hi all,
i've got recent cvs of ardour and jack and really wanna get a drum
machine plugged into it. so, hydrogen.
I got libhydrogen from cvs built fine, but
hydrogen appears to require qt 3.2 and gentoo only seems
to provider 3.1.2. D'OH!
I think i can get 3.2 if i update world, maybe... which
takes forever... ah the self-inflicted pain of gentoo :-)
anybody know exactly what I need to compile hydrogen cvs?
do i really need the whole thing of qt 3.2?
--
Paul Winkler
http://www.slinkp.com
Look! Up in the sky! It's MARAUDER MOVER!
(random hero from isometric.spaceninja.com)
Hi all, I've tried contacting the maintainer of the project (Andy), but
he never bothered to reply, so now I am taking this question to all of
you out there that might have had exposure to this interesting library.
Does anyone know what is the current status of the whole project anyhow?
Any help on this matter is greatly appreciated! When answering any of
the given questions, please include the question in your reply. Thank
you very much!
So here's the excerpt from the letter:
I've recently decided to incorporate libalsaplayer-like functionality
the upcoming version of my app RTMix. However, with having a particular
feature in mind I am wondering whether libalsaplayer provides that and
if not, whether you'd be willing to add it (or let me add it, although
that might get messy since I am not too familiar with the inner workings
of the lib). So, here it goes:
Apart from all the wonderful features libalsaplayer offers, I am looking
for some additional ones:
I would like to be able to indefinitely loop specific "ranges" of a
particular soundfile (i.e. 2.2secs-4.12secs or whatever).
1. Is it possible to do this and have music continually loop even if one
changes the direction of playing so that when the alsaplayer runs out of
the looping material in each direction that it just jumps back onto the
other end of the loop and contiues on (kind of like a ring-buffer)?
2. Is it also possible to do this kind of looping on the whole soundfile
(the gui version of alsaplayer always stops if I let it play 'til the
end or the beginning, depending in which direction I am going).
3. Is it possible to define loop points by addressing a particular
sample number rather than giving time in seconds?
4. How stable is alsaplayer when looping really small chunks of sound
(i.e. like 5 sample loop)?
5. Is alsaplayer capable of ramping such loop points by attenuating
let's say ending 20 samples (it would be cool if this number could be
user-selectable) and then ramping up the beginning 20 samples in order
to alleviate the "pop" that happens when looping a sound where waveforms
at the beginning and the end do not align?
6. If the feature in question 5 does not exist is there a way to control
output level of a player on a per-sample basis via callback so that one
can implement that outside of the lib?
7. If neither 5 or 6 are possible, would you be willing to implement
such functionality (i.e. a toggle_ramp( bool ); and set_ramp_length( int
); callbacks or something similar.
I would greatly appreciate your feedback on this matter as that will
greatly assist me in determining how to go about implementing such
functionality in my app.
Thank you very much! Looking forward to hearing from you. Sincerely,
Ivica Ico Bukvic, composer & multimedia sculptor
http://meowing.ccm.uc.edu/~ico
P.S. How do you implement reading of a sound faster and slower than it
sounds in such a gradual fashion? Do you adjust the sampling rate of the
DSP and if so does that affect other streams coming from the
libalsaplayer, or is this something that is sound-specific (and if so,
how)?
Hallo,
just a short information question: Is it true, that Redhat does not
provide any ALSA packages themselves so people who want to run some of
the many ALSA applications on Redhat will have to use third-party
rpm's like Fernando's or from freshrpms.net? Is this really true? If
yes, does someone know what RH's plans are regarding ALSA on kernel
2.6?
ciao
--
Frank Barknecht _ ______footils.org__
First: Using an equalizer, kill all frequencies below ~600 and above ~2k. Use a graphic EQ for this. (The two values are just examples, you can play with both frequencies to get the desired sound you're looking for.)
Second: get a recording of the inside blank track that comes after the last song of a noisy record. Loop this noise and mix it over top of the song once you've done the band pass filter from step one.
-Reuben
-----Original Message-----
From: Michal Seta [mailto:mis@creazone.32k.org]
Sent: Wednesday, October 22, 2003 8:32 PM
To: A list for linux audio users
Subject: [linux-audio-user] ... like an old vinyl
Hi all,
Does anyone know if it's possible, under linux, to turn a good quality audio
recording into a file that sounds like an old vinyl record. Any ideas?
Realtime and non-realtime will be equally useful.
Thanks.
--
_
__ __ (_)___ Michal Seta
/ \/ \ _/^ _|
/ V |_ \ @creazone.32k.org
(___/V\___|_|___/
http://www.[creazone]|[noonereceiving].32k.org
Hi all,
Does anyone know if it's possible, under linux, to turn a good quality audio
recording into a file that sounds like an old vinyl record. Any ideas?
Realtime and non-realtime will be equally useful.
Thanks.
--
_
__ __ (_)___ Michal Seta
/ \/ \ _/^ _|
/ V |_ \ @creazone.32k.org
(___/V\___|_|___/
http://www.[creazone]|[noonereceiving].32k.org
--- Anahata <anahata(a)treewind.co.uk> wrote:
On Wed, Oct 22, 2003 at 10:14:58PM -0500, Reuben Martin wrote:
> First: Using an equalizer, kill all frequencies below ~600 and above
> ~2k. Use a graphic EQ for this. (The two values are just examples, you
> can play with both frequencies to get the desired sound you're looking
> for.)
>
> Second: get a recording of the inside blank track that comes after the
> last song of a noisy record. Loop this noise and mix it over top of
> the song once you've done the band pass filter from step one.
>
That's a bit drastic for vinyl - more like an acoustically recorded
78rpm from about 1920. Those were pressed on shellac, not vinyl.
The priciple's about the same, though - add a few random background
ticks and clicks, a bit of white noise as many vinyl discs had tape
noise from the master, possibly a bit of distortion on peaks if you want
it to sound like a worn record/stylus, and also some gentle frequency
variation at 0.5 Hz (for 33 1/3 rpm) to simulate a less-than perfect
turntable or warped record.
I'm sure it can be done with Linux tools...
--
Anahata
anahata(a)treewind.co.uk Tel: 01638 720444
http://www.treewind.co.uk Mob: 07976 263827
------
I'd say its all possible with audacity:
http://audacity.sourceforge.net
Its an Multitrack-able Wave/Mp3/OGG-manipulation tool with LADSPA-support, I think. Anyway, there are a lot of "integrated plugins" that allows what you want to do. A real PROGRAMM can do what you want is not written, I think ... ;)
Sascha Retzki
_____________________________________________________________
Linux.Net -->Open Source to everyone
Powered by Linare Corporation
http://www.linare.com/
Hi all,
I was wondering if there is any application that can just act as a
standalone LADSPA host - with a jack input/output that can be inserted into
the signal chain. I've searched the net but couldn't find exactly what I'm
after. Does such an application exist - or are there any plans for one?
Cheers.
--- Benjamin Flaming <lau(a)solobanjo.com> wrote:
> I am the guy with the alsa problem you remember me ? Well, I recompiled alsa, this time without any unresolved symbols... . Don't work right,
What exactly doesn't work?
> but I installed a second card from which I know that it worked well,
How do you know this?
> but this card cannot output something ( the hardware-outport is broken [ ooups ] )... .
Do you mean the card is physically broken?
Do you get error messages when you try to use it?
Have you used a program such as alsamixer to bring up the volume level, which defaults to all the way off?
> Everything seems to work fine,
What is "everything"? You just said the first card didn't work right, and the second card had no output.
> except that alsa uses my broken card, the ES1371 for audio output.
If I understand ALSA correctly, this is because it is the default device -hw:0. I think you would need to explicitly tell ALSA to use a different device. Have you read up on .asoundrc files? (I haven't (I know I ought to) - but I think it might be the answer).
> If I set jack to Playback via the card with index 1 , so my ESS-Solo1 (snd-es1938):
> # jackd -d alsa -d hw:0 -P hw:1
Why do you say "-d hw:0" if you want it to use hw:1? Try this:
jackd -d alsa -d hw:1
Good luck :)
|)
|)enji
-----
LOL, hi !
I see that it is "irresponsible" to post a msg begins with 'you know iam the guy xy, I had x problem' ... .
Well:
I have had several problems with my ESS-Solo1 soundcard to start various sound apps which
using jack ... . The second card, the Es1371, IS able to start jack, the esssolo1 quits with some alsa-error msgs. I first thought the esssolo1-problem is a symbol-problem in the alsa-modules, but a recompilation ( a successfull ) did not change something with the "support" of the esssolo1 with jack.
So my Ess-Solo1 cannot be used via jack. My second card, Es1371, can , but it is physically broken. just the output-port, so this little "whole" where I put the speakers in, you understand ? I want to pipe jack-output to my first card, the Ess-Solo1. So I want to use my Es1371 for my MiDI-keyboard, for capturing AND for STARTING various soundapps like muse,ardour and Rosegarden4 and so on, but let the Ess-Solo1 output all the stuff jackd gets. (*puh* )
So, my cards are NOW:
Ess-solo1 <-> hw:0
Es1371 <-> hw:1
# jackd -d alsa -d hw:1 -P hw:0
Should set jack to USE the hw:1, so Es1371 and output via hw:0, my Ess-Solo1, right ? I think thats the question of my originally mail ;) . Sry, but I exchanged the order of both cards, I think in my first mail Ess-solo1 was hw:1 , but this is not good. You pointed it: My ess-solo1 is my normal sound-card for listening to music and so on, and it should be the first alsa card.
So, the above command prints out the help-menue which u get via:
# jackd -d alsa --help
Why ?
greetz, Sascha Retzki
_____________________________________________________________
Linux.Net -->Open Source to everyone
Powered by Linare Corporation
http://www.linare.com/
Hi!
I am the guy with the alsa problem you remember me ? Well, I recompiled alsa, this time without any unresolved symbols... . Don't work right, but I installed a second card from which I know that it worked well, but this card cannot output something ( the hardware-outport is broken [ ooups ] )... .
This is my modules.conf:
alias char-major-116 snd
alias snd-card-0 snd-ens1371
alias snd-card-1 snd-es1938
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0
alias sound-slot-1 snd-card-1
# Es1371
#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
# ESS-Solo1
alias sound-service-1-0 snd-mixer-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-1-12 snd-pcm-oss
#
options snd major=116 cards_limit=2
options snd-ens1371 index=0
options snd-es1938 index=1
----
Everything seems to work fine, except that alsa uses my broken card, the ES1371 for audio output.
If I set jack to Playback via the card with index 1 , so my ESS-Solo1 (snd-es1938):
# jackd -d alsa -d hw:0 -P hw:1
Jack just prints out the helpscreen, and if you have a look by yourself:
...
-P,--playback [name] playback output and optionally set the playback device (default: duplex)
...
[name] can be "optionally set (to) the playback device", in my case hw:1, so my second card with card index 1, see modules.conf. Why this jack-error ? And any idea why sound-output is not piped by alsa generally ??
Thx for attention
Sascha Retzki
_____________________________________________________________
Linux.Net -->Open Source to everyone
Powered by Linare Corporation
http://www.linare.com/