On Fri, 2005-12-30 at 22:44 -0700, Chad wrote:
> Evening!
>
> I don't really no where to start or what information to provide, so
> I'll start light and hope someone can guide me from there :)
>
> To start off:
> I have a fairly decent understanding of linux. I can follow a how-to
> pretty well,
Evening!
I don't really no where to start or what information to provide, so
I'll start light and hope someone can guide me from there :)
To start off:
I have a fairly decent understanding of linux. I can follow a how-to
pretty well, and understand most of the lingo.
I've read a lot (everything
> For starters, one may start measuring average frequency of interrupt
> requests generated by both cards provided both of the play the same
> length buffer.
>
> The measure of the average frequency will be the computers RTC.
>
> I am not saying that it is simple, and yes, prior knowledge, like NTP
> The way things are working now, I am only able to use two of the four
> input channels with this program; it sees them as a single stereo
> device, but doesn't see the other two channels at all.
So these are four mono ins/outs? Can you post the relevant sections of
your ALSA config file showing
> Is there some way to test the cards and measure their frequency? If
> so, you could do a pretty full-blown NTP for sound cards and that
> would be pretty freakin cool.
I know when programming the old SoundBlaster 16 cards they would trigger
an interrupt when they switched output buffers (so you
On Fri, Dec 30, 2005 at 05:42:32PM -0500, Lee Revell wrote:
> LDAS (the low delay audio streamer) does something similar - it has to
> compensate for the drift between two machines clocks' that exchange
> audio over the network but you could apply the algorithm to two cards on
> one box. It uses t
Craig Tinson <[EMAIL PROTECTED]> writes:
> new to the list so please bear with me.. have spent the last 7 hours
> trying to figure this out...
>
> this must be possible somehow - I have a pvr-250 capture card that
> dumps video *and* audio (via a 3.5m jack line-in) to an mpeg stream.
>
> I have a
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> incidentally, I used to work with LADSPA equalizer plugin, and I have
> a natively stereo version. Furthermore, my code is written as Perl/C
> combination, so with a flip of your fingers you can actually get
> whatever number of channels.
That sounds interesting... I think the problem is that t
> I have a cmedia card with optical input - and I want the resulting
> mpeg stream to have full surround..
Most optical/coax digital inputs (at least with common PC gear) are only
stereo. In order to have surround you'd need AC3 data sent over the
SPDIF connection, and then I'm not sure whether i
For starters, one may start measuring average frequency of interrupt
requests generated by both cards provided both of the play the same
length buffer.
The measure of the average frequency will be the computers RTC.
I am not saying that it is simple, and yes, prior knowledge, like NTP,
should be
On Fri, 2005-12-30 at 17:36 -0500, Ross Vandegrift wrote:
> On Fri, Dec 30, 2005 at 12:45:48PM +1000, Adam Nielsen wrote:
> > I've always wondered about this. Fair enough that two cards would play
> > back sound at slightly different speeds, but why can't you just drop a
> > sample or two every fe
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On Fri, Dec 30, 2005 at 12:45:48PM +1000, Adam Nielsen wrote:
> I've always wondered about this. Fair enough that two cards would play
> back sound at slightly different speeds, but why can't you just drop a
> sample or two every few milliseconds or so, to keep the sound roughly in
> sync?
How do
new to the list so please bear with me.. have spent the last 7 hours
trying to figure this out...
this must be possible somehow - I have a pvr-250 capture card that dumps
video *and* audio (via a 3.5m jack line-in) to an mpeg stream.
I have a cmedia card with optical input - and I want the re
Hi --
I've asked this question on the list twice before, but have never gotten
a response. Unfortunately, my need is getting urgent so I'd really
appreciate any help.
I have a Delta44 card (ice1712) with four ins/outs. The application I'm
using (Baudline spectrum analyzer) supports only the OSS
conrad berhörster gmx.de> writes:
>
> No, the driver is not finished right now
> sizu c~
Thanks .
but do you have idea to driver error ?
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No, the driver is not finished right now
sizu c~
Am Freitag 30 Dezember 2005 16:25 schrieb Eric Rantrua:
> Hello list , i have always same problem with Terratec sound card and
> alsa
>
> i use 8 channels of the terratec card , 6 works perfectly but 2 (front)
>
> have some distorsions.
>
>
> my ques
Hello list , i have always same problem with Terratec sound card and
alsa
i use 8 channels of the terratec card , 6 works perfectly but 2 (front)
have some distorsions.
my question , do you know person who use phase 28 with no problems on
front channels ?
my configuration: Debian Sarge , als
Adam,
incidentally, I used to work with LADSPA equalizer plugin, and I have
a natively stereo version.
Furthermore, my code is written as Perl/C combination, so with a flip
of your fingers you can actually get whatever number of channels.
The plugin is optimized in a sense that mutual for all ch
Does simple 'aplay -Dplug:spdif foo.wav' work?
=== On Friday 30 December 2005 13:36, Pieter-Jan Malfait wrote: ===
hello,
i'm trying to get all my sound throught the SP/DIF (coax) output of my
card. i'm using the latest development drivers (1.0.11rc1) because it
is the only version that
hello,
i'm trying to get all my sound throught the SP/DIF (coax) output of my
card. i'm using the latest development drivers (1.0.11rc1) because it
is the only version that supports the M-Audio Revolution 5.1
I have according to the wiki, set up a dmix device to adjust the sound
to proper settin
Hi,
it seems like OpenAL is broken with alsa-1.0.10. I am unable to compile
OpenAL since using kernel 2.6.14 which includes alsa-1.0.10rc1.
Sadly the hardware-accelerated version of OpenAL is broken, too. Despite
the fact it compiles without any errors, the created lib seems to be
unable to produ
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