Hi,

I've an Aureon Space 7.1 and would like to record from spidf in. I set my soundcard's control panel to use external master clock, so that it is syncronized with the digital signal.

Everything is ok, master clock is correctly set and if I want I can monitor the input. Obvioulsy there isn't any volume control for both recording and monitoring, since it is a digital signal .

My question is, how to perfectly record this stream? I tried Audacity but it doesn't "respect" the external clock, I have to manually set it to the correct clock (in my case 44,100Khz) in order to record correctly, but I don't think it is the best way to record it. I would prefer that the software could "read" the correct clock from the input, as the control panel does... In my opinion this is the only way to avoid any sample-rate conversion!

Am I wrong? Any suggestion?

Having used a number of different programs that perfectly record whatever is presented to them, I have yet to see one that did not require setting the sample rate in the program. Why would you suspect that is a problem? What kind of problem? If the program will let you record at a different sample rate, and some certainly will although this may be entirely dependent on the soundcard driver, there will be no sample rate "conversion." You will just get a stream of recorded data that will then play back at the wrong rate. The way to avoid problems is to take enough care to do it correctly.

some advanced audio programs (such as Cubase) have something like "External Sync.." maybe it's what you are looking for.

J.M.

As the author of a program that requires you to set the rate, I can explain why that is so in my case. I suspect it may be the same reason in Audacity and other programs....

In Wave Repair, setting the sample rate is not something you do in order to let the program know what sample rate is coming out of the soundcard. Rather, it is telling the program what rate you wish to record at. Then, when you actually start the recording, the program instructs the soundcard driver to switch to the chosen rate. So if any resampling is going to be done, it'll be in the soundcard driver, not the software. (For example, if you ask a SB Live to record at 44.1kHz, it'll initially sample at 48Khz then use DSP in the driver to resample to 44.1).

Ok, now I've understood! Thank you for your answers!

You are wrong. Surely that is clear enough?

/quote]You are wrong. Surely that is clear enough?[/quote]

Yes, now i'ts clear. In fact I've made some tests and they show that if I change the clock in the software it changes in the control panel too, as cliveb said... Thank you

So you can't record 44.1kHz sources without sample rate conversion even with a digital feed (at least with SBLive and AC97-cards)? Because the do work internally with 48kHz means they even resample 44.1 digital inputs to 48kHz and if i ask them to record 44.1kHz they resample it back to 44.1kHz, right? In this case it would be better to record everything in 48kHz, because that way you just have a single sample rate conversion (which is still unacceptable).

If you want a card that doesn't resample the S/PDIF input, don't buy Creative.