Linux-audio-user October 2004

linux-audio-user@lists.linuxaudio.org

153 participants
223 discussions

RE: [linux-audio-user] newbie to linux sound

by Reuben Martin

>> Just a quick question.... Did you compile your kernel with the CS46XX >> as a module? This may be a kernel issue rather than a sound issue. >> Stephen. > >> > I get the error: >> > /lib/modules/2.4.20-8/kernel/sound/pci/cs46xx/snd-cs46xx.o: init_module: >> > No such device > >How do I find out how cs46xx was compiled? In this case, it's quite simple: you look at the version of your kernel. ALSA was not merged into the kernel until early in the 2.5 development series. You are using a 2.4 kernel, therefore there is no way for it to be compiled any other way than as a module. -Reuben

20 years, 2 months

[linux-audio-user] 2.4.22 kernel patches

by Phil Carter

I'd like to try openMosix (openMosix.sf.net), which requires a kernel patch. Right now, I'm using 2.4.22 with low latency and preempt patches. Will I be able to use openMosix with the other patches? Has anyone tried this? Does it matter what order I apply the patches in? I'm not sure I even need preemption (I read about using it with the low latency patch once, but I can't remember where), but I'd like to have at least openMosix + low latency if I can. Phil Carter __________________________________ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail

20 years, 2 months

[linux-audio-user] xmms-jack 'couldn't open audio' problem

by Jussi Sainio

Hi, I have this problem when using xmms-jack output plugin in xmms.. If I run it as a normal user, I get usually "Couldn't open audio..." error box, but when running as root it works without problem. Could this be a permission issue or something? jack-rack, alsaplayer and ams don't have this kind of problem at all. jackd runs in realtime -mode, kernel is 2.6.9-rc2-mm4-VP-S7 xmms 1.2.10, xmms-jack 0.10 (0.9 has same issue, tested it too..) Here is some debugging output: xmms (xmms-jack verbose and trace mode enabled): [...] JACK_GetVolumeForChannel:deviceID(0), returning volume of 25 for channel 0 JACK_GetVolumeForChannel:deviceID(0), returning volume of 25 for channel 1 JACK_OpenEx:bits_per_channel=16 rate=44100, input_channels=2, output_channels=2 JACK_OpenEx:jack_port_name = NULL JACK_ResetFromThis:resetting this->deviceID(0) JACK_OpenEx:bytes_per_output_frame == 4 JACK_OpenEx:bytes_per_input_frame == 4 JACK_OpenDevice:creating jack client and setting up callbacks JACK_OpenDevice:client name 'bio2jack_0_7504' JACK_OpenDevice:setting up jack callbacks JACK_srate:the sample rate is now 48000/sec JACK_OpenDevice:engine sample rate: 48000 JACK_OpenDevice:creating output ports JACK_OpenDevice:port 0 is named 'out_0' JACK_OpenDevice:port 1 is named 'out_1' JACK_bufsize:the maximum buffer size is now 256 frames JACK_bufsize:expanding buffer from this->buffer_size == 0, to 1024 JACK_bufsize:called JACK_OpenDevice:calling jack_activate() ERR: JACK_OpenDevice:cannot activate client JACK_OpenEx:error opening jack device JACK_GetVolumeForChannel:deviceID(0), returning volume of 25 for channel 0 JACK_GetVolumeForChannel:deviceID(0), returning volume of 25 for channel 1 jackd verbose mode: [...] load = 0.3697 max usecs: 13.000, spare = 5320.000 new client: bio2jack_0_7504, id = 3 type 2 @ 0xb6555000 fd = 20 23:07:43.989 Audio connection graph change. registered port bio2jack_0_7504:out_0, offset = 3072 registered port bio2jack_0_7504:out_1, offset = 4096 gave capabilities to process 7504 23:07:44.152 Audio connection change. load = 0.3818 max usecs: 21.000, spare = 5312.000 Any help would be appreciated. :) Thanks. --jussi

20 years, 2 months

[linux-audio-user] newbie to linux sound

by Luke (Terry) Vanderfluit

Hi, I'm new to linux sound (but not to sound or linux) So I'm starting off on a journey and need some help. I have a dman pci card with cs46xx chipset. I'm using Redhat 9, when my machine booted it detected the card and configured it. However, all is not right. there are lots of crackles and pops when the volume is up. It can play wavs but it seems the volume needs to be quite high. I have downloaded the alsa tarballs (driver, lib, utils) what should be my first step in trouble shooting this setup thanks, kind regards, Luke -- ======================== Luke (Terry) Vanderfluit Mobile: 0421 276 282 ========================

20 years, 2 months

[linux-audio-user] Demudi looking good, soundcard advice?

by Alejandro Lopez

Chaps, Just installed Demudi, sorry I can't remember versions, it's some beta that came out something like 2-3 months ago which was the first to merge Debian + music studio on the same CD (requiring one installation only). Generally, I would recommend the distribution for an easy install (I've not used it yet), I'd like to make a couple of comments in case anyone is interested. -My PC won't boot from CD-ROM. By browsing the directory structure, I kind of guessed I had at least a couple of choices for DOS: booting from a rawrite2'd floppy or run loadlin. Fair enough, there's a vmlinuz file there as well as several .bin floppy images. But: loadlin is not included in the CD image, not it is rawrite2. Not a big deal for me as I had several other distros handy but not elegant either. In the same directory where these images reside, there's a little text file which lists loadlin and rawrite2: should read "loadlin", reads "lodlin" (in case anyone would like report this for future versions). -I was not able to select English language with a Spanish keyboard layout (I'm Spanish). You are supposed to select both in one go, the 2 different options in the menu (that you can see if you select back somewhere in the middle of the process) seem to be related and didn't work as such distinct options for me. -At one point, I went back and selected Spanish (for everything, that is), next step was repartitioning my hard disk which is inherently a dangerous operation. The translation of the partitioning software to Spanish was so poor (wrong actually) that I was scared to repartition in Spanish. Went back to English so I could understand what the software would do with my hard disk but now my keyboard layout doesn't match. I'm planning installing to a different machine I imported from Germany, so I'll have the same layout problem (I'd rather not choose German language to repartition my disk in order to have the keyboard properly installed). -From there on, the installation was as smooth as one could have asked for. All my hardware was detected and installed automagically including graphics card, monitor, mouse, sound card (ALSA), network card, modem, CD-ROM. So generally I'd say it's a very easy to install distribution, and definitely a huge improvement over the classic "2 installation steps" approach that all the other music distributions have or used to have. I'm very pleased with regards to that. The only other thing is I may need to buy a decent soundcard for this PC. It should have a GM synth (nothing special since I guess I'll be using soft synths anyway but I don't want to run a synth just to write a couple of arrangements), one stereo output and one stereo input both with good overall audio quality. Seemingly, the current trend is towards either USB 2.0, firewire or PCI. A colleague has just told me that USB 2.0 is supported by the Linux kernel starting from the latest version (2.6 I think?) only. Since music distributions are based on other distributions, chances are that they now run the 2.4 kernel or maybe older. Does that make sense or am I talking complete b*ll*cks? Also, this friend has heard about drops happening with audio over USB, but apparently this was on a Mac and the USB device was a hard disk rather than a soundcard. Still, he seems to think that USB 2.0 is not as good option as firewire for audio. (Which reminds me of SCSI vs IDE drives for audio a few years back, yes SCSI was the serious option for a couple of years but it was an 80% more expensive as well, and shortly afterwards IDE started to go "fast enough" and cheaper.) Is firewire better? Is it more expensive? Also, how does it compare to PCI? Lastly, if any of you has bought a soundcard (recently so the card is still in production) which is reasonably similar to what I need (wouldn't mind if it's slightly better, say 4 mono inputs and 4 mono outputs or something) and has succeed with having it running under a Linux based music studio, I'd be grateful if you drop a line. Many thanks!! _________________________________________________________________ Descarga gratis la Barra de Herramientas de MSN http://www.msn.es/usuario/busqueda/barra?XAPID=2031&DI=1055&SU=http%3A//www…

20 years, 2 months

[linux-audio-user] live moog madness

by Austin

It's time to ask for advice. :-) I'm playing a show next weekend with a rock band. I'm using a MIDI controller and a Roland Sound Canvas for almost everything, but there are two songs that need typical "Moog" sounds... highly distorted and heavy portamento (think Weezer or The Rentals). This the Sound Canvas cannot do (AFAIK). Without buying an old Moog, what are my options, and what do you suggest... looking for cheap, and flawless on-stage? Ideas: 1. Look for a cheap used MIDI sound module with great portamento. I've never seen one. 2. Alsa Modular Synth. I really like the portamento effect, but I've yet to find any great moog sounds, and I'm not good at making my own. It's quite stable, but gives me "clicky" artifacts sometimes. 3. ZynAddSubFX. Has some great synth sounds, but the portamento is too "over the top". Over dramatic. 4. VST. I have some *amazing* Moogy VST plugins, but no idea how to use them with linux. I'm aware of the wine/vst libraries, but though they were mainly for effects, not instruments. 5. Better ideas are welcome. Remember that simple and reliable is more important than sound. Thanks, Austin -- Austin <aacton(a)yorku.ca>

20 years, 2 months

[linux-audio-user] Problems with two Delta 1010LTs

by Thomas Wehrspann

After searching and reading nearly everthing about Delta 1010LTs and Alsa, i have still no clue how to solve my problem. I also posted on the Alsa mailinglist, but got no answer. I bought two Delta 1010LTs to do 16 channel multitrack recording with ardour and jack. And i have no problem doing so with either the first or the second one alone, despite regular messages of the form: "load = 0.2046 max usecs: 79.000, spare = 23140.000" My system is an Athlon TB 900 MHz, 512 MB RAM, both Deltas have a interrupt on there own (IRQ7 and IRQ5). At the moment it's just plain Xorg with twm. I use kernel 2.6.8-r3-gentoo with kernel ALSA driver and preemp and volutary preemp patch. ALSA-lib, -utils and -tools have version 1.06. In the meantime i tried a recent 2.4 kernel with realtime patches and Alsa driver 1.06 with different problems. PROBLEM 1: .asoundrc ------- Since Jack only supports one sound card, i tried to create a virtual 16 channel card through the .asoundrc file (appended at the end of this post). The Delta has 12 capture and 10 playback devices, so i created two multi devices, one for capture and one for playback and an asym device to put them together and some magic for jack. This is the only configuration that works a little. Is this the right configuration? Bevor jack accepts this card, the deltas had to be synced, either through S/PDIF or Wordclock (i tried both). Btw. is there a kernel sync like under Windows? Because, when i open two envy24control windows (one for each card) and change the clock rate in one envy24control the other changes as well when reset is selected. And what's the meaning of "reset" and "locked"? PROBLEM 2: kernel 2.6 ------- Without Syncing, jackd sometimes locks the entire system. When the cards are synced and i start jack there is an error message: "Alsa lib pcm.c: 1178: (snd_pcm_link) SNDRV_PCM_IOCTL_LINK failed: Operation already in progress" It seems to have no effect, though. Then i got a lot of XRUN callbacks. With jack Soft Mode i got "load = 0.2046 max usecs: 79.000, spare = 23140.000". I can work with it, but it shouldn'd be normal? If this where all, i won't have bothered you. Using the 16 channel virtual card with ardour works for the first 8 channels. The second 8 channels of the second card, which where synced to the first card, produces weird sound, both while recording and playback. It would be nice if someone can help me. Thanks in advance. PROBLEM 3: kernel 2.4 ------- Since i get no answer on the above question on the Alsa list, i changed the kernel back to 2.4 and the external ALSA driver 1.06. The configuration (hardware and .asoundrc) remained the same. When i start jackd with playback only, this works great now. The weird sound is no longer. With capture too, i get "jackd: pcm.c:5957: snd_pcm_mmap_commit: Assertion 'frames <= snd_pcm_mmap_avail(pcm)' failed" Sometime with a few "load = 2.3689 max usecs: 550.000, spare = 11059.000" before. Somewhere i read, that this is about unsyncronisation? With kernel 2.6, but without syncing, i got the same messages when using capture devices. Sorry for the big post and the many questions. Greetings Thomas Wehrspann ---.asoundrc--- ########################### # M-AUDIO DELTA 1010LT(1) # ########################### pcm.ice1712_1 { type hw card 0 } ctl.ice1712_1 { type hw card 0 } ########################### # M-AUDIO DELTA 1010LT(2) # ########################### pcm.ice1712_2 { type hw card 1 } ctl.ice1712_2 { type hw card 1 } ########################### # VIRTUAL 16 CHANNEL # ########################### pcm.delta8in { type multi slaves.a.pcm ice1712_1 slaves.a.channels 12 slaves.b.pcm ice1712_2 slaves.b.channels 12 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave a bindings.2.channel 2 bindings.3.slave a bindings.3.channel 3 bindings.4.slave a bindings.4.channel 4 bindings.5.slave a bindings.5.channel 5 bindings.6.slave a bindings.6.channel 6 bindings.7.slave a bindings.7.channel 7 bindings.8.slave b bindings.8.channel 0 bindings.9.slave b bindings.9.channel 1 bindings.10.slave b bindings.10.channel 2 bindings.11.slave b bindings.11.channel 3 bindings.12.slave b bindings.12.channel 4 bindings.13.slave b bindings.13.channel 5 bindings.14.slave b bindings.14.channel 6 bindings.15.slave b bindings.15.channel 7 } pcm.delta8out { type multi slaves.a.pcm ice1712_1 slaves.a.channels 10 slaves.b.pcm ice1712_2 slaves.b.channels 10 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave a bindings.2.channel 2 bindings.3.slave a bindings.3.channel 3 bindings.4.slave a bindings.4.channel 4 bindings.5.slave a bindings.5.channel 5 bindings.6.slave a bindings.6.channel 6 bindings.7.slave a bindings.7.channel 7 bindings.8.slave b bindings.8.channel 0 bindings.9.slave b bindings.9.channel 1 bindings.10.slave b bindings.10.channel 2 bindings.11.slave b bindings.11.channel 3 bindings.12.slave b bindings.12.channel 4 bindings.13.slave b bindings.13.channel 5 bindings.14.slave b bindings.14.channel 6 bindings.15.slave b bindings.15.channel 7 } pcm.delta { type asym playback.pcm delta8out capture.pcm delta8in } pcm.jack { type route slave.pcm delta ttable.0.0 1 ttable.1.1 1 ttable.2.2 1 ttable.3.3 1 ttable.4.4 1 ttable.5.5 1 ttable.6.6 1 ttable.7.7 1 ttable.8.8 1 ttable.9.9 1 ttable.10.10 1 ttable.11.11 1 ttable.12.12 1 ttable.13.13 1 ttable.14.14 1 ttable.15.15 1 } ctl.jack { type hw card 0 }

20 years, 2 months

[linux-audio-user] Motherboard for flawless low latencz operation recomendation ?

by Robert Rozman

Hi, I read a lot about having problems with latencies, interrupts on some motherboards under Linux. I originally asking for use in Asterisk PBX SW system under Linux, but since it's almost same low latency sensible application, I thought I'd ask also linux sound gurus - that have sometimes similar problems. Could you please be so kind to recomend me brand or type of motherboards that are working flawlessly for low latency applications under Linux (Gigabyte, Asus, Intel, some other...) ? Is there any site about PC motherboards particularly suitable for low latency applications under Linux ? Thanks in advance, regards, Robert.

20 years, 2 months

[linux-audio-user] dman pci and cs46xx driver

by Luke (Terry) Vanderfluit

Hi, I am new to this list and new to music making with linux, so I don't yet know much about alsa, jack, ladspa etc. I have a dman pci card that I would like to use. I've searched the net and found that there should be a driver for this card, the cs46xx. the card has the crystal cs4614-cm ep chipset. Does anyone know of a driver for this card? thanks, kind regards, Luke -- ======================== Luke (Terry) Vanderfluit Mobile: 0421 276 282 ========================

20 years, 2 months

[linux-audio-user] [OT] recording levels (mismatch)

by Erik Steffl

I am trying to record from a receiver (not sure if that's the right name, it's the amp/tuner, the thingy that all other components like CD player, tape deck, video etc. connect to) and it seems like the levels are too low (the soundcard is soundblaster live platinum, using alsa drivers that come with kernel 2.6.5). the receiver has two stereo audio outputs - one is marked as tape deck recording and the other one as VCR output but both seem to be about the same (and there doesn't seem to be a way to change them). I connect them to the RCA connectors on the front panel (live drive) but even if I set the mixer to 100% (on line in) the levels are too low. On vu-meter the levels only go to about 25-30% (max) while the 'normal' audio is somewhere between 50-75%. I believe both the receiver and soundcard work OK (I can record from receiver to tape deck, I can record to soundcard from other sources) so I guess there's mismatch between the signal from receiver and what soundcard expects. Is there anything I can do? do I need some pre-amp or something? Seems crazy but does it make any sense to try to record from speaker outputs? Is there any way to change what soundblaster expects on input? TIA erik

20 years, 2 months

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Linux-audio-user October 2004