I've run into a problem and its got me stumped, so i've signed up here, to see if any of you lot have any ideas.

I have an abit ax8 motherboard with realtek ac97 onboard sound. I also have an audigy sound card, although this does not contain any optical inputs. I am connecting up a set top box to record the audio via the optical spdif into the on board sound, this is where it gets interesting.

The stb outputs at 48kHz sample rate, however I cannot work out how to tell the soundcard this...It beleives the input it is receiving is 44.1kHz, I have tried recording this input using foobar and total recorder, but in both cases they just assume the data stream they are receiving are a speeded up 44.1kHz stream, therefore if i record in foobar for 2 minutes, i end up with a track that is about 2minutes 10seconds long an if i record using total recorder the same problem occurs.

I've installed the most up to date drivers, searched with google to no avail. Played with as many settings as i can find. I'm now at a lose. Does anyone here have any ideas?

Or can someone suggest a program that will speed up the file the required amount

I don't know this on-board soundcard but recording from an S/PDIF input in general requires settings in two places. One is in the soundcard's control panel or mixer software. You need to tell the control panel to use an external clock (which will be part of the input signal), not its own internal timing. Generally there is also a place to tell it what the external sample rate is. It doesn't make much sense for the soundcard to have digital input capability and omit these controls. If this things uses the Windows mixer, the necessary controls should be on the recording section of that.

Then there is a setting in the recording program. When you start a recording you tell it what sample rate the file is to be recorded at. This, of course, must be the same as the external clock if you want things to come out correctly.

I use several different recording programs, all work in much the same way, but none are freeware. I haven't used Audacity, which is freeware, but I am certain it has to have the proper settings. I know nothing about Foobar and Total Recorder, but they are extremely limited if they don't have a way to tell them to record at the sampling rate you want.

Most audio editors will allow you to resample or timestretch recorded audio to a different sample rate but it is much better to get it right while recording.

From what you said, i have looked into it a bit more, understanding more what the different parts do. Its made me rethink, the control software autosenses the sampling rate to 44.1khz, i can't imagine that is wrong, what i thought was an option to change it, is in fact jsut the program autodetecting, because when i remove the source, it "unlocks" and doesn't give any info. With that in mind i retested everything, recording sound at both 48 and 44.1khz, but both are still exhibiting the problem (not that i changed anything)

Despite me now understanding it a bit more, its just confusing me even more. I don't even know where to begin looking now. It can't be a software problem, because it is happening in two different programs. It also records fine when i record from a different source (non digital)

Thanks for your input andyh-ha

Edit: I've downloaded audacity, and that lets me change the speed, but as you said, thats not the ideal solution

Perhaps I misunderstand what you wrote, but if your soundcard control software is 'autosensing' the incoming clock and if that incoming signal is 48kHz and you are getting 44.1, then the autosensing is malfunctioning. Is there some part of the control panel software that displays what it 'believes' is the input sample rate? Does that match the actual input signal?

OR the recording software you are using is ignoring that information and recording as though the input was at 44.1 instead of at 48. Surely Audacity lets you select the recording sampling rate. If you record in Audacity with the file sample rate set at 48kHz, do you get the correct results?

Sorry, it is probably my poor explaination, but before i understood that the card was autosensing i was believing that the input was at 48000, which would have explained the slow down, however, now understanding that the input is being autosensed, I'm more inclined to believe the autosensing is correct than my previous assumption that the input is 48000 is incorrect.

I have set all of the pieces of software that i am recording with to both 44.1khz and with 48khz in a seperate recording, but both rates and all software experiences the problem, I would imagine that they are resampling in the software from what they are being told the source rate is but i don't know what the internal workings are.

I've caved with the problem and emailed abit tech support, see if they have heard of the problem, not sure on what sort of turnaround they have though. If i get a response, i will post it here

Recording programs do not resample as they record, at least no reasonable programs do. If you tell the program that a digital input is coming in at 44.1, it will proceed accordingly even if the input is at a different sample rate. Recording will be suscessful but playback will be strange, either too fast or too slow, depending upon what the actual sample rate was.

As I said, I don't know what recording facilities some progrms have but certainly Audacity must allow you to specify the input sample rate in order to establish the file into which you record. You should be able to make tests by making several short recordings, each at a different specified sample rate, and see which one comes out correct. That should tell you what the input actually is.

success, I feel an idiot now...can i claim stupidity through tiredness? I appears the two original programs don't allow you to change the input sample rate, which as you said is rather useless, and what i was doing in audacity was just changing the output sample rate, which obviously won't do much good.

It appears that the input is 44.1kHz which the soundcard then upsamples to 48kHz for some strange reason, if i then record at 48kHz it sounds fine

Thanks for your help

Resampling to 48k is the trademark of Creative cards. Ignorant wanna-bes emulate this fault to be 'compatible.' Genuine audio cards don't.

ah yes, the classic, "if we resample to a higher quality, then the final piece must be better", wonderful logic.

Well, thanks again for your help, it was driving me mad over the weekend, spending too much time on something that i thought would be a simple task