« Return to Thread: Re: [LAU] Fedora 12 & CCRMA

Re: Quality of sound on RHEL 6.0 beta and Fedora 13

by Niels Mayer :: Rate this Message:

On Mon, May 17, 2010 at 9:49 AM, Fernando Lopez-Lezcano <nando@...> wrote:

http://ccrma-mail.stanford.edu/pipermail/planetccrma/2010-March/016513.html
...

The HPET or other clock sources that are used for timing in the kernel
are _not_ physically connected (to my knowledge) to the clock source
that ships the samples through the sound card. That is a separate clock,
either a crystal in the sound card hardware itself or a clock signal
derived from the master crystal that drives the cpu clock. It does not
go through the kernel, it is a hardware feature. The kernel will
(should?) not affect where and how the hardware (a/d:d/a) gets the clock
from.

I agree. Thanks for clarifying this (again).

However, note http://linux.die.net/man/1/jackd

-c, --clocksource ( c(ycle) | h(pet) | s(ystem) )

Select a specific wall clock (Cycle Counter, HPET timer, System timer).

If you're telling jackd to run at a certain sample rate, e.g. 48k, how is it going to know what 48k actually is? Via the clock.

https://wiki.archlinux.org/index.php/User_talk:Hollunder#timers.2C_hpet.2C_hrtimer indicates the main improvement w/ using HPET in jack is better midi timing.

http://www.mail-archive.com/proaudio@.../msg03199.html also suggests the soundcard crystal oscillator is ultimately in control of the sample rate, and mentions the only time-related issue is the buffer running dry. Which could happen if there's a difference between what the soundcard thinks as 48k and what the system thinks of as 48k.

So If your clock isn't sampling often enough, or accurate-enough, couldn't that affect the sound? Although in jack, i guess you'd see that as a buffer overrun, and increase your buffers to fix it. Perhaps using a desktop audio system and a media player, you wouldn't get such indication. But then again pa and dmix in alsa use large buffers to begin with...

One last idea why a system update might improve the sound. Less resampling? For example if you formerly accessed through alsa dmix at a 48k rate for compatibility w/ legacy AC-97; now f13 supports native 44.1 and many other rates, and you can listen to your music at it's original 44.1K sample rate? Perhaps an improved pulseaudio-> ALSA configuration that just invokes "plug" to align the sample to whatever is required by the hardware, but not use "dmix" to resample?

I hope to be freeing up a machine for f13 in the next months so I guess I'll find out more about F13's ALSA configuration. I might even leave pulseaudio running

for a bit just to see how things work for everybody else :-).

Niels
http://nielsmayer.com

_______________________________________________
PlanetCCRMA mailing list
PlanetCCRMA@...
http://ccrma-mail.stanford.edu/mailman/listinfo/planetccrma

« Return to Thread: Re: [LAU] Fedora 12 & CCRMA