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.
Hallo,
Tom Szilagyi hat gesagt: // Tom Szilagyi wrote:
> On Sun, 25 Jan 2004, Frank Barknecht wrote:
> > I cannot hear it. :( I'm testing the reverb in Pd but although I crank
> > up decay times and played with dry/wet I cannot hear a reverb. I can
> > hear a bit of (seemingly high pass) filtering, but no real
> > reverbaration at all. I should hear a hall when selecting say "Smooth
> > Hall", shouldn't I? Changing presets seems to work, as the filtering
> > gets different when changing presets, but here I wouldn't call this a
> > reverb. Maybe someone could provide a small ogg which shows what I
> > should hear?
>
> Is it possible that the comb and/or allpass filters are turned off? I
> don't know how Pd displays plugins (i don't know Pd at all, maybe i should take
> a look) but there should be 4 toggle buttons somewhere saying "Comb
> Filters", "Allpass Filters", "Bandpass Filter", "Enhanced Stereo".
> If all buttons are on, i don't have an idea right now, but think, think...
In Pd, as it cannot use controls with spaces, you send messages like
"control #1 4" to set the first control to 4. I did set the Filter
controls this way to several values like 13 or 17. Should this turn
them on?
BTW: The other TAPs are working fine, http://footils.org/snd/echos.ogg
is a pluck playing through the stereo echo in Pd. (It's a bit big,
7.3M, currently uploading.)
ciao
--
Frank Barknecht _ ______footils.org__