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__
Hallo,
Tom Szilagyi hat gesagt: // Tom Szilagyi wrote:
> * TAP Reverberator
> Actually no less than 38 reverberator effects,
> ranging from Afterburn to Warehouse, including
> small/medium/large rooms, halls, plates...
> and more!
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?
ciao
--
Frank Barknecht _ ______footils.org__
Hi LAU,
OK, after 4 hours on this and enough Googling to break my keyboard, I
have to admit defeat.
What I Want To Do:
1. Load a web page containing a Flash Film
2. Play the Flash Film
3. Record any sound the Flash Film delivers to a WAV file.
4. Maybe post-process the WAV into Ogg, MP3, etc.
What I Have (Hardware + Software):
1. SB Audigy
2. Alsa kernel drivers, 1.0.0 RC2
3. Alsa libs, 1.0.0 RC2
4. Jack, compiled with the above Alsa libs
5. Ecasound, compiled with the Alsa libs and Jack
What works:
1. Sound from CD, System beeps, etc. works
What doesn't work:
1. Alsa: arecord -f cd -o test.wav, when playing an Ogg file, e.g.
2. Ecasound: ecasound -i:alsahw,0 -o test.wav, when playing an Ogg
file, e.g.
Am I missing something fundamental here? I can't imagine that this is an
especially arcane wish, simply wanting to record and play at the same
time, yet I can't find anything on this which helps. I have played with
alsamixer for about an hour, but I simply get silence all the time. All
the 'capture' devices are things like Mic, CD, Line-In, etc., which is,
obviously, NOT what I want. I simply want to record what I am currently
hearing -- although I'm testing this with playback of Ogg, actually I
clearly want to record the sound from the Flash Film mentioned.
I have read the various docs for Alsa, Ecasound and Jack, but I have to
admit that sound on Linux is *very* confusing. I am not really sure
whether I'm supposed to be using /dev/dsp, alsahw, or some other arcane
terminologies -- kernel compilation is easy in comparison! I'm not
getting any errors, by the way, everything *seems* to work, but when I
stop playback and check the output, all I hear is dead air.
I can (joyfully!) send the output of amixer if that would help, hope to
get a reply on this one soon. In Windows, this is a 5-minute no-brainer.
Install Total Recorder, and that's that. I can't imagine that it must be
so complex in Linux? First time I've really hit a wall in Linux -- which
clearly shows that sound is something which needs to be clarified a
*LOT* in Linux. (Having alsamixer start with everything mute is a good
example of this!)
Thanks a million if anyone can help: otherwise I'm rebooting into
Windows for now, since there's no other way of doing this at the moment.
Ed
Hello, I am trying to create a very simple midi filter client for the
alsa sequencer based on aseqview-0.1.4. I have alsa 0.9.8. This code
shows the input and the output port in aconnect when executed. And, when
the raw_midi client is connected The callback does get called when I
play keys on my piano. However, Its as though the message is getting
sent back to this client and not on to the next client. The
process_event function keeps getting called over and over again with the
same message when I hit one piano key. It's as though it is sending
itself the message. Any Ideas are greatly appreciated. -Garett
#include "portlib.h"
int process_event(port_t *p, int type, snd_seq_event_t *ev, int
*priate_data)
{
port_write_event(p, ev, 0);
}
int main()
{
unsigned int caps = SND_SEQ_PORT_CAP_WRITE |
SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_READ |
SND_SEQ_PORT_CAP_SUBS_READ;
port_client_t *client = port_client_new("MIDI filter",
SND_SEQ_OPEN_DUPLEX);
port_t *port = port_attach(client, "FILTER port", caps,
SND_SEQ_PORT_TYPE_MIDI_GENERIC);
int *priv_data;
/* add callback */
port_add_callback(port, PORT_MIDI_EVENT_CB,
(port_callback_t)process_event, priv_data);
port_client_do_loop(client);
return port_detach(port);
}
Hi all,
I will be setting up an audio Linux box this spring for a museum
exhibition that will run for about 4 weeks. It will probably run without
an xserver and use Pure Data in --nogui mode since a GUI is not needed,
and may turn itself on and off with a chronjob function.
My question is this: since I want to document the sound of the
exhibition, I would like ecasound to record from all four outputs
periodically [also as a chronjob]. I am not too familiar with ecasound,
but it seems like exactly the kind of low-overhead tool to do the job.
So can anyone tell me how to capture from *outputs* and not inputs?
thanks,
Derek
--
derek holzer ::: http://www.umatic.nl
---Oblique Strategy # 4:
"Abandon desire"