Hi,
I've just made the first alpha release of my DVD-Audio authoring
application. It's available at:
http://dvd-audio.sourceforge.net
The program is currently limited to 2-channel 16-bit audio, but all
DVD-Audio samplerates are supported (44.1KHz, 48KHz, 88.2KHz, 96KHz,
176.4KHz and 192KHz).
A simple "how-to" is available here, as well as documentation about
DVD-Audio is also available on the above site.
Any feedback or questions welcome - there is a dvd-audio-devel mailing
list linked from the above site, or you can email me privately.
Dave.
>> > At http://freepats.opensrc.org there is a mellotron sample in the flac
>> > format.
>>
>> Here is a link to a draft SoundFont. No loop points were set,
>> but looping is engaged. Approximately 12.3Mb.
>>
>> http://65.125.227.61/mellotronia.SF2
>
>As for looping, in the meanwhile a friendly guy from the italian
>newsgroup it.comp.musica has done a looped version of the reed:
>
>www.viandanze.com/materiale/mello.sf2 (~3.7 Mb)
Is this last sf2 based on the flac version? How complete the original
flac version is? Have the Mellotron tapes recorded from start to end?
Have the end been faded-out after digitization? Is the end of audio
the original end on the tapes or has somebody clipped the digitization?
I'm only curious. Main thing is that the digitization has the start and
as much as possible the sustained sound.
I would suggest to use the original non-looped version for most proper
sound. Not only because it is what Mellotron generated but also because
the non-looped sounds are better than the looped sounds.
Hmm... I have heard a Fairlight II sound library on CD with very bad
post-made looping. People who have no glue how to make the sample CDs
should not make and sell them.
Now, anyone have Fairlight II or III? Orchestron?
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
Quoting thewade <pdman(a)aproximation.org>:
> Before you publish a track, either on CD or on the net, how do you
> insure someone else wont put their name on your track and take credit
> for it?
There is a quite low-tech much used technique for this. Make a demo of the
song, or the whole song on a CD/cassette/anything. Put it in an envelope
with a written, signed and timestamped letter explaining the contents (and
maybe a usage license like creative commons). Seal the envelope so that
opening it would break the seal. Then mail the envelope to yourself,
preferably via certified mail.
Last, and the most important step, is to not open the letter after you
receive it.
In case of someone stealing your song, open the letter in court, or with a
clerk, notary etc. present who then can prove that the package wasn't opened
earlier and what the package contained and when the package was sealed.
This method is known in the popular music industry for having proof of when
you have made the song.
(just remember to get all the details right, I might have left something
important out)
Sampo
Both applications are capable of syncing to jack. Set both to do that, and
they will sync together.
In ardour: windows->options->sync->Positional sync = Sync with JACK
In hydrogen: in the main window, there is a "jack transport" setting which
should have a green "light" in it.
Sampo
Quoting ish(a)sarai.net:
> Can one sync Hydrogen with Ardour so they can play together( ie Play and
> stop together) i.e. we use the mixer of Ardour(with one track of
> Hydrogen)
> for the final mixdown with the Ardour Mixer.
>
> <>ISh
>
>
--- Clemens Ladisch <clemens(a)ladisch.de> wrote:
> R Parker wrote:
> > > > I'm trying to upgrade the software version of
> a Tascam
> > > > DM-24 digital mixing consol using amidi.
> >
> > Tried; 'aplaymidi -p 72:1 -d 5 flash.mid' but it
> > produces "MIDI file Dump Error ! -> Code = f1"
>
> This invocation looks OK.
>
> What does "Code f1" mean?
Not a clue. I went through the DM-24 manual trying to
find that message and couldn't. Even if I had it might
not have been informative--based on the other
explanations.
I broke down and used a Macintosh with a MIDI
sequencer application titled Midigraphy. It worked
fine but what I don't get is what is different between
midigraphy and aplaymidi or muse.
Anyway, I wish there was a clear method for me to
provide you with some useful information.
ron
> Regards,
> Clemens
>
>
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>
>
>If enough people use proprietary
>software, eventually (as was the case in the desktop OS market until
>recently) there will effectively be no choice.
>
well, enough people use proprietary software *now* - but there's still
choice ... in fact, a big reason why people would use proprietary
software over open-source is because there *is no choice*
the thing with skype is, there is no free alternative (the ones
mentioned in this thread are useless in any context i've tried them in)
... so anyone with programming skills who thinks we shouldn't be using
this software should be writing an open-source alternative ...
there is no point having a discussion on whether one should be using
open or proprietary software (the answer is bleeding obvious) when
clearly such a choice is not available in this case.
it's really stupid to expect people to forgo doing something *just
because* there is not a free alternative. i mean, i'd write an
open-source application that did what skype does if i knew how to
program - but i don't, so i'll keep using skype till someone does.
shayne
I have now two soundcards in my machine:
integrated VIA 686 soundcard (default) and SB Audigy 2
Which is the safest and trustworthy way to change Audigy as primary
card? Or then, "removing" the integrated card since I don't need it?
I edited the file /etc/modprobe.conf by changing
- "alias sound-slot-0 snd-via82xx" to "alias sound-slot-1 snd-via82xx"
and
- "alias sound-slot-1 snd-emu10k1" to "alias sound-slot-0 snd-emu10k1".
Didn't help, since for example alsamixer shows snd-via82xx as default.
My etc/modprobe.conf says:
# This file is autogenerated from /etc/modules.conf using
generate-modprobe.conf command
alias eth0 3c59x
remove snd-emu10k1 /sbin/modprobe -r snd-pcm-oss; /sbin/modprobe
--first-time -r --ignore-remove snd-emu10k1
install snd-emu10k1 /sbin/modprobe --first-time --ignore-install
snd-emu10k1 && { /sbin/modprobe snd-pcm-oss; /bin/true; }
alias sound-slot-1 snd-via82xx
install scsi_hostadapter /sbin/modprobe aic7xxx; /bin/true
alias sound-slot-0 snd-emu10k1
remove snd-via82xx /sbin/modprobe -r snd-pcm-oss; /sbin/modprobe
--first-time -r --ignore-remove snd-via82xx
install snd-via82xx /sbin/modprobe --first-time --ignore-install
snd-via82xx && { /sbin/modprobe snd-pcm-oss; /bin/true; }
install usb-interface /sbin/modprobe uhci-hcd; /bin/true
alias eth1 eth1394
alias ieee1394-controller ohci1394
alias char-major-195* nvidia
Hi, just spent a month getting a linux audio workstation going and I've got a load
newbie questions I have'nt been able to solve, or find answers to; some technical, some bigger picture :
Machine Specs
-------------
Uniwill 340s2 (Advent 5480DVD) laptop
384Mb RAM
PIII 800Mhz (8 speed steps)
SIS 7018 Integrated sound chip
Chipset mostly SIS I guess.
40Gb HD (Fujitsu, recent model)
(Yea I know it's a piece of junk but I could write a heafty tome about my money situation at the moment)
Software
--------
Win2K NTFS 20Gb
NTFS 10Gb
Fedora Core 2 + CCRMA (everything from rpms), CCRMA Low latency kernal latest stable (2.6) ext2 10Gb
Rosegarden4-1, built from source
Hokay,
1) I boot up the low latency kernal everything is fine until I run a midi app. The machine has no physical ports but presents an input port and 5 output ports in Qjackctl (SIS7018,trident driver). I think the first is raw, then Midi0-3. The clearest repeatable falut I can give is when I run Qjackctl, with jack either running or not running and try to connect to one of the SIS midi ports, the machine freezes when I connect (Caps lock does nothing). Why would I connect to one of these ports you ask ? Well I don't but some apps do it automatically (rosegarden, Kmid) and this has the same effect. Can anyone tell me how to fix the driver or just get rid of the midi ports in alsa (or any other part of the midi chain) so that apps don't try to connect to them ?
2) I have no modules.conf or conf.modules. tried creating them, putting stuff in them, all sorts.
3) Should I be building all my own software for i686 ? Most of the CCRMA stuff is i386 I can't remember at which point the 'big change' in intel architecture happend (was it 286 got protected mode or somthing). Anyone give me a ball park performace improvment percentage for rebuilding the music software ? Which leads me onto ...
4) Rosegarden. Without any midi tracks, or the low latency kernal, rosegarden was starting to grind along (Full CPU load) with 6 audio tracks (48K), no effects or anything. Is it possible to do anything useful (say 10midi, 10audio + usual effects) in rosegarden with my machine specs ? Has any one got a recent good spec machine which allows them to do this using rosegarden. Has anyone done this with Muse, should I switch ? Rosegardens code is really pretty but is it fast ?
5) Latency. Is there any way to do things in non-real time ? Other than realtime monitoring (which I'm happy to go without) is there any way to pre-render stuff and put up with a pause before the music starts ? Am I right in saying that if you have 30ms latency, and you record a track, no compensation for this latency will be applied and the track will be 30ms late ? Is there any time stamping built into jack/alsa ? You know what I'm getting at, have I miss-understood somthing ? For my 2c worth, as a drummer (amongst other things), you may not be able to hear 10ms (low latency, with your snare going through some effects), but you can feel it. It's THE difference between 'energetic' and 'Chill out' tracks.
6) When I plump for a new box, should I get multi-processor.
I've got no internet connection and have to go to the public library and download everything onto my 128Mb mp3 player at the moment. (Incidently if anyone in a similar position, I've got some tips on using yum as a package manager)
Any help (especially with the driver problem) grately appreciated, I'm nearly there !