Paul sent me a revised edition of the alsa home page with more of a current feel than
the previous version so you can find it here:
http://www.boosthardware.com/LAU/alsa/index4-pd.html
I have retained the link to the LAU-guide because I know it helps new users to find
needed information.
--- Paul Davis [EMAIL PROTECTED] wrote:
I disagree. I just went through the process of teaching a friend how to use li
nux and he needed every piece of information I could give him. It is my opinio
n that repetiton and hand holding is the key to success for all newbies. (Base
d on teaching ESL
I have a notebook with an es1969 which can mix 4 streams in hardware.
--
Patrick Shirkey - Boost Hardware Ltd
For the discerning hardware connoisseur
Http://www.boosthardware.com
Http://www.boosthardware.com/LAU/Linux_Audio_Users_Guide/
--- Ivica Bukvic [EMAIL PROTECTED] wrote:
First off,
--- Ivica Bukvic [EMAIL PROTECTED] wrote:
*There is esd, which is outdated and simply crappy.
*There is artsd, which is better, but not good enough, and again,
the
app must be made to be aware of it in order to utilize it.
*There is JACK project which has a huge potential but none of its
Hi,
How can an application retrieve the supported sample rates? (SNDRV_PCM_RATE* flags)
I know you can get the min and max sample rate, but it's more interesting
to know exactly which sample rates the hardware supports (if not continuous).
Maybe I didn't RTFM carefully enough, but I haven't
I do not mean to be hammering this issue into the ground, but Linux OS
as an audio workstation solution has been around for 3 years now, yet
the only soundcard I am aware of that is capable of doing hardware
mixing is SBLive!, and even that one is due to fact that Creative had
their hands in the
So what do think I should use instead? I need all 48 channels in perfect
sync (sample-accurate). Do you think that's impossible? The CPU-load of my
I think it might be very difficult if it involves two separate
cards. It would be easy on one card, for sure.
application is (with signal
i just got info on the internals of this interface
you're all gonna love it! even if it is expensive. peak and rms meters
in hardware, full matrix mixing with a range of -inf .. +2dB, 2 MIDI
ports. its going to be great, and hard to see why anyone would want to
use anything else for serious
Hello
AC3 passthru on the SB Live does not seem to work on all platforms with the
latest alsa-cvs. (26-2-2002)
It works on my system, which is a SB Live Rev 6, which only started working
recently.
It does not work on a SB Live rev 7 (I don't have this, but others have
reported.)
Can anyone
As a small extra, the emu10k1 kernel driver which support oss, has a few
bits of info which might be of help
From the emu10k1 linux kernel developers: -
We allocate memory with pci_alloc_consistent() which according to
DMA-mapping.txt assures 32-bits PCI addresses. On top of that
we set a
so, the matrix mixer on the hammerfall-dsp comes with 1456 independent
controls. each one represents the volume for routing an input (h/w
input or playback stream) to an output. there are 28 h/w inputs, 28
playback streams and 28 outputs.
it seems unwise to simply map this straight to the
At Wed, 27 Feb 2002 17:19:44 -,
James Courtier-Dutton wrote:
Or, it might be the problem of dma mask. ALSA uses 31bit mask while
Creative's driver uses 29bit. But anyway this won't affect unless you
have more than 512MB RAM and kernel ocasionally allocates the area
greater than
The trident cards can also do this. I believe that all cards that are
actually capable of doing this and for which the docs have been
provided have ALSA drivers that support it.
Well that's exactly where lies the problem. Most of the laptop
soundcards have marginal Alsa (and for that matter
I am happily outputting sound via spdif on my YMF744B based soundcard
(an AW744 Pro). What I'd like to be able to do, though is rotate the
left/right channels to come out of front/back instead. The reason is to
have stereo separation when I am at a different angle.
Is there any way to do this
I agree, my flawless statement was certainly overrated. Yet, the
question remains: what am I to do as a musician needing to utilize my
portable laptop while the apps/software I currently use get ported to
the JACK architecture (if they get ported at all)?
1a) Get Abramo to post (or point to)
this:
http://www.linuxpower.org/display.php?id=216
is an excellent article on mistakes that SGI made developing their
video APIs. much of what it says seems deeply relevant to audio as
well. from my perspective, its pretty confirming of a JACK approach,
but even if you don't see it that way,
1a) Get Abramo to post (or point to) detailed information on using a
share PCM device
1b) Get Abramo or Jaroslav to post (or point to) detailed information
on setting up libaoss (specifically, how to map /dev/dspN to a
specific
ALSA PCM device)
2) edit your ~/.asoundrc file to
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