Hi all,
This question is not really a problem with hardware or software per
se, but I would just like to know what people's favorite reverbs out
there are. I used to do all of my pro audio work on Mac and Windoze,
and I'm having great success getting things working with Linux, but
I've never really found any really decent reverbs, and this is
unfortunately keeping me from doing more final mixes with Linux. What
I'm looking for is something like a LADSPA or other plugin, or a
program that will process sound non-realtime (or even destructively,
like Audacity) with some decent sounding reverbs.
The TAP Reverberator plugin was okay, I guess, but really the only
other LADSPA reverb I've come across is Freeverb, which is not good,
at least using it for intimate vocal tracks and guitars. I mainly use
Ardour for tracking and mixing, Audacity for destructive editing and
cleaning up tracks, and Csound, which I've been known to use for DSP a
bit in the past.
Maybe I just have not stumbled upon the right plugins or programs.
Any suggestions of favorites?
One more related question: Does anyone know if any soundcards with
DSP in hardware work with Linux? I'm thinking of something along the
lines of the Creamware cards (the ones with the integrated DSP's), but
I'm assuming that most of these also need a software component to make
them work, and that is most definitely closed-source. I can dream,
though, right?
Jon M.
Hello list!
I'm having problems with my terratec phase 26 usb audiocard on ppc gentoo linux 2.6.10-r6 ...(on ibook 800Mhz)
the card is recognized but jack won't start (or starts 2 secs and then stops) and even aplay gives errors. with plughw:1 jack starts but not for long.
This is what jack says:
configuring for 48000Hz, period = 1024 frames, buffer = 2 periods
Couldn't open hw:1 for 32bit samples trying 24bit instead
Couldn't open hw:1 for 24bit samples trying 16bit instead
Sorry. The audio interface "hw:1" doesn't support any of the hardware sample formats that JACK's alsa-driver can use.
ALSA: cannot configure capture channel
cannot load driver module alsa
10:52:26.857 JACK was stopped successfully.
10:52:28.738 Could not connect to JACK server as client.
I found this on the list:
>> The device uses little endian samples, because this is what the USB
>> Audio Spec says. Jack supports only sample formats with native
>> endianness. Complain to the Jack developers, and/or use plughw.
>well, you can also try the portaudio layer ... my quattro won't work
>with jack / alsa ...
but portaudio doesn't work with pure data (and thats what i want to use the card with), as far as i know. Does anyone have the same problem with usb audio devices on ppc 2.6.x kernels or knows of a solution ;) ??? (if not, i'm returning my card asap and any suggestions of audiocards compatible with ppc linux + osx would be very welcome ;)
Thanks and grtz
Marloes
(Resending)
Hi Anthony,
> For those of us who are ignorant, could you point at a source of a demo
> of this? Ideally, a normal stereo and binaural version of the same thing?
I'm not aware of any direct comparisons, and I don't have space for my
large demo posted over a year ago, so here's a very short one:
http://home.earthlink.net/~davidrclark/linux_audio_users/binaural.mp3http://home.earthlink.net/~davidrclark/linux_audio_users/stereo.mp3
Rather than go completely OT here, I'd rather post my comments about this
demo on that page, which I hope to do in a day or so. Please be sure to
listen to both numerous times. The differences will become more and more
apparent as you do so.
Regards,
Dave.
I am posting this message a second time as I suspect I did something
wrong the first time. Excuse me if both messages arrive.
Hi all,
I am new to the list and new to Linux audio.
I wrote a simple QT program that uses sound, and it requires the NAS
sound server.
I installed NAS from Mandrake rpm's rebooted the box, and the program
works great.
The problem is that most of my other software, for example Amarok,
stopped producing sound.
Prior to installing NAS, I used KDE and there sound server aRts. When I
start Amarok I get a message from aRts, that it can't open device for
playback.
I thus have two questions:
1. Is there a way to resolve such a situation, in a way that allows two
programs that use different sound servers to play sounds at the same time?
2. A more basic question: Why do we have to use sound servers in Linux?
shouldn't it be the job of the operating system to allow sound
multiplexing through a set of standard API's? The situation is
especially bad, if there is no comfortable solution to the problem in my
first question.
Thanks,
Hillel.
Hi Flo,
First I want to thank you for taking the time to write up some advice
about headphones. All of the combined experience of headphone users
is helpful to get past the marketing hype.
------------------------
In your discussion, you completely left out a very important reason
why headphone listening is tiring. This reason is more important than
sound quality. This reason is that stereo images as currently produced
are fundamentally incompatible with headphone listening. They should be
binaural, as discussed in the URL below.
You also posted:
> Whenever possible, listening should be performed on good-quality
> loudspeakers.
I completely and emphatically disagree with this notion as well as do other
demanding audiophiles. But perhaps you are assuming professional mixing
rather that merely listening? Please see:
http://www.binaural.com/binfaq.html
for reasons why everyone should be *listening* with headphones rather than
speakers! Practical reasons are also extremely important. You simply
cannot take your monitor speakers everywhere you go.
My guess is that very few people here are familiar with binaural recordings
or know much about them. This is unfortunate. Once again, the main
reason that headphones don't sound very good and are tiring is because the
way in which the audio image is produced is fundamentally incompatible with
headphone use. The reason that "Use speakers" is considered by many to
be good advice is that most audio is *produced for speakers*, not headphones.
This does not have to be the case. This is why people are able to sell
headphone amplifiers to improve speaker mixes and why some people such as
Andrew have asked for software methods to create better audio images for
headphone listening.
Regards,
Dave.
P.S. I see that Arnold has posted something regarding attachments. I
must say I was quite puzzled when you said that my mails show up as
attachments, because I physically have to attach them and I hadn't ever
done that to this group. (Thanks, Arnold!)
Hi John,
> Does anyone know anything about the theory behind DTS Neo:6?
Not I. Are you familiar with the sursound site:
https://mail.music.vt.edu/mailman/listinfo/sursound
They have lots of technical info on surround sound topics. However, in order
to read the archives, you must subscribe --- or so it appears to me.
Personally I have not have good luck with surround sound, so I avoid it.
I see it as a Rich Man's Approximation to physical modelling.
Regards,
Dave.
First, I want to appologize for starting a new thread. I receive list
posts in "digest-mode" and for the life of me, I can't figure out how to
reply to an individual message to continue a thread. Instructions on
how to do this would be appreciated. (I also appologize if my attempts
lead to multiple posts of this content!...moderators feel free to
restructure/delete as needed.)
Now back to it:
Thanks for the pointers, Shayne, Lee.
I am still unable to get more than 2 capture channels. I've set the
"interface", "input device", & "output device" as described by Shayne.
I also renamed my .asoundrc file to .asoundrc.bak (this will prevent it
from being used, right?). This configuration gives me 2 capture and 16
playback channels. Further, it only works if I set the number of input
channels to 0 or 2 and the number of output channels to 0 or 16.
Otherwise, I get an error: ALSA cannot set channel count to X for
capture/playback.... Now, I have also just updated qjackctl to 0.2.15
and am getting the same behavior. BTW, I am only trying to use Audio
channels, no MIDI.
Any other ideas? Is this still an issue with the driver or my
configuration? I ask because the configuration seems so
straightforward. BTW, I am using ALSA-1.0.8 patched with Lee's
multichannel-v008 patch and jack is recent from cvs (0.99.49).
Another question: What output connectors/jacks do all 16 output channels
map to on my card/external connector box? I have experimented and it
seems that playback_11-_16 are unused for me. Is this correct? The
first 10 are mixed between analog/digital outputs. However, there is a
unique set of 6 specific playback channels that give me 6 analog output
streams that I can use for 5.1 sound, which is what I want.
Thanks guys.
-Rick
>From: Jamie Bullock <jamie(a)postlude.co.uk>
>Subject: [linux-audio-user] Triton Rosegarden bank
>Dear all,
>Does anyone have, or know of a Rosegarden MIDI bank map (rgd) file for
>the Korg Triton?
>Regards,
>Jamie
I'm working on one, but have other problems to deal with right now. Is
yours a Triton Rack, and how are you connecting it?
I had the same problem on Slackware 10/Audioslack no matter what wine I used.
Went to RH9 / CCRMA and it works ... perfectly for me & NI Kontakt. I did
Fedora Core 3/CCRMA last week and was able to build fst / jack_fst
successfully there too, though I have'nt tested it much yet.
HTH,
Cheers.