Linux-audio-user March 2006

linux-audio-user@lists.linuxaudio.org

215 participants
309 discussions

by Brett McCoy

Anyone used an M-Audio Midisport (like the 2x2 or 4x4)? How well do they work under Linux? I see they are supported in ALSA via EZ-USB. -- Brett -- Brett McCoy: Programmer by Day, Guitarist by Night http://www.alhazred.com http://www.cassandrasyndrome.com http://www.revelmoon.com

18 years, 8 months

[linux-audio-user] Muse and MIDI Syncing stuff

by Hiram Abiff

greetings! I've been messing with Muse a lot recently and there is one thing I haven't been able to manage completely. and that thing is, you've guessed it, MIDI syncing. I own a DR-880 Drum Machine which I connect via a USB cable to my computer and use it as a MIDI device in the qjackctl pathcbay I then proceed to add the DR-880 in Midi Ports/Soft Synths in Muse, after which Midi Ports/Soft Synths shows an Instrument named "generic midi" and a device named "DR-880" followed by: "Play:Device or resource busy". I don't understand why Muse says it's busy when nothing is playing on it. I set up MIDI Sync so that Muse is the Master and turn on "MIDI Clock" and "MIDI Machine Control" and set the port to 3 'cause that's the port I used in MIDI Ports/Soft Synths for the DR-880. So, finally when I press play in Muse the DR-880 starts playing. yay, great. But my enthusiasm didn't last for long as I was soon to find out that when I skip playback in Muse to a certain point or rewind it back the DR-880 keeps on playing. And this is EXTREMELY annoying and discouraging, especially for long songs where I have to play them from start each time just to record a MIDI solo on my keyboard at the last minute and it also makes music making tedious and boring. Now, what am I missing? Is Muse supposed to work this way and only sync with a MIDI device when played from the start? Or more probaly I misconfigured or my lack of knowledge is causing this. I kindly ask for any sort of help or some light on my dark path of playing the song from start to end each time I want to record a new part or edit it. Note: I figured out that I could place markers in certain points of the track where a pattern on the DR-880 is starting and then manually rewind on the DR-880 to that part when I want to start from a particular point in the song but this is also time consuming and driving me mad. Sorry for the extremely long mail but I am starting to lose my patience with this particular issue. Thnx, Hiram. -- "I happen to think that computers are the most important thing to happen to musicians since the invention of cat-gut which was a long time ago. " Robert Moog

18 years, 8 months

[linux-audio-user] Aligning audiofiles for different recorders.

by Phil Mendelsohn

> in terms of automating the alignment, it seems that ardour should > probably add a new feature: "position sync point at loudest sample". > then you just make sure you have a loud reference point in each file, > set up the sync points, and then align them all to a single reference > region. > > alas, it does not have that feature at this time. Hi, Paul, My worst days in audio and audio for video usually involved the kind of "wild sync" that is being discussed here. If the drift between channels is small enough to avoid problems, it will be easy enough to align by hand. If not, it won't help anyway. If you're ever going to marry the audio to video -- and the audio is longer than a minute or two -- then God have mercy on your soul! One wild clock is bad enough; two isn't even funny. Loudest sample is a criteria that can result in false positives -- especially if the recording is a field recording, not a studio recording. There is a point -- reached quite soon -- where it would be better to take the two sources back to analog, so that they can be rerecorded in the studio with a single A/D than to monkey with them in the asychronous state. Be a pal, a remember that good habits don't have to be broken! "One clock to rule them all..." ;) Don't encourage sloppy sync! Save yourself the work, and add a cooler feature instead -- like "Auto-Improve Lyrics"! I've been playing with Ardour, but am not sure if it has the equivalent of "Sync to Mark" that I used to use in Sonic, but that's a nice compromise between automated and manual. Cheers, Phil M -- Dept. of Mathematics, 342 Machray Hall U. of Manitoba, Winnipeg, Manitoba, Canada R3T 2N2 Office: 446 Machray Hall, 204-474-6470 http://www.rephil.org/ phil at rephil dot org

18 years, 8 months

[linux-audio-user] Re: Aligning audiofiles for different recorders.

by philicorda

> Date: Tue, 7 Mar 2006 14:22:48 +0100 > From: "Alex Polite" <notmyprivateemail(a)gmail.com> > Subject: [linux-audio-user] Aligning audiofiles for different > recorders. > Howdy > > I make a lot of interviews. Right now I have a two channel recorder. I > connect one headset mic to each channel, put one on myself and one on > the interviewee. Works kind of alright but I would rather have the > interviewee wear one recorder and wear another myself. This gives us > more freedom to move around during the interview. > > The tricky part will be to align the two separate recordings. I could > probably do it manually in ardour, dragging the regions back and forth > and stretching them until they line up exactly. > > I've googled a bit to find a tool that does this aligning > automatically but haven't come up with anything. > > Does anybody out here know of something? You could try using a clapper board, or anything that makes a sharp click. It should get picked up by both mics. Then, when you come to align the two recordings, you just line up the clicks at the start and they will be in sync. If both recorders are digital then they should stay in sync for quite a while. You should not need to do any time stretching. This is the old way of getting two recorders (a camera and location sound) to sync, and works quite well. It's still used nowadays when people are too cheap (like me) to do proper time code. > > alex

18 years, 8 months

[linux-audio-user] Re: Aligning audiofiles for different recorders.

by Phil Mendelsohn

> I make a lot of interviews. Right now I have a two channel recorder. I > connect one headset mic to each channel, put one on myself and one on > the interviewee. Works kind of alright but I would rather have the > interviewee wear one recorder and wear another myself. This gives us > more freedom to move around during the interview. > > The tricky part will be to align the two separate recordings. I could > probably do it manually in ardour, dragging the regions back and forth > and stretching them until they line up exactly. It's a lot easier to syncronize the word clocks when recording! The best solution is *one* wireless mic (for the interviewee) and you get to keep the recorder, your own mic, and the receiver. That way you can still all walk around. With two recorders, you really want a way to share word clock. You might get away with two recorders, but if you *do* develop a problem, that will _really_ drive you nuts in the editing room! If you have a budget for a second recorder, I'd get it to have a spare (or redundant recording!) Cheers, Phil M -- Dept. of Mathematics, 342 Machray Hall U. of Manitoba, Winnipeg, Manitoba, Canada R3T 2N2 Office: 446 Machray Hall, 204-474-6470 http://www.rephil.org/ phil at rephil dot org

18 years, 8 months

[linux-audio-user] [ANN] QjackCtl 0.2.20 released!

by Rui Nuno Capela

Greetings, So here comes the time for another public release of the (cute) JACK Audio Connection Kit - Qt Interface: QjackCtl 0.2.20 is out! Just as one can read from the change log: - Server path setting now accepts custom command line parameters (after a kind suggestion from Jussi Laako). - The internal XRUN callback notification statistics and reporting has been changed to be a bit less intrusive. - Patchbay socket dialog gets some more eye-candy as icons have been added to the client and plug selection (combobox) widgets. - Connections and patchbay lines coloring has changed just slightly :) - New patchbay socket forwarding feature. Any patchbay socket can now be set to have all its connections replicated (i.e. forwarded) to another one, which will behave actively as a clone of the former. Forward connections are shown by vertical directed colored lines, and can be selected either on socket dialog or from context menu (currently experimental, only applicable to input/writable sockets). - Optional specification of alternate JACK and/or ALSA installation paths on configure time (after a patch from Lucas Brasilino, thanks). Available from the usual places: http://qjackctl.sourceforge.net http://sourceforge.net/projects/qjackctl Enjoy. -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

18 years, 8 months

[linux-audio-user] Converting Unsigned Byte Audio to 2-Channel WAV with SOX

by Martin McCormick

I was once able to convert a 8K samples per second unsigned audio file gotten by sending /dev/dsp >thefile to a 44,100 samples per second stereo .wav file suitable for burning on to an audio CD. I found what I believe to have been the command I used in the form of a shell script but I get the following errors: sox: Do not support unsigned with 16-bit data. Forcing to Signed. sox: Invalid options specified to avg for this channel combination The script is: #! /bin/sh sox -r8000 -t ub cdda.ub -t wav -c 2 -w -r44100 output.wav resample .95 avg 1,1 Just for laughs, I removed the avg flag and actually got a .wav file which was the correct pitch and all, but which was full of clicks and missing pieces of sound, obviously not usable. According to the sox manual, the -r8000 isn't actually necessary since sox defaults to a 8-K sample rate. The only thing I can think of is that I have forgotten some step that I put in to the script originally or that sox has tightened up some of the syntax in the last couple of years. All I remember was that the resulting CD worked and didn't sound any worse than the original file gotten from /dev/dsp. The avg flag makes the levels correct for both channels which, in this case, are the same. Thanks for any help. Martin McCormick WB5AGZ Stillwater, OK Systems Engineer OSU Information Technology Department Network Operations Group

18 years, 8 months

[linux-audio-user] linuxsampler question

by Julien Claassen

Hi LinuxSamplers! I've just seen a nice gigasample library and I wonder, if it could be used with LS? For those, who wanna have a look themselves: http://www.postpiano.com Search for products, pianos "OLD LADY". The library has I think 12 velocity layers, same goes for pedal-down- and key-release-samples. It _CAN_ also use IR to give body-resonance and concert-hall reverb (but that's not so important. Anyone any ideas about that? Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net - the Linux TextBased Studio guide

18 years, 8 months

[linux-audio-user] Web-based collaborative band/musicians app?

by Jacob

Hello, I'm looking for a web-based application that supports working on and discussing of music and pieces within an band. What do I would like to do with such an app: - Upload of mp3 takes which has been recorded during rehearsals or sessions. - Click somewhere in a timeline based image (e.g. amplitude/time or spectrum/time) and the playback starts there. - Define views on parts of an mp3 recording, because they are sometimes several hours long and often contain different songs and lots of talk - Attach Comments (of different musicians) to certain parts or timestamps of a recording - Discuss pieces, eventually with inclusion of some staves (which should be rendered on the server). - All editing should be stored in a versioning repository, like most wikis do. - It should be usable be musicians who don't have a degree in information sciences and who don't want to spent to much time with the platform itself. Perhaps something like a wiki for musicians. - Hardware requisites on the user side should not be more than be a web browser (graphics based) and a mp3 playing app that is capable to play back streams. What I'm _not_ looking for: - Something like Ardour with a web frontend (ok, it wouldn't be a show stopper if it's features were available as well). So, before starting to write something like that on my own from scratch I would like to know, if there is some similar project out there in the world that needs some contribution instead and that would be usable in the short term. Thanks for reading, Yours, Jacob

18 years, 8 months

[linux-audio-user] Midi?? What Midi??

by Renato Fabbri

I´ve tryed getting my Radium usb keybord to work for days now, its not nice. where can the probelm be? well.. I dont have that modules.conf ou conf.modules file in /etc. There is a modprobe.conf and I mess with it. Second, in console: #amidi -l Device Name and nothing more #aconnect -i -o gives me a cliente 0: ´system´[type kernel] an client 62: ´midi through´ [type kernel] and nothing more.... I am using alsa for my delta 66 and it is doing fine. My Fedora Core 3 with planet ccrma is doing fine too. when alsa is turned on, it starts snd-usb-audio as it should. And I have hotplug installed. Help? thanx Renato __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 8 months

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Linux-audio-user March 2006