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Hello List
I'm looking forward to get the Creative Extigy for my Notebook but there is
one think I want to know before.
My Notebook has a SiS 7012 SoundChip, working with ALSA on Gentoo Linux with
Kernel 2.6.1-rc1. I want to use both Soundcards. Is this possible? As I read
on http://www.cs.umass.edu/~cochran/ is the Extigy driver based on OSS.
Did anyone of you have any problems with using the Extigy together with other
Soundcards? Or has anyone of you experience with the Extigy and other
Soundcards?
Bye
Olli
- --
Registered Linux User Nr. 259631
http://counter.li.org/cgi-bin/runscript/display-person.cgi?user=259631
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Hi all,
This is my first message on this mailinglist. :D
I don't know too much english so sorry for all mistakes i'm going to
make in this message.
I would like to set up my pc to record a cd of a band.
I would like to rec single instrument at time and also all the band at
the same time.
What software i've to use?!
What hardware i need?!
Thanks for the replys
bye
Livio
PS. In Italian "se c'è qualche italiano che mi possa dare una mano
sarebbe molto gradito"
In English: "Is there an Italian who can help me?"
Hello,
for some reason, my sound input is redirected directly to the output
instead of being captured by alsa/jack/ardour.
I suppose there are many possible explanations, but are there easy
checks to get an idea about where my sound input gets lost? Or a
detailed guide about sound input and alsa and jack?
Sound input has been working with ecamegapedal before I installed
ardour, so I suppose it's no hardware problem.
My setup:
gentoo kernel 2.4.23
ASUS A7N8X-X nforce mainboard
alsa snd-intel8x0 driver (working for output)
jack 0.91 (output OK)
jack readable clients: alsa_pcm (capture_1 & capture_2)
thanks,
Wouter
I Have outputs from Quattro work great, but no inputs, and no way to
adjust volume.
KAMix shows proper inputs and outputs for the PCM card (second card), but
when set to the Quattro card. the KAMix Window is blank for inputs
and outs.
Qinit shows a segmentation fault.
Alsapart of Modules.conf:
# Alsa sound support.
# Warning: please don't modify comments over aliases 'snd-card-#'
alias char-major-116 snd
#module options should go here
#OSS/Free portion
alias char-major-14 off
options snd-intel8x0 snd_ac97_clock=0 snd_enable=1 snd_index=1
options snd snd_cards_limit=2 snd_major=116
# YaST2: sound system dependent part
#
alias sound-slot-0 snd-card-0
alias sound-service-0-0 snd-mixer-oss
alias sound-slot-1 snd-card-1
alias sound-service-1-0 snd-mixer-oss
alias sound-slot-2 off
alias sound-service-2-0 off
alias sound-slot-3 off
alias sound-service-3-0 off
# uniq.virtual:USB Audio Quattro
alias snd-card-0 snd-usb-audio
# H0_h.AyAc5dtQ3r5:nForce2 AC97 Audio Controler (MCP)
alias snd-card-1 snd-intel8x0
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-0-11 snd-mixer-oss
alias sound-service-0-12 snd-pcm-oss
alias sound-service-1-1 snd-seq-oss
alias sound-service-1-3 snd-pcm-oss
alias sound-service-1-8 snd-seq-oss
alias sound-service-1-11 snd-mixer-oss
alias sound-service-1-12 snd-pcm-oss
# end of i386 part for modules.conf
# --- END: Generated by ALSACONF, do not edit. ---
I'm comparing the RME Multiface with the M-Audio Delta 1010 (the full
model, not the LT).
Both things provide 8 balanced analog ins/outs (and balanced is very
important to me), some digital ins/outs (the M-Audio provides only
SPDIF, RME has SPDIF and optical) and MIDI. Both are fully supported by
ALSA. Both can do 96kHz/24bit.
M-Audio Delta doesn't have headphones output.
The big difference is the price.
RME Multiface can be bought in the US for like $920 (the breakout box
plus the PCI card).
M-Audio Delta 1010 is $600 total; even if i stick to it a 4-way
headphone amp, it's still only $700.
What do i lose if i go the M-Audio way, instead of the RME?
I know the RME stuff are more like sound routers (can combine the
channels in a very flexible fashion), plus they do a lot of things with
0% CPU load.
How's the M-Audio Delta in this regard? Can i do the same clever things
with it?
--
Florin Andrei
http://florin.myip.org/
Hi,
I have set up two alsa devices with dmix plugin for each. I can now send
several independent audio streams to two destinations - different sound
cards (dmix_main and dmix_kids).
I wonder if I can set up additional pcm device that could receive single
audio stream and send it to two cards simultaenously ? Can this device be
used in parallel to both dmix devices ?
Thanks,
Robert.
Hi
I'm working on a sf2 of the MIS piano and need an efficient way to setup
all the looppoints. Would be nice is swami was able to truncate the
audio, but it seems not. Would
it be possible to truncate/loop the files outside swami and have the
looppoints imported along with the samples? If so what software should I
use for setting loops??
Thanks in advance.
--
peace, love & harmony
Atte
http://www.atte.dk
Hallo,
CK hat gesagt: // CK wrote:
> analyseplugin says:
>
> "Comb Filters" input, control, toggled, default 1
> so #4 - #7 should be set to 1 I put a patch that works for me here:
> http://test.pilot.fm/nolink/ladspa_tap_rev.pd
Ah, thank you, now it is working. Somehow I was under the impression,
that the filter controls set *how many* filters are in use... It
seems, also the other values should be set to decent values.
I do get warnings with some of your example values which exceed the
boundaries set by tap_reverb. I changed this and added a preset name
display (using the Pd external "pool") to an archive here:
http://footils.org/pkg/ladspa_tap_rev.tgz
Tom, although I only shortly listened to it: This is a wonderful
sounding and highly flexible reverb, something that was sorely missing
before. I don't like freeverb and gverb very much, but this will
become a favourite, a classic, I'm confident. So cool. Thank you a
lot.
ciao
--
Frank Barknecht _ ______footils.org__
Hi all,
This is my first message on this mailinglist. :D
I don't know too much english so sorry for all mistakes i'm going to
make in this message.
I would like to set up my pc to record a cd of a band.
I would like to rec single instrument at time and also all the band at
the same time.
What software i've to use?!
What hardware i need?!
Thanks for the replys
bye
Livio
PS. In Italian "se c'è qualche italiano che mi possa dare una mano
sarebbe molto gradito"
In English: "Is there an Italian who can help me?"
Hi,
I'm designing integrated home automation/entertainment system based on Linux
and other open source apps. I have several possible sound sources (like
Festival as speech synthesis, music players, and also possible voip or
ordinary telephony applications).
I'd like to deal with all this stuf using signal routing and mixing. Some
possbile scenarions:
- when internet voice call comes in, then I connect to local audio card for
certain room
- when watching TV, speech synthesis would like to announce something
- from one room I'd like to talk to another...
If I think ideally - best would be to have range of "virtual" sound
destinations, that could be dinamically routed,mixed to physical devices. As
far as my novice knowledge goes I was thinking of using Alsaplayers as
music/wav players (they have software volume control) , Jack (as sound
servers) and Ecasound (for routing, mixing) ....
Any other advice in apps to use, more info or any other opinion would be
more than grateful.
Thanks in advance,
Robert.