Wrote a bit of "documentation" based on my reading of the sequencer
code. It's an attempt to answer the question "how does one send and
receive sys-ex using the sequencer?" Read and use at will, but don't
trust it until Takashi, Frank, and / or Jaroslav give their approval..
==
I've been poking through the sequencer code a bit, and noticed two
things which puzzle me. Both are to do with the way variable-length
events are transmitted:
1. Why does snd_seq_event_output_direct() allocate a temporary buffer
and copy the data to it? Wouldn't it be more efficient to read
Is there a way to patch the ALSA drivers along with ALSA OSS emulation
into a static kernel so that no modules have to be loaded for sound to
work? I am currently running 2.5.1-pre1 but also run 2.4.17-pre2
sometimes and would like to do this for both kernels if possible. Thank
you for any i
well, jaroslav's patch (to change the ESO value) was most of the
solution. in the diff below, i have also cleaned up the spurious
interrupt detection. with these two changes, i can run my trident with
a configuration that previously caused endless xruns in full duplex
mode (44.1kHz, 64 frame perio
>I would like to use the sound card as a clock source for a DVD player.
>I have a SB Live with alsa drivers 0.9.x.
>
>Which API call can I make to the sound card which is equivalent to the
>gettimeofday call?
>Also, what will be the units of time received from the call?
there is no direct API cal
Hello
I would like to use the sound card as a clock source for a DVD player.
I have a SB Live with alsa drivers 0.9.x.
Which API call can I make to the sound card which is equivalent to the
gettimeofday call?
Also, what will be the units of time received from the call?
Cheers
James
--
Nothing
>>From the RME documentation it appears that it is possible to detect when a
>source is connected to the RME 9652 card, and whether the source has the
>sampling rate expected. But when I try to read from the card using ALSA, I
>get data whether or not the card has any input. Is there a way to dete
> -Original Message-
> From: Jaroslav Kysela [mailto:[EMAIL PROTECTED]]
> Sent: 04 December 2001 13:07
> To: James Courtier-Dutton
> Cc: Ricardo Colon; [EMAIL PROTECTED]
> Subject: RE: [Alsa-devel] linux sound card w/ digital output
>
>
> On Mon, 3 Dec 2001, James Courtier-Dutton wrote:
>
>It's really discarded as spurious? All voices has separate spurious
>interrupt checking based on the global frame counter (stimer). Please, can
>you verify it? I still think, that it would be better to call
>snd_trident_capture_pointer() and see the actual ptr. My patch should
>fix the problem, i
>From the RME documentation it appears that it is possible to detect when a
source is connected to the RME 9652 card, and whether the source has the
sampling rate expected. But when I try to read from the card using ALSA, I
get data whether or not the card has any input. Is there a way to detect
t
At 04 Dec 2001 15:33:41 +,
Alec Edworthy wrote:
>
> Hi,
>
> just grabbed the latest alsa from cvs and did a ./cvscompile in
> alsa-drivers and got:
>
> card-intel8x0.c: In function `intel8x0_suspend':
> card-intel8x0.c:1075: structure has no member named `pcm_misc'
> make[1]: *** [card-inte
Hi,
just grabbed the latest alsa from cvs and did a ./cvscompile in
alsa-drivers and got:
card-intel8x0.c: In function `intel8x0_suspend':
card-intel8x0.c:1075: structure has no member named `pcm_misc'
make[1]: *** [card-intel8x0.o] Error 1
Just thought I would let everyone know.
How do I cure
>> well, if there are > 2 threads, one only polling and others doing
>> management work -- and this is probably the most useful approach to low-
>> latency PCM IO -- it is imaginable that a non-polling thread starts and
>> stops the device.
>
>In this case pthread_cond_wait and pthread_cond_signal
Emmanuel Fleury wrote:
> Takashi Iwai wrote:
>
>>
>>
>> There is "audigy" branch on cvs at opensource.creative.com.
>> We can check it.
>
>
>
> Yes, here is how to get it:
>
>
> export
> CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/local/cvsroot
> cvs login
>
>
> cvs -z3 co -r audigy emu10k1
>
Takashi Iwai wrote:
>
>
> There is "audigy" branch on cvs at opensource.creative.com.
> We can check it.
Yes, here is how to get it:
export
CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/local/cvsroot
cvs login
cvs -z3 co -r audigy emu10k1
Regards
--
Emmanuel
Sometimes one should just look at
Hi,
I can't find a difference in the way audioengine and jack open / configure
the alsa device, but audioengine works, jack does not. Jack DOES work on
a single card, just not when both cards are linked. Abramo, do you have
time to look at this?
_J
In the new year, Jeremy Hall wrote:
> yeah I
Hi,
At Tue, 4 Dec 2001 14:00:56 +0100 (CET),
Jaroslav Kysela wrote:
>
> On Mon, 3 Dec 2001, Chuck Busby wrote:
>
> > Hi - I got an answer back from my buddy at EMU. He says he hasn't been
> > working directly with the SBLive! guys for a while, but still has
> > connections. (He is one of the f
On Tue, 4 Dec 2001, Roman Razilov wrote:
> Does anymody works on subj?
> Is it really difficult to adapt VIA's AC97 driver for SIS?
> As far as I understand it, it must not be very complicated, and I could ever
> try to do it, if I'l have some time.
> Is there any documentation avaible?
> I just
One thing I would like to know about the emu10k1 (SB Live).
I noticed a register in the emu10k1 which has
#define SPBYPASS0x5e/* SPDIF BYPASS mode register
*/
#define SPBYPASS_ENABLE 0x0001 /* Enable SPDIF bypass mode
On Mon, 3 Dec 2001, James Courtier-Dutton wrote:
> Sort off. SPDIF-In PCM mode only.
> You cannot record AC3 or DTS SPDIF digital data.
> The SB Live sends all the samples through an interpolator, which is fine for
> PCM, but plays havoc with AC3 and DTS.
The interpolator is not used for incomin
On Mon, 3 Dec 2001, Chuck Busby wrote:
> Hi - I got an answer back from my buddy at EMU. He says he hasn't been
> working directly with the SBLive! guys for a while, but still has
> connections. (He is one of the founders of EMU) He asked me "What
> exactly are you looking for? I don't think thi
On Mon, 3 Dec 2001, Tom Prado wrote:
> On Sun, 2 Dec 2001, marco trevisani wrote:
>
> > now is getting weird.
> > from var/log/message
> > PCI: Enabling device 00:1f:5 ( -> 0001)
> > PCI: No IRQ known for interrupt pin B of device 00:1f.5. Please try using
> > pci=biosirq (tried it hangs the
Tim Goetze wrote:
>
> Today Abramo Bagnara wrote:
>
> >Paul Davis wrote:
> >>
> >> >Don't make this mistake: poll *have* to return immediately in *all*
> >> >cases where to wait is useless (i.e. when no non-user driven event may
> >> >happen).
> >> >
> >> >This is the rule you need to remember.
On Mon, 3 Dec 2001, Maarten de Boer wrote:
> Hello,
>
> I am trying to run the alsa-lib/test/latency test on an
> ESS Maestro 2 (Dell Inspiron 3700), but it seems that
> this card cannot do interleaved audio?
>
> Jaroslav, could you update latency.c for these kind of
> cards?
Use 'plughw:0,0' de
On Mon, 3 Dec 2001, Abramo Bagnara wrote:
> Jaroslav Kysela wrote:
> >
> > On Mon, 3 Dec 2001, Alexander Ehlert wrote:
> >
> > > Hi,
> > >
> > > can it be, that snd_pcm_open in the alsa lib is not threadsafe?
> >
> > You're right, the snd_config_update() function is not thread safe (it uses
> > g
>
> I think only few cards support the detection of non-audio data format
> on hardware. And as mentioned before, on some chips, the stream is
> resampled on other rates..
>
>
> Takashi
>
Can anyone tell me which cards do NOT do resampling on SPDIF IN so Recording
AC3 works.
Cheers
James
Today Abramo Bagnara wrote:
>Paul Davis wrote:
>>
>> >Don't make this mistake: poll *have* to return immediately in *all*
>> >cases where to wait is useless (i.e. when no non-user driven event may
>> >happen).
>> >
>> >This is the rule you need to remember.
>>
>> where does this rule come from?
Did you notice that I was talking about SPDIF Digital IN not OUT. (IN =
RECORD, OUT=PLAY).
Most cards can do Digital OUT for AC3 etc, but fewer can do Digital IN for
AC3 etc.
Cheers
James
> -Original Message-
> From: Takashi Iwai [mailto:[EMAIL PROTECTED]]
> Sent: 04 December 2001 10:52
At Tue, 4 Dec 2001 12:02:37 +0100 (CET),
Erik Inge Bolsø wrote:
>
> >> Another thing to watch for is there are two types of SPDIF in.
> >> 1) SPDIF PCM Audio in. Recording normal stereo at different rates.
> >> 2) SPDIF non-audio in. (AC3, DTS etc.)
> >> The SB Live does (1) but NOT (2)
> >
> >Hm
Does anymody works on subj?
Is it really difficult to adapt VIA's AC97 driver for SIS?
As far as I understand it, it must not be very complicated, and I could ever
try to do it, if I'l have some time.
Is there any documentation avaible?
I just got a SIS735 mainboard with sound onboard, but it wor
At Tue, 4 Dec 2001 10:24:51 -,
James Courtier-Dutton wrote:
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]]On Behalf Of Takashi Iwai
> > Sent: 04 December 2001 09:31
> > To: Joshua Jacobs
> > Cc: Dan Hollis; [EMAIL PROTECTED]
> > Subject: Re: [Alsa-
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]]On Behalf Of Takashi Iwai
> Sent: 04 December 2001 09:31
> To: Joshua Jacobs
> Cc: Dan Hollis; [EMAIL PROTECTED]
> Subject: Re: [Alsa-devel] linux sound card w/ digital output
>
>
> Hi Josh,
>
> At Mon, 03 Dec 20
Hi Josh,
At Mon, 03 Dec 2001 14:59:38 -0500,
Joshua Jacobs wrote:
>
>
> I had posted earlier about finding a sound card which supports toslink output
> under linux. It appears that Dan's page http://www.alsa-project.org/~goemon/
> has been updated to include an entry on the Fortissimo II and
On Mon, 3 Dec 2001, Dan Hollis wrote:
> Er... the matrix just lists connectors, that doesn't mean those outputs
> are supported. (I don't know if 4624 spdif is supported with the driver,
> or if fortissimo II toslink is supported)
thanks for this clarification. could you add that to the matrix
On Mon, Dec 03, 2001 at 03:33:54AM +0800, Kyle Centers wrote:
>
> You're right. I added flag to xmms, and the plugin works perfecty now.
> With your permission (since you made the suggestion), I'll send an
> email off to the xmms team about it.
>
I've already report this bug in the bugzilla sy
Paul Davis wrote:
>
> >Don't make this mistake: poll *have* to return immediately in *all*
> >cases where to wait is useless (i.e. when no non-user driven event may
> >happen).
> >
> >This is the rule you need to remember.
>
> where does this rule come from? i was under the impression that
> pol
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