Does emu10k1 work at a user settable sample rate? 44100 Hz? 32000Hz?
MS Windows driver has a good sounding reverb. Could this reverb dsp
code be extracted from the MS Windows driver and used as is in Alsa?
It's not so easy and I'm sure that we'll violate some patent doing things
Hi all:
On Fri, 2002-10-11 at 11:53, torben hohn wrote:
Does emu10k1 work at a user settable sample rate? 44100 Hz? 32000Hz?
I answer to the original poster first. The Emu10k1 is *locked* at a
48Khz rate. All sources and outputs will be converted to and from this
rate for processing.
At 11 Oct 2002 09:16:00 +1100,
Zenaan Harkness wrote:
Forwarde as requested. Sorry if it gets through twice or not at all...
zen
-Forwarded Message-
From: Paul Davis [EMAIL PROTECTED]
To: Zenaan Harkness [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: Re:
At Thu, 10 Oct 2002 20:12:12 +0200,
Helmut Obertanner wrote:
Am Donnerstag, 10. Oktober 2002 17:14 schrieben Sie:
At Thu, 10 Oct 2002 16:20:31 +0200,
Helmut Obertanner wrote:
How can i tell my soundcard to play 1 stereochannel on Frontspeakers
and another stereochannel on
I might be a complete idiot, but when I load up the usb-audio drivers
for my extigy out of the cvs tree, I'm not even getting the devices I
need to test the stuff. I'm using devfs, and and hwC0D0 doesnt pop
up.
I the same guy that once claimed that he was going to work on drivers
for the
At 10 Oct 2002 11:11:06 -0500,
Jack O'Quin wrote:
Peter L Jones [EMAIL PROTECTED] writes:
I think it's not so nice. Anyone running ALSA 0.9 should know it's
not a release version and be happy to have (some breakage). If
all that's needed is a quick edit of modules.conf, it
At Fri, 11 Oct 2002 09:13:12 -0500,
[EMAIL PROTECTED] wrote:
I might be a complete idiot, but when I load up the usb-audio drivers
for my extigy out of the cvs tree, I'm not even getting the devices I
need to test the stuff. I'm using devfs, and and hwC0D0 doesnt pop
up.
hwCxDx is a hwdep
On Fri, 11 Oct 2002, James Courtier-Dutton wrote:
Thankyou, I will use snd_pcm_drop(), but as a side note, what actually
does snd_pcm_reset() do.
Just resetting delay to 0 does not make much sense to me.
It drops all samples in the ring buffer (thus reseting delay to 0). Note
David Sankel wrote:
Hello ALSA developers,
I was wondering if there is a requirement list for
1.0.0 release of ALSA. If not, what would be expected
in a 1.0.0 release? (I've got a few brain/cpu cycles
to spare towards a 1.0.0 release)
I know the topic of sound deamons has really
Hi Mark,
At Fri, 11 Oct 2002 09:36:59 -0500,
[EMAIL PROTECTED] wrote:
I don't feel like spamming the list with newbie junk, but I still want
to help as much as possible, so I'm just going to email you
personally. If you think this stuff should still go to the list
anyway, just say.
why
On Fri, Oct 11, 2002 at 05:07:25PM +0200, Takashi Iwai wrote:
Here is everything syslog spits out at me when I start up alsa (the
cant locate module stuff is obviously nothing, along with one or two
other things, but I'll leave them in for completeness sake):
Oct 11 09:30:39 atrophy
At Fri, 11 Oct 2002 10:34:11 -0500,
[EMAIL PROTECTED] wrote:
On Fri, Oct 11, 2002 at 05:07:25PM +0200, Takashi Iwai wrote:
Here is everything syslog spits out at me when I start up alsa (the
cant locate module stuff is obviously nothing, along with one or two
other things, but I'll
At Fri, 11 Oct 2002 18:02:09 +0200,
I wrote:
hmm, could you run alsactl store and show the generated
/etc/asound.state (only for the second card is enough)?
at least we can see whether the controls are parsed properly.
also, please check the kernel message after running the commands
above.
On 11 Oct 2002 10:34:23 +0200, Martin Soto wrote:
Hi all:
On Fri, 2002-10-11 at 11:53, torben hohn wrote:
Does emu10k1 work at a user settable sample rate? 44100 Hz? 32000Hz?
I answer to the original poster first. The Emu10k1 is *locked* at a
48Khz rate. All sources and outputs
On Fri, Oct 11, 2002 at 06:02:09PM +0200, Takashi Iwai wrote:
% aplay -Dplughw:1,0 foo.wav
and for capture
% arecord -Dplughw:1,0 -fcd bar.wav
I'd love to do this, but:
atrophy:/etc/modutils# amixer -c1
amixer: Mixer load error: hw:1
atrophy:/etc/modutils#
At Fri, 11 Oct 2002 13:00:23 -0500,
[EMAIL PROTECTED] wrote:
On Fri, Oct 11, 2002 at 06:27:17PM +0200, Takashi Iwai wrote:
At Fri, 11 Oct 2002 18:02:09 +0200,
I wrote:
hmm, could you run alsactl store and show the generated
/etc/asound.state (only for the second card is enough)?
On Fri, Oct 11, 2002 at 08:12:54PM +0200, Takashi Iwai wrote:
please rebuild the alsa drivers with configure option
--with-debug=detect ? this will add more verbose debug outputs
(ususally annoying).
Small typo in the patch, nothing big just a . that should be a
the attached patch will
Jack O'Quin wrote:
Can someone please explain what terrible problem we're trying to solve
that justifies introducing *any* breakage at all?
ALSA is part of the 2.5 kernel now. It is mainstream Linux software,
good technology, needed by many users. Isn't it about time to start
Hi all,
these enhancements are in CVS for PCM API:
- added snd_pcm_avail() function - this function returns count of
available frames for write or read operations, the position
is read directly from hardware; the snd_pcm_avail_update() call
is still mandatory before any I/O!!!
-
However, one small buglet. When recording from line1 (aux in the
current driver), the sound sometimes sounds very tinny and scratchy. I
can fix this by muting/unmuting the CD and adjusting the CD volume.
Yes, the CD slider in alsamixer. Not the Aux one or the Capture one or
the DAC one. The
Hi Helmut,
At Fri, 11 Oct 2002 11:53:42 +0200,
Helmut Obertanner wrote:
how is that with other cards, is the mastervolume also respoonnsible for the
rearchannels ???
it depends. the later models of cmipci support 4/6 channels playback
via one DMA, i.e. interleaved output. in this case,
At Thu, 10 Oct 2002 20:32:30 -0500,
Neill Bell wrote:
I've read a few postings on here about recording through the S/PDIF
input on CMI8738-based cards, but I haven't seen any solutions. I've
been able to get recording working with my Midiman card by modifying the
cmipci.c file to include the
At Fri, 11 Oct 2002 12:45:11 +1000,
James Courtier-Dutton wrote:
Hello
I was wondering how easy it would be to add a classification to each
control element. (switches, volume, capture on/off etc.)
The classification would be as follows: -
1) Used during capture. I.E. Switches and volume
Hi,
can anyone test the latest cvs snd-usb-audio driver with the SB
Extigy?
now the names of mixer controls on this device became more
understandable.
the mixer topology of extigy is depicted in
alsa-kernel/usb/usbmixer_maps.c. unlike other usb devices, this is a
really complicated one.
i have
Hi,
At Thu, 10 Oct 2002 22:31:33 -0700 (PDT),
Albert Jongkit Wong wrote:
i don't think all of them are broken.
possibly this behavior might be influenced by the period size, etc.
can you check /proc/asound/card0/pcm0p/sub0/hw_params at each case and
compare them?
xmms (oss)
...
Thankyou, I will use snd_pcm_drop(), but as a side note, what actually
does snd_pcm_reset() do.
Just resetting delay to 0 does not make much sense to me.
It drops all samples in the ring buffer (thus reseting delay to 0). Note
that everybody are welcome to improve the current
Hi all!
As promissed, here goes a quick and dirty example showing how to program
the emu10k1 chip (found in the popular Sound Blaster Live! card) with
the ALSA drivers. The example requires ALSA CVS from today and the
alsalib source code in order to compile and run.
The DSP program already
Jaroslav Kysela wrote:
Hi all,
these enhancements are in CVS for PCM API:
- added snd_pcm_avail() function - this function returns count of
available frames for write or read operations, the position
is read directly from hardware; the snd_pcm_avail_update() call
is still
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