Linux-audio-user January 2004

linux-audio-user@lists.linuxaudio.org

148 participants
168 discussions

[linux-audio-user] Re: [linux-audio-dev] [ANN] new LADSPA plugins

by Frank Barknecht

Hallo, Tom Szilagyi hat gesagt: // Tom Szilagyi wrote: > On Sun, 25 Jan 2004, Frank Barknecht wrote: > > I cannot hear it. :( I'm testing the reverb in Pd but although I crank > > up decay times and played with dry/wet I cannot hear a reverb. I can > > hear a bit of (seemingly high pass) filtering, but no real > > reverbaration at all. I should hear a hall when selecting say "Smooth > > Hall", shouldn't I? Changing presets seems to work, as the filtering > > gets different when changing presets, but here I wouldn't call this a > > reverb. Maybe someone could provide a small ogg which shows what I > > should hear? > > Is it possible that the comb and/or allpass filters are turned off? I > don't know how Pd displays plugins (i don't know Pd at all, maybe i should take > a look) but there should be 4 toggle buttons somewhere saying "Comb > Filters", "Allpass Filters", "Bandpass Filter", "Enhanced Stereo". > If all buttons are on, i don't have an idea right now, but think, think... In Pd, as it cannot use controls with spaces, you send messages like "control #1 4" to set the first control to 4. I did set the Filter controls this way to several values like 13 or 17. Should this turn them on? BTW: The other TAPs are working fine, http://footils.org/snd/echos.ogg is a pluck playing through the stereo echo in Pd. (It's a bit big, 7.3M, currently uploading.) ciao -- Frank Barknecht _ ______footils.org__

20 years, 5 months

[linux-audio-user] Echo cancelling utility

by Robert Rozman

Hi, does anyone know of any free or open source echo cancellation software... Regards, Robert.

20 years, 5 months

[linux-audio-user] Re: [linux-audio-dev] [ANN] new LADSPA plugins

by Frank Barknecht

Hallo, Tom Szilagyi hat gesagt: // Tom Szilagyi wrote: > * TAP Reverberator > Actually no less than 38 reverberator effects, > ranging from Afterburn to Warehouse, including > small/medium/large rooms, halls, plates... > and more! I cannot hear it. :( I'm testing the reverb in Pd but although I crank up decay times and played with dry/wet I cannot hear a reverb. I can hear a bit of (seemingly high pass) filtering, but no real reverbaration at all. I should hear a hall when selecting say "Smooth Hall", shouldn't I? Changing presets seems to work, as the filtering gets different when changing presets, but here I wouldn't call this a reverb. Maybe someone could provide a small ogg which shows what I should hear? ciao -- Frank Barknecht _ ______footils.org__

20 years, 5 months

[linux-audio-user] Recording Internet Stream, Duplex, Help!?

by Ed

Hi LAU, OK, after 4 hours on this and enough Googling to break my keyboard, I have to admit defeat. What I Want To Do: 1. Load a web page containing a Flash Film 2. Play the Flash Film 3. Record any sound the Flash Film delivers to a WAV file. 4. Maybe post-process the WAV into Ogg, MP3, etc. What I Have (Hardware + Software): 1. SB Audigy 2. Alsa kernel drivers, 1.0.0 RC2 3. Alsa libs, 1.0.0 RC2 4. Jack, compiled with the above Alsa libs 5. Ecasound, compiled with the Alsa libs and Jack What works: 1. Sound from CD, System beeps, etc. works What doesn't work: 1. Alsa: arecord -f cd -o test.wav, when playing an Ogg file, e.g. 2. Ecasound: ecasound -i:alsahw,0 -o test.wav, when playing an Ogg file, e.g. Am I missing something fundamental here? I can't imagine that this is an especially arcane wish, simply wanting to record and play at the same time, yet I can't find anything on this which helps. I have played with alsamixer for about an hour, but I simply get silence all the time. All the 'capture' devices are things like Mic, CD, Line-In, etc., which is, obviously, NOT what I want. I simply want to record what I am currently hearing -- although I'm testing this with playback of Ogg, actually I clearly want to record the sound from the Flash Film mentioned. I have read the various docs for Alsa, Ecasound and Jack, but I have to admit that sound on Linux is *very* confusing. I am not really sure whether I'm supposed to be using /dev/dsp, alsahw, or some other arcane terminologies -- kernel compilation is easy in comparison! I'm not getting any errors, by the way, everything *seems* to work, but when I stop playback and check the output, all I hear is dead air. I can (joyfully!) send the output of amixer if that would help, hope to get a reply on this one soon. In Windows, this is a 5-minute no-brainer. Install Total Recorder, and that's that. I can't imagine that it must be so complex in Linux? First time I've really hit a wall in Linux -- which clearly shows that sound is something which needs to be clarified a *LOT* in Linux. (Having alsamixer start with everything mute is a good example of this!) Thanks a million if anyone can help: otherwise I'm rebooting into Windows for now, since there's no other way of doing this at the moment. Ed

20 years, 5 months

[linux-audio-user] alsa sequencer filter client

by Garett Shulman

Hello, I am trying to create a very simple midi filter client for the alsa sequencer based on aseqview-0.1.4. I have alsa 0.9.8. This code shows the input and the output port in aconnect when executed. And, when the raw_midi client is connected The callback does get called when I play keys on my piano. However, Its as though the message is getting sent back to this client and not on to the next client. The process_event function keeps getting called over and over again with the same message when I hit one piano key. It's as though it is sending itself the message. Any Ideas are greatly appreciated. -Garett #include "portlib.h" int process_event(port_t *p, int type, snd_seq_event_t *ev, int *priate_data) { port_write_event(p, ev, 0); } int main() { unsigned int caps = SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_SUBS_READ; port_client_t *client = port_client_new("MIDI filter", SND_SEQ_OPEN_DUPLEX); port_t *port = port_attach(client, "FILTER port", caps, SND_SEQ_PORT_TYPE_MIDI_GENERIC); int *priv_data; /* add callback */ port_add_callback(port, PORT_MIDI_EVENT_CB, (port_callback_t)process_event, priv_data); port_client_do_loop(client); return port_detach(port); }

20 years, 5 months

[linux-audio-user] ecasound ???'s

by derek holzer

Hi all, I will be setting up an audio Linux box this spring for a museum exhibition that will run for about 4 weeks. It will probably run without an xserver and use Pure Data in --nogui mode since a GUI is not needed, and may turn itself on and off with a chronjob function. My question is this: since I want to document the sound of the exhibition, I would like ecasound to record from all four outputs periodically [also as a chronjob]. I am not too familiar with ecasound, but it seems like exactly the kind of low-overhead tool to do the job. So can anyone tell me how to capture from *outputs* and not inputs? thanks, Derek -- derek holzer ::: http://www.umatic.nl ---Oblique Strategy # 4: "Abandon desire"

20 years, 5 months

[linux-audio-user] Can't install a noteedit dependency

by Glenn McCord

One of the dependencies for noteedit is tse3 but during 'make' it cans out with this error: /usr/include/bits/stdio.h: In function `__ssize_t getline(char**, size_t*, FILE*)': /usr/include/bits/stdio.h:103: declaration of `__ssize_t getline(char**, size_t*, FILE*) throw ()' throws different exceptions /usr/lib/gcc-lib/i686-pc-linux-gnu/3.2.3/include/stdio.h:583: than previous declaration `__ssize_t getline(char**, size_t*, FILE*)' make[4]: *** [Demidify.lo] Error 1 make[4]: Leaving directory `/home/glenn/apps/tse3-0.2.7/src/tse3/util' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/glenn/apps/tse3-0.2.7/src/tse3' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/glenn/apps/tse3-0.2.7/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/glenn/apps/tse3-0.2.7' make: *** [all] Error 2 Does anyone hava an ideas or should I just dig up a binary.

20 years, 5 months

[linux-audio-user] jack compiling problems with kernel 2.6

by Alexander Klosch

hi, I'm running gentoo/ppc on ibook rev 2.2 kernel 2.6.1_rc3 [including snd-powermac] and alsa-lib 1.0.1 when i compile jack I get following errors: [snip] checking for gethostent... yes checking for setsockopt... yes checking for connect... yes checking for on_exit... yes checking for atexit... yes JACK will use System V shm API (shmget() and friends) checking for capgetp in -lcap... no configure: WARNING: *** no libcap present configure: WARNING: *** required program md5sum not found configure: error: *** Capabilities support not present. Run configure again without --enable-capabilities. . but alsa is there and gives sound, no problems there. how to solve this? regards alex

20 years, 5 months

[linux-audio-user] Jack and SHM

by Christophe Vescovi

Since there has been a lot of post concerning jack usage and optimisation recently it has remind me a problem I got that I never been able to understand. I have never been able to run jackstart with -R option when compiling jack with Sys V shm support. jackd -R as root work with SysV shm but if I want to use jackstart I have to use POSIX shm. The error message I am getting with SysV shm is the classical "cannot lock down memory for RT thread (Cannot allocate memory)". I am running a Mandrake 9.2 multimedia kernel (2.4.22, low latency, preemption and capability patches). Since most of jack users seem to use SysV shm I don't understand why I cannot (I am the only exception ....) Christophe

20 years, 5 months

[linux-audio-user] 2nd LAD conference

by Matthias Nagorni

Hello, as announced earlier, the 2nd Linux Audio Developers conference will take place from 29. April to 2. May 2004 at ZKM (Center for Art and Media), Karlsruhe, Germany. In addition to the "Call for Papers" there is also a "Call for Music". Both are open until 29. February. We are looking for music that has been done with Linux Audio software and intend to present it in a concert at the "Kubus" concert hall of ZKM. "Live" performance with Linux Audio software is welcome as well. More information on the conference, registration, accomodation and the "Call for Music" is available from www.zkm.de/lad The current list of speakers (in alphabetical order) and talks is as follows: Fons Adriaensen, 1) AEOLUS - a church organ in your PC 2) Using JAAA for audio measurements Frank Barknecht, Rapid and Reuseable Audio Development With Pd Julien P. Claassen, Linux console - a textbased studio Paul Davis, Adding MIDI to a Digital Audio Workstation Orm Finnendahl, 1) Linux as a Workstation for Composers 2) PD Workshop Bob Ham, LADCCA Steve W. Harris and Jörn Nettingsmeier, Audio Engineering in a Nutshell Steve W. Harris, JACK, JAMin, Mastering Takashi Iwai, ALSA Sucks? - Trouble Shooting for Your Healthy Music Life Jaroslav Kysela, ALSA project - the last year Victor Lazzarini, Developing Spectral Signal Processing Applications Matthias Nagorni, Introduction to Modular Synthesis Han-Wen Nienhuys, LilyPond, a modular, extensible and programmable compiler for high-quality music notation Dave Phillips, 1) Keynote: One Year After: Notable Linux Audio Development in 2003 2) Still Nailing Jelly: An Update on Linux Audio Documentation Projects Abstracts and links to the respective projects are available at www.zkm.de/lad Matthias -- Dr. Matthias Nagorni SuSE Linux AG Maxfeldstr. 5 phone: +49 911 74053375 D - 90409 Nuernberg fax : +49 911 74053483

20 years, 5 months

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Linux-audio-user January 2004