Hi all!
Although still in the early - very early - stages, I finally decided to
announce it: my webpage, http://ltsb.sf.net .
I try to demonstrate how you can use a text-based linux environment
(console) as a nicely equipped recording-studio. At the moment only the basic
chapters of this guide and a few early audio-examples are online. But as time
allows, I'll continue.
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net - the Linux TextBased Studio guide
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Hi all,
I'm looking for methods in Linux to add reverberation to sounds (not
necessarily in real time) using controls based on room size and
specifications. I've found some LADSPA plugins that have simple controls
- - GVerb, Freeverb, and Stereo reverb - which would work if I can't find
anything else. Does anyone know of any options of any sort that provide
more complex controls (room dimensions in three directions, damping,
placement of source and mic)? Thanks for any advice.
matt
- --
Matthew Shanley
E-mail: mshanley(a)alum.rpi.edu
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> From: Clemens Ladisch <clemens(a)ladisch.de>
> > timo:~$ aplay -L
> > multi {
> > ...
> > is this as it should be?
>
> Yes. Does "aplay -D multi something.wav" work?
It plays the song through one of the soundcards. I am not sure that if
that qualifies as working?
Timo
Hello.
I'm developing a jack application. The test files are played to
the application with aplay and mpg123.
How do I route the played audio to my application and
then the output to Alsa?
Note that aplay plays to the "wave" control in alsamixer.
Now my application connects to
alsa_pcm:in_1
alsa_pcm:in_2
alsa_pcm:out_1
alsa_pcm:out_2
and it is the same "wave" control.
Puzzle: To prevent aplay from going to speakers, I have to turn
the "wave" to zero. But then how do I route anything to speakers
if my application outputs to the same "wave"?
You may freely assume the aplay and mpg123 are not changed in
any way.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
>From: Erik Steffl <steffl(a)bigfoot.com>
>
> I am trying to record from a receiver (not sure if that's the right
[ ... ]
>soundcard expects. Is there anything I can do? do I need some pre-amp or
>something?
Are you recording the radio programs? What are the levels when
you play a CD? Or one of the properly working devices, you listed
in your mail, through the receiver:
ipod --> VCR in --> tape rec --> live
What are the levels then?
If you record radio, then you need nothing. Radio has lower
amplitude range and all fits to the 14-bit you seem getting in.
Non-HiFi video audio is also quite poor.
It is totally different problem if the recorded audio plays
at too low volume or shows tiny in audio editor. Audio player
authors always forget to make the output faders to range above
the 0 dB. You need to set the output fader to +12 dB to get
normal sound out. This output fader is a software gain: y = gain*x;
not a hardware gain. Or you need to increase the volume somewhere
else, like some Jack app or an editor plugin.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
Hello.
For LiveDrive one needs to have
enable_ir=1
in "/etc/modules.conf" (or equivalent file in other Linuxes).
However, backpanel midi port could be a different thing.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
>From: Mikhail Ramendik <mr(a)ramendik.ru>
>
>But now I want to use ALSA. Which program can I use to monitor input
>levels, and which (possibly different) program can I use for simple,
>buffered, preferrably command-line recording?
What did you find?
I have used years my own Alsashmrec. It uses two processes/threads
one for Alsa reading and one for disk writing. A few years ago
this was absolutely necessary for skipless recording. You want
this feature in the recorder you choose.
Alsashmrec computes max values per a disk buffer and prints them like
this:
Alsashmrec 1.0
Recording format: 16-bit/44100 Hz/Stereo
Buffers: A/D + Shared + Disk == 1024 + 2621440 + 163840 bytes
0000000000000000000003646564556767777677787767656856585787757678767787
7677
The print has "x" for the overflow. The screen is eventually filled
with the max values. That is good because I can take a break and
may still see at what point the overflow occured. Perhaps Alsashmrec
should additionally print the time values at the begin of each row.
When two programs are used there is danger that you monitor different
signal plug than what is being recorded.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
hi all,
i'm trying to figure out how to synchronize a hammerfall card
with a delta1010, and have them both running with jack. (on
2 different machines, or on one machine both running under 1
jack instance) anyone tried this before?
the hammerfall has a 9-pin adat-sync port, the delta1010 has
2 wordclock bnc's. the hammerfall has optical digi, the delta1010
has rca. seems they are from different worlds. is this solvable
with cabling only? or do i need to buy yet another magic box for this?
in realation to this, i am looking for a video i/o card which can
sync to this setup. is there anything out there that is not very
very expensive?
cheers
tom
Hello,
I have an Aztec 2320 sound card.It is ISA and not PCI. I have
recently installed Fedora core 2 on my machine. I could not get the
sound to work, so I checked and found that I have indeed sound support
compiled in my kernel (2.6). Then I had visited the alsa-project.org
site and according to their suggestion, loaded the aztec 2320 module -
snd_azt2320. Now my machine correctly detects the card but when I play
a sound file, no sound is heard. I checked that the volume is high by
running kmix and alsamixer program. Also earlier I was able to pop an
audio CD and play it successfully. But after loading the Aztec module,
I am not able to hear that too, though the music is being played.
I have included the lines as specified in the alsa-project.org site
into the /etc/modules.conf file.
The 'lsmod' listing as well as the contents of the '/etc/modules.conf'
file has been included below for your viewing. Please tell me where I
could have gone wrong. Is it that the details given on the site were
for a PCI based card and is different for an ISA based card of the
same chipset?
I checked for the files - /dev/dsp, /dev/mixer, /dev/sequencer and
/dev/midi - but I found that /dev/midi is missing. Could it be because
of that too?
Sorry for this long email.
Thanks in advance
Ravi
//My machine's lsmod listing is as given below
Module Size Used by
snd_azt2320 9768 0
snd_opl3_lib 7424 1 snd_azt2320
snd_hwdep 6276 1 snd_opl3_lib
snd_cs4231_lib 18948 1 snd_azt2320
snd_pcm 68872 1 snd_cs4231_lib
snd_timer 17156 3 snd_opl3_lib,snd_cs4231_lib,snd_pcm
snd_page_alloc 7940 2 snd_cs4231_lib,snd_pcm
snd_mpu401_uart 4864 1 snd_azt2320
snd_rawmidi 17184 1 snd_mpu401_uart
snd_seq_device 6152 2 snd_opl3_lib,snd_rawmidi
floppy 47440 0
snd_mixer_oss 13824 0
snd 38372 10
snd_azt2320,snd_opl3_lib,snd_hwdep,snd_cs4231_lib,snd_pcm,snd_timer,snd_mpu401_uart,snd_rawmidi,snd_seq_device,snd_mixer_oss
soundcore 6112 1 snd
sermouse 3712 0
ipv6 184288 8
parport_pc 19392 1
lp 8236 0
parport 29640 2 parport_pc,lp
ipt_REJECT 4736 1
ipt_state 1536 5
ip_conntrack 24968 1 ipt_state
iptable_filter 2048 1
ip_tables 13440 3 ipt_REJECT,ipt_state,iptable_filter
nls_utf8 1536 1
nls_cp437 5376 1
vfat 10496 1
fat 33472 1 vfat
dm_mod 33184 0
uhci_hcd 23708 0
ext3 102376 1
jbd 40216 1 ext3
//Contents of the modules.conf file
# ALSA portion
alias char-major-116 snd
alias snd-card-0 snd-azt2320
# module options should go here
# 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
--
----------------------
I hear and I forget, I see and I remember, I DO and I understand.
-Confucius-
Hi Free,
Thank you for your comments.
>Please try the final 1.2.0, all the tools you mention are there.
I'm glad to hear that. As I said, I had no problem after all since I had
rawrite2 somewhere anyway, but having this corrected makes Demudi neater.
>This is related with the new Debian installer mechanism, AFAIR if you
>press the back button you'll be prompted language and keyboard in
>different screens. I'll try to address the problem for the next 1.2.1
Actually, you can select language and keyboard in different screens by
selecting back just as you describe. Problem is, somehow X will be
configured for US layout keyboard anyway if you select Spanish keyboard and
English language (kind of strange). Since the computer is always run under
X, all I had to do really is to edit /etc/X11/XF86Config-4 (not 100% sure
about the path where this file resides) to change the layout from "" to
"es".
>Translations for the Debian installer are still in progress. Please
>please please send any suggestion to debian-boot(a)lists.debian.org,
>such contribution is very appreciated.
Thanks for pointing to the relevant forum. The thing here is I was scared to
read the partition manager in Spanish, but I'm not sure that this is part of
the Debian installer? Can you clarify so I can get in touch with the
relevant people? I noticed it's a different tool to what it used to be (some
fancy flavour of fdisk I think), but I don't know wether it is an
application in itself or it is just part of the Debian installer. Whatever
it is, if I could get hold of a list of messages (in English) this software
has, I'd be happy to translate as accurately as possible and provide the
translated list of messages. Especifically, the "finish and write partition
table" reads "abort and write partition table" in the Spanish translation,
not good.
Cheers,
Alex
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