Linux-audio-user January 2007

linux-audio-user@lists.linuxaudio.org

187 participants
180 discussions

[linux-audio-user] audio record/upload box

by rene christen

greetings, got a peculiar problem for the list... we're trying to set up a website to record and upload roughly hour long audio to for streaming. we'd be interested in any ready to go solutions if anyone can recommmend any but if we need to build our own we'd prefer to use linux. what we'd need is a linux box with a really simple interface to start/stop recording, then maybe automatically trim silence off the end of the file and upload it to a web server in a format ready for streaming. any advice on ways to go about this would be greatly appreciated. what we'd most likely do is have a web-interface for some perl scripts which would then control some command line apps like ecasound. any advice greatly appreciated, rene.

17 years, 8 months

[linux-audio-user] dssi-vst

by osbusters＠softhome.net

Hi Well...I've exhausted myself trying to get this to work, and naturally a new poster to a mailing list should introduce himself etc But it's 5:45AM, I've got a wicked cold, my throat's on fire, and because I can't sleep...I've decided (unwisely) to keep trying to get this to work. I'm using an rpm (mdk) of rosegarden I rebuilt from a source rpm...it seems to work fine, and I can get those cruddy "trivial" synths to work fine. I guess I should be happy after hoping to record in linux now for 5 years...but it won't be a studio till I can get my Arturia vsti's, my native instruments purring along in time. So, How do I get dssi-vst to see my vsti's? I've tried everything I can think of, but nothing seems to work. here's the message from my last attempt: [ ~]# jack-dssi-host dssi-vst.so:Pro-53.dll jack-dssi-host: Warning: DSSI path not set jack-dssi-host: Defaulting to "/usr/local/lib/dssi:/usr/lib/dssi:/root/.dssi" VST_PATH not set, defaulting to /root/vst:/usr/local/lib/vst:/usr/lib/vst DSSI_PATH not set, defaulting to /root/.dssi:/usr/local/lib/dssi:/usr/lib/dssi RemoteVSTClient: executing /usr/local/lib/dssi/dssi-vst/dssi-vst-scanner /tmp/rplugin_qry_pxhrvE RemoteVSTClient: executing /usr/lib/dssi/dssi-vst/dssi-vst-scanner /tmp/rplugin_qry_pxhrvE DSSI VST plugin scanner v0.3 Copyright (c) 2004 Chris Cannam - Fervent Software Plugin scanner version mismatch VST_PATH not set, defaulting to /root/vst:/usr/local/lib/vst:/usr/lib/vst DSSI VST plugin scanner v0.3 Copyright (c) 2004 Chris Cannam - Fervent Software dssi-vst-scanner: Failed to open output file /tmp/rplugin_qry_pxhrvE : No such file or directory dssi-vst-scanner: Defaulting to stdout ÀVST_PATH not set, defaulting to /root/vst:/usr/local/lib/vst:/usr/lib/vst jack-dssi-host: Error: Plugin label "Pro-53.dll" not found in library "dssi-vst.so" The things that stand out, of course are: VST_PATH not set & DSSI_PATH not set, + the "plugin-scanner mismatch" I am at my wits end with this. A few more days and I slog back to my windows DAW. Any help would be great!

17 years, 8 months

[linux-audio-user] Re: Patch one channel to both channels in ALSA?

by Peter Clarke

Patrick, thanks for the input. Just to answer your question, I'm using tvtime. There are other tv viewer apps available, but I haven't tried them yet. Actually, my gut feeling is that the problem is in the driver (saa7134), not the app. Peter C

17 years, 8 months

[linux-audio-user] How Does One get 2 8-KB Streams (/dev/dsp*2)?

by Martin McCormick

I have ALSA on a Linux system with a 2.6.5 kernel and a CS4236 sound card. I have written a VOX or Voice Actuated Relay recording program that receives audio from a radio such as a police scanner or amateur radio receiver and stores it in a file as unsigned 8-bit audio of the type you get if you cat /dev/dsp >somefile. My program just opens /dev/dsp and sits there looking for samples that indicate audio being received and begins recording until there is a period of silence at which time it stops recording and continues to keep /dev/dsp open for more audio. It occurred to me that it should be possible to open either /dev/dsp or some other device for 2 8-K streams so that a similar program could read them both and store audio in files corresponding to the right and left channels of the card. Basically, I am not sure what is the best practice for opening the 16-bit, 8-K device. I think once that is solved, my program could be made to act as 2 independent VOX's. While doing a Google search, I found a OSS-sound document that is several years out of date and mentioned such devices as /dev/dspw which is a 16-bit version of /dev/dsp but modern Linux does not appear to have anything other than /dev/audio and /dev/dsp. Since this application needs to be portable, I want to set up the opening of the 16-bit 8-K channel in a standard way. Thanks for any pointers on documentation you might have. I have found the How-To document for Linux sound which mentions the standard /dev sound devices, but it is geared more toward helping folks get their Linux sound running rather than writing programs using the devices. Many thanks. Martin McCormick WB5AGZ Stillwater, OK Systems Engineer OSU Information Technology Department Network Operations Group

17 years, 8 months

[linux-audio-user] Re: SB450 any good? why does jack fail?

by sam＠bristolwireless.net

> Hi all > > 1, firstly i was wondering is the sb450 is any good? is it worth me trying to > use this card for audio mixing? > > 2, I have installed 64studio, by default the sound works and with a small > ammount of tweaking i can capture audio using audacity. However sound does > not > work from firefox unless i first run alsaconf. Is there a way to have working > sound on firefox and other programs using alsa? and how can i make it so that > i > do not have to run alsaconf on every boot? > > 3, Jack fails to start; > 15:12:56.334 Could not open ALSA sequencer as a client. MIDI patchbay will be > not available. > ALSA lib seq_hw.c:457:(snd_seq_hw_open) open /dev/snd/seq failed: No such > file > or directory > How can i fix this please? > > Thanks > > Sam > > This is my full hardware spec: http://hp-nx6325.pbwiki.com/ ---------------------------------------------------------------- This message was sent via Slackmail, web mail from Psand.net.

17 years, 8 months

[linux-audio-user] KMix for Audiophile 24/96

by millward

For a few months now I've been using Audacity with an on-board sound circuit, the ASUS A7N8X motherboard's onboard AC97 audio controller. It's been OK, but maybe I can get a better sound with an M-Audio Audiophile 24/96. I got one used, with no documentation, and I can't see any at the M-Audio site either. My problem is I don't understand the KMix mixer for the Audiophile 24/96. I'm just a simple guitar player, not an audio technician. The AC97's mixer with it's Volume, PCM, LineIN, and LineGain ... it was easy to figure out. But this mixer lists stuff I've no idea what it means. Could anyone tell me what these KMix settings do? Output: DAC ( two of them ) Multi ( 10 of them ) MultiTrackPeak MultiTrackVolumeRate ( I've got the Audiophile's two RCA outputs plugged into my sterio amp ) Input: IEC 958 Multi ( two of them ) DAC ( two of them... shouldn't these be ADC ? ) H / W Multi ( two of them ) MultiTrackVolume Rate Switches: IEC 958 ( two of them set to PCM Out ) Deemphasis ( set to off ) H / W ( two of them set to PCM Out ) MultiTrack Internal Clock ( set to 44100 ) MultiTrack Internal Clock Default ( set to 44100 ) And one last question: Is this M-Audio Audiophile 24/96 sound card really any good? If not, I may as well remove it and use what I already understand. Thank you for your patience, and ANY information would be very welcome!

17 years, 8 months

[linux-audio-user] Patch one channel to both channels in ALSA?

by Peter Clarke

Is there a neat software solution for patching, say, the left channel of PCM output to both channels of the line output of my soundcard in ALSA? I need this because I've just installed a TV card, which works great except that the dual language function doesn't seem to be supported. So I get English in one speaker and Chinese in the other. I'd rather have the same in both. I thought of doing it with the JACK patchbay, but the TV output unfortunately doesn't show up in JACK. Thanks for any help, Peter C

17 years, 8 months

[linux-audio-user] music made with linux

by Atte André Jensen

Hi and merry Christmas + happy new year to all! This is a song I wrote last week and recorded using muse, specimen, zyn and ardour. It's the first song I did with my new SE usb2200a usb microphone. http://web4490.web03.talkactive.net/demo/ro/ro.mp3 The dust haven't settled, so some things (for instance the synth solo part) are subject to change, but I think it's gonna end up sounding something along these lines... (Christian) lyrics in danish, feedback welcome. -- peace, love & harmony Atte http://www.atte.dk | quintet: http://www.anagrammer.dk | compositions: http://www.atte.dk/compositions

17 years, 8 months

[linux-audio-user] Om/Ingen status

by rob

Hi all Does anyone known the current status of Om/Ingen? The http://www.nongnu.org/om-synth/ page says Om _was_, but not that Ingen _is_ :) Anyway, in case anyone is interested.... Exec summary: Om engine segfaults when attempting to load the vocoder.so (http://www.sirlab.de/linux/descr_vocoder.html) plugin. The plugin appears OK since it loads and executes fine under Jack-Rack. cheers R (gdb) run Starting program: /usr/bin/om (no debugging symbols found) (no debugging symbols found) (no debugging symbols found) (no debugging symbols found) [Thread debugging using libthread_db enabled] [New Thread -1212094800 (LWP 27822)] [SlowEventQueue] Starting. [New Thread -1212789856 (LWP 27825)] [PostProcessor] Starting. [New Thread -1221182560 (LWP 27826)] [AlsaDriver] Successfully opened ALSA sequencer. [OSC] Started OSC server at osc.udp://castro.ntlworld.com:16180/ [Main] Successfully locked all memory. [New Thread -1229759584 (LWP 27827)] lash_open_socket: could not connect to host 'localhost', service '14541' lash_comm_connect_to_server: could not create server connection lash_init: Not attempting to start daemon server automatically lash_init: could not connect to server 'localhost' - disabling LASH [LashDriver] Failed to connect to LASH. Session management will not function. [New Thread -1238152288 (LWP 27828)] [New Thread -1246544992 (LWP 27837)] [JackDriver] Activated Jack client. [New Thread -1254937696 (LWP 27838)] [AlsaDriver] Started realtime MIDI thread (SCHED_FIFO, priority 20) [OSC] Registered client osc.udp://castro.ntlworld.com:15799/ (1 clients) [JackDriver] Enabling. [NodeFactory] DSSI_PATH is empty. Assuming /usr/lib/dssi:/usr/local/lib/dssi:~/.dssi Now load the vocoder LADSPA plugin... Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1212789856 (LWP 27825)] 0xb7c7c2c3 in strlen () from /lib/libc.so.6 (gdb) thread apply all bt Thread 7 (Thread -1254937696 (LWP 27838)): #0 0xffffe410 in __kernel_vsyscall () #1 0xb7cc5553 in poll () from /lib/libc.so.6 #2 0x08077964 in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #3 0xb7f1f294 in start_thread () from /lib/libpthread.so.0 #4 0xb7cce32e in clone () from /lib/libc.so.6 Thread 6 (Thread -1246544992 (LWP 27837)): #0 0xffffe410 in __kernel_vsyscall () #1 0xb7cc5553 in poll () from /lib/libc.so.6 #2 0xb7f3d181 in jack_client_close () from /usr/lib/libjack.so.0 #3 0xb7f40716 in jack_drop_real_time_scheduling () from /usr/lib/libjack.so.0 #4 0xb7f1f294 in start_thread () from /lib/libpthread.so.0 #5 0xb7cce32e in clone () from /lib/libc.so.6 Thread 5 (Thread -1238152288 (LWP 27828)): #0 0xffffe410 in __kernel_vsyscall () #1 0xb7cc5553 in poll () from /lib/libc.so.6 #2 0xb7f332b9 in lo_server_recv_noblock () from /usr/lib/liblo.so.0 #3 0xb7f33f23 in lo_server_thread_start () from /usr/lib/liblo.so.0 #4 0xb7f1f294 in start_thread () from /lib/libpthread.so.0 #5 0xb7cce32e in clone () from /lib/libc.so.6 Thread 4 (Thread -1229759584 (LWP 27827)): #0 0xffffe410 in __kernel_vsyscall () #1 0xb7f257f6 in __nanosleep_nocancel () from /lib/libpthread.so.0 #2 0x0806e967 in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #3 0xb7f1f294 in start_thread () from /lib/libpthread.so.0 #4 0xb7cce32e in clone () from /lib/libc.so.6 Thread 3 (Thread -1221182560 (LWP 27826)): #0 0xffffe410 in __kernel_vsyscall () #1 0xb7f257f6 in __nanosleep_nocancel () from /lib/libpthread.so.0 #2 0x08074141 in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #3 0xb7f1f294 in start_thread () from /lib/libpthread.so.0 #4 0xb7cce32e in clone () from /lib/libc.so.6 Thread 2 (Thread -1212789856 (LWP 27825)): #0 0xb7c7c2c3 in strlen () from /lib/libc.so.6 #1 0xb7dcb206 in std::string::compare () from /usr/lib/gcc/i686-pc-linux-gnu/4.1.1/libstdc++.so.6 #2 0x08078ec5 in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #3 0x0804fad6 in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #4 0x0804fbfa in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #5 0x0806fa77 in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #6 0x0806e454 in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #7 0xb7f1f294 in start_thread () from /lib/libpthread.so.0 #8 0xb7cce32e in clone () from /lib/libc.so.6 Thread 1 (Thread -1212094800 (LWP 27822)): #0 0xffffe410 in __kernel_vsyscall () #1 0xb7c9d316 in nanosleep () from /lib/libc.so.6 #2 0xb7cc859a in usleep () from /lib/libc.so.6 #3 0x08053672 in std::operator+<char, std::char_traits<char>, std::allocator<char> > () #4 0x0804b114 in ?? () #5 0x08080480 in ?? () #6 0x08080480 in ?? () #7 0x08080428 in ?? () #8 0x0804a541 in _init () #9 0xb7c28878 in __libc_start_main () from /lib/libc.so.6 #10 0x0804af11 in ?? () #0 0xb7c7c2c3 in strlen () from /lib/libc.so.6 Portage 2.1.1-r2 (default-linux/x86/2006.1/desktop, gcc-4.1.1, glibc-2.4-r3, 2.6.16-rt29 i686) ================================================================= System uname: 2.6.16-rt29 i686 Intel(R) Pentium(R) D CPU 3.40GHz Gentoo Base System version 1.12.6 Last Sync: Fri, 05 Jan 2007 03:30:01 +0000 app-admin/eselect-compiler: [Not Present] dev-java/java-config: [Not Present] dev-lang/python: 2.4.3-r4 dev-python/pycrypto: 2.0.1-r5 dev-util/ccache: [Not Present] dev-util/confcache: [Not Present] sys-apps/sandbox: 1.2.17 sys-devel/autoconf: 2.13, 2.60 sys-devel/automake: 1.4_p6, 1.5, 1.6.3, 1.7.9-r1, 1.8.5-r3, 1.9.6-r2 sys-devel/binutils: 2.16.1-r3 sys-devel/gcc-config: 1.3.13-r4 sys-devel/libtool: 1.5.22 virtual/os-headers: 2.6.17-r2 ACCEPT_KEYWORDS="x86" AUTOCLEAN="yes" CBUILD="i686-pc-linux-gnu" CFLAGS="-O2 -march=i686 -pipe" CHOST="i686-pc-linux-gnu" CXXFLAGS="-O2 -march=i686 -pipe" MAKEOPTS="-j3" PORTDIR_OVERLAY="/usr/portage/local/layman/pro-audio"

17 years, 8 months

[linux-audio-user] Fedora 5 and mp3...

by Russell Hanaghan

I have, for the first time, set up FD5 and CCRMA for a look see. I like how it all works and the RT kernel runs sweet and gives very low Xrun free latency. My only curiosity is that Redhat seem to be not interested in pushing the mp3 licensing envelope. Xmms and Amarok wont play my extensive list of MP3's citing Redhat does not support the format due to said licensing issues. Other than building from source, are there any players out there that do support MP3 under FC5? I know Ogg is better, etc but car stereos, other players in my house are all using mp3. I'd prefer to stay with the one format for now. TIA. Russell

17 years, 8 months

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Linux-audio-user January 2007