Well, I got the new device subsystem working in Audiality, although
I've only finished the SDL audio driver. ALSA 0.9 rawmidi more or
less finished, OSS audio is a mess, and there's no OSS rawmidi yet.
(Although that's rather trivial; basically a stripped down version of
ALSA rawmidi.)
Anyway, I got sidetracked - as usual. ;-)
First, I started playing with a noise modulated delay line, but didn't
find the results very interesting. So I decided to run a bunch of
fixed feedback delay lines in parallel - and that sounded a lot more
interesting: It became a pretty neat reverb. :-)
With 16 delays and exponential or exponential +/- Golomb ruler, it
sounds great for "normal" sounds, but has some problems with
transients; it's not dense enough. However, with these uneven delay
time distributions, it doesn't sound metallic at all. It rather turns
the transients into some sort of "noise" (sounds like random granular
synthesis, basically), that soon becomes this "shhhh" most of us
desire.
With even/linear delay time distribution, it still sounds pretty good,
but it starts to remind of a gong simulator or something. Plate
reverb, maybe... Interesting, but not what I was looking for -
although I'm throwing all interesting variants in, so they can be
selected through a plugin control.
With 32 delays, it starts to sound *real* interesting - but it also
starts to abuse my PC133 RAM pretty seriously. It won't run real time
on my P-III 933 above some 48 delay lines, although this is with 32
bit buffers of 8192 samples each. This can be improved upon a lot,
since I'm only using a tiny fraction of the maximum delay times with
that dense settings. (Like 300 +/- 100 samples.) Also, it's all
integer, so I could probably use 16 bits.
I have no feedbacks in beetween delays; just one local, LP filtered
feedback per delay. Every other pair of delays is stereo reversed.
Tried some network feedback setups, but I couldn't figure out any
truly useful configurations.
Untested idea: Have multiple feedbacks for each delay line. (The old
"reverb" of Audiality is actually just one stereo delay line with
multiple feedbacks, BTW.) The idea is that, while not as effective as
separate delays, this is a lot cheaper, so I might get more
density/cycle.
Oh, and I'll try the well known delay time modulation trick, of
course. Maybe using filtered noise instead of periodic LFO waveforms?
Thoughts? Any ideas for useful cross feedback configurations?
//David Olofson - Programmer, Composer, Open Source Advocate
.- The Return of Audiality! --------------------------------.
| Free/Open Source Audio Engine for use in Games or Studio. |
| RT and off-line synth. Scripting. Sample accurate timing. |
`---------------------------> http://olofson.net/audiality -'
--- http://olofson.net --- http://www.reologica.se ---
Changes:
- Bug when .ecasoundrc was not in ~/ directory (Ecasound 2.2.0)
- Bug when no ladspa directory defined in ecasoundrc
http://www.sourceforge.net/projects/tkeca
Regards,
Luis Pablo
Ahora pod�s usar Yahoo! Messenger desde tu celular. Aprend� c�mo hacerlo en Yahoo! M�vil: http://ar.mobile.yahoo.com/sms.html
Hi all,
I've been messing with hdsp and found out an interesting issue. After
installing the driver for the first time onto the xp machine I also had
no sound coming out of the soundcard even though totalmix app showed
levels to be up. After clicking on the levels and/or moving them a bit,
suddenly the sound was there (as if the default values were there only
in the app itself, but did not affect the actual levels on the card
until they were moved for the first time). I was wondering maybe this is
the problem we're having in Linux, where the card loads just fine, but
for some reason there is no sound until a reboot has taken place.
I know this may be completely stupid thing to pursue, but since I've
heard of no fix for this issue, I thought this may be a possible
culprit. I am still struggling with the Linux driver (I got the card
working on an Dell Inspiron 8200 where the card posts and everything
supposedly works, but there is no sound coming from it), so perhaps as
dumb as this sounds, this may be the solution to the problem we've been
having with the hdsp in Linux.
I should also mention that I am using hdsp pcmcia interface and that
I've flashed it using the latest update from the rme-audio website
(could this also have something to do with it?).
Any info on this issue is greatly appreciated. Sincerely,
Ivica Ico Bukvic
I'm getting errors when I call snd_seq_open in a plugin, which go away in a standalone test app:
ALSA lib dlmisc.c:100:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load
ALSA lib conf.c:2655:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null)
ALSA lib conf.c:3066:(snd_config_update_r) hooks failed, removing configuration
calling: snd_seq_open(&seq_handle, "default", SND_SEQ_OPEN_INPUT, 0)
What could be the reason for this?
cheers,
dave
The SooperLooper LADSPA plugin now comes with PD patches!
For those that might not remember, SooperLooper is a LADSPA plugin that
emulates the Gibson-Oberheim Echoplex Digital Pro looping sampler. Go get
more info and download it at its new address:
http://essej.net/sooperlooper/
See the screenshot of the PD patches in action here:
http://essej.net/sooperlooper/sooperlooper_pd_shot.png
It is controllable via the PD GUI or MIDI program change messages
from your footcontroller (the ideal interface).
Unfortunately, there are bugs in the plugin~-0.2 external that seems to
be around. I made a source patch that fixes the output control message
bugs, and extended them to include the output control port index as well.
You will need this patched version of plugin~ to run SooperLooper in PD
properly.
Get the patch here:
http://essej.net/sooperlooper/pd-plugin0.2-patch.diff
If someone on the pd-devel list could forward this, I would much
appreciate it.
Dave Phillips, could you also update the web address on your LM&S pages
with the new address above?
Enjoy... please post with any problems/suggestions.
jlc
>What I think could be possible is using (writing a driver for) the
>scratchamp with OSS or ALSA drivers, as they seem to be USB soundcards
>by creative. Those will have standard chipsets.
>But that wasn't the question I guess...
If you are interested in trying to get it working with alsa then you need
the usb-audio driver. Installing it and inserting the module will tell
you pretty quickly how easy it will be to get working. The alsa usb-audio
driver is significantly more advanced than the oss version which it was
based on.
If it doesn't work for you then sending in the info from lsusb to the
alsa-devel list will help a lot.
Have a look at the generic instructions for usb in the alsa-docs.
http://www.alsa-project.org/alsa-doc/
Patrick.
--
Hi!
Do you think it's possible that Final Scratch could use two internal
Soundcards instead of the two external ones?
I bought this Final Scratch package and opened the external box. This
looks like a USB-Hub witch two external Creative USB sound cards.
But I don't like external sound cards and especially not the creative
ones!
Do you think it's possible to route the USB sound cards to my internal
sound cards?
I could use the sound driver library from linux. But my problem is how
to link the software to other soundcards?
Thanks for an answer!
Modnogg
I'm rewriting the audio and control I/O code right now, and I'm adding
ALSA 0.9 audio and MIDI support while I'm at it. ALSA 0.5 support
will probably be ditched in the process, as I can't test it, and
don't care much. OSS I/O and SDL audio output will still be there.
The new design is based on a common device struct for all I/O devices,
and drivers and APIs are supported by adding implementations behind
that, C++ polymorphic style. I think this will make it a lot easier
to add support for other I/O APIs. It also comes with a string based
selection interface that can easily be hooked up to the command line,
config files or whatever.
//David Olofson - Programmer, Composer, Open Source Advocate
.- The Return of Audiality! --------------------------------.
| Free/Open Source Audio Engine for use in Games or Studio. |
| RT and off-line synth. Scripting. Sample accurate timing. |
`---------------------------> http://olofson.net/audiality -'
--- http://olofson.net --- http://www.reologica.se ---
ZynAddSubFX is a open-source software synthesizer for
Linux.
It is available at :
http://zynaddsubfx.sourceforge.net
or
http://sourceforge.net/projects/zynaddsubfx
Please send me instruments done by you with
zynaddsubfx.
news:
1.0.5 -The bug that crashed ZynAddSubFX if you
change some effect parameters, it is really removed (I
forgot to update the file before upload)
- Other bugfixes and code clean-ups
- Added a Global Filter to SubSynth
- Added keyresponse limits to Part
- Added presets to Effects
- The fade is smaller on high frequecy
content and larger on low frequecies; so you'll don't
hear starting clicks on basses and audible fadeins on
higher pitched sounds
- Added tunnings to Reverb: you can
choose Random of Freeverb. Freeverb is written by
Jezar at Dreampoint; I used the tunings from it.
__________________________________________________
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