[linrad] Re: Sampling speed.
Hello Alberto, Long ago you sent the mail about resampling in Windows. I have not tested until today for various reasons. The result is somewhat unexpected. I can see no difference whatsoever when running a Delta 44 at nominal 100 kHz. Neither performance nor cpu load is affected at all. The choice is available for Delta 44 multi on one of my Windows 2000 installations but not on the other. Presumably the drive routines from M-Audio differ. The two figures here http://www.sm5bsz.com/linuxdsp/install/uiparm.htm do truly represent the Windows performance and one should avoid non-integer resampling (silently) done by the Windows operating system. Another thing, Linrad now has a network by which one can split processing between several computers and also add various post-processors like MAP65 (hopefully) I have been able to do this for Linux only - using some prototype code for multicasting supplied by ON4IY and some code for a connected socket that I got from W3SZ. I think it would be nice to add network capabilities in the Windows version as well but I do not know how to proceed. Do you know where I can find some prototype code for receiving multi-cast under Windows ? (Data rates may be rather high so speed is essential.) 73 Leif Leif Asbrink wrote: My concern is non-integer rate conversion. A soundcard that actually samples at 3.072 MHz and sends 48 kHz to the PC has to be down-sampled by 1.088435 times to produce 44.1 kHz. If it were properly done in the PC it would be perfectly OK, but judging from my experiment, the Windows device driver is not very accurate. My problem: How come that the computers do not ask do you really want a rate conversion ? when the user sets a speed that will cause a non-integer conversion? In Windows you can choose the compromise between quality of the resampling and resources utilized for that. Click on Control Panel | Sound and Audio Devices | Audio | Advanced | Performance and in the bottom part of the panel you are now in, you will find a slider labeled Sample rate conversion quality. It is advisable to keep it at its extreme right, the best quality. Anyway in Winrad I coded a non-integer resampling routine using the method of interpolation on a windowed sinc. The results seems to be quite good, the spurs generated by the resampling are usually between -90 and -100 dBc (if I recall correctly). I wonder which algorithm does use Windows. The interpolation is quite fast, the CPU used is minimal. 73 Alberto I2PHD # This message is sent to you because you are subscribed to the mailing list linrad@antennspecialisten.se. To unsubscribe, E-mail to: [EMAIL PROTECTED] To switch to the DIGEST mode, E-mail to [EMAIL PROTECTED] To switch to the INDEX mode, E-mail to [EMAIL PROTECTED] Send administrative queries to [EMAIL PROTECTED] # This message is sent to you because you are subscribed to the mailing list linrad@antennspecialisten.se. To unsubscribe, E-mail to: [EMAIL PROTECTED] To switch to the DIGEST mode, E-mail to [EMAIL PROTECTED] To switch to the INDEX mode, E-mail to [EMAIL PROTECTED] Send administrative queries to [EMAIL PROTECTED]
[linrad] Re: Sampling speed.
Hi All, If you have a sound card sampling at 48khz with no anti-aliasing filter, and you have a tone spurious or otherwise coming into that card at 38khz the sampled data will have a tone at 10khz in its output that did NOT EXIST at its input. Thus the name aliasing. It is MUCH simpler and cheaper to have the audio chip provide anti-aliasing than to build multipole L-C filters!! Thus my comment about ISA cards being useful because they have the filters built into the chip! Happy holidays to all, john On Mon, 18 Dec 2006, Alberto di Bene wrote: John Harrison, NI1B wrote: Hi All, The one REAL change in recent (PCI) sound cards seems to be that the input low pass filter AKA anti-aliasing filter HAS BEEN ELIMINATED on many (all?) of the cheaper PCI sound cards. This probably was done because the cheap AC'97 compliant chipsets found on the mainboards, or in the less expensive PCI sound cards do sample at a _fixed_ rate of 48 kHz, so the anti-aliasing filter is now fixed, no need to adjust it when changing sampling speed. The other sampling speeds are produced by Windows itself, as said by Leif, by a software downsampling routine, and the anti-aliasing filtering is also done in software, before the downsampling. This under Windows. Under Linux, I don't know... 73 Alberto I2PHD # This message is sent to you because you are subscribed to the mailing list linrad@antennspecialisten.se. To unsubscribe, E-mail to: [EMAIL PROTECTED] To switch to the DIGEST mode, E-mail to [EMAIL PROTECTED] To switch to the INDEX mode, E-mail to [EMAIL PROTECTED] Send administrative queries to [EMAIL PROTECTED] # This message is sent to you because you are subscribed to the mailing list linrad@antennspecialisten.se. To unsubscribe, E-mail to: [EMAIL PROTECTED] To switch to the DIGEST mode, E-mail to [EMAIL PROTECTED] To switch to the INDEX mode, E-mail to [EMAIL PROTECTED] Send administrative queries to [EMAIL PROTECTED]
[linrad] Re: Sampling speed.
John Harrison, NI1B wrote: Hi All, If you have a sound card sampling at 48khz with no anti-aliasing filter, and you have a tone spurious or otherwise coming into that card at 38khz the sampled data will have a tone at 10khz in its output that did NOT EXIST at its input. Yes, that's right, but are you sure they don't have an anti-alias filter at all ? Probably, given the fixed sampling frequency of 48 kHz, they have just a fixed, simple RC network. If you don't need to switch the filter along with the sampling frequency, you can build it simply and cheaply with just a few passive components. 73 Alberto I2PHD . # This message is sent to you because you are subscribed to the mailing list linrad@antennspecialisten.se. To unsubscribe, E-mail to: [EMAIL PROTECTED] To switch to the DIGEST mode, E-mail to [EMAIL PROTECTED] To switch to the INDEX mode, E-mail to [EMAIL PROTECTED] Send administrative queries to [EMAIL PROTECTED]
