Linux-audio-user December 2002

linux-audio-user@lists.linuxaudio.org

109 participants
111 discussions

[linux-audio-user] New relase of ZynAddSubFX (1.0.3)

by Nasca Paul

ZynAddSubFX is a opensource software synthesizer for Linux. It is available at : http://zynaddsubfx.sourceforge.net or http://sourceforge.net/projects/zynaddsubfx News: 1.0.3 - small bugfixes: "Bypass Global Filter" from ADnoteUI dissapears sometimes ; removed the low amplitude noise produced by the reverb; if you "acconect" zynaddsubfx with aseqview no note was processed a long time. - added Notch Filter - added the option to randomize the ressonance function - added VU-Meter - Change the Insertion effect modes behaves (it sounds a bit louder) - Added to the project an external program called Spliter that splits the keyboard and alows you to play two instruments same time. You can use this program with ZynAddSubFX or any other synthesizer. - Added a new function to OscilGen Paul. __________________________________________________ Do you Yahoo!? Yahoo! Mail Plus - Powerful. Affordable. Sign up now. http://mailplus.yahoo.com

21 years, 12 months

[linux-audio-user] once in a blue moon usb-audio

by iriXx

ok this is getting weird. usb audio decided to run the other day when i rebooted into 2.4.18 kernel (from mandrake 8.2 install... the one that used to work fine). i thought i'd recompiled alsa for that kernel by mistake. but no - it turns out i've recompiled for both 2.4.18 and 19. sooo... why did it work a couple of days ago and wont work yesterday and today????? the modules *are* loading... this is output of lsmod: snd-seq-oss 25376 0 (unused) snd-seq-midi-event 3400 0 [snd-seq-oss] snd-seq 37388 2 [snd-seq-oss snd-seq-midi-event] snd-pcm-oss 38756 0 (unused) snd-mixer-oss 10496 0 [snd-pcm-oss] snd-usb-audio 34208 0 (unused) snd-pcm 56928 0 [snd-pcm-oss snd-usb-audio] snd-timer 9952 0 [snd-seq snd-pcm] usbcore 59072 0 [snd-usb-audio] snd-rawmidi 12224 0 [snd-usb-audio] snd-seq-device 4000 0 [snd-seq-oss snd-seq snd-rawmidi] snd 26728 0 [snd-seq-oss snd-seq-midi-event snd-seq snd-pc m-oss snd-mixer-oss snd-usb-audio snd-pcm snd-timer snd-rawmidi snd-seq-device] soundcore 4068 2 [snd] i have these devices: [root@hamish miriam]# ls -l /dev/dsp* lr-xr-xr-x 1 root root 9 Dec 22 19:43 /dev/dsp -> /dev/dsp0 crwxrwxrwx 1 miriam audio 14, 3 Dec 22 19:43 /dev/dsp0 crw------- 1 miriam audio 14, 19 Dec 22 19:43 /dev/dsp1 crw------- 1 miriam audio 14, 35 Dec 22 19:43 /dev/dsp2 crw------- 1 miriam audio 14, 51 Dec 22 19:43 /dev/dsp3 [root@hamish miriam]# ls -l /dev/s* crwxrwxrwx 1 miriam audio 14, 1 Dec 22 19:43 /dev/sequencer lr-xr-xr-x 1 root root 10 Dec 22 19:43 /dev/sequencer2 -> /dev/music lr-xr-xr-x 1 root root 4 Dec 22 19:43 /dev/stderr -> fd/2 lr-xr-xr-x 1 root root 4 Dec 22 19:43 /dev/stdin -> fd/0 lr-xr-xr-x 1 root root 4 Dec 22 19:43 /dev/stdout -> fd/1 /dev/scsi: total 0 /dev/shm: total 0 /dev/snd: total 0 crw------- 1 miriam audio 116, 0 Jan 1 1970 controlC0 crw------- 1 miriam audio 116, 32 Jan 1 1970 controlC1 crw------- 1 miriam audio 116, 64 Jan 1 1970 controlC2 crw------- 1 miriam audio 116, 96 Jan 1 1970 controlC3 crw------- 1 miriam audio 116, 128 Jan 1 1970 controlC4 crw------- 1 miriam audio 116, 160 Jan 1 1970 controlC5 crw------- 1 miriam audio 116, 192 Jan 1 1970 controlC6 crw------- 1 miriam audio 116, 224 Jan 1 1970 controlC7 lr-xr-xr-x 1 root root 16 Dec 22 19:50 dev -> /proc/asound/dev/ crw------- 1 miriam audio 116, 1 Jan 1 1970 seq crw------- 1 miriam audio 116, 33 Jan 1 1970 timer /dev/sound: total 0 crw------- 1 miriam audio 14, 1 Jan 1 1970 sequencer crw------- 1 miriam audio 14, 8 Jan 1 1970 sequencer2 so whats going on???????? -- iriXx www.iriXx.org copyleft: creativity, technology and freedom? info(a)copyleftmedia.org.uk www.copyleftmedia.org.uk _ ( ) ascii ribbon against html email X / \ cat /dev/sda1 > /dev/dsp *** stopping make sense ***

21 years, 12 months

[linux-audio-user] Getting SPDIFF Output to work

by bosr＠xs4all.nl

Hi, I'm trying to get my digital output (SPDIFF) to work. I have a Creative SB Audigy. I managed to get the digital output to work with the OpenSource emu10k1 drivers at sourceforge (http://sourceforge.net/projects/emu10k1/), but I'm trying to switch to ALSA drivers, because the documentation with mplayer suggested switching to them for AC3 passthrough functionality. So, I built the ALSA stuff from CVS, and got the analog part to work fine. However, the digital output is not working... Tweaking with the alsamixer settings does not seem to help. Currently, the XMMS output plugins I've tried to use are both Esound Output Plugin 1.2.7 and the OSS Driver 1.2.7 (the former of which worked fine to produce the digital output with the emu10k1 driver). Does anyone have any idea how to get this working? This is the sound portion of lsmod: Module Size Used by Tainted: P snd-pcm-oss 43492 1 (autoclean) snd-mixer-oss 15288 1 (autoclean) [snd-pcm-oss] snd-emu10k1 79920 2 (autoclean) snd-pcm 81472 0 (autoclean) [snd-pcm-oss snd-emu10k1] snd-timer 15272 0 (autoclean) [snd-pcm] snd-util-mem 3096 0 (autoclean) [snd-emu10k1] snd-hwdep 5760 0 (autoclean) [snd-emu10k1] snd-rawmidi 18592 0 (autoclean) [snd-emu10k1] snd-seq-device 6252 0 (autoclean) [snd-emu10k1 snd-rawmidi] snd-ac97-codec 37896 0 (autoclean) [snd-emu10k1] snd 42572 0 (autoclean) [snd-pcm-oss snd-mixer-oss snd-emu10k1 snd-pcm snd-timer snd-util-mem snd-hwdep snd and this is the sound portion of my modules.conf: # ALSA portion alias char-major-116 snd alias snd-card-0 snd-emu10k1 post-install snd-emu10k1 /usr/sbin/alsactl restore pre-remove snd-emu10k1 /usr/sbin/alsactl store # module options should go here options snd-emu10k1 index=0 extin=0x0fc3 extout=0x1f0f # OSS/Free portion alias char-major-14 soundcore alias sound-slot-0 snd-card-0 # card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss Some versioning info: - cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 0.9.0rc6. Compiled on Dec 1 2002 for kernel 2.4.20 with versioned symbols. TIA, Reinier.

21 years, 12 months

Re: [linux-audio-user] Gentoo users?

by Jonathan Kraut

>>Yeah, I'm up for this. I'm not too familiar with making ebuilds, but it >>looks to be fairly straightforward. At the moment, I've just set up a >>hierarchy under /opt, where I manually (configure, make) compile + >>install stuff. The trouble with this is that not all packages give the >>option to specify a location (--foo-lib=DIR), and assumes stuff is >>installed in /usr (grrr :) > >Alrighty then. I've started to ask questions on the gentoo-user alias about >how best to deal with cvs ebuilds. Once there there are some answers i'll >put together a webpage for the files. > >I considered going down the /opt path ;-) but it just isn't the gentoo way! Jon/Mike - Greetings! I'm new to the list and a gentoo user. I've been playing with ebuilds. I made one for jack-cvs and, which I'll email it to you, and to anyone else who wants. I'm working on some others so if you find a place to throw them up I'd be happy to contribute. Jonathan Kraut jkraut1(a)nyc.rr.com

21 years, 12 months

[linux-audio-user] soft synth manager gui

by Chris Cannam

A number of people using the Rosegarden sequencer for MIDI (myself included) are principally using it to drive ALSA soft synths such as iiwusynth or timidity. Unlike (say) MusE, which allows you to start and stop soft synths from the sequencer itself, Rosegarden treats soft synths just like any other MIDI device: it knows nothing special about them except the name that ALSA returns. It's the user's responsibility to start and stop synths and ensure that the right patches and so forth are available. This makes for quite a bit of flexibility, but means that the average user who runs a small number of synths but changes the patch sets (soundfonts or whatever) frequently has a little bit of a management problem in ensuring that the correct setup for a given composition is always available. What I imagine would be useful is a generic probably-GUI-driven app that simply starts and stops soft synths, particularly synths that are driven using the ALSA MIDI API, that output to JACK, and that might not have GUIs of their own (admittedly the only one of those I can think of right now is iiwusynth, but I'm sure there are others and even if you only use iiwusynth you could have any number of different soundfont configurations). As well as starting and stopping synths on command, it might be able to store and recall a selection of multi-synth studio setups of the user's design. It'd be particularly neat if the soft-synth manager could itself be driven remotely through some sort of MIDI control event. Is there such a thing as this? (Anyone feel like writing one?) Is there any other way of arranging synths and suchlike so that this sort of application is not actually necessary, assuming that we want to maintain the current arrangement where Rosegarden does not take responsibility for running the synths itself? Chris

22 years

[linux-audio-user] Audio RPMS for Mandrake users

by Robert Jonsson

Hi guys, I just thought I would mention, in case someone missed it. There are some kind people who have started supplying RPMS for common applications for various distributions. PlanetCCRMA (http://ccrma-www.stanford.edu/planetccrma/software/) being the most well known I think (PlanetCCRMA is for Redhat though). Now there are a few places for Mandrake users to go: - http://rpm.nyvalls.se/sound9.0.html - which I have successfully used to test new versions of both Ardour and MusE. This site keeps _very_ up to date, keeps many of the applications I'm interested in right now. - http://www.esm.rochester.edu/kevine/turnkey/home.html - which I haven't tried but looks promising. Scope here is more towards completeness than having the newest of everything, this is the feeling I get anyway. 2 öre-worth Regards, /Robert

22 years

[linux-audio-user] ymfpci external midi.

by Allan Baker

Hi. I have a Yamaha ymf740c and installed 0.9.0rc6 "the debian way." Audio is working but can someone help me in setting up the mpu401? Also, is there a doc describing module options? As a fairly newbie "parm:mpu_port long array (min = 1, max = 8), description "MPU-401 Port." (from modinfo and from the alsa web site) doesn't have any useful meaning to me. I found some settings that worked when I installed 0.9.0b12 but I have no idea why they worked, or where I was supposed to find them. Any help will be appreciated

22 years

[linux-audio-user] recording (analog) from live drive

by rob

Hi, I just got a sblive platinum 5.1. And am trying to get recording (analog) from the live drive working. I can record fine from the line and mic on the back and can here stuff coming in on live drive but I can't for the life of me get it recorded :(. If anyone has managed could they send me the mixer settings they're using and the command line to arecord (or whatever program there using to record). Thanks (alsa 0.9rc3, debian sid, 2.4.19 + low latency) -- rob <mailingLists(a)pangolin.org.uk>

22 years

[linux-audio-user] RME 96/8 Pad on RH8

by Reuben Martin

Hello all, I'm just now putting together my first linux box and need somebody to hold my hand and walk me through the setup of my sound card. (BTW if there is already a thread on this that I didn't catch on my quick search of the list archives, please point me to it.) I'm currently using a RME 96/8 PAD on a RH8 distro. I installed the alsa packages available from freshrpms.net. After getting everything installed I started up alsamixer and the only controls available is a volume control for the DAC. Please tell me that I'm missing something and that there is much more control functionality available than just levels for the DAC. Shouldn't I have at least some sort of controls for the digital IOs and the ADAT? Is there some sort of fancy config I need to do in /etc/asoundrc? Have I missed something in configuration of devices, subdevices, some package not installed etc? Also, (slightly off topic) this card is very picky about the buffer settings (i.e. fixed buffersize of 64KBytes). I know what settings work when setting number of periods and frame size, but how do I work that out when programs only let you set the buffer size in units of time rather than actual size? Any help is appreciated. Reguards, -Reuben

22 years

[linux-audio-user] Re: [linux-audio-dev] RME Hammerfall DSP Cardbus+Multiface

by D. Sen

Hi Mark, Thanks very much for your those hints. Do you have any idea (or where I can find the documentation) on how to play through the SPDIF or ADAT ports? Where is the matrix mixer controls (amixer) documented for these cards? DS Mark Knecht wrote: dsen, Hi. I'm messing with similar things this evening. (And not doing great, but not failing either.) I'm using a HDAP 9652, not the Multiface, but it's the same driver I believe. The difference is I have no 'headphone output'. Just 26 outputs. I'm in KDE. Inside of a Konquerer window, I have set the file associations for wav and x-wave to be aplay -device="plughw" This seems to work all the time in a Konquerer window, but doesn't work on a file just sitting on my desktop. Don't ask me why. FYI#1 - make sure you don't have a blocking aplay sitting around. Do a ps -aux | grep aplay and make sure nothing is there. FYI#2 - My headphones and main speakers are running off of playback_1 & 2. Good luck, Mark On Wed, 2002-12-18 at 20:29, D. Sen wrote: /> Hi,/ /> / /> I have just bought myself the Cardbus+Multiface RME system. I think I / /> have finally gotten the alsa modules (snd-hammerfall-mem and snd-hdsp) / /> to load correctly along with the OSS emulation modules. BUT I cant seem / /> to play anything out of the Lineout/Headphone output. I have tried / /> various 'amixer cset' commands but all I get is silence (trying to play / /> a simple wav file using aplay)./ /> / /> Any suggetsions?/ /> / -- --------------------------------------- D. Sen, PhD 21 Woodmont Drive Randolph NJ 07869 Home Email: dsen(a)homemail.com Tel: 973 216 2326 Work Email: dsen(a)ieee.org Web: http://www.auditorymodels.org/~dsen

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Linux-audio-user December 2002