On Sun, 6 Jun 2004, Glenn Maynard wrote:
On Sun, Jun 06, 2004 at 12:40:31PM +0200, Jaroslav Kysela wrote:
Any relationship to the fact that I can only allocate 21 subdevices with
ALSA, but 31 with DirectSound?
Yes, 64 / 3 = 21 .
That stinks (but if it's necessary for decent
Roc Wu wrote:
# ./aplay -t wav -f U8 -r 22050 alarm.wav
Playing WAVE 'alarm.wav' : Unsigned 8 bit, Rate 22050 Hz, Mono
ALSA lib pcm_plug.c:727:(snd_pcm_plug_hw_refine_schange)
Unable to find an usable access for 'default'
aplay: set_params:832: Sample format non available
And the
Florin Andrei wrote:
What's the max number of cards in a system that can be used by ALSA
simultaneously?
8
What's the max number of MIDI ports that's supported by ALSA?
There can be up to 8 rawmidi devices per card, but each device can
have an unlimited number of subdevices.
OSS emulation
On Mon, 7 Jun 2004, Clemens Ladisch wrote:
Florin Andrei wrote:
What's the max number of cards in a system that can be used by ALSA
simultaneously?
The answers to the questions above - are they in the docs?
Use the Source, Luke! :-)
Note that applications shouldn't rely on these
--- Clemens Ladisch [EMAIL PROTECTED]
Roc Wu wrote:
# ./aplay -t wav -f U8 -r 22050 alarm.wav
Playing WAVE 'alarm.wav' : Unsigned 8 bit, Rate
22050 Hz, Mono
ALSA lib
pcm_plug.c:727:(snd_pcm_plug_hw_refine_schange)
Unable to find an usable access for 'default'
aplay:
On Mon, Jun 07, 2004 at 05:25:22PM +0800, Roc Wu wrote:
--- Clemens Ladisch [EMAIL PROTECTED] µÄÕýÎÄ£º
Roc Wu wrote:
# ./aplay -t wav -f U8 -r 22050 alarm.wav
Playing WAVE 'alarm.wav' : Unsigned 8 bit, Rate
22050 Hz, Mono
ALSA lib
pcm_plug.c:727:(snd_pcm_plug_hw_refine_schange)
On Mon, 7 Jun 2004, Russell King wrote:
Actually, I disagree. It's an ALSA bug. The warning is created if
the AACI close method is called while the DMA or IO is still running.
If DMA is still running here, we've already freed the DMA buffer, so
we're either reading from or writing to memory
Roc Wu wrote:
Unable to find an usable access for 'default'
aplay: set_params:832: Sample format non
available
Yes. Thanks for your replay. Maybe I should send the
mail to arm-linux mailist.
PS. Could you recommend some docs about the ALSA
internals and Low level drivers? There are too many
docs
At Mon, 7 Jun 2004 12:43:01 +0200 (CEST),
Jaroslav wrote:
On Mon, 7 Jun 2004, Russell King wrote:
Actually, I disagree. It's an ALSA bug. The warning is created if
the AACI close method is called while the DMA or IO is still running.
If DMA is still running here, we've already freed
At Mon, 7 Jun 2004 14:08:17 +0100,
Russell King wrote:
On Mon, Jun 07, 2004 at 02:45:20PM +0200, Takashi Iwai wrote:
i guess so, too. as you can see in the original post, the error
returned from hw_params callback (sample not available), thus it
doesn't call trigger(START) callback yet
At Mon, 7 Jun 2004 14:51:13 +0100,
Russell King wrote:
On Mon, Jun 07, 2004 at 03:40:23PM +0200, Takashi Iwai wrote:
At Mon, 7 Jun 2004 14:08:17 +0100,
Russell King wrote:
On Mon, Jun 07, 2004 at 02:45:20PM +0200, Takashi Iwai wrote:
i guess so, too. as you can see in the
Russell King wrote:
But unfortunately I don't have the driver code myself to be able to
comment, so its probably been fscked.
If the code was posted publically, the author of the code would get a
lot more useful help from more eyes.
---
This
I've been trying to write an ALSA driver for the AC97 port on an
embedded AMD au1000 MIPS processor but am having some difficulties. The
processor's DMA controller has two buffers which automatically toggle
back and forth once the buffer is full. My problem is that when I
playback a wave file
On Mon, Jun 07, 2004 at 04:18:55PM +0200, Takashi Iwai wrote:
You're right. The error was not txcr, but in another WARN_ON() for
checking chan-tx_substream (line 404)! (Russell, you mislead this,
too ;)
Well I don't have the exact source which this guy is using, so I can
only guess.
The
On Mon, Jun 07, 2004 at 03:24:46PM +0100, James Courtier-Dutton wrote:
Russell King wrote:
But unfortunately I don't have the driver code myself to be able to
comment, so its probably been fscked.
If the code was posted publically, the author of the code would get a
lot more useful
What's the max number of cards in a system that can be used by ALSA
simultaneously?
What's the max number of MIDI ports that's supported by ALSA?
What are other similar limits?
The answers to the questions above - are they in the docs?
If yes, where? (i couldn't find anything related)
--
Florin
On Sat, 5 Jun 2004, Chris Purnell wrote:
For PCM playback the EMU10K1 driver is allocating an extra voice.
This is somewhat wastefull and I kind of need all 64 voices.
Is appears to be using it to generate the period interrupts.
Does anyone know what it would take to rewrite the driver to
On Sun, 6 Jun 2004, Glenn Maynard wrote:
On Sat, Jun 05, 2004 at 12:59:36PM +, Chris Purnell wrote:
For PCM playback the EMU10K1 driver is allocating an extra voice.
This is somewhat wastefull and I kind of need all 64 voices.
Is appears to be using it to generate the period
On Sat, 5 Jun 2004, Owen Fraser-Green wrote:
Hi,
After some tinkering around with ALSA's mixer with a couple of different
cards it's become apparent to me that there's not too much
standardization when it comes to track naming.
It's work in progress. We need to create an abstract layer on
On Sun, Jun 06, 2004 at 12:40:31PM +0200, Jaroslav Kysela wrote:
Any relationship to the fact that I can only allocate 21 subdevices with
ALSA, but 31 with DirectSound?
Yes, 64 / 3 = 21 .
That stinks (but if it's necessary for decent latency, which it doesn't
get in Windows, oh well).
Hi folks:
I don't want to post my question again. But there is
not any response about my question. Maybe I did some
thing wrong about asking question, please let me know.
I will change my method, Thanks a lot.
Please give me some help.
Best Regards
--- Roc Wu [EMAIL PROTECTED] Hi
folks:
Hi,
After some tinkering around with ALSA's mixer with a couple of different
cards it's become apparent to me that there's not too much
standardization when it comes to track naming.
What I'd like to be able to do is adjust the different levels on
different channels and to adjust the mast mixer
I found these broken links in the page
http://www.alsa-project.org/documentation.php3 :
http://www.alsa-project.org/~iwai/*
http://www.alsa-project.org/~frank/alsa-sequencer/
http://www.alsa-project.org/~valentyn/*
http://www.alsa-project.org/~jfulmer/alsa-faq.html
--
Giuliano.
On Saturday 05 June 2004 09:51, Owen Fraser-Green wrote:
Now, I know an envy24mixer exists to make life easier for me but surely
it would be a bit nicer if the mixer interface provided a more
consistent abstraction. In my situation, I want my software to be useful
without a screen so the
On Sat, Jun 05, 2004 at 12:59:36PM +, Chris Purnell wrote:
For PCM playback the EMU10K1 driver is allocating an extra voice.
This is somewhat wastefull and I kind of need all 64 voices.
Is appears to be using it to generate the period interrupts.
Does anyone know what it would take to
On Sun, 30 May 2004, Giuliano Pochini wrote:
On Wed, 26 May 2004 09:52:06 +0200 (CEST)
Jaroslav Kysela [EMAIL PROTECTED] wrote:
ALSA does not know about this. All period sizes must be equal.
I thought about this again. Are you sure all periods must be aqual ?
When I record or
At Mon, 31 May 2004 12:58:53 +0200,
Pedro Lopez-Cabanillas wrote:
On Monday 31 May 2004 03:07, I wrote:
My Gigabyte motherboard's integrated sound interface used to work with this
driver, but it doesn't with 1.0.5
Any program (amixer info, alsactl restore, alsamixer ...) trying to use
At Thu, 27 May 2004 21:26:22 +0200,
Martin Langer wrote:
Hi,
if someone is interested in a free (I mean GPL) firmware for the Tascam
US-X2Y devices I can offer an open source replacement for the second stage
loader alsa-firmware/usx2yloader/tascam_loader.ihx. All three US-X2Y devices
At Sat, 29 May 2004 16:05:05 +0100,
James Courtier-Dutton wrote:
Hi,
I am trying to do work on the Audigy LS driver.
I have now discovered that I can send sound to the Front, Rear and
Center/LFE.
I have not found out how to set the amount of interleaved channels that
the sound card
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
The install.txt tells one how to patch the current alsa-driver 1.0.5a
with it, and also explains where some other files should be put.
The indentation might need correcting before
James Courtier-Dutton wrote:
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
Are you going to work on any other Audigy-series drivers, e.g. Audigy 2 ZS?
I'm finding the emu10k1 driver in alsa-driver-1.0.5a has serious problems
with
At Tue, 1 Jun 2004 13:18:26 GMT,
[EMAIL PROTECTED] wrote:
both are ok for ALSA native apps. you can specify the substream index
in the configuration.
the latter would be easier for OSS compatible layer, though.
Do you have a preference? I can change the emu10k1x driver to do (2)
as
William wrote:
James Courtier-Dutton wrote:
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
Are you going to work on any other Audigy-series drivers, e.g. Audigy 2 ZS?
I'm finding the emu10k1 driver in alsa-driver-1.0.5a has serious
James Courtier-Dutton wrote:
William wrote:
James Courtier-Dutton wrote:
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
Are you going to work on any other Audigy-series drivers, e.g. Audigy 2 ZS?
I'm finding the emu10k1 driver
At 31 May 2004 20:24:26 -0700,
Fernando Pablo Lopez-Lezcano wrote:
I'm trying to build Takashi's version of latencytest (0.5.3) and
apparently something has changed in the kernel (or I have something that
I need to turn on in the config options?). This is what I get:
known problem... i'll
At Mon, 31 May 2004 16:59:47 +0200,
Robert Rozman wrote:
Hi,
I'd like to have device with 4 independent stereo channels. I wonder what
card is lower cost, 7+1 channel, widely known, that could be also used for 4
independent stereo devices under Alsa - it's important to work flawlessly in
At Sun, 30 May 2004 13:29:23 -0400,
Ivica Ico Bukvic wrote:
Hi all,
I've recompiled my 2.6.5 kernel with rtc compiled in and was able to install
latency-test module. However, now when I run the run_tests the program goes
through initial 2 draw 500x500 square tests and then every following
William wrote:
James Courtier-Dutton wrote:
William wrote:
James Courtier-Dutton wrote:
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
Are you going to work on any other Audigy-series drivers, e.g. Audigy 2 ZS?
I'm finding the
James Courtier-Dutton wrote:
I have played some .mid files now, but I have not noticed any problems
with my Audigy 2.
When I downloaded the fonts file, mine had a different name to yours.
Yours: WST25FStein_00Aug14.SF2
Mine: WST25FStein_00Sep22.SF2
It makes no difference which of these
At Tue, 1 Jun 2004 14:58:38 +0100,
William wrote:
James Courtier-Dutton wrote:
William wrote:
James Courtier-Dutton wrote:
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
Are you going to work on any other
Takashi Iwai wrote:
William wrote:
Random intermittent distortion means, e.g. the sound during MIDI playback
becomes muffled for a minute or two and then returns to normal sound
quality, or the sound wrongly becomes mono for a few seconds, and then
returns to stereo.
Also, it seems emu10k1
At Tue, 1 Jun 2004 16:46:45 +0100,
William wrote:
Experiment 2:
-
Load the manufacturer's Standard GM Midi file:
$ asfxload /etc/synthgm.sbk
$ cat /proc/asound/card0/wavetableD*
Device: Emu10k1
Ports: 4
Addresses: 65:0 65:1 65:2 65:3
Use Counter: 0
Max Voices: 64
Takashi Iwai wrote:
William wrote:
Experiment 2:
-
Load the manufacturer's Standard GM Midi file:
$ asfxload /etc/synthgm.sbk
$ cat /proc/asound/card0/wavetableD*
Device: Emu10k1
Ports: 4
Addresses: 65:0 65:1 65:2 65:3
Use Counter: 0
Max Voices: 64
Allocated Voices:
At Tue, 1 Jun 2004 17:43:08 +0100,
William wrote:
Takashi Iwai wrote:
William wrote:
Experiment 2:
-
Load the manufacturer's Standard GM Midi file:
$ asfxload /etc/synthgm.sbk
$ cat /proc/asound/card0/wavetableD*
Device: Emu10k1
Ports: 4
Addresses: 65:0
Takashi Iwai wrote:
William wrote:
Takashi Iwai wrote:
only 4096 bytes (= 1 page) allocated. i guess this file is for ROM
soundfonts on SB AWE boards, not for SB Live/Audigy?
That's odd because the file is 34832 bytes long
(see http://christian.datzko.ch/computer/synthgm.sbk)
a
On Tue, Jun 01, 2004 at 10:39:38AM +0200, Takashi Iwai wrote:
At Thu, 27 May 2004 21:26:22 +0200,
Martin Langer wrote:
Hi,
if someone is interested in a free (I mean GPL) firmware for the Tascam
US-X2Y devices I can offer an open source replacement for the second stage
loader
On Mon, May 31, 2004 at 11:26:21PM +0200, JoDaY wrote:
Martin Langer wrote:
On Mon, May 31, 2004 at 05:45:24PM +0200, JoDaY wrote:
Hi all,
I've 2 install of linux, the first one is the oldest where hotplug
is able to load firmware when I plug a Tascam us-122.
But if I use snd-usb-usx2y.o
On Tuesday 01 June 2004 10:40, Takashi Iwai wrote:
Pedro Lopez-Cabanillas wrote:
Anybody using the same hardware should avoid the 1.0.5 driver, IMO.
Relevant sources are ac97_codec.c and ac97_patch.c. The problem was
related to a mutex deadlock.
yep, sorry, my bad.
the bug hits AC97
When I attempt to do a 4 channel output using the aboce chip set, and I
control it with 'alsamixer' I note that when Alternate Level to
Surround Out is [Off] it is really on. I really don't know if this
is a fault in 'alsamixer', or the driver. It appears that these are
reversed in sense. I
On Mon, 31 May 2004, Roc Wu wrote:
Hello alls:
Sorry for post the mail again. I posted it several
days ago, but no response. Anybody can give me some
hints?
I cross compiled the alsa-lib-1.04 to arm platform.
The 2.6.6 kernel including alsa driver is ok on our
ARM board, so I want to
Hi,
I have an Audigy2 el cheepo edition, and an Audigy LS.
The Audigy2 luckily has a separate digital output jack, so I can easily
connect a mono or stereo jack into the Audigy2, have an RCA plug on the
other end, and plug it into an external AC3 decoder, and AC3 passthru works.
The Audigy LS
On Monday 31 May 2004 03:07, I wrote:
My Gigabyte motherboard's integrated sound interface used to work with this
driver, but it doesn't with 1.0.5
Any program (amixer info, alsactl restore, alsamixer ...) trying to use the
control interface hardly hangs, and it is impossible to kill it at
On Mon, 31 May 2004, Pedro Lopez-Cabanillas wrote:
Mobo: Gigabyte 7VT600 with chipset VIA KT600
Sound: Integrated Realtek ALC655 AC97 codec
http://tw.giga-byte.com/MotherBoard/Products/Products_Spec_GA-7VT600.htm
Anybody using the same hardware should avoid the 1.0.5 driver, IMO.
Hi,
I'd like to have device with 4 independent stereo channels. I wonder what
card is lower cost, 7+1 channel, widely known, that could be also used for 4
independent stereo devices under Alsa - it's important to work flawlessly in
this mode (I'm running alc650 as 3 stereo channels but do get
With regard to my soundcard problems. I actually have two cards in my
system. The first is an AC97 thing on an Intel motherboard, judging by
this dmesg output I'd say its an Intel 8xx chip...
Intel 810 + AC97 Audio, version 0.24, 15:50:18 Oct 29 2003
PCI: Found IRQ 9 for device 00:1f.5
PCI:
On Wed, 26 May 2004 09:52:06 +0200 (CEST)
Jaroslav Kysela [EMAIL PROTECTED] wrote:
ALSA does not know about this. All period sizes must be equal.
I thought about this again. Are you sure all periods must be aqual ?
When I record or play something using unequal periods, sound is
Hi,
I've got my header file installed now. I had to install two rpms
alsa-lib-1.0.4-12.rhfc1.at.i386.rpm
alsa-lib-devel-1.0.4-12.rhfc1.at.i386.rpm
on my default Fedora Core 1 install. I restarted and my cdplayer is
still working so thats good. I can also do my compiles with
alsa/asoundlib.h
Martin Langer wrote:
On Mon, May 31, 2004 at 05:45:24PM +0200, JoDaY wrote:
Hi all,
I've 2 install of linux, the first one is the oldest where hotplug
is able to load firmware when I plug a Tascam us-122.
But if I use snd-usb-usx2y.o (seen in depmod), hotplug is not able to
load the right
Zack Borschuk wrote:
I was wondering if a petition asking Creative Labs to release the needed
information to create efficient support for the Audigy LS would be
plausible or not. Please let me know, so that if it is plausible, I
could start working on getting the petition signed by enough
I'm trying to build Takashi's version of latencytest (0.5.3) and
apparently something has changed in the kernel (or I have something that
I need to turn on in the config options?). This is what I get:
# make -f Makefile.module
make -C /lib/modules/`uname -r`/build SUBDIRS=`pwd` modules
make[1]:
Hello,
I would like to report that I have successfully now recorded @ 48KHz and
24 bit using arecord. This previously didn't work with the alsa 0.9
version on a usb device. Works fantasticly now !
Full duplex is also working well.
Matt
--
http://flatmax.org
WSOLA TimeScale Audio Mod :
Hi Tim and all,
On Sat, 29 May 2004 16:34:49 +0200 (CEST)
Tim Goetze [EMAIL PROTECTED] wrote:
[Remi Bernhard]
I tried with kernel 2.4.25 + lowlatency patch.
I have the same xruns :-/
Any other idea ?
checklist:
* running jackd as root? with -R option?
Tried - failed.
* tried
Someone may wish to look into the cause of this... CONFIG_PCI is
unselected in this case.
CC [M] sound/core/oss/mixer_oss.o
In file included from sound/core/oss/mixer_oss.c:26:
include/sound/core.h:215: warning: `struct pci_dev' declared inside parameter list
include/sound/core.h:215: warning:
Am Donnerstag 27 Mai 2004 21:26 schrieb Martin Langer:
Hi,
if someone is interested in a free (I mean GPL) firmware for the Tascam
US-X2Y devices I can offer an open source replacement for the second stage
loader alsa-firmware/usx2yloader/tascam_loader.ihx. All three US-X2Y
devices need this
On Sat, 29 May 2004, Frank W. Miller wrote:
using ALSA. When I try to run my existing binary that makes OSS calls on
top of Fedora that is using an ALSA driver for my soundcard, I get some
strange behaviour, some of the calls don't work the same. For example, the
card supports full-duplex
On Sun, 2004-05-30 at 04:09, Remi Bernhard wrote:
Hi Tim and all,
On Sat, 29 May 2004 16:34:49 +0200 (CEST)
Tim Goetze [EMAIL PROTECTED] wrote:
[Remi Bernhard]
I tried with kernel 2.4.25 + lowlatency patch.
I have the same xruns :-/
Any other idea ?
checklist:
*
Hi everyone,
Is there any documentation on writing alsa plugins?
I could not find anything really helpful om the web site.
I have a system which I am working on where it would be very helpful to do
some specialised processing below the level of jackd, and it seems that an
alsa plugin may well
Hello list,
after compiling the new 1.0.5 packages, the sound modules
do nor start correctly, resulting in an unbootable system,
or more correctly an unrebootable system.
The system hangs after, in my case, both sound card modules
loaded (loading snd-intel8x0 done loading snd-ice1712 done).
ALSA driver available from:
http://www.superbug.demon.co.uk/alsa/
* FEATURES currently supported:
*Front, Rear and Center/LFE.
*Surround40 and Surround51.
*Capture from MIC input.
*
* BUGS:
*--
*
* TODO:
*Need to add a way to select capture source.
*4 Capture channels,
Hi all,
I've recompiled my 2.6.5 kernel with rtc compiled in and was able to install
latency-test module. However, now when I run the run_tests the program goes
through initial 2 draw 500x500 square tests and then every following test
simply does something like this:
starting diskread (or
Hello alls:
Sorry for post the mail again. I posted it several
days ago, but no response. Anybody can give me some
hints?
I cross compiled the alsa-lib-1.04 to arm platform.
The 2.6.6 kernel including alsa driver is ok on our
ARM board, so I want to test the driver. how to do it?
I think maybe
Hello all,
ALSA 1.0.5 release is available for download.
Jaroslav
Changes:
* alsa-driver
- use the new module_param*() functions
- clean up of power-management codes
- removed superfluous warning messages after pci_module_init()
- fixed the
Hello,
I read a lot a great thing about the audiophile 24/96 on the net, and i
decided to buy it to get better latency than my old sblive! I just
received the audiophile 24/96 and unfortunatly, i have a lot of XRuns
with when i run jackd.
I already tried the following :
- i changed irq (several
Jaroslav Kysela wrote:
ALSA 1.0.5 release is available for download.
URL from http://www.alsa-project.org/:
a href=ftp://ftp.alsa-project.org/pub/firmware/alsa-firmware-1.0.5.tar.bz2;1.0.5/a
$ wget ftp://ftp.alsa-project.org/pub/firmware/alsa-firmware-1.0.5.tar.bz2
No such file
[Remi Bernhard]
I read a lot a great thing about the audiophile 24/96 on the net, and i
decided to buy it to get better latency than my old sblive! I just
received the audiophile 24/96 and unfortunatly, i have a lot of XRuns
with when i run jackd.
my old system had various 2.4.x-ll kernels, asus
On Sat, 2004-05-29 at 06:16, Remi Bernhard wrote:
So i have 2.6.5 kernel, and not 2.4.x. Do you think i should install
2.4.25 instead ?
(i gather that 2.4 is still the better choice for low-latency
operations.)
I read the alsa-* archives, and i saw a thread with two people who
A linux user, Greg Turpin (from Colorado, USA), kindly donated an Audigy
LS to me, so that I could try to provide an Audigy LS driver to the ALSA
project.
As you all know, I have already been relatively successful and now have
sound coming from the Front Speakers, and hope to improve support
[Remi Bernhard]
So i have 2.6.5 kernel, and not 2.4.x. Do you think i should install
2.4.25 instead ?
i do think so because i never saw any hard facts about latency posted
for the 2.6 series, while 2.4.x-ll is well documented and proven to
work quite well here and elsewhere.
it _could_ also be
Hi,
As far as minimizing buffer size is concerned - if
you are doing live audio or you want to monitor recording with inboard
effects turned on you need to have a small buffer size. With the
envy24 chipset cards you can use hardware monitoring so you can set
the buffer size to 2048 and
[Remi Bernhard]
To be more sharp, i need to reduce latency because i use softsynth
(zynaddsubfx, fluidsynth, etc.) that needs a low latency when playing
from midi keyboard, and if not, there is a lag time, that man can
hear, between the time when you hit the key, and the time it is played.
I
On Sat, 29 May 2004 13:44:40 +0200 (CEST)
Tim Goetze [EMAIL PROTECTED] wrote:
[Remi Bernhard]
So i have 2.6.5 kernel, and not 2.4.x. Do you think i should install
2.4.25 instead ?
i do think so because i never saw any hard facts about latency posted
for the 2.6 series, while 2.4.x-ll is
On Sat, 2004-05-29 at 07:13, Tim Goetze wrote:
[Remi Bernhard]
To be more sharp, i need to reduce latency because i use softsynth
(zynaddsubfx, fluidsynth, etc.) that needs a low latency when playing
from midi keyboard, and if not, there is a lag time, that man can
hear, between the time when
[Jan Depner]
On Sat, 2004-05-29 at 07:13, Tim Goetze wrote:
[Remi Bernhard]
To be more sharp, i need to reduce latency because i use softsynth
(zynaddsubfx, fluidsynth, etc.) that needs a low latency when playing
from midi keyboard, and if not, there is a lag time, that man can
hear, between
[Remi Bernhard]
I tried with kernel 2.4.25 + lowlatency patch.
I have the same xruns :-/
Any other idea ?
checklist:
* running jackd as root? with -R option?
* tried running jackd without any clients?
* IDE-DMA enabled?
* ll-sysctl interface chosen at kernel build time? turned it on?
* X
Hi,
I am trying to do work on the Audigy LS driver.
I have now discovered that I can send sound to the Front, Rear and
Center/LFE.
I have not found out how to set the amount of interleaved channels that
the sound card can do, so it is fixed at 2 channels per stream.
The sound card has 4 voices
On Sat, 2004-05-29 at 09:31, Tim Goetze wrote:
depends on what 'system' we're talking about, which isn't really clear
in the first place (nor is it too important, but here goes anyway ...
:)
you can see the whole setup as the system, then latency is the time
from keypress to voltage change
[Jan Depner]
Agreed. But the problem that keeps popping up on the lists is that
people who are not doing live sound, have cards that do hardware
monitoring, and don't need to use a tiny buffer size waste their time
trying to get the minimum buffer size because they think they need to
get
Greetings,
I'm writing a softphone for Linux under Fedora Core 1. I've had this
client working well using OSS under Red Hat 9 for awhile now. I've run
into some problems however when trying to use the OSS calls under
Fedora. I thought I would try to just write an ALSA driver and went to
the
Your best bet is to download the sources and build the packages. That
way you'll know exactly what you've got. Also, at some point you may
have to go into the libraries and do some debugging.
Jan
On Sat, 2004-05-29 at 19:02, Frank W. Miller wrote:
Greetings,
I'm writing a softphone for
Thanks for the reply...
I probably need to be more specific here. I have a driver that works well
with OSS under Red Hat 9. When I reinstalled my system using Fedora Core 1,
I did not need OSS any more. I have nice sound coming from my speakers
using ALSA. When I try to run my existing
I would be willing to bet that the problem is that you don't have the
development libraries. You can play via ALSA without having them. I
don't think Fedora is very big on loading all of the development
libraries. I run Fedora Core 1 on three systems at my house. When I
load them I load
The Fedora Core should have some RPM with the ALSA header files. Since
it is not installed, try looking in the CDs with whatever tool Fedora
provides for searching the RPMS. From a quick look at the names of the
RPMS I cannot find where the ALSA header files are but since I do not
run Fedora
Cournapeau David [EMAIL PROTECTED] writes:
Hi there,
For my research, I need to use audio with matlab under linux, and
sound support of matlab is kind of... well, crappy (basically, it
is opening the /dev/audio file and write to it; on my computer, it
doesn't seem to work).
Måns Rullgård wrote:
If you are playing fairly short pieces of sound you might be able to
use this function I wrote some time ago:
function playsnd(y,fs,bits)
wf = tempname;
ws = warning;
warning off
wavwrite(y, fs, bits, wf);
warning ws
wf = [wf '.wav'];
[s,o] = unix(sprintf('aplay %s', wf));
Cournapeau David [EMAIL PROTECTED] writes:
Måns Rullgård wrote:
If you are playing fairly short pieces of sound you might be able to
use this function I wrote some time ago:
function playsnd(y,fs,bits)
wf = tempname;
ws = warning;
warning off
wavwrite(y, fs, bits, wf);
warning ws
wf = [wf
Is there a way to make alsa-lib stop spamming the terminal with error
messages, for instance when a non-blocking open fails?
--
Måns Rullgård
[EMAIL PROTECTED]
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At Thu, 27 May 2004 16:54:38 +0200 (CEST),
Thomas Charbonnel wrote:
Thanks Thomas, I really appreciate the work you have done making this
all work. I will try and add some notes on the alsa site for the RME
9632 on the mixer settings
Ed W
You're welcome. It would be indeed nice
Hi,
At Thu, 27 May 2004 22:43:06 -0400,
Trond Myklebust wrote:
It appears to boil down to changeset
[EMAIL PROTECTED]|ChangeSet|20040517133203|52763.
More specifically to the line which adds the AC97_SCAP_SKIP_MODEM flag
to the mixer probe. That flag causes the mixer probe to exit with an
Takashi Iwai wrote:
At Thu, 27 May 2004 20:17:17 +0100,
James Courtier-Dutton wrote:
Here is my first go at Audigy LS support.
It can play sound to the front speakers.
great!
/* hardware definition */
static snd_pcm_hardware_t snd_audigyls_playback_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
At Fri, 28 May 2004 12:16:09 +0100,
James Courtier-Dutton wrote:
Takashi Iwai wrote:
At Thu, 27 May 2004 20:17:17 +0100,
James Courtier-Dutton wrote:
Here is my first go at Audigy LS support.
It can play sound to the front speakers.
great!
/* hardware definition */
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