Linux-audio-user November 2004

linux-audio-user@lists.linuxaudio.org

188 participants
181 discussions

[linux-audio-user] a little more music made with Ardour

by Dave Phillips

Greetings: As I'm learning more about Ardour I plan to assemble some minor projects and put them on-line as further examples. My most recent addition is a Prelude & Fugue in G for TX802 organ. It's an old piece that I had previously recorded to CD many moons ago. I ripped the track, remixed it in Ardour, the sound is now much more to my liking, and it's now on-line for your listening pleasure here: http://linux-sound.org/ardour-music.html Minimal work with Ardour, but the sound was so much better afterwards. Btw, you keyboard players will have to forgive me, the pieces aren't really meant to be humanly playable (though they may be, I never checked with an actual player). :) Best, dp

19 years, 10 months

[linux-audio-user] debugging usb audio (silence from imic)

by Anthony Baxter

I've plugged a Griffin imic into my Dell 5150 (running FC3-ish). The machine recognised the card and modprobe snd-usb-audio, followed by alsamixer -c 1, showed the card. I can set and get the settings and the like all fine. The one thing I _can't_ get out is audio. None at all. aplay -device=hw:1,0 foofile.wav seems to run fine, except for the bit where it puts out sound. I've not got anything muted in alsamixer that I can see. This is kernel 2.6.8-1.541. (Hopefully) relevant bits from /proc/asound: bonanza# ls card0 cards I82801DBICH4 oss seq timers card1 devices modules pcm system version bonanza# cat cards 0 [I82801DBICH4 ]: ICH - Intel 82801DB-ICH4 Intel 82801DB-ICH4 at 0xf4fff800, irq 7 1 [system ]: USB-Audio - iMic USB audio system Griffin Technology, Inc iMic USB audio system at usb-0000:00:1d.0-1, full speed bonanza# cat version Advanced Linux Sound Architecture Driver Version 1.0.6 (Sun Aug 15 07:17:53 2004 UTC). Compiled on Sep 1 2004 for kernel 2.6.8-1.541. bonanza# cd card1 bonanza# ls id oss_mixer pcm0c pcm0p stream0 usbbus usbid bonanza# cat id system bonanza# cat stream0 # during 'aplay --device=hw:1,0 fooofile.wav' Griffin Technology, Inc iMic USB audio system at usb-0000:00:1d.0-1, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 2 URBs = 5 [ 4 4 4 4 4 ] Packet Size = 200 Momentary freq = 44100 Hz Interface 1 Altset 1 Format: S16_LE Channels: 1 Endpoint: 1 OUT (ADAPTIVE) Rates: 6400 - 48000 (continuous) Interface 1 Altset 2 Format: S16_LE Channels: 2 Endpoint: 1 OUT (ADAPTIVE) Rates: 6400 - 48000 (continuous) Capture: Status: Stop Interface 2 Altset 1 Format: S8 Channels: 1 Endpoint: 4 IN (SYNC) Rates: 6400 - 48000 (continuous) Interface 2 Altset 2 Format: S16_LE Channels: 1 Endpoint: 4 IN (SYNC) Rates: 6400 - 48000 (continuous) Interface 2 Altset 3 Format: S16_LE Channels: 2 Endpoint: 4 IN (SYNC) Rates: 6400 - 48000 (continuous) bonanza# lsmod | egrep 'snd|sound' snd_intel8x0 29709 0 snd_usb_audio 55457 1 snd_usb_lib 10305 1 snd_usb_audio snd_ac97_codec 61329 1 snd_intel8x0 snd_pcm_oss 42361 0 snd_mixer_oss 14529 1 snd_pcm_oss snd_pcm 84681 3 snd_intel8x0,snd_usb_audio,snd_pcm_oss snd_timer 25413 1 snd_pcm gameport 3777 1 snd_intel8x0 snd_mpu401_uart 7233 1 snd_intel8x0 snd_rawmidi 21733 2 snd_usb_lib,snd_mpu401_uart snd_seq_device 6217 1 snd_rawmidi snd 44709 11 snd_intel8x0,snd_usb_audio,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer,snd_mpu401_uart,snd_rawmidi,snd_seq_device soundcore 7457 1 snd snd_page_alloc 6985 2 snd_intel8x0,snd_pcm One thing I've noticed is that lsmod is telling me that the snd_usb_audio device is being used, but it's not saying _what_ it's being used by. I can't see any processes running that have anything in /dev/snd open. I'd love to get some tips on where to look next for continuing to debug this - one of my own projects is shtoom (a voip phone) and I'm always getting pleas for help with ALSA - it'd be excellent if I knew a bit more about what I'm doing here so I can help them in turn <wink> Thanks, Anthony

19 years, 10 months

[linux-audio-user] NMI 34 From Delta 1010 Causes Reboot

by David L. Craig

I'm scratching my head about where to go next with a problem I'm having with my Delta 1010 on my IBM Netfinity dual Xeon box. I'm runniing DeMuDi 1.2.0 with the 2.4 kernel. The kernel hiccups on an NMI reason code 34 and reboots. The PowerPC hypervisor records a SERR event from the 1712 device with status 4210. M-Audio tech support says they won't support the card since it's under Linux, not even to let me know if the hardware or software is at fault. They say go to alsa-project.org. I think angula.org should be the next stop but their web server seems to be out of service. Any ideas will be greatly appreciated. -- May the LORD God bless you exceedingly abundantly! Dave Craig - - - - - - - - - - - - - - - - - - - - "'So the universe is not quite as you thought it was. You'd better rearrange your beliefs, then. Because you certainly can't rearrange the universe.'" --from _Nightfall_ by Asimov/Silverberg

19 years, 10 months

[linux-audio-user] nvidia drivers

by Florin Andrei

Something apparently unrelated to Linux audio, but in fact it is. Let me explain. 1. Is there a way to run an NVidia card dual-head (two monitors) using the open source drivers? 2. Is there a way to silence the fan on an NVidia-based card other than loading up the proprietary drivers? (which silences it as long as there is no hard graphics work) Being able to extend the desktop across two monitors, and not having to bear the obnoxious fan on the graphics card are the reasons why i'm using the NVidia proprietary driver, because: - i want to use two monitors because audio apps are hungry for desktop real estate - the noisy fan is very disturbing when doing any kind of audio work But there's a nasty side-effect of the proprietary driver - the system seems to freeze for a few seconds every once in a while (about once a day). There are no side-effects of the open source driver, but i don't know how to tell it to work in dual-head mode and i don't know how to silence the fan. -- Florin Andrei http://florin.myip.org/

19 years, 10 months

Re: [linux-audio-user] Instrument Tuner: fmit 0.9.0

by Gilles Degottex

On Thursday 25 November 2004 15:17, Austin wrote: > On Thu, 2004-11-25 at 08:32, Gilles Degottex wrote: > > On Wednesday 24 November 2004 03:50, you wrote: > > > Any chance of making the binary both jack and alsa compatible in the > > > future? > > what you exactly mean by compatible ? > > merged in only one prog with an option in the settings to choose between > > them ? > > Well, you could do it that way, but much simpler would be just to detect > jackd process owned by same user, use jack interface if it's found, and > if not, default to alsa. > It's just that I can't compile both versions for Mandrake, why you can't ? I did both rpm 'fmit-jack' and 'fmit-alsa' (http://download.gna.org/fmit/) they should be compatibles for Suse and Mandrake, maybe more ditros. > so for now I > uploaded the alsa version. I just thought it would be nice in the > future if automatic jack detection was there too. yes, it could be usefull, but If I do so, fmit becomes dependent to JACK, whatever we use it or not. that's why I did two versions. but I can do a third mixed version. to get optional dependency at running-time, I should do an input plugin support ....... Gilles

19 years, 10 months

[linux-audio-user] help

by Javier TM

On Sat, 9 Oct 2004 12:00:20 -0400, <linux-audio-user-request(a)music.columbia.edu> wrote: > Send linux-audio-user mailing list submissions to > linux-audio-user(a)music.columbia.edu > > To subscribe or unsubscribe via the World Wide Web, visit > http://music.columbia.edu/mailman/listinfo/linux-audio-user > or, via email, send a message with subject or body 'help' to > linux-audio-user-request(a)music.columbia.edu > > You can reach the person managing the list at > linux-audio-user-owner(a)music.columbia.edu > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of linux-audio-user digest..." > > > Today's Topics: > > 1. [ANN] QSynth 0.2.2 released! (Rui Nuno Capela) > 2. [ANN] QjackCtl 0.2.12 released! (Rui Nuno Capela) > 3. ALSA OSS emulation? (Mikhail Ramendik) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 8 Oct 2004 17:00:49 +0100 (WEST) > From: "Rui Nuno Capela" <rncbc(a)rncbc.org> > Subject: [linux-audio-user] [ANN] QSynth 0.2.2 released! > To: linux-audio-user(a)music.columbia.edu > Message-ID: <33348.192.168.1.5.1097251249.squirrel(a)192.168.1.5> > Content-Type: text/plain;charset=iso-8859-1 > > Hi everyone, > > After a great long time, lurking on CVS, here comes another step to this > fluidsynth's Qt/GUI frontend: > > Qsynth has been released: 0.2.2 is now public! > > Taken from the changelog: > > - Minor configure fixes. > > - Meanwhile, XPM icon(s) were brainlessly converted to PNG format. > > - Engine panel settings are now properly saved on stop/restart. > > - Icons were added to the engine tab selector context menu. > > - Master gain front panel control gets rescaled and now ranges from > 0..200, with midpoint at 100 (unit gain). > > - Added Mac OS X build instructions (README-OSX, by Ebrahim Mayat). > > - Soundfont bank offset option gets its trial time (EXPERIMENTAL); > please > note that fluidsynth 1.0.5 is needed to build on this feature, which is > being properly detected and only enabled at configure time. > > - Output level peak meters are now featured as an option (EXPERIMENTAL), > which must be explicitly enabled on setup for those to show up; in > addition, overall GUI refresh cycle period has been reduced from 200 to > 100 msec. > > - Top level sub-windows are now always raised and set with active focus > when shown to visibility. > > As usual, grab it from: > > http://qsynth.sourceforge.net > > Cheers, and enjoy, -- Using Opera's revolutionary e-mail client: http://www.opera.com/m2/

19 years, 10 months

[linux-audio-user] Re: es1370 and the el cheapo myth

by Timo Sivula

Hello, thanks for the help. > Date: Tue, 19 Oct 2004 12:45:16 +0200 (METDST) > From: Clemens Ladisch <clemens(a)ladisch.de> > Timo Sivula wrote: > > > Does "aplay -D multi something.wav" work? > > > > It plays the song through one of the soundcards. I am not sure that if > > that qualifies as working? > > It found the "multi" device. > > The reason that Jack didn't is that it requires a control device, too. > Please add the following: > > ctl.multi { > type hw > card 0 > } I did that. In addition I removed the crystals from two of the three cards and linked them to the first card, so that all three cards now run on the crystal from the first. Thanks to the advice from Jaroslav Kysela I was able to test and verify that all three cards run simultaneously and can play separate files at the same time. When I now run Jackstart the following happens: - clipeti clip - localhost:~$ jackstart -R -dalsa -dmulti -r44100 -p128 -n3 -S back from read, ret = 1 errno == Success jackd 0.99.0 Copyright 2001-2003 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details loading driver .. apparent rate = 44100 creating alsa driver ... multi|multi|128|3|44100|0|0|nomon|swmeter|-|16bit configuring for 44100Hz, period = 128 frames, buffer = 3 periods You appear to be using the ALSA software "plug" layer, probably a result of using the "default" ALSA device. This is less efficient than it could be. Consider using a hardware device instead rather than using the plug layer. Usually the name of the hardware device that corresponds to the first soun You appear to be using the ALSA software "plug" layer, probably a result of using the "default" ALSA device. This is less efficient than it could be. Consider using a hardware device instead rather than using the plug layer. Usually the name of the hardware device that corresponds to the first soun ALSA lib pcm.c:1178:(snd_pcm_link) SNDRV_PCM_IOCTL_LINK failed: Operation already in progress jackstart: pcm.c:5957: snd_pcm_mmap_commit: Assertion `frames <= snd_pcm_mmap_avail(pcm)' failed. - clip clip - a) I do not understand why the "You appear to be using ..." text appears twice? I only start Jackstart once. b) I also do not understand the error messages at the end. The first one: "ALSA lib pcm.c:1178...." I can not find on the net. What does this mean? The other one: "pcm.c:5957: snd_pcm_mmap_commit:" error is referred to at http://alsa.opensrc.org/index.php?page=TwoCardsAsOne and the reason is said to be the cards running out of sync. My cards are hard-synced on clock level, and should therefore be also sample accurately in sync, so this error should not appear. Any idea what is going on? Here is my ~/.asoundrc - clip - pcm.multi { type plug slave.pcm { type multi slaves.a { pcm "hw:0" channels 2 } slaves.b { pcm "hw:1" channels 2 } slaves.c { pcm "hw:2" channels 2 } bindings [ { slave a channel 0 } { slave a channel 1 } { slave b channel 0 } { slave b channel 1 } { slave c channel 0 } { slave c channel 1 } ] } } ctl.multi { type hw card 0 } - clip clip - br, Timo

19 years, 10 months

[linux-audio-user] Re: RME is no more

by Brian Redfern

Hmmm, with Digidesign buying Maudio, and now this news, its entirely possible that there won't be any high end audio cards for linux at all.

19 years, 10 months

[linux-audio-user] Edirol UM-1S weird behaviour

by mchristoph.eckert＠t-online.de

Hi, I have problems using my Edirol UM-1S: http://www.edirol.com/press/hirez_photos/um1_1s.html When I boot while it is plugged in, it doesn't appear in kaconnect. Additionally, my DSL connection will not work. When I plug it out and reboot, the DSL connection will still not work, I have to reboot again and then my box behaves correctly again. When I boot and plug the UM-1S in after booting, the mouse will hang for some seconds. After that, the UM-1S will be ready to use in kaconnect in most (but unfortunately not all) cases. When I do a tail -f /var/log/messages while pluggin it in, I get: Nov 21 00:11:33 Grandevitesse usb 1-1: new full speed USB device using address 2 Nov 21 00:11:34 Grandevitesse snd-usb-audio: probe of 1-1:1.0 failed with error -5 Nov 21 00:11:34 Grandevitesse snd-usb-audio: probe of 1-1:1.1 failed with error -5 Nov 21 00:11:34 Grandevitesse midi: probe of 2-1.1:1.0 failed with error -5 Nov 21 00:11:34 Grandevitesse midi: probe of 1-1:1.0 failed with error -5 Nov 21 00:11:34 Grandevitesse midi: probe of 1-1:1.1 failed with error -5 Nov 21 00:11:34 Grandevitesse usbcore: registered new driver midi A lsusb | grep -i Roland gives Bus 001 Device 002: ID 0582:0009 Roland Corp. When I do a cat /lib/modules/2.6.7-gentoo-r6/modules.usbmap | grep 0009 | grep 0582 I get two lines starting with snd-usb-audio usb-midi each. So, the kernel seems to know the device. The question is, if really both of the modules are needed. AFAIK, snd-usb-audio contains both the usb and the midi driver. What is usb-midi for? I'm currently using a Gentoo, 2.6.7-gentoo-r6 #5 SMP Sat Oct 9 23:31:33 CEST 2004 i686 Intel(R) Pentium(R) 4 Mobile CPU 1.60GHz GenuineIntel GNU/Linux. The problem has been similar on my previous Mandrake box, but at least the DSL connection has not been affected. It also seems that not the DSL connection directly is affected but the network interface. pppoe tries to reconnect (I can see it in the syslog), but it can reconnect as soon as I do a /etc/init.d/net.eth0 restart Has anyone had similar problems, or does anyone use the UM-1S and has some tips? Why does my syslog give an »probe of 1-1:1.0 failed with error -5« ? Every wee small hint is highly welcome. Best regards ce

19 years, 10 months

[linux-audio-user] Problems recording using soundcard's 'mic' jack

by Sebastian Tennant

Hi all. I'm running Debian 'Sid' and a working ALSA sound driver. My soundcard is a Cirrus Logic CS4281. The GNOME 'CD Player' app plays CDs no problem. 'Muine' plays mp3s no problem and all the GNOME desktop sound effects work fine. However when I plug a line-level audio signal into the soundcard's 'mic' mini-jack and try to record a wav audio file I get nothing. (I can hear the sound fine through the headphones/laptop speakers). I've tried 'wavtools' and 'ecasound' but when I try to play back the resulting wav file using 'Alsaplayer' all I get is silence. No doubt the thing I'm doing wrong is glaringly obvious but be damned if I can figure out what it is! Any help much appreciated. sebyte -- CC me by all means but a follow-up will usually do.

19 years, 10 months

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Linux-audio-user November 2004