Hi!
After checking out cvs yesterday , I have now errors when running level
plugin from Abramo Bagnara
or ameter plugin ( which is here:
http://perso.club-internet.fr/lgeorget/ameter.html
) , when buffer size is low:
aplay -D ameter -B 5000 -v glametest.wav
...
aplay: pcm_write:1014: write erro
Paul Davis wrote:
>
> >The server based approach has an added cost of an extra context switch
> >every period (about 1500 cycles on my machine i.e.), but this is fully
> >amortized by such an huge difference.
>
> recall that (1) the context switch time is not a fixed cost but
Mine was only a ver
Quoting Takashi Iwai <[EMAIL PROTECTED]>:
> > > hmm.. ICH4 has a separate SPDIF channel and it looks like we don't use
> > > this properly. that is, SPDIF is routed via the normal slot 3/4.
> > > perhaps we should route the spdif via slot 10/11 (in GLOB_CNT bit
> > > 30:31 to 1:1), and set SPSA t
On Thu, 20 Feb 2003, tomasz motylewski wrote:
> Do I understand it correctly that the server stores data in 32 bit buffer and
> then puts it in 16 bit DMA buffer of the card? This is one operation more
> compared with mixing directly in DMA buffer.
There is no server and 32-bit buffer is used for
Jaroslav:
> I think that we can lose more in the client/server model. Also, note that
client/server will have higher latency. The server has to copy the samples
"last minute" to DMA buffer and the client has to manage before the server
copies the data. In the direct model only the client's timing
On Thu, 20 Feb 2003, Jaroslav Kysela wrote:
> > Then after ./build config I get in alsa-lib/src/pcm/Makefile
> >
> > @HAVE_JACK_TRUE@else !HAVE_JACK
> > @HAVE_JACK_TRUE@endif !HAVE_JACK
> >
> > @HAVE_JACK_TRUE@all: libpcm.la
>
> Could you try to remove !HAVE_JACK string?
>From where?
I have j
On Thu, 20 Feb 2003, Takashi Iwai wrote:
> At Thu, 20 Feb 2003 15:14:45 +0100 (CET),
> Jaroslav wrote:
> >
> > If there is a way to determine the connection between sequencer port and
> > rawmidi device, I'm ok with any solution. Otherwise, please, solve this
> > problem.
>
> one of the advant
What is the difference between serialmidi and serial-u16550?
Is it possible to use a serial midi interface from a USB->serial device
with serialmidi?
Is it possible to use the MTPAV through a PCMCIA parallel port?
I have a MIDI interface that I can attach to my Palm Pilot. It relies on
the palm
On Thu, 20 Feb 2003, tomasz motylewski wrote:
>
> Debian woody, current cvs:
>
> ./build prep
> Pre-configuring alsa-driver
> make: Nothing to be done for `all-deps'.
> Pre-configuring alsa-lib
> src/pcm/Makefile.am:6: JACK_PLUGIN multiply defined in condition
> automake: src/pcm/Makefile.am: u
On Thu, 20 Feb 2003, Abramo Bagnara wrote:
> Now I'm able to get the same results you see.
>
> However I think that we need to extract some results from this data.
>
> I'll leave alone MMX optimizations because I want to compare apples with
> apples.
>
> The distributed saturation (also when it
On Thursday 20 February 2003 15:14, Jaroslav Kysela wrote:
> On Thu, 20 Feb 2003, Takashi Iwai wrote:
> > At Thu, 20 Feb 2003 09:55:18 +0100 (CET),
> >
> > Jaroslav wrote:
> > > On Thu, 20 Feb 2003, Pedro Lopez-Cabanillas wrote:
> > > > On Wednesday 19 February 2003 14:51, Takashi Iwai wrote:
> > >
On Thu, 20 Feb 2003, Paul Davis wrote:
> $prefix/include/alsa/iatomic.h defines LOCK but doesn't #undef it. it
> would be nice if it did.
Done.
Jaroslav
-
Jaroslav Kysela <[EMAIL PROTECTED]>
Linux Kernel Sound Maintainer
ALSA Project, SuSE Lab
Debian woody, current cvs:
./build prep
Pre-configuring alsa-driver
make: Nothing to be done for `all-deps'.
Pre-configuring alsa-lib
src/pcm/Makefile.am:6: JACK_PLUGIN multiply defined in condition
automake: src/pcm/Makefile.am: unterminated conditionals: @HAVE_JACK_TRUE@
src/pcm/Makefile.am:9:
$prefix/include/alsa/iatomic.h defines LOCK but doesn't #undef it. it
would be nice if it did.
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>The server based approach has an added cost of an extra context switch
>every period (about 1500 cycles on my machine i.e.), but this is fully
>amortized by such an huge difference.
recall that (1) the context switch time is not a fixed cost but
depends on the memory behaviour between switches an
In it's infinite wisdom, Dell has started shipping a new variant of
the SB live value - they call it the SB live 5.1(whatever), creative
model # CT0200.
It is _not_ a standard emu10k1 chipset.
Is anyone on the list working with this beast? If so, do you need some
help? If not, can anyone point
Just for informations sake, I just submitted patches to the helix
project to add native alsa support to the client.
Might also add alsa to the producer.
and am also thinking of Jackit support for both, perhaps.
--
My cat's a debugger
Potter, Lorn, "ljp"
core developer / Web Administra
Jaroslav Kysela wrote:
>
> On Thu, 20 Feb 2003, Abramo Bagnara wrote:
>
> > Jaroslav Kysela wrote:
> > >
> > > On Wed, 19 Feb 2003, Abramo Bagnara wrote:
> > >
> > > > The results are amazing and I'd say Jaroslav has done some mistakes in
> > > > his handmade asm.
> > >
> > > I don't think so. It
>At Thu, 20 Feb 2003 06:48:12 -0500,
>Paul Davis wrote:
>>
>> >Is there a functioning driver available for the new Hammerfall HDSP9652
>> >yet?
>>
>> the hdsp driver works, but the mixer register configuration is
>> different and so the volumes cannot be set. i have not yet found
>> the time to m
At Thu, 20 Feb 2003 06:48:12 -0500,
Paul Davis wrote:
>
> >Is there a functioning driver available for the new Hammerfall HDSP9652
> >yet?
>
> the hdsp driver works, but the mixer register configuration is
> different and so the volumes cannot be set. i have not yet found
> the time to merge the
On Thu, 20 Feb 2003, Abramo Bagnara wrote:
> Jaroslav Kysela wrote:
> >
> > On Wed, 19 Feb 2003, Abramo Bagnara wrote:
> >
> > > The results are amazing and I'd say Jaroslav has done some mistakes in
> > > his handmade asm.
> >
> > I don't think so. It seems that my brain still remembers assemb
On Thursday 20 February 2003 16.40, Giuliano Pochini wrote:
[...]
> According to official docs, 20 and 24bit versions have only a
> different DAC. The driver always sends data in the same format.
> I've not looked at the sources yet.
Yes, that seems to be the case. All models use 24 bit signal pat
On 20-Feb-2003 David Olofson wrote:
> On Thursday 20 February 2003 10.17, Giuliano Pochini wrote:
>> Is someone writing drivers for Echoaudio cards ? Perhaps
>> I'll buy one soon and I can try to write drivers if nobody
>> alse is working on it.
>
> I have an old Layla20 and intend to write a dri
On Thu, Feb 20, 2003 at 10:21:17AM +0100, Takashi Iwai wrote:
> AC3 passthrough is ok for some cs46xx cards, hercules game theater XP
> and terratec x-fire 1024. on these cards, so far, 2 codecs are
> supported.
>
> for supporting 3 codecs, we likely need another dsp codes...
I am sorry to both
I have a rme digi96 pst, and was wondering 2 things, how do I access the
digital channels 1-8 both from users space and programmatically?
ljp
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At Thu, 20 Feb 2003 15:14:45 +0100 (CET),
Jaroslav wrote:
>
> If there is a way to determine the connection between sequencer port and
> rawmidi device, I'm ok with any solution. Otherwise, please, solve this
> problem.
one of the advantages of the sequencer is that it hides the lowlevel
implem
On Thursday 20 February 2003 10.17, Giuliano Pochini wrote:
> Is someone writing drivers for Echoaudio cards ? Perhaps
> I'll buy one soon and I can try to write drivers if nobody
> alse is working on it.
I have an old Layla20 and intend to write a driver for it. It
shouldn't be too much work to
On Thu, 20 Feb 2003, Takashi Iwai wrote:
> At Thu, 20 Feb 2003 09:55:18 +0100 (CET),
> Jaroslav wrote:
> >
> > On Thu, 20 Feb 2003, Pedro Lopez-Cabanillas wrote:
> >
> > > On Wednesday 19 February 2003 14:51, Takashi Iwai wrote:
> > > > there is the name field for each rawmidi substream, and the
>Is there a functioning driver available for the new Hammerfall HDSP9652
>yet?
the hdsp driver works, but the mixer register configuration is
different and so the volumes cannot be set. i have not yet found
the time to merge the "code" from RME into the existing hdsp
driver. if someone else wants
At Thu, 20 Feb 2003 01:01:52 -0800,
Jaroslaw Sobierski wrote:
>
> Quoting Takashi Iwai <[EMAIL PROTECTED]>:
>
> > At Mon, 17 Feb 2003 08:01:27 -0800,
> > Jaroslaw Sobierski wrote:
> > >
> > >
> > > Another question though. I still cannot get the AC3 pass-through to work.
> > > I assume the spdi
At Thu, 20 Feb 2003 09:55:18 +0100 (CET),
Jaroslav wrote:
>
> On Thu, 20 Feb 2003, Pedro Lopez-Cabanillas wrote:
>
> > On Wednesday 19 February 2003 14:51, Takashi Iwai wrote:
> > > there is the name field for each rawmidi substream, and the sequencer
> > > looks up this string.
> > >
> > > this
At Wed, 19 Feb 2003 14:20:51 +0100,
Gregoire Favre wrote:
>
> Hello again,
>
> is what written under http://alsa.opensrc.org/index.php?page=cs46xx
>
> "Dolby Digital and DTS passthrough is currently un-implemented."
>
> right?
AC3 passthrough is ok for some cs46xx cards, hercules game theater
At Wed, 19 Feb 2003 18:06:22 +0100,
Marc Titinger wrote:
>
> Hi All,
>
> I can't figure out from the code what are the implications of having
> the flag SNDRV_PCM_INFO_BLOCK_TRANSFER set in a
> snd_pcm_hardware_t.info field ?
this is an inherited flag from the older version, and plays no big
ro
Is someone writing drivers for Echoaudio cards ? Perhaps
I'll buy one soon and I can try to write drivers if nobody
alse is working on it.
Bye.
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At Thu, 20 Feb 2003 05:51:01 +0300,
Anton Worshevsky wrote:
>
> Now i made tests with last cvs for 4 channel too :
> VIA8235 + ALC650
> 0x6a.14=0 : 4ch - OK; 5.1 - swaped
> 0x6a.14=1 : 4ch - swaped; 5.1 - OK
>
> As you see - setting bit 14 of 0x6a reg is not an option.
>
> We need to apply follo
At Thu, 20 Feb 2003 07:19:03 +0100,
p z wrote:
>
> I tink, there is still small bug (I think) - via82xx.c
>
> else
>/* slot mapping: 3,4,6,9,7,8 */
>val = 0x4000;
>snd_ac97_update_bits(chip->ac97, AC97_ALC650_MULTICH, 0xc000, val);
>
> is this correct ???
> val = 0x4000; (I thi
Quoting Takashi Iwai <[EMAIL PROTECTED]>:
> At Mon, 17 Feb 2003 08:01:27 -0800,
> Jaroslaw Sobierski wrote:
> >
> >
> > Another question though. I still cannot get the AC3 pass-through to work.
> > I assume the spdif source must be set to AC-Link (which is hw-reset default
>
> > anyway), and I
On Thu, 20 Feb 2003, Pedro Lopez-Cabanillas wrote:
> On Wednesday 19 February 2003 14:51, Takashi Iwai wrote:
> > there is the name field for each rawmidi substream, and the sequencer
> > looks up this string.
> >
> > this reminds me the former patch by Clemens, and i've forgotten to
> > check thi
Jaroslav Kysela wrote:
>
> On Wed, 19 Feb 2003, Abramo Bagnara wrote:
>
> > The results are amazing and I'd say Jaroslav has done some mistakes in
> > his handmade asm.
>
> I don't think so. It seems that my brain still remembers assembler ;-)
> You passed wrong values to my code so it did unali
Jaroslaw Sobierski wrote:
>
>
> s16 s=sample;
> if (unlikely(sample != (s32)s))
>
I've verified exactly this yesterday evening, but it's less efficient
than ordinary boundary check.
--
Abramo Bagnara mailto:[EMAIL PROTECTED]
Opera Unica
Quoting Takashi Iwai <[EMAIL PROTECTED]>:
> At Mon, 17 Feb 2003 11:36:34 -0800,
> Jaroslaw Sobierski wrote:
> >
> > OK, here is the unified diff. Two of them actually, i separated the
> > different options for better clarity.
> ...
> >
> >Option 2 : The enumerated (more readable but more co
Quoting Jaroslav Kysela <[EMAIL PROTECTED]>:
> I don't think so. It seems that my brain still remembers assembler ;-)
...
> sample = *sum;
> s16 s;
> - if (unlikely(sample & 0x))
> + if (unlikely(sample & 0
Jaroslav Kysela wrote:
>
> On Wed, 19 Feb 2003, Abramo Bagnara wrote:
>
> > The results are amazing and I'd say Jaroslav has done some mistakes in
> > his handmade asm.
>
> I don't think so. It seems that my brain still remembers assembler ;-)
I've no doubts about that ;-)
> You passed wrong v
Hi all !
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