Subject: 4. low-latency audio I/O for Windows (3) From: Lorenzo Picinali <LPicinali@xxxxxxxx> Date: Thu, 4 Oct 2007 18:30:04 +0100 List-Archive:<http://lists.mcgill.ca/scripts/wa.exe?LIST=AUDITORY>Dear Dan, If you want a "standard" interface, I think that you'll have the best results using RME or Motu products, and buying the interfaces with a PCI card (or PCI Express) and an external rack unit (communicating each other using a proprietary bus and format), and not with firewire or, even worst, USB connections. RME and Motu offer products with an integrated DSP (for example, in the Motu products is called Cuemix DSP) for the routing of the IO: this is usually useful if you want to automatically route an input to an output without processing it, and in this case has absolutely no perceivable latency. Although, these DSP for the routing can help even when you need to do some internal processing, but they can't achieve 0 latency... The problem is that the latency is also given by your system (computer) and by the software you are using. The input-output buffer size is essential: if you have a really short buffer, you system will have to work much more, but the latency will be less, while if you have a big buffer, the system will be more "relaxed", but you'll have bigger latency. Nowadays, it's a good habit, for playback systems or for processing which does not require realtime performances, to use a IO buffersize of 512 samples, which means nearly 10ms of latency: this is of course audible if you are trying to do direct processing and monitoring in realtime. Then, the software you are using is another really important thing to consider: using MaxMSP, for example, you'll have to choose a vector size, and this means the same thing of the buffer size explained before. I think that with one of the two interfaces listed before (RME and Motu, with the PCI or PCIe card), with a fast PC and a good amount of RAM, and with a well calibrate software for the processing, you can easily achieve a latency of 128 samples (which means, in your case, at 48kHz, a bit less than 3 ms). The only other solution to achieve real time processing with no latency, is to use a DSP processing system based on PC, such as Digidesign Pro Tools, Sadie, ecc... For example, with a Digidesign Pro Tools TDM or HTDM (it means that you don't only have to buy the interface, but even the dsp card for the processing) you can easily achieve a latency < 1ms. The problems of these systems is that they are a bit expensive (the basic setting could be around 6000 euros)... Anyway, if you have a good audio card (as the RME or Motu), it depends everything on the speed and memory of your computer, on the software and on the kinof processing you need to perform! Yours Lorenzo -- Lorenzo Picinali PhD Student Music, Technology and Innovation Research Centre De Montfort University Clephan building, CL 0.19 The Gateway LE1 9BH Leicester UK tel. +44.0116.2551551, internal 6770 e-mail lpicinali@xxxxxxxx web http://www.mti.dmu.ac.uk/members-postgrad.htm