Hi,

Does anyone know the link between sampling rates of music programs and the processing speed of computers ?

I am trying to record music, tuned to exactly 432hz (and assosciated harmonics), but cannot get accurate results once the sounds are saved as wave files. I have tried different sample rates, which do make a difference, but I would like to be as exact as possible. My guess is that the speed of the processor is out of sinc with the wavelengths of the sounds i am producing. (in a similar way that mp3's corrupt the original sound of .wav files) I would appreciate any comments or ideas.

thanks

kula

What source are you using for the sound? A similiar frustration is "ripping" songs from vinyl to HD. The anologue\digital thing just makes the sound inaudible. A quick workaround to that problem was the introduction of digital outputs to decks. I never was aware of any other way. Maybe the problem with sampled sounds & processors is the interface between them?

i can only guess, but I don't see how it could be a processor speed problem except as a small fraction of a percent. most likely it's your sound hardware/software.

I can produce a pure tone (using wavelab sound editor to generate it) of 432hz and when saved to a .wav it has become 432.02hz, not a big difference, but big enough to make a difference to what i want to achieve. When I recorded 144hz (the note 'D' when 'A" is tuned to 432hz) as a pure tone, it appeared unaffected, but that may be the accuracy of wavelab to measure it.If i write a whole tune using 432hz, the more complex it is , the less accurate the outcome when it is saved. I have tried saving at low quality up to 48khz, but it still looses the accuracy.

Thanks for replying

kula

Sounds like software. If you go to an even lower frequency, I'd guess the error is still there? Yah I'd guess there is a "accuracy vs. frequency" thing, that sets a band where the card or software can create within a tolerance of freq.xx but as you leave the band on either end you go to freq.x tolerance. Just guessing though.

It your audio hardware that does the sampling. Check the max caps of your sound card.

SB Live!Audio [E800]
dwMaxSecondarySampleRate 191999

This is for DirectX. It should be posible to sample up to this rate but I have not tried it yet, it may spit an error. THis is very high for audio but would improve your error.
I assume it can play back at the same rate.

Thanks for the advice, i'm going to have a fiddle. The faster the sampling rate I use, the more accuracy I get but as you say, it may be the accuracy of the sound generator in the first place.

The lower the frequency I generate, the less distortion I get and my guess (I know little about computers, a bit more more about sound, so apologies if this analogy doesnt work) but, the reason i want to preserve an exact frequency of 432hz is to enable the wavepeaks of sound to be in sync with each other, even over a long period of time. With even a small disprepency, the wavepeaks gradually go out of phase. Lower frequecies are less effected so i was thinking that say the computer is sampling at 4321 times a second, and the frequency I want is 4320, that the small difference amounts to the computer sampling slightly behind each wavepeak and that over the period of a song, this adds up to being a larger difference.

Ive only just woken up, so i hope this is co-herent, (I'm not !)

wave peaks ^--^--^--^--^--^--^--^--^--^--^
sample time s. s. s. s. s. s. s. s. s. s. s. s. s. s. s. s

11 peaks, 16 'moments' of sample, but all the wavepeaks are not sampled/saved as they do not line up.

Thanks again
kula