Hello!
Does somebody know what I can do to get my soundcard to work? The
manufacturer is not specified, but the chip is YMF744B. I have been
told that this might not be compatible, as the alsa-site only cites a
YMF744 as compatible. Does anybody have a definite answer on this?
The card is detecte
The plugin compiles again - I removed a number of unnecessary
things from it.
After having some sleep I'll try to launch it.
What center frequencies would you like to have ?
It's configurable through the plugin's Perl part, I'd like to enter them
the way you like the because you'll have other is
> I'll try to deliver it on Sunday evening in terms of GMT+2 timezone.
Excellent!
> First a childish question: are you sure the speakers are in phase ?
Yes, I can't stand out of phase speakers (at least by 180 degrees) so
I'm sure that's not the problem. Given that the speakers aren't
equidista
> At present, I don't have an ALSA config file -- I'm using the stock
> configuration under Debian.
I suspect in this case that the ALSA default is just to map the first
two channels to /dev/dsp0.
> The Delta44 card appears to be recognized with channels 1 and 2
> mapping to the left and right ch
Jon B wrote:
You're just running both devices off the same crystal. That's
certainly one way to do it, but wouldn't work with devices that use
different circuits or crystals, and I don't think it can be extended
to more than 2 or 3 cards.
sure it can.. just build a small timer circuit on a pro
> Hi. The only solution I saw to this problem was in a HOW-TO that a guy posted.
> Sadly I can't find it, but it was a soldering iron job on the soundcards.
Right here:
http://www.djcj.org/LAU/quicktoots/toots/el-cheapo/
You're just running both devices off the same crystal. That's
certainly on
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It doesn't matter actually.
If read pointers of the sound cards are accessible, then one of the
soundcards can be considered as the reference one and "absolute"
time can be measured as
(buffer_size * number_of_fully_read_buffers) + read_pointer
- I mean, read_pointer has the range of 0..buffer_s
As I understand it, the new HPET timers are sufficiently different from the
old RTC chips, that the RTC timer has gotten screwed up in the transition.
Ie, you could use the new hpet fuctionality and it would work well.
Unfortunately no software does. It uses the old RTC interface, and that as
I un
Hi. The only solution I saw to this problem was in a HOW-TO that a guy posted.
Sadly I can't find it, but it was a soldering iron job on the soundcards. For
3 cards, disable the crystals on 2 cards, and link the crystal from the first
card to the 2 cards with the disabled crystals. this logically
On Sat, 2005-12-31 at 12:36 -0800, Bill Unruh wrote:
> But the problem is getting those ticks out. In particular, with the
> new timer chips on the newer chipsets, rtc works by polling, which is
> notoriously bad at accurate timing. Even with interrupts, you cannot
> have them too often or your co
On Sat, 31 Dec 2005, Ross Vandegrift wrote:
On Sat, Dec 31, 2005 at 02:11:19PM +0200, Sergei Steshenko wrote:
- I have already suggested to measure average interrupt request frequency.
If a card is playing N samples buffer with actual sampling frequency Fs, then
time between interrupts per buf
As a second thought regarding calibration - if we temporarily cross-connect
analog outputs and inputs of two cards, we can measure their relative
clock frequencies much more directly and precisely - no random factor.
But I agree, the best way is an HW solution, possibly removing
xtals/disabling cl
On Saturday 31 December 2005 17:54, Jon B wrote:
> > I would've thought though that if you had a software buffer of X
> > milliseconds, then you should be sending that buffer to the card every X
> > milliseconds (according to some other timer in the system.) If it takes
> > longer before the card
Technology has known calibration for quite a long time, so the cards
(the sampling rate correction code rather) can be precalibrated.
That is. before real palyback certain numbers of junk (zero filled ?)
can be player in order to measure sampling frequency difference.
Now, even in real time of rea
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On Sat, Dec 31, 2005 at 02:11:19PM +0200, Sergei Steshenko wrote:
> - I have already suggested to measure average interrupt request frequency.
>
> If a card is playing N samples buffer with actual sampling frequency Fs, then
> time between interrupts per buffer empty is (N / Fs), i.e. for two card
> I would've thought though that if you had a software buffer of X
> milliseconds, then you should be sending that buffer to the card every X
> milliseconds (according to some other timer in the system.) If it takes
> longer before the card is ready for the next buffer then you need to
> drop some
On Saturday 31 December 2005 16:30, Lorenz Hopfmüller wrote:
> Hello folks,
> I wonder if I can play/use MIDI with Vivanco 6C PCI without using timidity.
> In the description they write about a onboard-DSL-wavetable-synthesizer.
> Arent these wavetables for playing Midi without a software synth?
>
On 31/12/05, Tony Lill <[EMAIL PROTECTED]> wrote:
Craig Tinson <[EMAIL PROTECTED]> writes:> 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 resulting>
Hello folks,
I wonder if I can play/use MIDI with Vivanco 6C PCI without using timidity.
In the description they write about a onboard-DSL-wavetable-synthesizer. Arent
these wavetables for playing Midi without a software synth?
# aplaymidi -l
PortClient name Port name
6
[EMAIL PROTECTED] said the following on 12/31/2005
07:11 AM:
>Message: 5
>Date: Sat, 31 Dec 2005 15:10:43 +1000
>From: Adam Nielsen <[EMAIL PROTECTED]>
>To: alsa-user@lists.sourceforge.net
>Subject: Re: [Alsa-user] Need help mapping Alsa to OSS devices (3rd try)
>
>
>> The way things are working n
[EMAIL PROTECTED] wrote:
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someone please unsubscribe this moron, please
Thanks,
--
Morten
:wq
---
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I'll try to deliver it on Sunday evening in terms of GMT+2 timezone.
I haven't been working with it for about 9-10 months; more things have
been developed based on it, etc.
First a childish question: are you sure the speakers are in phase ?
I once remotely helped a guy in urgent desire of an equ
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"
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 have already suggested to measure average interrupt request frequency.
If a card is playing N samples buffer with actual sam
There is excellent alasa mixer called 'qamix'. You can find it here:
http://www.suse.de/~mana/kalsatools.html
I was frustrated with multiple alsa controls until have created config file
for the qamix (I use Terratec Aureon 7.1 Space with the ICE1724 driver).
You can try to play with this ICE1724
when i try aplay -Dplug:spdig mywav.wav , it doens't give an error
anymore, but it doesn't play sound either... i rechecked alsamixer and
i have turned all my DAC sliders to the max, enabled everything
concerncing IEC958 output, and user all H/W (there are five of them,
i don't know exactly what t
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