Hi,
I'm trying to get my digital output (SPDIFF) to work. I have a Creative
SB Audigy. I managed to get the digital output to work with the OpenSource
emu10k1 drivers at sourceforge (http://sourceforge.net/projects/emu10k1/),
but I'm trying to switch to ALSA drivers, because the documentation with
mplayer suggested switching to them for AC3 passthrough functionality.
So, I built the ALSA stuff from CVS, and got the analog part to work fine.
However, the digital output is not working... Tweaking with the alsamixer
settings does not seem to help.
Currently, the XMMS output plugins I've tried to use are both Esound Output
Plugin 1.2.7 and the OSS Driver 1.2.7 (the former of which worked fine to
produce the digital output with the emu10k1 driver).
Does anyone have any idea how to get this working?
This is the sound portion of lsmod:
Module Size Used by Tainted: P
snd-pcm-oss 43492 1 (autoclean)
snd-mixer-oss 15288 1 (autoclean) [snd-pcm-oss]
snd-emu10k1 79920 2 (autoclean)
snd-pcm 81472 0 (autoclean) [snd-pcm-oss snd-emu10k1]
snd-timer 15272 0 (autoclean) [snd-pcm]
snd-util-mem 3096 0 (autoclean) [snd-emu10k1]
snd-hwdep 5760 0 (autoclean) [snd-emu10k1]
snd-rawmidi 18592 0 (autoclean) [snd-emu10k1]
snd-seq-device 6252 0 (autoclean) [snd-emu10k1 snd-rawmidi]
snd-ac97-codec 37896 0 (autoclean) [snd-emu10k1]
snd 42572 0 (autoclean) [snd-pcm-oss snd-mixer-oss snd-emu10k1 snd-pcm snd-timer snd-util-mem snd-hwdep snd
and this is the sound portion of my modules.conf:
# ALSA portion
alias char-major-116 snd
alias snd-card-0 snd-emu10k1
post-install snd-emu10k1 /usr/sbin/alsactl restore
pre-remove snd-emu10k1 /usr/sbin/alsactl store
# module options should go here
options snd-emu10k1 index=0 extin=0x0fc3 extout=0x1f0f
# OSS/Free portion
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0
# card #1
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss
Some versioning info:
- cat /proc/asound/version
Advanced Linux Sound Architecture Driver Version 0.9.0rc6.
Compiled on Dec 1 2002 for kernel 2.4.20 with versioned symbols.
TIA,
Reinier.
>>Yeah, I'm up for this. I'm not too familiar with making ebuilds, but it
>>looks to be fairly straightforward. At the moment, I've just set up a
>>hierarchy under /opt, where I manually (configure, make) compile +
>>install stuff. The trouble with this is that not all packages give the
>>option to specify a location (--foo-lib=DIR), and assumes stuff is
>>installed in /usr (grrr :)
>
>Alrighty then. I've started to ask questions on the gentoo-user alias about
>how best to deal with cvs ebuilds. Once there there are some answers i'll
>put together a webpage for the files.
>
>I considered going down the /opt path ;-) but it just isn't the gentoo way!
Jon/Mike -
Greetings! I'm new to the list and a gentoo user. I've been playing with ebuilds. I made one for jack-cvs and, which I'll email it to you, and to anyone else who wants. I'm working on some others so if you find a place to throw them up I'd be happy to contribute.
Jonathan Kraut
jkraut1(a)nyc.rr.com
A number of people using the Rosegarden sequencer for MIDI (myself
included) are principally using it to drive ALSA soft synths such
as iiwusynth or timidity. Unlike (say) MusE, which allows you to
start and stop soft synths from the sequencer itself, Rosegarden
treats soft synths just like any other MIDI device: it knows
nothing special about them except the name that ALSA returns.
It's the user's responsibility to start and stop synths and ensure
that the right patches and so forth are available.
This makes for quite a bit of flexibility, but means that the
average user who runs a small number of synths but changes the
patch sets (soundfonts or whatever) frequently has a little bit
of a management problem in ensuring that the correct setup for a
given composition is always available.
What I imagine would be useful is a generic probably-GUI-driven app
that simply starts and stops soft synths, particularly synths that
are driven using the ALSA MIDI API, that output to JACK, and that
might not have GUIs of their own (admittedly the only one of those
I can think of right now is iiwusynth, but I'm sure there are others
and even if you only use iiwusynth you could have any number of
different soundfont configurations). As well as starting and
stopping synths on command, it might be able to store and recall a
selection of multi-synth studio setups of the user's design. It'd
be particularly neat if the soft-synth manager could itself be
driven remotely through some sort of MIDI control event.
Is there such a thing as this? (Anyone feel like writing one?) Is
there any other way of arranging synths and suchlike so that this
sort of application is not actually necessary, assuming that we want
to maintain the current arrangement where Rosegarden does not take
responsibility for running the synths itself?
Chris
Hi guys,
I just thought I would mention, in case someone missed it. There are
some kind people who have started supplying RPMS for common applications
for various distributions.
PlanetCCRMA (http://ccrma-www.stanford.edu/planetccrma/software/) being
the most well known I think (PlanetCCRMA is for Redhat though).
Now there are a few places for Mandrake users to go:
- http://rpm.nyvalls.se/sound9.0.html - which I have successfully used
to test new versions of both Ardour and MusE.
This site keeps _very_ up to date, keeps many of the applications I'm
interested in right now.
- http://www.esm.rochester.edu/kevine/turnkey/home.html - which I
haven't tried but looks promising.
Scope here is more towards completeness than having the newest of
everything, this is the feeling I get anyway.
2 öre-worth
Regards,
/Robert
Hi.
I have a Yamaha ymf740c and installed 0.9.0rc6 "the debian way." Audio
is working but can someone help me in setting up the mpu401?
Also, is there a doc describing module options? As a fairly newbie
"parm:mpu_port long array (min = 1, max = 8), description "MPU-401 Port."
(from modinfo and from the alsa web site) doesn't have any useful
meaning to me. I found some settings that worked when I installed 0.9.0b12
but I have no idea why they worked, or where I was supposed to find them.
Any help will be appreciated
Hi,
I just got a sblive platinum 5.1. And am trying to get recording
(analog) from the live drive working. I can record fine from the line
and mic on the back and can here stuff coming in on live drive but I
can't for the life of me get it recorded :(.
If anyone has managed could they send me the mixer settings they're
using and the command line to arecord (or whatever program there using
to record).
Thanks
(alsa 0.9rc3, debian sid, 2.4.19 + low latency)
--
rob <mailingLists(a)pangolin.org.uk>
Hello all,
I'm just now putting together my first linux box and need somebody to hold
my hand and walk me through the setup of my sound card. (BTW if there is
already a thread on this that I didn't catch on my quick search of the list
archives, please point me to it.)
I'm currently using a RME 96/8 PAD on a RH8 distro. I installed the alsa
packages available from freshrpms.net. After getting everything installed I
started up alsamixer and the only controls available is a volume control for
the DAC. Please tell me that I'm missing something and that there is much
more control functionality available than just levels for the DAC. Shouldn't
I have at least some sort of controls for the digital IOs and the ADAT? Is
there some sort of fancy config I need to do in /etc/asoundrc? Have I missed
something in configuration of devices, subdevices, some package not
installed etc?
Also, (slightly off topic) this card is very picky about the buffer settings
(i.e. fixed buffersize of 64KBytes). I know what settings work when setting
number of periods and frame size, but how do I work that out when programs
only let you set the buffer size in units of time rather than actual size?
Any help is appreciated.
Reguards,
-Reuben
Hi Mark,
Thanks very much for your those hints. Do you have any idea (or where I
can find the documentation) on how to play through the SPDIF or ADAT ports?
Where is the matrix mixer controls (amixer) documented for these cards?
DS
Mark Knecht wrote:
dsen,
Hi. I'm messing with similar things this evening. (And not doing
great, but not failing either.) I'm using a HDAP 9652, not the
Multiface, but it's the same driver I believe. The difference is I have
no 'headphone output'. Just 26 outputs.
I'm in KDE. Inside of a Konquerer window, I have set the file
associations for wav and x-wave to be
aplay -device="plughw"
This seems to work all the time in a Konquerer window, but doesn't work
on a file just sitting on my desktop. Don't ask me why.
FYI#1 - make sure you don't have a blocking aplay sitting around. Do a
ps -aux | grep aplay
and make sure nothing is there.
FYI#2 - My headphones and main speakers are running off of playback_1 &
2.
Good luck,
Mark
On Wed, 2002-12-18 at 20:29, D. Sen wrote:
/> Hi,/
/> /
/> I have just bought myself the Cardbus+Multiface RME system. I think I /
/> have finally gotten the alsa modules (snd-hammerfall-mem and snd-hdsp) /
/> to load correctly along with the OSS emulation modules. BUT I cant seem /
/> to play anything out of the Lineout/Headphone output. I have tried /
/> various 'amixer cset' commands but all I get is silence (trying to play /
/> a simple wav file using aplay)./
/> /
/> Any suggetsions?/
/> /
--
---------------------------------------
D. Sen, PhD
21 Woodmont Drive
Randolph
NJ 07869
Home Email: dsen(a)homemail.com Tel: 973 216 2326
Work Email: dsen(a)ieee.org Web: http://www.auditorymodels.org/~dsen
Hej to all,
I used to work with the really nice loopbased timestretching music
application ACID on windows. Is there any equivalent on linux ?
Thanks a lot
-- A l e x
Paul,
Great info. I didn't know about modinfo. Is there a way to query what the
current setting of these parameters are? It seems that I'm told I have a
range to work in, like 1-8, but no indication of how they are currently set
or what they do. (Maybe only read the code?)
I'm sure that many are just on/off sorts of things, but some clearly are
not.
Mark
-----Original Message-----
From: linux-audio-dev-admin(a)music.columbia.edu
[mailto:linux-audio-dev-admin@music.columbia.edu]On Behalf Of Paul Davis
Sent: Thursday, December 19, 2002 6:21 AM
To: linux-audio-dev(a)music.columbia.edu
Cc: Alsa-List
Subject: Re: [linux-audio-dev] RME Hammerfall DSP Cardbus+Multiface
>dsen,
> Hi. I'm messing with similar things this evening. (And not doing
>great, but not failing either.) I'm using a HDAP 9652, not the
>Multiface, but it's the same driver I believe. The difference is I have
>no 'headphone output'. Just 26 outputs.
>
> I'm in KDE. Inside of a Konquerer window, I have set the file
>associations for wav and x-wave to be
>
>aplay -device="plughw"
>
actually, i doubt that this is the issue at all. the hdsp driver has a
module option:
snd_line_outs_monitor
(possibly changed in recent versions to just "line_outs_monitor"). if
set to a non-zero value, all other outputs will be monitored on the
headphone/line out channels. you can find out about module options
using the modinfo command. for example:
% modinfo snd-hdsp
filename: /lib/modules/2.4.19/kernel/sound/pci/rme9652/snd-hdsp.o
description: "RME Hammerfall DSP"
author: "Paul Davis <pbd(a)op.net>"
license: "GPL"
parm: snd_index int array (min = 1, max = 8), description "Index
value for RME Hammerfall DSP interface."
parm: snd_id string array (min = 1, max = 8), description "ID string
for RME Hammerfall DSP interface."
parm: snd_enable int array (min = 1, max = 8), description
"Enable/disable specific Hammerfall DSP soundcards."
parm: snd_precise_ptr int array (min = 1, max = 8), description
"Enable precise pointer (doesn't work reliably)."
parm: snd_line_outs_monitor int array (min = 1, max = 8), description
"Send all input and playback streams to line outs by default."
parm: snd_force_firmware int array (min = 1, max = 8), description
"Force a reload of the I/O box firmware"
the parameter defaults to zero, so by default, you hear nothing.
there is no way to use any existing linux audio mixer to control
*what* you hear - the hdsp has 1542 mixer controls arranged in a
matrix, and its just not possible to make sense of this with mixers
designed around a few channels with direct routing. work is proceeding
on an equivalent to RME's TotalMix to offer ways of controlling this.
--p
