jitter are apalling. Heck, I can make a clock that is more accurate and stable than any soundcard clock I’ve come across. The technology is not difficult or expensive. You can even improve your own clock for free, right now…

Beefy’s soundcard clock tip:

1. Locate the clock on your soundcard. This will look like a small metal box. It may have two wires at one end, or it may be soldered directly to the pcb (no wires will be visible) If you don’t see a metal box, then your card probably has a ceramic resonator instead. Throw the card away and get a half-decent one, come on, soundblaster cards are cheap enough, even for me!

2. Find a small piece of white expanded polystyrene foam (not the black stuff, which can be conductive!!!)

3. Make an impression in the foam so that it will be a tight fit over your clock, with the bottom resting on the pcb. The idea is to thermally insulate it, so you want to keep out any draughts. Obviously, the larger the piece of foam, the better, just get it as large as the pcb can accomodate.

4. Secure in place. Use a bit of double-sided adhesive tape, or some impact adhesive. Don’t go mad.

5. In order to allow your clock to stabilise, turn on your computer at least 30 minutes (really, that long!) before you want to use it for DAW work. Ideally, don’t ever turn it off.

6. If you don’t have a clue what you’re doing - dont try.

The above mod will not do anything for jitter, that is inherent in the design of the clock and the electrical conditions in the circuitry. It will improve short-term drift, but may also lower the clock frequency slightly.


Bill’s issue is caused by a different phenomenon. There are several reasons why this can happen. It can even happen on some systems by just loading the same wave file. This is why we always recommend mixing the count-in click into any posted tracks.

In normal cases, this is just a sync problem and as soon as the tracks are realigned everything’s OK. In other cases (broken ones), no sync is possible. This is usually caused by using a SBLive card in 44.1kHz mode.

It’s interesting that when number DAWs are working correctly, there is never a timing problem other than possibly lining up initial sync when recording different tracks on different machines – as long as all additional tracks are recorded to a mix including some original track on one of the machines. Differences in soundcard timing actually cause pitch differences – but not enough to detect. (Even poor clocks are closer than 1% – imagine being off by over 7 minutes per day on your system clock.)

I’ve mixed loads of songs with tracks from all sorts of sources…

I’ve also mixed about 20 songs converted from a “real” studio’s 2" tape (with Jeff’s “click” so I could like them up).

No problems encountered.

Hi Guys:
Thanks for these expainations. I’m reading between the lines. Is this a topic that could be placed on the “wikie”. ???

Refining this topic, for adding it to the wikie would amount to keeping collaborators from falling into poor habits of shareing tracks, for a project…

For example… A configeration list of “Standards” to lessen the chances of “errors” that might cause the “Project” and “Shared Tracks” from creating ir-reversable issues… and head-aches… Well…

This wouldn’t be ment to burden you with a lot of extra work… BUT… You and or Mark would have the right way of communicating this receipy into the “Wickie”… as “Things to avoid when creating tracks for shareing and collaborating”… Or… Things to consider when setting up your DAW for “Shareing Tracks for Collaborating”… :O ???


It never ceases to amaze me how wonderfully much more I get on this forum than I bargained for…

Outside of the wonderful technical journey you chaps have taken me on, can it well be said then that I won’t be doing any harm if I, say, record with a firewire interface, and then conduct some rough mastering which may involve destructive editing of envelopes or effects or rendering (yes I backup as well…and I realize the rendering does not delete the master tracks)?

In other words, will the native quality of the rendered files/destructively edited tracks be the same whether my interface is connected, or I’m just using headphones and the onboard soundcard? I realize the audio playback quality won’t be as good with onboard soundcard, and I don’t do any critical editing without the interface connected and my nearfields as the primary listening source, but what I’m getting at here is will I mucky things up by editing/mastering without connecting the interface used during the record process?

The answer I’m seeing in the above replies is “No.” Is that correct? ???

I greatly appreciate the input. :)

Yes, the answer is “No”. :)

Right – I think you have the right idea.

When N or a wave editor is manipulating audio data, the soundcard isn’t involved at all. (Note: there are some exceptions, for folks who have hardware DSPs and use them – but if you do this I’m sure you know it!)

So, the only affect the playback soundcard has on the audio quality is if it affects your judgements on what to do (which, of course, it does … but you already know that based on your post above).

Now, if you’re not confused enough already, we can go on! :wink:

Ok, just seeding it!
Like this.