Linux-audio-user

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20662 discussions

Re: [LAU] Internal speaker set as default playback, not my usb soundcard.

by Andrew C

Excellent, thanks for that tip, Phillip! Got it fixed now. Put "autospawn = no" into $HOME/.config/pulse/client.conf, killed PA as you suggested and everything worked as expected, alsamixer controls the volume (.asoundrc works perfectly again) and mplayer/flash/MOCP plays through my usb card. Andrew. On Tue, Aug 19, 2014 at 2:47 PM, Philipp Überbacher <murks(a)tuxfamily.org> wrote: > On Tue, 19 Aug 2014 13:48:22 +0100 > Andrew C <countfuzzball(a)gmail.com> wrote: > > > Speaking of .asoundrc, I had set up my like so (which predictably no > > longer works, even though my usb card is still at index 0): > > > > pcm.!default { > > type plug > > slave.pcm "softvol" #make use of softvol > > } > > > > pcm.softvol { > > type softvol > > slave { > > pcm "dmix" #redirect the output to dmix (instead > > of "hw:0,0") > > } > > control { > > name "PCM" #override the PCM slider to set the > > softvol volume level globally > > card 0 > > } > > } > > > > I doubt this would cause a mess up (even so, even *that* is no longer > > working, my soundcard ignores any changes made by alsamixer). > > > > At this point, would I guess there's some pulseaudio weirdness > > happening? Any ideas how I can use alsa straight up, unless I've > > somehow been using a pulseaudio-alsa wrapper this entire time. > > > > To my mind, it makes absolutely no sense why stuff would be playing > > out of Card 1, and not Card 0. Or at least why that has suddenly > > changed to be the default with this upgrade. > > > > Andrew. > > I'm not sure whether you get direct alsa access while PA is running. > > You could disable pulseaudio and try again. I have configured PA > so that it is only ever started manually by me and I simply use > 'pulseaudio --start' and 'pulseaudio --kill'. However, depending on > your distributions setup it might be a PITA to stop PA and keep it > stopped. You can use 'pulseaudio --check' to see whether it runs, it > returns 0 if it runs, 1 otherwise. Or just use 'pulseaudio --kill' and > see whether it throws an error or succeeds because something started PA > again. This can happen because PA is set to respawn or because some > program started PA on startup. That's a PITA I don't want to deal with, > hence my entirely manual setup. > > Regards, > Philipp >

11 years, 3 months

Re: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable

by david

Well, I've heard that Gnome 3's default configuration makes it a major resource hog - particularly for memory. Like KDE4. I avoid Gnome 3. Both it and KDE4 launch a bunch of system services that I think can really impact RT use. Sorry, I don't know anything about how Fedora might be configured differently from Debian. On 08/18/2014 12:12 PM, Sam Tuke wrote: > For what its worth, audacity is unstable on Fedora with Gnome 3 as well. > Routine crashes and corrupted recovery files make it a real headache to > use. Maybe its the versions of the packages we're using? > > Sam. > > On 18 August 2014 20:07:49 CEST, david <gnome(a)hawaii.rr.com> wrote: > > On 08/18/2014 04:45 AM, James Stone wrote: > > On Mon, Aug 18, 2014 at 3:34 PM, Fons Adriaensen > > On Mon, Aug 18, 2014 at 02:21:56PM +0200, Philipp Überbacher > wrote: > > You can also try jack. Audacity uses jack in a very > weird way, as soon > as you roll audacity autoconnects to the first outputs > it finds and > disconnects once you stop rolling, > > > That's only one of the many apps that claim to support Jack but > get it completely wrong. In many cases, but not always, > portaudio > is to blame. > > Perso nally I think the way Audacity handles audio on linux is > very bad > - doesn't manage to do Alsa, Jack or Pulseaudio right as far as > I can > see (if I don't run jack it endlessly changes the sample rate on my > card - making lots of clicks and pops as it takes over 1 minute to > start up!). I tried discussing problems on their forums but to no > avail. > > > I use Audacity on 2 different machines, both 64-bit Debian Sid, with a > UCA-202 USB sound card on the laptop and the now-working-again (YAY!) > Audiophile on the desktop. With or without JACK, Audacity never changes > the sample rate as you mention above. Doesn't take a minute to start up, > either. > > I think there's some more fundamental problem with your system setup > than Audacity. Maybe Audacity's difficulties handling audio just make it > more sensitive to the fundamental problem than other apps. -- David W. Jones gnome(a)hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com

11 years, 3 months

Internal speaker set as default playback, not my usb soundcard.

by Andrew C

Hello all, Recently did an upgrade to Ubuntu 14.04 from 12.04. All went well except I've encountered a very weird problem. My usb audio is no longer the default sound card. I had to some some trickery (made snd-usb-audio=-1 and snd-hda-intel=-2) in /etc/modprobe.d/alsa-base.conf to get the usb card as #0 as listed by /proc/asound/cards and then I got all sound going out through my usb card. Fun fun fun. Now with Ubuntu 14.04, my usb card is still listed as #0 in asound/cards, but mplayer defaults to my internal speakers when issued 'mplayer sound.wav'. Also tried: mplayer -Dalsa=hw=0.0 sound.wav Plays through my usb soundcard mplayer -Dalsa=hw=1.0 sound.wav Plays through my internal speakers mplayer -Dalsa=plughw sound.wav Plays through my usb soundcard mplayer -Dalsa=default sound.wav Plays through my internal speakers. Also I tried opening 'pavucontrol' fwiw and disabling the internal speaker. Playing a sound file without any other options via mplayer sends the sound through my usb soundcard. This is really strange and perhaps I'm not entirely understanding all the "wonderfulness" that is alsa. Cheers, Andrew.

11 years, 3 months

Ubuntu Studio (was: Audacity unstable)

by Alan McKay

Hey folks, I'm thinking I might just install Ubuntu Studio tonight and run with that. The website does not seem to tell me the essential differences from Ubuntu. Which desktop is used? Are there any other major differences I'll notice aside from this? thanks, -Alan -- "Don't eat anything you've ever seen advertised on TV" - Michael Pollan, author of "In Defense of Food"

11 years, 3 months

JACK not working with Audiophile 2496 anymore

by david

I get these messages from JACK when I start it using QJackCtl, trying to use my AudioPhile 2496. This is running on Debian Sid, uname -a reports "3.14-2-amd64 #1 SMP Debian 3.14.15-2 (2014-08-09) x86_64 GNU/Linux" (but I was getting the same error on kernel 3.02.4 before that.) I am a member of the audio group. JACK starts and runs fine if I pick the "default" interface, but that doesn't play any audio through the Audiophile. (I have no idea what it's playing through.) While parts of Pulse are installed, according to "killall pulse" Pulse audio is not running. Trying to remove the installed Pulse packages wanted to also remove a bunch of other apps, such as csound, GIMP, some KDE4 apps, etc. I found an old thread on a forum where the poster said installing "dbus-python" fixed it, but the closest package I could find similar to that name is "python-dbus" and it's installed. This is a fresh Debian Sid install and my first experience with Debian's move to systemd instead of good old reliable init, so could that be messing things up? Or that it's using PAM (my prior Debian experience didn't include PAM) and htop shows a number of (sd-pam) processes running)? jackdmp 1.9.10 Copyright 2001-2005 Paul Davis and others. Copyright 2004-2014 Grame. jackdmp 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 JACK server starting in realtime mode with priority 10 self-connect-mode is "Don't restrict self connect requests" audio_reservation_init Acquire audio card Audio2 creating alsa driver ... hw:M2496|hw:M2496|512|2|48000|0|0|nomon|swmeter|-|32bit configuring for 48000Hz, period = 512 frames (10.7 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 32bit integer little-endian ALSA: use 2 periods for playback 21:08:30.707 Could not connect to JACK server as client. - Overall operation failed. - Server communication error. Please check the messages window for more info. JackPosixProcessSync::LockedTimedWait error usec = 5000000 err = Connection timed out Driver is not running Cannot create new client Cannot read socket fd = 14 err = Success CheckRes error JackSocketClientChannel read fail Cannot open qjackctl client ALSA: poll time out, polled for 15999022 usecs JackAudioDriver::ProcessAsync: read error, stopping... 21:10:14.445 JACK is stopping... Jack main caught signal 15 Released audio card Audio2 audio_reservation_finish 21:10:14.463 JACK was stopped successfully. 21:10:14.463 Post-shutdown script... 21:10:14.463 killall jackd jackd: no process found 21:10:14.879 Post-shutdown script terminated with exit status=256. aplay -l lists the following audio devices: **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: VT1828S Analog [VT1828S Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: VT1828S Digital [VT1828S Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 2: VT1828S Alt Analog [VT1828S Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: M2496 [M Audio Audiophile 24/96], device 0: ICE1712 multi [ICE1712 multi] Subdevices: 1/1 Subdevice #0: subdevice #0 These problems originally began when I had to reset the BIOS, which re-enabled the motherboard audio. I tried disabling the motherboard audio, but then ALSA wouldn't load at all. My replacement desktop system is too modern to support such old equipment as the Audiophile, it would be nice to have it working in the old desktop system where it worked fine for many years, but now no longer works. -- David W. Jones gnome(a)hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com

11 years, 3 months

dynamic range analysis tool

by Raffaele Morelli

Hi, is there any linux tool which can produce plots like this? http://i46.tinypic.com/351iyqq.jpg I am actually using a combination of audacity (export sample data) and R to plot. regards /r

11 years, 3 months

Seeking computer musicians for a survey on how do computer musicians develop computer musical instruments or applications?

by Abram Hindle

Hello, I apologize for the intrusion but the linux audio users list hosts exactly the people we are looking for: computer musicians who code. We're trying to figure out how software engineering research can help computer musicians or if computer musicians need any help. Are you a computer musician? Do you code in languages geared towards music (e.g., Pure Data, Max MSP, Chuck, SuperCollider, etc.) or make music in other programming languages like C, Java, C++, Javascript, etc.? If so, we would like to hear from you! We the PIs, Gregory Burlet and Abram Hindle, are from the Department of Computing Science at the University of Alberta and are conducting a survey of computer musicians to investigate how this demographic of software developers program musical instruments or applications. Please visit the survey invitation website <http://webdocs.cs.ualberta.ca/%7Egburlet/musiccoders_survey.html> and click the "I consent, take me to the survey" button to complete the survey (if you consent). The survey will take 5 to 10 minutes. http://webdocs.cs.ualberta.ca/%7Egburlet/musiccoders_survey.html Thanks for your time! Gregory Burlet and Abram Hindle Graduate Student & Assistant Professor Department of Computing Science University of Alberta CANADA

11 years, 3 months

jack: interface selection has no effect

by Max

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 selecting a different interface in qjackctl has no effect. i can't use the external soundcard, because jack stays always with the internal one. any ideas what causes this symptom? i am running pulse on top of jack like this: http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack m -----BEGIN PGP SIGNATURE----- Version: GnuPG v1 iEYEARECAAYFAlPlEr0ACgkQ3EB7kzgMM6IMsgCfXLjIAxmgzISXQeGRpF9QMoef c0AAn3Hm+OvgLZtTWH/sKCSBhWTZpVrt =PX5E -----END PGP SIGNATURE-----

11 years, 3 months

vsts and wine startup times

by James Stone

I have been playing around with a new program that does win->linux vsts called airwave. It seems pretty nice as it does xembed properly so all menus etc. appear correctly, and the vst performance seems acceptable (fewer xruns than vst-bridge). I hit a problem when first running it which turned out that it was due to the host not waiting long enough for the vst to initialise. It hard coded a maximum wait of 3 seconds.. Whereas on my system, wine takes around 24 seconds to start first time or around 4 seconds on subsequent times. I have been in discussion with the dev about increasing the timeout. He changed it to 8s which works OK apart from first run on my system. So my questions: 1) what is the longest startup time for wine that would be reasonable to assume? 2) is 24 seconds abnormal for a reasonably recent (last 3years - AMD A6-3670 ) desktop system? James

11 years, 3 months

Header missing to build Zita AT 1

by Ralf Mardorf

Hi, building http://kokkinizita.linuxaudio.org/linuxaudio/downloads/zita-rev1-0.2.1.tar.… did work, but building http://kokkinizita.linuxaudio.org/linuxaudio/downloads/zita-at1-0.2.3.tar.b… failed. [rocketmouse@archlinux source]$ make g++ -O2 -ffast-math -Wall -MMD -MP -DVERSION=\"0.2.3\" -DSHARED=\"/usr/local/share/zita-at1\" -march=native -I/usr/X11R6/include `freetype-config --cflags` -c -o zita-at1.o zita-at1.cc In file included from jclient.h:28:0, from zita-at1.cc:29: retuner.h:27:28: fatal error: zita-resampler.h: No such file or directory #include <zita-resampler.h> ^ compilation terminated. <builtin>: recipe for target 'zita-at1.o' failed make: *** [zita-at1.o] Error 1 Regarding to the info given by the INSTALL text only clthreads and clxclient are needed dependencies. Even the headers provided by zita-resampler are missing zita-resampler.h [rocketmouse@archlinux zita-at1-0.2.3]$ pacman -Ql zita-resampler | grep resampler.h zita-resampler /usr/include/zita-resampler/resampler.h zita-resampler /usr/include/zita-resampler/vresampler.h zita-resampler /usr/share/doc/zita-resampler/resampler.html The RME HDSPe AIO still doesn't provide all ADAT IOs as jack ports, so I can't use a 19" reverb, that's why I will give Zita Rev 1 a shot, IOW I already got what I need and since I'm in the middle of a home studio production and I don't need AT 1, I won't trial by error how to build AT 1, but it would be nice to test it. How can I get rid of this fatal error? Regards, Ralf

11 years, 3 months

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Linux-audio-user