From cjdarken at nps.edu Mon Nov 1 07:50:11 2004 From: cjdarken at nps.edu (Darken, Chris USA) Date: Mon, 1 Nov 2004 07:50:11 -0800 Subject: [CM] RE: clm Message-ID: Bill: config.windoze was exactly what I needed. all.lisp loaded without problems and the violin examples, and Cook's singer, which is really what I wanted to play with, now seem to work fine. Maybe I misled by calling the compiler Visual C++ 6. That's what I call it and what it's called in my program menu, but it really is just Microsoft's C compiler. Clearly clm has been used with this compiler before, since there are references to "cl" in the code, and no one calls their C compiler that except Microsoft, that I am aware. Anyhow, thanks a million for the file! Am CC'ing the list, so that the next fellow searching the mailing list archive will find it. Chris -----Original Message----- From: Bill Schottstaedt [mailto:bil at ccrma.Stanford.EDU] Sent: Monday, November 01, 2004 4:41 AM To: Darken, Chris USA Subject: clm I think it's saying that "configure" did not run -- here's a copy of config.windoze (not sure how up-to-date it is, ot whether it has any hope of working with VisualC++): /* config.h. Generated by configure. */ #ifndef CONFIG_H_LOADED #define CONFIG_H_LOADED /* #undef const */ /* #undef CLOSEDIR_VOID */ #define RETSIGTYPE void /* Define to `int' or something if doesn't define. */ /* #undef mode_t */ /* #undef pid_t */ /* #undef size_t */ /* #undef off_t */ /* #undef WORDS_BIGENDIAN */ /* #define HAVE_GETCWD 1 */ /* #define HAVE_GETWD 1 */ #define HAVE_STRFTIME 1 #define HAVE_STRERROR 1 #define HAVE_VPRINTF 1 /* #define HAVE_READLINK 1 */ /* #define HAVE_SETLOCALE 1 */ /* #define HAVE_SLEEP 1 */ /* #define HAVE_ACCESS 1 */ /* #define HAVE_OPENDIR 1 */ /* #define HAVE_SIGNAL 1 */ /* #define HAVE_FINITE 1 */ /* #define HAVE_ISNAN 1 */ /* #define HAVE_FSTATFS 1 */ /* #define HAVE_CLOCK 1 */ /* #define HAVE_VSNPRINTF 1 */ #define HAVE_SNPRINTF 1 #define HAVE_MEMMOVE 1 #define HAVE_STRDUP 1 /* #define HAVE_FILENO 1 */ /* #define HAVE_LSTAT 1 */ #define STDC_HEADERS 1 /* #undef TM_IN_SYS_TIME */ /* #define HAVE_SYS_MOUNT_H 1 */ /* #define HAVE_DIRENT_H 1 */ /* #undef HAVE_NDIR_H */ /* #undef HAVE_SYS_DIR_H */ /* #undef HAVE_SYS_NDIR_H */ #define HAVE_FCNTL_H 1 #define HAVE_LIMITS_H 1 #define HAVE_STRING_H 1 /* #define HAVE_UNISTD_H 1 */ /* #define HAVE_STDBOOL_H 1 */ /* #undef HAVE_SYS_TIME_H */ /* #define HAVE_DLFCN_H 1 */ /* #undef HAVE_SOUNDCARD_H */ /* #define HAVE_SYS_SOUNDCARD_H 1 */ /* #undef HAVE_MACHINE_SOUNDCARD_H */ /* #undef HAVE_SYS_MIXER_H */ /* #undef USR_LIB_OSS */ /* #undef USR_LOCAL_LIB_OSS */ /* #undef OPT_OSS */ /* #undef VAR_LIB_OSS */ /* #undef HAVE_LIBC_H */ /* #define HAVE_SYS_VFS_H 1 */ /* #define HAVE_SYS_STATFS_H 1 */ /* #define HAVE_FPU_CONTROL_H 1 */ /* #define HAVE_SETJMP_H 1 */ /* #define TIME_WITH_SYS_TIME 1 */ /* #define HAVE_GNU_LIBC_VERSION_H 1 */ /* #define HAVE_PWD_H 1 */ /* #define HAVE_LOCALE_H 1 */ /* #undef HAVE_SYS_FPU_H */ /* #define HAVE_SYS_PARAM_H 1 */ /* #define HAVE_ALSA_ASOUNDLIB_H 1 */ /* #define HAVE_BYTESWAP_H 1 */ /* */ /* #define _FILE_OFFSET_BITS 64 */ /* #undef _LARGE_FILES */ #define SIZEOF_INT 4 #define SIZEOF_CHAR 1 #define SIZEOF_LONG 4 #define SIZEOF_SHORT 2 #define SIZEOF_INT_P 4 #define SIZEOF_FLOAT 4 #define SIZEOF_VOID_P 4 #define SIZEOF_OFF_T 4 /* #define LINUX 1 */ /* #undef SGI */ /* #undef ALPHA */ /* #undef HPUX */ /* #undef SUN */ /* #undef SOLARIS */ /* #undef OPENBSD */ #define WINDOZE 1 /* #define HAVE_OSS 1 */ /* #undef HAVE_ALSA */ /* #undef HAVE_JACK */ /* #undef HAVE_SAM_9407 */ /* #undef MAC_OSX */ /* #undef ESD */ /* #undef BSDI */ #define HAVE_EXTENSION_LANGUAGE 0 /* #undef SND_CONFIG_GET_ID_ARGS */ #define Float double #define SNDLIB_USE_FLOATS 1 /* #undef MUS_SAMPLE_BITS */ /* #undef ESD_VERSION */ /* #undef AUDIOFILE_VERSION */ /* #undef WITH_MODULES */ /* #undef HAVE_KAUDIODEVICEPROPERTYTRANSPORTTYPE */ /* #undef HAVE_KLINEARPCMFORMATFLAGISNONINTERLEAVED */ #define USE_SND 0 #define CLM 1 #endif From hgudenus at fh-sbg.ac.at Mon Nov 1 06:26:15 2004 From: hgudenus at fh-sbg.ac.at (Heinrich Gudenus) Date: Mon, 1 Nov 2004 15:26:15 +0100 Subject: [CM] [CLM] eval within the run-body. References: <20041029190002.16056.13174.Mailman@cm-mail.stanford.edu> Message-ID: <000e01c4c01e$bea1e790$1c00a8c0@LAPHENNE> hello, I am trying to write an instrument in clm, where it is possible to use a fir-filter, formant and/or delay.. in every possible combination .. i thought of a variable which holds the output-call and the filters that are turned on e.g. [when fir=t and formant=t and delay=nil generate: (fir-filter fir (formant formant (locsig loc i ...)))] ... this should be evaluated within the run-body ... doesnt work .. of course .. apart from the fact that eval is not supporte within the run-body, i think that my suggestion for solution is incorrect .. one working solution would be to write down every possible combination .. (this seems to be cumbersome) .... -- is it possible to evaluate a variable within the run-body.. .. -- does anyone have a suggestion how to solve my (little) problem .. (the idea doesnt seem to be so unusual, does it?) ... thx. h.g. From thomasbechteler at iyte.edu.tr Tue Nov 2 02:06:05 2004 From: thomasbechteler at iyte.edu.tr (Thomas F. Bechteler) Date: Tue, 2 Nov 2004 12:06:05 +0200 Subject: [CM] snd - channel number during installation Message-ID: <200411021206.06182.thomasbechteler@iyte.edu.tr> Dear snd-experts! I am using SuSE 9.0 pro. (and ALSA) which is shipped with snd-6.11, and as a hardware I am using the Hoontech DSP24 value. After installing "snd" I got for the record window 12 input channels and 10 output channels. However, the DSP24 Value has only 2 input and 2 ouput channels. To the best of my knowledge, I configured the soundcard correct and I can play sound. (The record feature I didn't test extensively yet.) Can I choose the displayed channels or can I configure during the installation the channel number of snd? Thank you in advance for your solutions and comments. Thomas P.S.: I used previously RedHat 8.0 together with the same soundcard but an older version of snd (maybe 6.0). At that time the channel number was correct. From fbar at footils.org Tue Nov 2 08:51:15 2004 From: fbar at footils.org (Frank Barknecht) Date: Tue, 2 Nov 2004 17:51:15 +0100 Subject: [CM] snd - channel number during installation In-Reply-To: <200411021206.06182.thomasbechteler@iyte.edu.tr> References: <200411021206.06182.thomasbechteler@iyte.edu.tr> Message-ID: <20041102165115.GA2662@fliwatut.scifi> Hallo, Thomas F. Bechteler hat gesagt: // Thomas F. Bechteler wrote: > I am using SuSE 9.0 pro. (and ALSA) which is shipped with snd-6.11, You should consider upgrading manually. Snd7 is much more fun. > and as a hardware I am using the Hoontech DSP24 value. > After installing "snd" I got for the record window 12 input channels > and 10 output channels. However, the DSP24 Value has only > 2 input and 2 ouput channels. To the best of my knowledge, > I configured the soundcard correct and I can play sound. > (The record feature I didn't test extensively yet.) > > Can I choose the displayed channels or can I configure > during the installation the channel number of snd? No, you cannot use them. I get the same effect with my M-Audio Audiophile, which has the same chipset as your card. This chipset itself is capable of 10/12 operation, but it is built onto several cards which don't wire all those channels to real in/out-channels. Unfortunatly at a driver level, ALSA doesn't recognize the hardware differences between those cards, as the driver is chipset specific. You get the same, if you start a mixer like envy2control or alsamixer: It will show much more channels than actually available. envy24control has an option to reduce the displayed channels: -o, --outputs Limit number of outputs to display -i, --input Limit number of inputs to display -s, --spdif Limit number of spdif outputs to display IMO something like this would be useful for Snd as well, because the mixer window shown for our cards is really HUGE. (It doesn't fit on my screen at least.) Ciao -- Frank Barknecht _ ______footils.org__ From thomasbechteler at iyte.edu.tr Tue Nov 2 23:19:46 2004 From: thomasbechteler at iyte.edu.tr (Thomas F. Bechteler) Date: Wed, 3 Nov 2004 09:19:46 +0200 Subject: [CM] snd - channel number during installation In-Reply-To: <20041102165115.GA2662@fliwatut.scifi> References: <200411021206.06182.thomasbechteler@iyte.edu.tr> <20041102165115.GA2662@fliwatut.scifi> Message-ID: <200411030919.46438.thomasbechteler@iyte.edu.tr> Hello (and Servus), On Tuesday 02 November 2004 18:51, Frank Barknecht wrote: > Hallo, > > Thomas F. Bechteler hat gesagt: // Thomas F. Bechteler wrote: > > I am using SuSE 9.0 pro. (and ALSA) which is shipped with snd-6.11, > > You should consider upgrading manually. Snd7 is much more fun. Yep, I tried that too but I faced some problems with the GUI. (And this was the same for older versions of snd I tried to install for testing.) The "configure" script couldn't find "motif" or "gtk" on my SuSE distribution - and this for my two computers. It seems SuSE put the important files to some strange folders, or not at all. "configure" was claiming not to find the necessary file "gtk-config", nor did I. Hence, snd was installed without GUI. Now I am trying to figure out how to solve that problem. [...] > No, you cannot use them. I get the same effect with my M-Audio > Audiophile, which has the same chipset as your card. This chipset > itself is capable of 10/12 operation, but it is built onto several > cards which don't wire all those channels to real in/out-channels. > Unfortunatly at a driver level, ALSA doesn't recognize the hardware > differences between those cards, as the driver is chipset specific. Hmmm, although the setup tool YaST allows to choose among several soundcards with the same chip-set ... > You get the same, if you start a mixer like envy2control or alsamixer: > It will show much more channels than actually available. > > envy24control has an option to reduce the displayed channels: > > -o, --outputs Limit number of outputs to display > -i, --input Limit number of inputs to display > -s, --spdif Limit number of spdif outputs to display Thanks for this information! > IMO something like this would be useful for Snd as well, because the > mixer window shown for our cards is really HUGE. (It doesn't fit on my > screen at least.) Oh yeah, that is the same for me: 10+12 channels are too many to be properly displayed on my screen. It would be a nice improvement of "snd" to be able to choose the number of displayed channels! I hope that will be a feature in the near future. Until this time I have to cope with the many many channels ;-) Thomas From fbar at footils.org Wed Nov 3 00:25:07 2004 From: fbar at footils.org (Frank Barknecht) Date: Wed, 3 Nov 2004 09:25:07 +0100 Subject: [CM] snd - channel number during installation In-Reply-To: <200411030919.46438.thomasbechteler@iyte.edu.tr> References: <200411021206.06182.thomasbechteler@iyte.edu.tr> <20041102165115.GA2662@fliwatut.scifi> <200411030919.46438.thomasbechteler@iyte.edu.tr> Message-ID: <20041103082507.GA9692@fliwatut.scifi> Hallo, Thomas F. Bechteler hat gesagt: // Thomas F. Bechteler wrote: > On Tuesday 02 November 2004 18:51, Frank Barknecht wrote: > > Hallo, > The "configure" script couldn't find "motif" or "gtk" on my SuSE > distribution - and this for my two computers. It seems SuSE > put the important files to some strange folders, or not at all. > "configure" was claiming not to find the necessary file "gtk-config", > nor did I. Hence, snd was installed without GUI. Now I am trying > to figure out how to solve that problem. Do you have the "-devel" packages for Gtk (or whatever they are called on Suse) installed? > > Unfortunatly at a driver level, ALSA doesn't recognize the hardware > > differences between those cards, as the driver is chipset specific. > > Hmmm, although the setup tool YaST allows to choose among > several soundcards with the same chip-set ... You will also get the card's name, if you do a "cat /proc/asound/cards", but nevertheless, for some reason, ALSA doesn't care about the actual channel counts. This is not a Snd problem, but a more flexible mixer window surely would be a good additional feature. Ciao -- Frank Barknecht _ ______footils.org__ From tim at orford.org Tue Nov 2 11:28:20 2004 From: tim at orford.org (Tim Orford) Date: Tue, 02 Nov 2004 20:28:20 +0100 Subject: [CM] Re: snd jack playback mode bug In-Reply-To: References: <20041029190002.16056.13174.Mailman@cm-mail.stanford.edu> Message-ID: <20041102192820.GH2729@orford.org> On Sun, Oct 31, 2004 at 06:16:15PM +0100, Kjetil Svalastog Matheussen wrote: > > If you are running jack in Playback mode, snd 7.8 will > > imediately segfault in mus_audio_read_buffers(). > > > > i assume that this is easily repeatable. I can provide > > any additional information if needed. > > > > Yes, please provide some code that use mus_audio_read_buffers. > I wrote the jack-code, but have never used mus_audio_read_buffers. yes thanks for adding this v useful functionality! I must apologise for the incorrect information regarding mus_audio_read_buffers - that was indicated by gdb but appears to be false. By adding printfs i beleive the segfault happens here: static int sndjack_getnuminchannels(void){ int lokke=0; const char **ports=jack_get_ports(sndjack_client,"alsa_pcm:capture_*","",0); while(ports[lokke]!=NULL){ ////// ****appears to crash here with lokke=0 lokke++; } if(lokke<2) return 2; return lokke; } I'm using jack 9.8, but am still assuming that this segfault is reproducable on any system when jack is started with '--playback'. If others have no time or inclination to fix, i will gladly do so, but am reluctant as i have little experience of jack coding or knowledge of Snd. Once again, thanks Kjetil for adding jack support, and regards to all. -- Tim Orford From bil at ccrma.Stanford.EDU Wed Nov 3 06:51:02 2004 From: bil at ccrma.Stanford.EDU (Bill Schottstaedt) Date: Wed, 03 Nov 2004 06:51:02 -0800 Subject: [CM] [CLM] eval within the run-body. In-Reply-To: <000c01c4bdc0$22933620$1c00a8c0@LAPHENNE> References: <418229A5.3050008@ccrma> <000c01c4bdc0$22933620$1c00a8c0@LAPHENNE> Message-ID: <4188F056.7010600@ccrma> examp.scm has a similar instrument named chain-dsps (for Snd/CLM, not CL/CLM), but in this case it might be simpler to use a loop and a case statement: (let ((current-output 0.0)) (do ((i 0 (1+ i))) ((= i num-gens)) (case (aref gens i) (0 (setf current-output (fir-filter fir current-output))) (1 (setf current-output (delay dly current-output))) (2 (setf current-output (formant frm current-output))) ... )) (locsig locs samp current-output)) where we obviously set up the various generators in advance. (There's an example in the test file ug.ins: "s4"). From bil at ccrma.Stanford.EDU Wed Nov 3 07:00:56 2004 From: bil at ccrma.Stanford.EDU (Bill Schottstaedt) Date: Wed, 03 Nov 2004 07:00:56 -0800 Subject: [CM] snd - channel number during installation In-Reply-To: <200411030919.46438.thomasbechteler@iyte.edu.tr> References: <200411021206.06182.thomasbechteler@iyte.edu.tr> <20041102165115.GA2662@fliwatut.scifi> <200411030919.46438.thomasbechteler@iyte.edu.tr> Message-ID: <4188F2A8.1080300@ccrma> > I hope that will be a feature in the near future. Until this time I have to > cope with the many many channels ;-) I could easily add a variable for this -- one exists already in a hidden (debugging) context, but I wasn't sure how it would interact with the various audio choices -- I'll try to get around to it later today. I assume you'd want to say something like "record n chans and ignore the rest", or "record chan 3 and 123..."? From peter at peterlutek.com Wed Nov 3 08:17:40 2004 From: peter at peterlutek.com (Peter Lutek) Date: Wed, 03 Nov 2004 11:17:40 -0500 Subject: [CM] snd - channel number during installation In-Reply-To: <4188F2A8.1080300@ccrma> References: <200411021206.06182.thomasbechteler@iyte.edu.tr> <20041102165115.GA2662@fliwatut.scifi> <200411030919.46438.thomasbechteler@iyte.edu.tr> <4188F2A8.1080300@ccrma> Message-ID: <1099498659.3422.2.camel@localhost.localdomain> On Wed, 2004-11-03 at 10:00, Bill Schottstaedt wrote: > > I hope that will be a feature in the near future. Until this time I have to > > cope with the many many channels ;-) > > I could easily add a variable for this -- one exists already > in a hidden (debugging) context, but I wasn't sure how it would > interact with the various audio choices -- I'll try to get around > to it later today. I assume you'd want to say something like > "record n chans and ignore the rest", or "record chan 3 and 123..."? i have a similar issue with my RME Multiface -- 18 ins and 18 outs overflow my display space. personally, i would suggest your latter example, so that the user may choose any arbitrary (and not necessarily contiguous) set of channels. thanks, bill! -p From thomasbechteler at iyte.edu.tr Wed Nov 3 23:54:06 2004 From: thomasbechteler at iyte.edu.tr (Thomas F. Bechteler) Date: Thu, 4 Nov 2004 09:54:06 +0200 Subject: [CM] snd - channel number during installation Message-ID: <200411040954.06730.thomasbechteler@iyte.edu.tr> Hello! I finally installed snd-7.0 and it shows me 2 input and 2 ouput channels. It may depend on the snd-version how many channels are being displayed. I tested the recording feature and it was working. Also playback works fine ... up to one point, but that is no snd-specific problem. This problem occurs also with e.g. "audacity". But snd, since it is a more professional program, gave me some information why it crashes. I posted the following message to "alt.os.linux.suse" since it seems a SuSE-specific problem. For the interested reader I copied my message at the end of this message. Cordially Thomas ============================================= Dear Reader! Recently I posted here a message about my problem with sound. I am using SuSE 9.0 pro. Now I came closer to specify my problem, and it seems not a KDE specific problem, as I assumed in my previous posted message. Problem: When I am using recording software ("audacity" or "snd") the sound is being recorded. However, when playing back and the playback is finished (either by pressing stop or the sound file is simply finished) the programs are crashing (they close). After this, there is no sound at all! And I could not get it back, even by playing around with YaST. Opening the recording software and trying to play back results then in an immediate crash. The program "snd" gave me fortunately at least this message: open read /dev/dsp0: Device or resource busy [audio.c[1956] linux_audio_open_with_error] With my limited knowledge in Linux I could only get around by rebooting the computer. During the shutdown the screen showed me the following information (I give just an excerpt): snd-mixer busy ... snd-* busy failed After rebooting the sound was there. (Just a remark: after crashing the audio it is not possible to restart the X-server by "ctrl-alt-backspace". The computer hangs, no GUI, and also to switch to any console is of no use. I only could brutally switch off the computer.) What is the reason that the device is busy? And how can I solve this problem? Thanks for any hints and comments Cordially Thomas From bil at ccrma.Stanford.EDU Fri Nov 5 03:32:29 2004 From: bil at ccrma.Stanford.EDU (Bill Schottstaedt) Date: Fri, 05 Nov 2004 03:32:29 -0800 Subject: [CM] snd - channel number during installation Message-ID: <418B64CD.4090106@ccrma> I added the variable "recorder-in-chans" which is currently just a number -- if 0 (the default), you get as many inputs as the recorder can find; otherwise it sets the number of displayed inputs (apparently some audio cards insist that you read 12 channels no matter what). It should be set before you start the recorder the first time, and not changed thereafter. (I'll add various features like a list of desired channels and so on once I've goofed around with this a bit). From howell at ucsc.edu Fri Nov 5 09:41:12 2004 From: howell at ucsc.edu (dave cope) Date: Fri, 5 Nov 2004 09:41:12 -0800 Subject: [CM] WACM Message-ID: Dear Colleague, This summer, for the third year, we'll be offering the Workshop in Algorithmic Computer Music (WACM) at UC Santa Cruz. Student evaluations from the past two years have been uniformly positive and we're excited about doing it again. One of the things the students benefit most from is limiting the workshop to fifteen students, which allows for a very high student/teacher ratio. But it also means that people applying later in the year may not get in. So I'm sending you this email to remind you about the workshop and to give you an opportunity to apply before our main publicity goes out. I won't barrage you with details here (a general blurb appears below), but if you're interested you can visit the website, which is at: http://summer.ucsc.edu/wacm ... and please don't hesitate to contact me if you have any questions. If you know anyone else who might benefit from the workshop, please feel free to forward this message. All best, Dave Cope Workshop in algorithmic computer music to be offered summer 2004 The first Workshop in Algorithmic Computer Music (WACM) will take place from June 25 through July 9, 2004 at the University of California, Santa Cruz (UCSC). Participants will learn the Lisp computer programming language and create their own composition and analysis software. The workshop will be held at UCSC's new Music Center, which features state of the art facilities as well as sweeping vistas of the Monterey Bay. David Cope, Paul Nauert, Peter Elsea, and Soren Goodman will be among those on hand to teach and advise workshop attendees. Participants will take classes on the basic techniques of algorithmic composition and algorithmic music analysis, learning and using the computer programming language Lisp. Participants will create three significant software projects: a Markov-based rules program, a genetic algorithm, and software modeled on the Experiments in Musical Intelligence program. Music analysis software and techniques will also be covered in depth. Many compositional approaches will be discussed in detail including rules-based techniques, data-driven models, genetic algorithms, neural networks, fuzzy logic, mathematical modeling, and sonification. Software programs such as Common Music, Max, Open Music, and so on will also be presented and used. The workshop offers 66 hours of class instruction and over 120 hours of individual consultation. Each participant will receive a free class reference manual, class software in source code, and other printed and software tools and guides relevant to the workshop's goals. Access to the world class computer music facilities at UCSC will allow performances and special lectures by invited guests. The workshop includes five daily hours of individual computer time (with an advisor close at hand). Each week will culminate in performances of participant works and presentations of analysis and composition software. By the workshop's end, each participant will have gained facility with Lisp, have written compositional and analytical software programs, and be presented with a CD of their compositions. Prospective students must have an ability to read music and understand basic music theory as well as have at least a basic facility with computers (text editing, etc.). Students need not be active musicians or programmers. Fee is $1750. Room and board is available on campus if desired. 5 units of credit is available. For more information or to download an application, please go to http://summer.ucsc.edu/wacm. -- All Experiments in Musical Intelligence scores are now available from http://www.spectrumpress.com Information on the third Workshop in Algorithmic Computer Music (WACM) is available from http://summer.ucsc.edu/wacm. And be sure to check out my website at http://arts.ucsc.edu/faculty/cope David Cope Department of Music University of California Santa Cruz, CA 95064 -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.s.matheussen at notam02.no Sun Nov 7 05:53:39 2004 From: k.s.matheussen at notam02.no (Kjetil Svalastog Matheussen) Date: Sun, 7 Nov 2004 14:53:39 +0100 (CET) Subject: [CM] Re: snd jack playback mode bug Message-ID: Tim Orford: >> > If you are running jack in Playback mode, snd 7.8 will >> > imediately segfault in mus_audio_read_buffers(). >> > >> > i assume that this is easily repeatable. I can provide >> > any additional information if needed. >> > >> >> Yes, please provide some code that use mus_audio_read_buffers. >> I wrote the jack-code, but have never used mus_audio_read_buffers. > >yes thanks for adding this v useful functionality! >I must apologise for the incorrect information regarding >mus_audio_read_buffers - that was indicated by gdb but appears to >be false. By adding printfs i beleive the segfault happens here: > > >static int sndjack_getnuminchannels(void){ > int lokke=0; > const char **ports=jack_get_ports(sndjack_client,"alsa_pcm:capture_*","",0); > > while(ports[lokke]!=NULL){ ////// ****appears to crash here with lokke=0 > lokke++; > } > if(lokke<2) return 2; > return lokke; >} Oh yes, thank you. jack_get_ports returns NULL when there are no ports. I'll fix it ASAP. Something like this is a quick fix: static int sndjack_getnuminchannels(void){ int lokke=0; const char **ports=jack_get_ports(sndjack_client,"alsa_pcm:capture_*","",0); if(ports==NULL) return 0; while(ports[lokke]!=NULL){ ////// ****appears to crash here with lokke=0 lokke++; } if(lokke<2) return 2; return lokke; } -- From k.s.matheussen at notam02.no Sun Nov 7 09:58:50 2004 From: k.s.matheussen at notam02.no (Kjetil Svalastog Matheussen) Date: Sun, 7 Nov 2004 18:58:50 +0100 (CET) Subject: [CM] Re: snd jack playback mode bug In-Reply-To: References: Message-ID: On Sun, 7 Nov 2004, Kjetil Svalastog Matheussen wrote: > > Oh yes, thank you. jack_get_ports returns NULL when there are no ports. > I'll fix it ASAP. > > Something like this is a quick fix: > > > static int sndjack_getnuminchannels(void){ > int lokke=0; > const char **ports=jack_get_ports(sndjack_client,"alsa_pcm:capture_*","",0); > > if(ports==NULL) return 0; > > while(ports[lokke]!=NULL){ ////// ****appears to crash here with lokke=0 > lokke++; > } > if(lokke<2) return 2; > return lokke; > } > Oops. That was not enough. Below and attached is a working patch. Thanks for your excellent bug-report Tim. Please apply Bill. --- audio.c~ 2004-11-07 12:47:56.000000000 +0100 +++ audio.c 2004-11-07 18:51:49.000000000 +0100 @@ -9023,7 +9023,7 @@ static int sndjack_getnumoutchannels(void){ int lokke=0; const char **ports=jack_get_ports(sndjack_client,"alsa_pcm:playback_*","",0); - while(ports[lokke]!=NULL){ + while(ports!=NULL && ports[lokke]!=NULL){ lokke++; } if(lokke<2) return 2; @@ -9033,7 +9033,7 @@ static int sndjack_getnuminchannels(void){ int lokke=0; const char **ports=jack_get_ports(sndjack_client,"alsa_pcm:capture_*","",0); - while(ports[lokke]!=NULL){ + while(ports!=NULL && ports[lokke]!=NULL){ lokke++; } if(lokke<2) return 2; @@ -9137,8 +9137,7 @@ ) ) { - fprintf (stderr, "Error. Cannot connect jack output port %d: \"%s\".\n",ch,temp); - goto failed_connect; + printf ("Warning. Cannot connect jack output port %d: \"%s\".\n",ch,temp); } } @@ -9153,8 +9152,7 @@ ) ) { - fprintf (stderr, "Error. Cannot connect jack input port %d: \"%s\".\n",ch,temp); - goto failed_connect; + printf ("Warning. Cannot connect jack input port %d: \"%s\".\n",ch,temp); } } -- -------------- next part -------------- --- audio.c~ 2004-11-07 12:47:56.000000000 +0100 +++ audio.c 2004-11-07 18:55:56.000000000 +0100 @@ -9023,7 +9023,7 @@ static int sndjack_getnumoutchannels(void){ int lokke=0; const char **ports=jack_get_ports(sndjack_client,"alsa_pcm:playback_*","",0); - while(ports[lokke]!=NULL){ + while(ports!=NULL && ports[lokke]!=NULL){ lokke++; } if(lokke<2) return 2; @@ -9033,7 +9033,7 @@ static int sndjack_getnuminchannels(void){ int lokke=0; const char **ports=jack_get_ports(sndjack_client,"alsa_pcm:capture_*","",0); - while(ports[lokke]!=NULL){ + while(ports!=NULL && ports[lokke]!=NULL){ lokke++; } if(lokke<2) return 2; @@ -9137,8 +9137,7 @@ ) ) { - fprintf (stderr, "Error. Cannot connect jack output port %d: \"%s\".\n",ch,temp); - goto failed_connect; + printf ("Warning. Cannot connect jack output port %d: \"%s\".\n",ch,temp); } } @@ -9153,8 +9152,7 @@ ) ) { - fprintf (stderr, "Error. Cannot connect jack input port %d: \"%s\".\n",ch,temp); - goto failed_connect; + printf ("Warning. Cannot connect jack input port %d: \"%s\".\n",ch,temp); } } From fd106 at york.ac.uk Tue Nov 9 08:22:57 2004 From: fd106 at york.ac.uk (Fergal Dowling) Date: Tue, 9 Nov 2004 16:22:57 +0000 Subject: [CM] osx-10.3-snd Message-ID: <9DC022DC-326B-11D9-A56C-00039367667A@york.ac.uk> Hi, I've downloaded osx-10.3-snd and when I type the command... [PowerBook:~] username% ./osx-10.3-snd I get the following message... dyld: ./osx-10.3-snd can't open library: /usr/local/lib/libintl.1.dylib (No such file or directory, errno = 2) Trace/BPT trap Any ideas? Thanks in advance, Fergal Dowling. From bigswift at ufl.edu Tue Nov 9 08:57:15 2004 From: bigswift at ufl.edu (shreeswifty) Date: Tue, 9 Nov 2004 11:57:15 -0500 (EST) Subject: [CM] osx-10.3-snd Message-ID: <1225217155.1100019435867.JavaMail.osg@osgjas04.cns.ufl.edu> where does one attain the file? On Tue Nov 09 11:22:57 EST 2004, Fergal Dowling wrote: > Hi, > I've downloaded osx-10.3-snd and when I type the command... > [PowerBook:~] username% ./osx-10.3-snd > > I get the following message... > dyld: ./osx-10.3-snd can't open library: > /usr/local/lib/libintl.1.dylib (No such file or directory, > errno = 2) > Trace/BPT trap > > Any ideas? > > Thanks in advance, > Fergal Dowling. > > _______________________________________________ > Cmdist mailing list > Cmdist at ccrma.stanford.edu > http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist > > Patrick Pagano, B.S.,M.F.A Candidate Research And Development Assistant Digital Worlds Institute University Of Florida (352) 294-2070 From taube at uiuc.edu Tue Nov 9 09:46:11 2004 From: taube at uiuc.edu (Rick Taube) Date: Tue, 9 Nov 2004 11:46:11 -0600 Subject: [CM] osx-10.3-snd In-Reply-To: <1225217155.1100019435867.JavaMail.osg@osgjas04.cns.ufl.edu> References: <1225217155.1100019435867.JavaMail.osg@osgjas04.cns.ufl.edu> Message-ID: <3E3A534E-3277-11D9-8419-000A95CA28FC@uiuc.edu> >> Hi, >> I've downloaded osx-10.3-snd and when I type the command... >> [PowerBook:~] username% ./osx-10.3-snd >> I get the following message... >> dyld: ./osx-10.3-snd can't open library: >> /usr/local/lib/libintl.1.dylib (No such file or directory, errno = >> 2) Did you install X11 and all the libraries it needs? you must also have the x11 server running before you try to start snd. if the snd binary uses gtk then you will also need those libs from Fink: $ pkg-config --libs gtk+-2.0 -L/sw/lib -lgtk-x11-2.0 -lgdk-x11-2.0 -latk-1.0 -lgdk_pixbuf-2.0 -lm -lpangoxft-1.0 -lpangox-1.0 -lpango-1.0 -lgobject-2.0 -lgmodule-2.0 -lglib-2.0 -lintl -liconv From fd106 at york.ac.uk Tue Nov 9 10:17:45 2004 From: fd106 at york.ac.uk (Fergal Dowling) Date: Tue, 9 Nov 2004 18:17:45 +0000 Subject: [CM] osx-10.3-snd In-Reply-To: <3E3A534E-3277-11D9-8419-000A95CA28FC@uiuc.edu> References: <1225217155.1100019435867.JavaMail.osg@osgjas04.cns.ufl.edu> <3E3A534E-3277-11D9-8419-000A95CA28FC@uiuc.edu> Message-ID: Thanks, It's working fine now. The problem seems to be a result of Fink installing libiconv and guile libraries in /sw/lib instead of /usr/local/lib Fergal. On 9 Nov 2004, at 17:46, Rick Taube wrote: > >>> Hi, >>> I've downloaded osx-10.3-snd and when I type the command... >>> [PowerBook:~] username% ./osx-10.3-snd >>> I get the following message... >>> dyld: ./osx-10.3-snd can't open library: >>> /usr/local/lib/libintl.1.dylib (No such file or directory, errno = >>> 2) > > Did you install X11 and all the libraries it needs? you must also > have the x11 server running before you try to start snd. if the snd > binary uses gtk then you will also need those libs from Fink: > > $ pkg-config --libs gtk+-2.0 > -L/sw/lib -lgtk-x11-2.0 -lgdk-x11-2.0 -latk-1.0 -lgdk_pixbuf-2.0 -lm > -lpangoxft-1.0 -lpangox-1.0 -lpango-1.0 -lgobject-2.0 -lgmodule-2.0 > -lglib-2.0 -lintl -liconv > From rsegnini at ccrma.Stanford.EDU Thu Nov 11 09:21:12 2004 From: rsegnini at ccrma.Stanford.EDU (Rodrigo Segnini) Date: Thu, 11 Nov 2004 09:21:12 -0800 (PST) Subject: [CM] no gui snd & multichannel playback In-Reply-To: <1097516295.2535.13.camel@linux.LUTEKGROUP.lan> Message-ID: Hi all, two questions: * no gui I recall you can compile snd w/o a gui. But can you, once compiled with one, launch it without? * multichannel playback (having only 2 out) Snd has been helping me playback n-channels files on my laptop; each time it would say something like: n-channels file, folding into stereo. Today I started tweaking jack through qjackctl to get less xrun errors, and now snd complains: error, n-channel file, only 2 available. I have tried to return to default values to no avail. Any help is appreciated From rsegnini at ccrma.Stanford.EDU Thu Nov 11 09:33:49 2004 From: rsegnini at ccrma.Stanford.EDU (Rodrigo Segnini) Date: Thu, 11 Nov 2004 09:33:49 -0800 (PST) Subject: [CM] no gui snd & multichannel playback (add) In-Reply-To: Message-ID: Perhaps I should have included the following info: Thanks in advance for your responses. R. >snd --help Snd is a sound editor; see http://ccrma.stanford.edu/software/snd/.This is Snd version 7.5 of 19-May-04: Xen: 1.11, Guile: 1.6.4 jack Sndlib 19.1 (27-Apr-04, int24 samples) CLM 2.48 (11-May-04) fftw-3.0.1 Motif 2.2.3 X11R6 OpenGL Xpm 3.4.11 LADSPA 1.1 with large file support with gettext: ja_JP Compiled Jun 1 2004 18:41:53 C: 3.2.2 20030222 (Red Hat Linux 3.2.2-5) Libc: 2.3.2.stable configured via: ./configure --prefix=/usr --with-guile=yes --with-run=yes --with-jack --with-static-xm=yes --with-alsa=yes --with-ladspa --with-gl On Thu, 11 Nov 2004, Rodrigo Segnini wrote: > Hi all, two questions: > > * no gui > I recall you can compile snd w/o a gui. > But can you, once compiled with one, launch it without? > > * multichannel playback (having only 2 out) > Snd has been helping me playback n-channels files on my laptop; each time > it would say something like: n-channels file, folding into stereo. > Today I started tweaking jack through qjackctl to get less xrun errors, > and now snd complains: error, n-channel file, only 2 available. > > I have tried to return to default values to no avail. > > Any help is appreciated > > _______________________________________________ > Cmdist mailing list > Cmdist at ccrma.stanford.edu > http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist > From rsegnini at ccrma.Stanford.EDU Thu Nov 11 09:56:57 2004 From: rsegnini at ccrma.Stanford.EDU (Rodrigo Segnini) Date: Thu, 11 Nov 2004 09:56:57 -0800 (PST) Subject: [CM] compositions made with tools from this list In-Reply-To: Message-ID: Hi all, I am wondering if anyone has compiled a list of pieces composed -for the very most part- using the tools discussed on this list. I have in fact, done pieces where sounds and other structural elements where done from code, but in whose formal layout a standard DAW app. like ProTools or Logic intervened. The intervention reflected more my relative proficiency in handling finishing aspects (final mastering, notation, media packaging -cd,multitrack tape-) through those apps, than limitations (specially nowadays) of c{m,lm,mn}+snd (potentially recruiting help from other linux scriptable apps). I am interested in presenting examples from fully fledged compositions to an -more musically than technically experienced- audience interested in learning about c{m,lm,mn}+snd. Unless this information already exists, I would like to compile it for the benefit of all. Depending on the amount and type of responses, we can have a somewhat fancy repository, with sound and code links, descriptions, or I would just collate back to the list your responses. Thanks, Rodrigo rsegnini at ccrma.stanford.edu From juanig at ccrma.Stanford.EDU Thu Nov 11 12:39:27 2004 From: juanig at ccrma.Stanford.EDU (Juan Reyes) Date: Thu, 11 Nov 2004 15:39:27 -0500 Subject: [CM] compositions made with tools from this list In-Reply-To: References: Message-ID: <1100205567.5883.49.camel@strawberri.maginvent.org> I consider Snd a great composition tool (I am afraid to say ). I have used it on my last compositions with no other editors involved. CM+CLM+CMN combination still a favorite. I remember few years ago before Rick or Mary Simoni were writing their books on the subject, the issue of a "compositions compilation" was brought. That list might somewhere around if I am not mistaken. --* Juan On Thu, 2004-11-11 at 12:56, Rodrigo Segnini wrote: > Hi all, > > I am wondering if anyone has compiled a list of pieces composed -for the > very most part- using the tools discussed on this list. > > I have in fact, done pieces where sounds and other structural elements > where done from code, but in whose formal layout a standard DAW app. like > ProTools or Logic intervened. > > The intervention reflected more my relative proficiency in handling > finishing aspects (final mastering, notation, media packaging > -cd,multitrack tape-) through those apps, than limitations (specially > nowadays) of c{m,lm,mn}+snd (potentially recruiting help from other linux > scriptable apps). > From taube at uiuc.edu Thu Nov 11 15:43:11 2004 From: taube at uiuc.edu (Rick Taube) Date: Thu, 11 Nov 2004 17:43:11 -0600 Subject: [CM] compositions made with tools from this list In-Reply-To: References: Message-ID: <72ED405F-343B-11D9-93D9-000A95CA28FC@uiuc.edu> > Unless this information already exists, I would like to compile it for > the > benefit of all. Depending on the amount and type of responses, we can > have > a somewhat fancy repository, with sound and code links, descriptions, > or I > would just collate back to the list your responses. I had exactly the same idea several years ago and in fact as part of my tenure documentation I did compile a sizable list of pieces. Unfortunately, once I got tenure I didn't continue to pursue the project :/ and so I'm afraid that I'm to blame for the current lack of a repository . Its still a great idea though and I would contribute recordings of pieces and source code if you establish one. It would also be a good place to make instrument samples available and so on. Im sure I can dig up the list of pieces but since its several years old its probably a better idea for you to make an explicit call for pieces again. If you do Ill certainly send in all the stuff I have. -rick From bil at ccrma.Stanford.EDU Fri Nov 12 04:32:15 2004 From: bil at ccrma.Stanford.EDU (Bill Schottstaedt) Date: Fri, 12 Nov 2004 04:32:15 -0800 Subject: [CM] no gui snd & multichannel playback In-Reply-To: References: Message-ID: <4194AD4F.9050101@ccrma> > But can you, once compiled with one, launch it without? Use the -b or -batch switch (grfsnd.html). From howell at ucsc.edu Tue Nov 16 09:53:39 2004 From: howell at ucsc.edu (dave cope) Date: Tue, 16 Nov 2004 09:53:39 -0800 Subject: [CM] new DMA in computer music Message-ID: Dear Colleague: We are pleased to announce that the University of California at Santa Cruz now offers a new D.M.A. in computer-assisted composition and in world music composition. The newly established Doctorate of Musical Arts Degree (D.M.A.) in Music Composition has tracks in computer-assisted composition and in world music composition. The track in computer-assisted composition includes algorithmic techniques for the generation of musical materials and structures to be realized in the creation of instrumental, vocal, and digitally synthesized music. The track in world music composition addresses a variety of compositional approaches influenced by indigenous non-Western musics, with a focus upon those musics taught by faculty composers, ethnomusicologists, and applied instructors. The D.M.A. program seeks to develop accomplished, active, and articulate composers with a broad awareness of the diverse styles, cultural influences, media, venues, and technical means available to them in the 21st century. Applications will be accepted in fall 2004 for entrance to the program in fall 2005. More information can be accessed at the web site http://music.ucsc.edu/grad The deadline for receipt of applications is February 1. Please contact Yalenda Listmann at yalenda at ucsc.edu or (831) 459-3199 for additional information about the program. -- All Experiments in Musical Intelligence scores are now available from http://www.spectrumpress.com Information on the third Workshop in Algorithmic Computer Music (WACM) is available from http://summer.ucsc.edu/wacm. And be sure to check out my website at http://arts.ucsc.edu/faculty/cope David Cope Department of Music University of California Santa Cruz, CA 95064 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bil at ccrma.Stanford.EDU Wed Nov 17 06:06:42 2004 From: bil at ccrma.Stanford.EDU (Bill Schottstaedt) Date: Wed, 17 Nov 2004 06:06:42 -0800 Subject: [CM] snd 7.8 Message-ID: <419B5AF2.9060101@ccrma> Snd 7.8: new version of gfm package (sndlib/gfm) thanks to Mike Scholz, now included (along with sndins) in the Snd tarball. also many other improvements various bugfixes and improvements thanks to Kjetil S. Matheussen added: recorder-in-chans vct->string normalize-channel controls->channel sound-data->sound-data (for oscope) rms-envelope in env.scm (based on Bret Battey's rmsenv.ins) edit history pane popup menu (popup.scm and gtk-popup.scm) (unfinished) oscope.scm: oscilloscope dialog removed: track-id arg to mix-selection edpos arg to play-selection mus_audio_save|restore (audio.c) and Scheme/Ruby correlates the dumb buttons at the bottom of the controls (Apply etc) these are now in various popup menus moved scale-sound-by and scale-sound-to to snd7.scm. changed handling of recorder-out-chans -- it now sets the max out chans displayed in the recorder dialog, as well as the default output file chans. checked: gtk 2.5.4|5, sbcl 0.8.16, ruby 1.8.2-preview3 with much help from: Mike Scholz, Heinrich Gudenus, Kjetil S. Matheussen, Dave Phillips, Michal Seta, Thomas F. Bechteler, Frank Barknecht, Peter Lutek, Tim Orford, Juan Reyes From joshp at u.washington.edu Wed Nov 17 09:43:36 2004 From: joshp at u.washington.edu (Joshua Parmenter) Date: Wed, 17 Nov 2004 09:43:36 -0800 Subject: [CM] problem with defobject? Message-ID: <351F26AE-38C0-11D9-BED7-000393ACA176@u.washington.edu> Hi everyone, I'm sorry if this has been discussed, but it seems like there is a problem with defobject in CM 2.4.2, using openmcl on MacOS X. Here is the code example with CM's output: (defobject ex3 (i) (freq) (:parameters ins time dur freq)) (new ex3 :freq 440) ;; returns this ? ? #i(ex3 freq 440) ;; However, if I try to re-define the object with a change ;; (just a simple example) (defobject ex3 (i) (freq amp) (:parameters ins time dur freq amp)) (new ex3 :freq 440 :amp 1) ;; this error is returned ? ? > Error in process listener(2): :AMP is an invalid initarg to INITIALIZE-INSTANCE for #. > Valid initargs: #(:TIME :INS :DUR :FREQ). > While executing: CCL::CHECK-INITARGS > Type :POP to abort. Type :? for other options. 1 > Any ideas why re-evaluating the defobject doesn't seem to overwrite the previous one? Thanks in advance, Josh From taube at uiuc.edu Wed Nov 17 11:43:45 2004 From: taube at uiuc.edu (Rick Taube) Date: Wed, 17 Nov 2004 13:43:45 -0600 Subject: [CM] problem with defobject? In-Reply-To: <351F26AE-38C0-11D9-BED7-000393ACA176@u.washington.edu> References: <351F26AE-38C0-11D9-BED7-000393ACA176@u.washington.edu> Message-ID: this is a bug, thanks for the clear example -- ill take a look at it. -rick > Any ideas why re-evaluating the defobject doesn't seem to overwrite > the previous one? > > Thanks in advance, > > Josh > > _______________________________________________ > Cmdist mailing list > Cmdist at ccrma.stanford.edu > http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist From joshp at u.washington.edu Wed Nov 17 11:46:51 2004 From: joshp at u.washington.edu (Joshua Parmenter) Date: Wed, 17 Nov 2004 11:46:51 -0800 Subject: [CM] problem with defobject? In-Reply-To: References: <351F26AE-38C0-11D9-BED7-000393ACA176@u.washington.edu> Message-ID: <6D406223-38D1-11D9-9AC4-000D933431A0@u.washington.edu> Thanks Rick, Josh ****************************************** Joshua D. Parmenter Graduate Student, Music Composition http://students.washington.edu/joshp "...Some people think a composer's supposed to please them, but in a way a composer is a chronicler... He's supposed to report on what he's seen and lived." -Charles Mingus On Nov 17, 2004, at 11:43 AM, Rick Taube wrote: > this is a bug, thanks for the clear example -- ill take a look at it. > -rick > >> Any ideas why re-evaluating the defobject doesn't seem to overwrite >> the previous one? >> >> Thanks in advance, >> >> Josh >> >> _______________________________________________ >> Cmdist mailing list >> Cmdist at ccrma.stanford.edu >> http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist > From taube at uiuc.edu Tue Nov 23 07:45:30 2004 From: taube at uiuc.edu (Rick Taube) Date: Tue, 23 Nov 2004 09:45:30 -0600 Subject: [CM] Plotter GUI on Linux available Message-ID: The Plotter GUI is now available for x86/Linux (CMU19a and SBCL 0.8.16) in addition to Darwin/OpenMCL. The latest sources also include plot editing, region selection, conditional selection, bug fixes and some other stuff. The code is now integrated into CM's head CVS branch and should build "out of the box" on properly configured systems. However, if you use SBCL or OS X then there may be a bit of pre-installation depending on your system. On the mac it is possible to open a plotter window from inside Emacs.app or Terminal.app in addition to xterm and xemacs. For more information on plotter configurations see: http://pinhead.music.uiuc.edu/~hkt/cm/doc/install.html#configure_plotter CM's new install.html also contains an expanded explanation of how to get the sources using CVS, see: http://pinhead.music.uiuc.edu/~hkt/cm/doc/install.html#restore_cvs After downloading the source, type 'cm.sh -l cmucl' or whatever and then use my temporary (foo) funcion to open a test display. You might consider reading the updated main plotter documentation in the dictionary before attempting any of the examples (yeah, right): http://pinhead.music.uiuc.edu/~hkt/cm/doc/dict/plotter-topic.html I think the next main task(s) will be to add lots of sequencing commands so that it can be a general tool for editing/mixing music, sort of a gui version of the old Stella commad interpreter. Copy/paste and undo/redo would be nice too. A big thanks to Dave Phillips, Anders Vinjar, Tobais Kunze and Michael Klilngbeil for testing out Plotter on their various systems. Please let me know of any problems, bugs, misfeatures, or ideas for new features. --Rick From taube at uiuc.edu Tue Nov 30 07:59:41 2004 From: taube at uiuc.edu (Rick Taube) Date: Tue, 30 Nov 2004 09:59:41 -0600 Subject: [CM] gtk breakout package Message-ID: ive taken the support code that cm uses for gtk display and made it available as a completely separate package released under the llgpl. the homepage for the code is: http://common-lisp.net/project/lambda-gtk/ --rick From shulmang at colorado.edu Tue Nov 30 08:17:18 2004 From: shulmang at colorado.edu (Garett Shulman) Date: Tue, 30 Nov 2004 09:17:18 -0700 Subject: [CM] SND specifying audio devices Message-ID: <41AC9D0E.2060708@colorado.edu> Hello, I am trying to use snd with an maudio 66 which has 10 channels of I/O. This causes the record window to extend well below my monitor (which is running at 1280x1024). I have a plug device defined in my asoundrc named 66_12 which is a 2 channel subset of the I/Os on my audio hardware. I have been trying: export SNDLIB_ALSA_PLAYBACK_DEVICE=66_12 and: export SNDLIB_ALSA_RECORD_DEVICE=66_12 However, the record window still includes the full 10 channels for input and output. I noticed that you can adjust the size of the meters. However, I am not sure this will solve the problem as the sliders for all the channels also take up a significant amount of screen realestate. Questions: Are environment variables the best way to specifiy an alsa device for audio I/O? Is there a list of environment variables that snd recognizes documented anywhere? Is there another mechanism to limit the channels that snd uses for audio I/O? Thanks for any suggestions that anyone might have. -Garett BTW, snd seems very cool: an audio editor based on emacs that you can extend with lisp and have to recompile to configure... it's so geeky. ;) -------------- next part -------------- An embedded message was scrubbed... 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