Linux-audio-user

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20661 discussions

by Jack Bowling

Anybody successfully compiled the latest Audacity v1.2.1? It breaks spectacularly in the portaudio section for me with all kinds of wxGTK errors. I have the whole wxGTK environment on my box including the devel stuff and I haven't had these errors with previous versions. -- Jack Bowling mailto: jbinpg(a)shaw.ca

21 years, 6 months

Re: [linux-audio-user] audacity 1.2.1

by Juhan Leemet

The main part of my post was.... >> FWIW, I have successfully built audacity-1.2.1 on SuSE 8.2... I added the note... >> message about it using portaudio 15 which restricts it to oss, and it said >> would have to change to use the experimental (?) portaudio 16 to get alsa Where I made my mistake... >> ...On my audio machine, I've got an M-Audio >> Audiophile-2496, and I think (?) I'm restricted to using oss (because of >> envy24control?) anyway? I'm a bit unclear on that. I'll play more later (on >> another machine). Just another "data point" for you guys. > >just curious ... how does envy24control restrict you to using oss? >envy24control is an ALSA tool for controling an ice1712 chip's mixing, >routing and other features. You pretty much have to be using alsa to use >envy24control. Many people, myself included, are using envy24control and >the alsa-drivers on cards (delta-66 for me) based on this chipset. > >-Eric Rz. Thanks for the correction. As I indicated, I'm a bit fuzzy on the interactions. What confuses me more is that these days I alternate between two PC machines: one with Intel 82801AA-ICH and the other with ICE1712 - M Audio Audiophile 24/96. I was shooting from the hip... and I missed... (I should have checked everything before posting). I remember having to mess a lot with the audio driver settings on the ice1712 machine in xmms to get anything to play, but that may have been a problem with xmms(? see below). On that envy24control thing viz. alsa... I do find it annoying that there is a lack of "integration" between such apps as audacity and xmms and the audio volume control. I'm not sure what the problem is. I would have expected that the volume control on audacity or xmms should be "connectible" to the main (which one? I would suggest analog volume, as "master"? or PCM1 & PCM2?) volume control on the envy24control panel. With the 82801 it works that way: I can control volume either with xmms slider, or from kmix (or any other?). However, I think the xmms volume slider changes only the PCM volume (on my i82801 machine), leaving the master where it was. With ice1712, I have to bring up both audacity (dead slider) or xmms (dead slider) and envy24control (to adjust volume, on analog output panel). I don't have "normalization" in xmms working to my satisfaction yet, so it's even more annoying. I don't know if it's a permanent problem, or whether it will (eventually) be fixed? How? (getting a bit farther afield from audacity...) I've had other problems with applications (such as xmms) storing machine specific configuration stuff into my home directory (shared across machines). Hopefully, all that stuff should also be fixed. In principle, I should be able to move between machines (with different audio, etc.) and be able to run my applications without having to tweak everything (yet again). An "initial tweak" on a new platform is acceptable, but shouldn't mess up others. This may have been the main confusing factor for me, making me distrust things. (back to audacity!) Oh, and a couple of observations about audacity-1.2.1: 1) Mono recording on ice1712 is (again/still) transposed 1 octave down: the default recording is mono, and it seems to mess up the data stream coming from the M-Audio Audiophile-2496. I get my voice transposed 1 octave down, must be some sample/buffer mapping thing. I've seen this before with previous versions, and I've seen others mention it. Workaround is to record stereo. 2) (slight anomaly) recorded track identification panel missing until recording stops. I don't know if this is "by design" or "by mistake". When I start recording (in mono or stereo), the wavefore panels come up (empty, natch), but the track identification panel on the LHS is missing. When I stop recording, this panel appears (with the L/R panning slider, etc.). Maybe this is "by design", to prevent people trying to tweak that stuff? Only for playback? Not a problem (except you can't see mode info), but unexpected. BTW, great mailing list! I'm picking up much useful stuff here. I still don't have my audio stuff setup to my liking, but it's a background activity for me. These days soliciting paying work is taking priority (unfortunately). I do use audacity to record/playback to (re)learn some guitar and singing. It's pretty bad (my performance that is), but it's getting better. The gear helps. -- Juhan Leemet Logicognosis, Inc.

21 years, 6 months

Re: [linux-audio-user] audacity 1.2.1

by Juhan Leemet

>Jack Bowling <jbinpg(a)shaw.ca> escribió: > >> Anybody successfully compiled the latest Audacity v1.2.1? It breaks >> spectacularly in the portaudio section for me with all kinds of wxGTK >> errors. I have the whole wxGTK environment on my box including the devel >> stuff and I haven't had these errors with previous versions. > >mmm... it's funny, it fails for me too. never had had problem with audacity >before. i searched the audacity mailing list and nobody complained about >this, though... > >anyone? FWIW, I have successfully built audacity-1.2.1 on SuSE 8.2, but I haven't tested it yet. All I had to do was adjust some package (dependency) naming, since SuSE has different package names. While building, I did notice a message about it using portaudio 15 which restricts it to oss, and it said I would have to change to use the experimental (?) portaudio 16 to get alsa support. I have not done that yet. On my audio machine, I've got an M-Audio Audiophile-2496, and I think (?) I'm restricted to using oss (because of envy24control?) anyway? I'm a bit unclear on that. I'll play more later (on another machine). Just another "data point" for you guys. -- Juhan Leemet Logicognosis, Inc.

21 years, 6 months

RE: (Erado Alert - Objectionable content) Re: [linux-audio-user] Newbie checks in

by Matthew Allen

> -----Original Message----- > From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio- > user-bounces(a)music.columbia.edu] On Behalf Of Russell Hanaghan > Sent: Monday, May 10, 2004 6:54 PM > To: A list for linux audio users > Subject: (Erado Alert - Objectionable content) Re: [linux-audio-user] > Newbie checks in > > I have not messed with USB audio cards specifically. I do use a midiman > 2x2 USB midi interface with no probs...Frank has much on USB sound > devices...and none of it good! :) Actually if you check the archives, Frank recommends a couple of cheap USB sound devices. The ones he will tell you to stay away from are the non complient ones like the Midiman USB stuff (their PCI however is stellar). m. _________________________________________________ Scanned on 11 May 2004 15:58:26 Scanning by http://erado.com

21 years, 6 months

[linux-audio-user] Designing system for 'virtual' whole house audio system that deals with all kinds of audio sources - is ecasound the answer?

by Robert Rozman

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, voip or ordinary telephony applications, intercom) and several sound destinations (rooms in my house - can be either remote desktop running some network audio client or separate output on local audio card). Now I'd like to implement 'virtual' audio router/mixer in software that can be dynamically controlled from other program language (Perl is preffered in my case). I'd need to combine several audio sources to each sound destination (like big software switching/mixing/routing black box), dynamically change volumes, add/remove chains etc... I'd like to deal with all this with Ecasound (it can be controlled from Perl) - or should I use some other program ? Some possible scenarions: - when internet voice call comes in, then I connect to certain channel on audio card for certain room (route two way audio stream that comes from Internet to certain audio destination/source) - when watching TV (sound going to some audio card output), speech synthesis would like to announce something (I'd like to volume down TV audio and mix speech, and then go with TV volume to normal level) - 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) , Festival as speech synthesis, some softphone for IP telephony (that could output to ecasound or Jack) and every other valuable suggestion for software package I get. I don't know much of Jack, maybe its also part of solution.... Any other advice in apps to use, more info or any other opinion would be more than grateful. Also if anyone made some effort or thinking in this direction - it'll be of great help... Thanks in advance, Robert.

21 years, 6 months

[linux-audio-user] [ANN] galan-0.3.0-beta6 "vst(i) in the modular synth" released.

by torbenh＠gmx.de

hi... i just wanted to announce the release of galan-0.3.0_beta6. This release has vst(i) support through libfst. So if you ever wanted to wire up networks of vst plugins and instruments, you can do this now. fst is available here: http://linuxaudiosystems.com/fst/fst-1.5.tar.gz we have some issues with the embedding of windows into the app. it will work if you set managed = "N" in your wine config. and it will work with IcwWM and fluxbox. for other windowmanagers i cant tell. i hope to find this issue so that it works with every windowmanager soon. and for those who dont know. gAlan is a mixture of pd and reaktor. there is eventprocessing, and there are two windows: one for the schematics and one for the controls. in the controls window you can have several panels with custom background images. look here for an example of an instrument built with gAlan: http://galan.sourceforge.net/anti-aliased-knobs.png galan supports subpatches. and polyphony is already possible (but i will refine that a lot in the future) the documentation is not very good, and the example patches are a little old. but i hope the stuff which you can add from the Lib/ menu gets you started quite easyly. the download page is here: http://sourceforge.net/projects/galan -- torben Hohn http://galan.sourceforge.net -- The graphical Audio language

21 years, 6 months

Re: [linux-audio-user] Newbie checks in

by eviltwin69＠cableone.net

Welcome back! Check out my web site: http://myweb.cableone.net/eviltwin69/ALSA_JACK_ARDOUR.html Ardour is the best multitrack available but it takes some setup to get running. I've documented the scratch setup on my web site (since you already know your way around Linux this shouldn't be a problem). Don't forget to check out JAMIn for mastering (http://jamin.sourceforge.net/). Jan On Mon, 10 May 2004 00:16 , Jos Laake <jos(a)radiks.net> sent: >Hi all, > >I'm a lifelong musician turned geek who is now returning to the music >arena. I'm not new to technical issues. My day job until recently was >working on real-time Linux for Sony, so I know my way around a Linux >system fairly well. The music part I've got down. But music >technology, well, the last time I recorded anything was on a Tascam >4-track reel-to-reel and, whoa! Things have changed! And I'm gonna >need some help getting up to speed with all this cutting edge digital >audio stuff available today. > >As an avid open source supporter, I intend to use a Linux box as a >multitrack digital studio to help me bring 20 years of sidetracked >musical talent back to life. > >But so far I've had nothing but trouble so I went looking for help and >ended up here. So here's what I've got... > >On my main machine, I'm using RedHat 9 (kernel version = 2.4.20-6) on a >little cube Pentium4 machine (Ice Cube). The audio module that RedHat >came up with at configuration is: 'i810-audio' > >I'm also working with an old Gateway dual Pentium II runing RedHat 9 >with SMP (kernel version = 2.4.20-6SMP) and this one is using a Creative >Labs SoundBlaster MP3+ USB Audio unit. It comes up with an audio module >called 'audio'. (I'd like to set this up for my son who is a budding >guitarist). > >I was thinking about using the latest ALSA drivers (1.0.4) and Audacity, >but, first thing first. I'm having trouble getting the ALSA drivers >working. > >First question: Is this the best way to go? I mean, I want to do pro >or near-pro quality audio, but I don't want to spend boatloads on sound >cards without knowing there's good open source software to use. > >Thanks in advance for any and all help. >~Jos~ > > >Jos Laake

21 years, 6 months

[linux-audio-user] rather OT: transporting monitors?

by miriam clinton (iriXx)

hi all, rather OT, but i thought the best place to ask. heading out on a plane flight back to the UK, and need a means of transporting a computer monitor, preferably without paying for shipping as we have plenty of baggage weight available. we're planning to pack it inside its cardboard box with loads of bubble wrapping and lots of 'Fragile: handle as eggs' stickers on the side. does anyone have any experience of doing this, and have any sugestions on the best way to go about it? thanx muchly, mC~ -- "Do not engage me in irrelevant conversation" -- Seven of Nine, Star Trek: Voyager www.iriXx.org www.copyleftmedia.org.uk

21 years, 6 months

[linux-audio-user] VSTserver on MDK 10.0

by Russell Hanaghan

Has anyone got Thacs vstserver and libvst working on mandrake? I have tried and tried to no avail. Wine is running and seems to fire up the standard stuff like filemanager, windblows games, etc. I suspect it's to do with libvst but don't know. I'd really like to get this working as I have many vst plugins. thanks R~

21 years, 6 months

[linux-audio-user] xruns, ksoftirqd, alsa, jack

by Wade Winright

Hello all, I have the following set up: 1.8Ghz CPU 512 RAM VXPocket 440 (Digigram) kernel with low latency patch applied Planet CCRMA software installed When I fire up jack, I get xruns. At the same time, the ksoftirqd grabs about 50% of the CPU. Is this normal? I've tried tuning the jackd settings, and the only thing that seems to work is -C (capture only) but that seems fairly unproductive in a real multitrack recording situation. Any suggestions? Thanks! Wade

21 years, 6 months

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Linux-audio-user