Linux-audio-user

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13 participants
20678 discussions

[linux-audio-user] Re: [linux-audio-dev] Soundcard spotting

by Kai Vehmanen

On Tue, 22 Oct 2002, Peter L Jones wrote: >>> * MidiMan Delta Audiophile 2496 (Envy24) >>> * Creative SB PCI 128 (ES1371) >> I've used both of these extensively with JACK and numerous other ALSA apps >> and they work really well (full-duplex, low-latency use). Other > Heh. Now, one of these I have in my machine ((PII vintage) Celeron 400) > already. The other would set me back £150. Your comment makes me think > there's little to choose between them. So, simply upgrading my soundcard > from a £15 low end consumer-oriented unit to something costing 10 times the > price looks like getting me nothing. Or am I missing something? :-( Well, yes. ES1371 brings you 2ch in+out with max 48000Hz sampling rate, and 16bit sample resolution. Midiman 2496 on the other hands provides up to 96kHz sampling rate, 24bit sample resolution, 2ch in+out and digital in+out. Check the specs from manufacturer's site. And btw, I confused Audiophile with Delta44 (which I have, has 4ins + 4outs, no digital in/out). Both are based on the envy24 chipset, should perform equally well. Please, correct me if I'm wrong. >> - GUS MAX (this very, very old ISA-card can still beat a number of >> today's crappy chipsets... I don't know whether to cry or laugh ;)) > I noticed that the ENS1371 seems to have a better rating on one site I looked > than to EMU10K, so this doesn't surprise me! Yup, I'll probably never get tired of the following slogan: "sb128 (ens1371) is the best creative card as it's the one they didn't make themselves". :) Ok, maybe the current SB cards are better, but I'll never forgive the company the disappointment their AWE64Gold caused me. Such waste of money! ;) >> All in all, most of the PCI-cards supported by ALSA have fairly good >> drivers. > But how do I compare one card with another? What should I be looking for? > How can I tell which will reduce the load on my computer and which will > increase the load? Is there any difference? Well, it depends on what you want to do. How many channels you need in and/or out, do you need high-quality recording, do you need digital ins/out, do you need hardware support for multi-open, etc, etc? I'm not a hardware expert so I can't answer to all these questions, but I can tell about the criteria I used when I selected my last card. My primary use is multitrack recording and mixing. I needed capability to record >2 channels, high-quality a/d and good support for low-latency and full-duplex. My choice was midiman delta44. It has 4 ins, an external a/d&a/d box (important for high-quality conversion), good ALSA drivers and wasn't too expensive (ie. a lot cheaper that the RME cards for instance). So far I've been very satisfied with this purchace. PS Let's cross-post to linux-audio-user. That and alsa-user are probably the best forums for this discussion. -- http://www.eca.cx Audio software for Linux!

23 years, 1 month

[linux-audio-user] routing oss midi to second alsa midi port

by Olivier

Hi all, I'm trying to get the oss midi port emulation to use the second alsa midi port, and I guessed that must be possible with the .asoundrc, module parameters or some other configuration. But I cannot find how, does anyone have an example? thanks, Olivier

23 years, 1 month

[linux-audio-user] DJCJ Linux Audio Developers Tech Support database.

by Patrick Shirkey

I'm pleased to anounce that a database for people willing to provide Tech Suppport to their local community has been setup. The service is free of charge and hosted at http://www.djcj.org/ The purpose of this database is to promote the professional arm of the Linux Audio Developers community. It is intended to be of use to potential clients who may be interested in getting a Linux Audio system working but don't have the time or background knowledge to do the installation or system maintainance? The people and businesses presented in the database are not endorsed or guaranteed by DJCJ.org but they are active members of the Linux Audio Community. Payment for services received is encouraged. Rates are decided by the parties involved. This is intended to be a database for professional tech support. Please let people know about it so that we can show the world we are more than just a bunch of amateurs hacking in our spare time. The database currently provides a very simple interface for adding your contact details and there is also a contact form provided for potential clients to easily get hold of you. Special thanks to Steve Harris and Antti Boman for assistance with the internal code. It is guaranteed that the interface will become much more user configurable over time. Any feedback is welcome and appreciated. -- Patrick Shirkey - Boost Hardware Ltd. For the discerning hardware connoisseur Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide ======================================== "Um...symbol_get and symbol_put... They're kindof like does anyone remember like get_symbol and put_symbol I think we used to have..." - Rusty Russell in his talk on the module subsystem

23 years, 1 month

[linux-audio-user] Sweep 0.5.9 -- Scrubby Speex!

by Conrad Parker

Sweep 0.5.9 Development Release ------------------------------- Sweep is an audio editor and live playback tool for GNU/Linux, BSD and compatible systems. It supports many music and voice formats including WAV, AIFF, Ogg Vorbis, Speex and MP3, with multichannel editing and LADSPA effects plugins. Inside lives a pesky little virtual stylus called Scrubby who enjoys mixing around in your files. This development release is available as a source tarball at: http://prdownloads.sourceforge.net/sweep/sweep-0.5.9.tar.gz?download Sweep now supports Speex, a special purpose speech codec designed for efficient Voice over IP (VoIP) and file-based compression. Speex is free, open and unpatented; more information is available at http://www.speex.org/. This release also includes improved handling of the main volume and pitch controls, contributed by Zenaan Harkness. Screenshots: http://www.metadecks.org/software/sweep/screenshots/ Audio demos: http://www.metadecks.org/software/sweep/demos.html Sweep is designed to be intuitive and to give you full control. It includes almost everything you would expect in a sample editor, and then some: * precise, vinyl like scrubbing * looped, reverse, and pitch-controlled playback * playback mixing of unlimited independent tracks * looped and reverse recording * internationalisation * multichannel and 32 bit floating point PCM file support * support for Ogg Vorbis, MP3 and Speex compressed audio files * LADSPA 1.1 effects support * multiple views, discontinuous selections * easy keybindings, mouse wheel zooming * unlimited undo/redo with fully revertible edit history * multithreaded background processing * shaded peak/mean waveform rendering, multiple colour schemes Sweep is Free Software, available under the GNU General Public License. More information is available at: http://www.metadecks.org/software/sweep/ Thanks to Pixar Animation Studios and CSIRO Australia for supporting the development of this project. enjoy :) Conrad.

23 years, 1 month

RE: [linux-audio-user] changing speed without changing pitch

by Stuart Allie

Thanks Luis, I'll give it a try. And I'll check out the csound list too. Stuart > > Can anybody out there tell me what I (or csound :) did wrong? > > well, i won't try to tell you what might be wrong with your > orchestra, but > if you are interested in time-stretching using csound, try my > instrument > below. with it you can do time-stretch and/or pitch transpose > independently. [snip] > > good luck, > > lj > > (ps. aren't you interested in joining the csound mailing-list?)

23 years, 1 month

RE: [linux-audio-user] changing speed without changing pitch

by Stuart Allie

Hi, I *finally* got around to trying the csound method of changing the speed of an audio file without changing the pitch, but it didn't work for me :( I used the following "orc" file: > sr = 44100 > kr = 4410 > ksmps = 10 > nchnls = 1 > > instr 1 > kfreqscale = 1 > ispecwp = 0 > ktime line 1, p3, 0 > apvl pvoc ktime, kfreqscale, "file.pvc", ispecwp > out apvl > endin and then the "sco" file: > i1 0 12 > e with a 6 second sample, so this should double the audio to 12 seconds. I did the PV analysis like this: > csound -U pvanal -n 1024 -w 4 file.wav file.pvc and then ran csound via: > csound -W -R -d file.orc file.sco -o slow.wav and then I waited for a little while :) When csound was done, the ourput file was, well, interesting... Basically the input file was 6 seconds of single-note guitar playing, and the output was a weird (though not unpleasant :) whooshing-sweeping-phasing sound for 2 seconds, then a faint LFO-type sound to the end of the file. No sign of the original clip, slowed or otherwise. Can anybody out there tell me what I (or csound :) did wrong? Thanks in advance, Stuart

23 years, 1 month

[linux-audio-user] Evolution UC16 problems, modules.conf for usb-midi and usb-audio together

by m

hi all thanks for your help with compiling alsa, i've now got both usb-audio and usb-midi compiled... but i'm having problems getting midi running. the results of modprobe snd-usb-midi is a load of unresolved symbols... (as listed below)... i will also need to write a modules.conf which supports both my m-audio quattro *and* the Evolution UC16 - which i am somewhat stumped by... any help would be greatly appreciated, thanks m~ [root@hamish miriam]# modprobe snd-usb-midi /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_new_Rbc1a629f /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_transmit_peek_R2f3ef431 /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_receive_R4b303a51 /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_set_ops_R246c97c8 /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_transmit_ack_Rb639865e modprobe: insmod /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi .o failed /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_new_Rbc1a629f /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_transmit_peek_R2f3ef431 /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_receive_R4b303a51 /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_set_ops_R246c97c8 /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o: unresolved sy mbol snd_rawmidi_transmit_ack_Rb639865e modprobe: insmod /lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi .o failed modprobe: insmod snd-usb-midi failed -- 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 ***

23 years, 1 month

[linux-audio-user] RE: [music-dsp] linux and midi

by Joachim Backhaus

> What about using midi-thru boxes? they don't add latency to > the midi signal, > they just 'amplify' and 'split'. If you run 14 MIDI devices through one MIDI Port using MIDI Thru Boxes you get a worse timing compared to using a multiple Output Interfaces like the Steinberg Midex 8 oder the Midiman Midisport 8x8. But I don't know if there is a 8x8 Interface with Linux drivers. All of them implement a feature to reduce latency and I think that's a problem for the driver developers because the companies don't want to show the specifications for that in public. Regards, Joachim PS: I added the Linux-Audio-User group to the cc because this is linux specific question. > ----- Original Message ----- > From: "Tjeerd Sietsma" <tsietsma(a)hotmail.com> > To: <music-dsp(a)shoko.calarts.edu> > Sent: Tuesday, October 22, 2002 9:03 > Subject: [music-dsp] linux and midi > > > > > > > > I need to connect 14 midi devices (each using 1 > programmable midi channel) > > to a PC runnig linux. Midi thru isn't an option, since the > latency will be > > too high. Does anyone has hardware recommendations? > > > > dupswapdrop -- the music-dsp mailing list and website: > subscription info, > FAQ, source code archive, list archive, book reviews, dsp links > http://shoko.calarts.edu/musicdsp/ >

23 years, 1 month

[linux-audio-user] problem converting mp3 to wav

by Howard Sanner

I just searched the archives on this and the suggestions there didn't help. I'm trying to convert some mp3 files into wav files. First I tried having Audacity export a wav file. Then I tried loading the mp3 into Broadcast 2000 and having it render it to a wav. Finally I tried the suggestion in the archives of mpg123 -w filename.wav filename.mp3. In each case, when I tried to burn a CD from the resulting wav file, cdrecord reports "inappropriate audio coding in [filename]." I'm using Eroaster 2.0.10 as a front end to cdrecord. Audacity and Broadcast 2000 both load all these wav files and play them just fine. Anybody know what's going on here? Thanks. Howard Sanner flagstad(a)mindspring.com

23 years, 1 month

[linux-audio-user] problem converting mp3 to wav

by Howard Sanner

Apparently my subscription to the list got dropped. I guess that's why things have seemed so quiet lately in Linux audio land. I just searched the archives on this and the suggestions there didn't help. I'm trying to convert some mp3 files into wav files. First I tried having Audacity export a wav file. Then I tried loading the mp3 into Broadcast 2000 and having it render it to a wav. Finally I tried the suggestion in the archives of mpg123 -w filename.wav filename.mp3. In each case, when I tried to burn a CD from the resulting wav file, cdrecord reports "inappropriate audio coding in [filename]." I'm using Eroaster 2.0.10 as a front end to cdrecord. Audacity and Broadcast 2000 both load all these wav files and play them just fine. Anybody know what's going on here? Thanks. Howard Sanner flagstad(a)mindspring.com

23 years, 2 months

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Linux-audio-user