Linux-audio-user October 2005

linux-audio-user@lists.linuxaudio.org

174 participants
148 discussions

[linux-audio-user] audio software review

by Vedran Vucic

Dear colleagues, I wrote audio software review about some free software audio packages for audio recording, mixing, managing radio station. You can find it on http://www.gnucentar.org.yu/Audio_Tools_Scan.pdf please feel free to send me your comments. With best wishes, Vedran Vucic

18 years, 7 months

[linux-audio-user] Re: HDSP9652 and maybe driver problems

by Mark Knecht

> > > > Hi Michele, > > I know you've had this problem for awhile. Sorry about that. > > > > Right now one of my HDSP 8652 outputs is hooked to an ADAT AI-3. > > Not as nice as your RME converter but reasonable. I think I could > > test it if you could give me some clear instructions on how to do this > > quickly. I don't have enough time to figure it out by myself. IS it > > really as easy as connecting an output to an input? > > Hi Mark, > > it is really easy in fact! > Just download rec_imp > (http://www.duffroomcorrection.com/wiki/Simple_Automated_IR_Measuring_Tool) > and compile it (simply tipe Make). Yes, I tried this yesterday. It doesn't compile because my machine is AMD64... > After this run it with ' rec_imp rec.pcm 44100 10 21000 45 0:0 0:0' where > rec.pcm will be the recorded impulse response, 10 the starting frequency, > 21000 the end frequency and 45 the duration in seconds of the log sine sweep > signal. 0:0 and 0:0 are input and output channels: I do not remember which > is the input and output but simply running 'rec_imp' with no parameters > display the syntax. > > Leave HDSPMixer oper while you are running rec_imp, so you can see if > recording channel starts with some delay. > > Also if you can send me the recorded impulse responses at > michele.spinoloATgmail.com I can run the boring part for you. Will do if I ever get it working. > > > > > One question for you. Is the problem identical no matter what ADAT > > port you use on the HDSP 9652? > > Did not try this: do you think it could solve the problem? One other idea. Remove the two ADAT cables going to your ADI-8 Pro and then loop on back on itself so that the ADAT output goes back to the ADAT input. This will return whatever signal you send and allow to you determine if the problem is in the HDSP 9652 or in the ADI-8 Pro. Cheers, Mark > > Thanks! > Michele >

18 years, 7 months

[linux-audio-user] Problem with es1371 card - no PCM volume controls!

by Stuart Allie

Hi All, I just upgraded from RedHat 9 to Fedora Core 3 + PlanetCCRMA. FC3 correctly identified both the onboard sound (intel8x0) and the pci card (ens1371). While the onboard sound works perfectly, I can't get any sound out of the pci card. The mixer for the pci card doesn't show any PCM controls, so I can't unmute and set the PCM volume. There's a master volume, phono, aux, CD, ... but no PCM. Some info that might help: - /etc/modprobe.conf is pretty simple: alias char-major-116 snd alias snd-card-0 snd-ens1371 option snd-card-0 index=0 alias snd-card-1 snd-intel8x0 option snd-card-1 index=1 alias char-major-14 soundcore alias sound-slot-0 snd-card-0 alias sound-slot-1 snd-card-1 - Alsa is 1.0.9 (rc1) from the last planet cdrom image for FC3. - /proc/asound/card0/oss_mixer has a line like this: PCM "" 0 - amixer doesn't list a "PCM" simple mixer control. - when installing FC3, it successfully played a sample sound through the ens1371 card. (FC3 uses alsa 1.0.6) Any suggestions? Thanks, Stuart ------------------------------------------------ Stuart Allie stuart -dot- allie -at- hydro -dot- com -dot- au musicians.opensrc.org/StuartAllie allie.opensrc.org ------------------------------------------------

18 years, 7 months

[linux-audio-user] Re: linux-audio-user Digest, Vol 25, Issue 12

by Gavin Stevens

> Date: Wed, 5 Oct 2005 11:05:59 +0100 (BST) > From: "Dave Griffiths" <dave(a)pawfal.org> > Subject: [linux-audio-user] [LAM] some music > To: linux-audio-user(a)music.columbia.edu > Message-ID: > <54989.193.203.82.226.1128506759.squirrel(a)www.webmail.pawfal.or > g> > Content-Type: text/plain;charset=iso-8859-1 > > Hi all, > > Some new tracks from me here: > http://www.pawfal.org/patterncascade/ > > cheers, > > dave Hi Dave, I really enjoyed pattern A. Pattern B might take a bit more getting used to regarding the tuning used. However, I wrote some pieces for harpsichord a few years ago & I spent a lot of time listening to unequal temperaments like quarter-comma meantone & the beautiful Vallotti tuning (in which tuning the pieces were performed). After this, I - a hardened "equal-temperament" pianist - found the equal tuning of the piano rather sour for a good few weeks. The Live track sounded great, once I adjusted to the industrial sound-world. Thanks for sharing these tracks & I look forward to more. Gavin.

18 years, 7 months

[linux-audio-user] AGNULA nightmares, again

by Dave Phillips

Greetings: How do I set up the networking in AGNULA/Demudi 1.2.1 ? I've installed the system on a second disk in my desktop system. I want to sh*tcan the 2.6.12 kernel asap, but I have no network connection. I fixed /etc/network/interfaces and can see eth0 in ifconfig's report, but where is the tool to configure the interface ?? Best, dp

18 years, 7 months

[linux-audio-user] Freecycle 0.51alpha and agnula

by Hector Centeno

Hi, I've been trying to build Freecycle 0.51alpha and I can't get to solve a series of errors related to libinstpatch. I've followed all the instructions in the README regarding code changes to portmidi, libinstpatch, etc. I wonder if anyone here has succeeded in building it using agnula? I found a deb package but it is for Debian unstable. There are also static builds available but again they are built using higher versions of libraries. Anyone knows of a deb for Sarge? thanks! Hector

18 years, 7 months

[linux-audio-user] Re: New Virtual Keyboard Proposition

by philicorda

On Wed, 2005-10-05 at 09:19 -0400, linux-audio-user-request(a)music.columbia.edu wrote: > From: Carlo Capocasa <capocasa(a)gmx.net> > Subject: [linux-audio-user] New Virtual Keyboard Proposition > To: linux-audio-user(a)music.columbia.edu > Message-ID: <di054s$lp7$1(a)sea.gmane.org> > Content-Type: text/plain; charset=ISO-8859-15; format=flowed > > Hi there! > > I've been using the Virtual Keyboard quite often and it's great, a > regular computer keyboard can be a great two-octave non-dynamic > keyboard. I don't see why it couldn't be used in concert settings, > especially for electronic/synth-samples based music. I enjoy using this too. With monophonic sounds it's a bit like tapping on a guitar. :) > > The currently available Virtual Keyboard does have its issues though: > > * No more than about three notes can be played at once Is this a limitation of the keyboard scanning electronics, or a software thing, I wonder? I have some vague memory that it's possible to change the baud rate of ps/2 devices too. There are many good points in your post. I would add some more... I think it would be interesting to try a layout that does not mimic a piano keyboard. There are twelve keys from 'a' to '#', from 'q' to ']', and '1' to '=' on my keyboard. These could be chromatic notes, with 'a','q',and '1' being the lowest note of each octave. This would give three octaves on the keyboard, and I don't think it would be harder to play. I had a go at doing this in pd, but got stuck converting the key codes into midi note numbers. I think that with the mouse it would be good if there was an option for the controllers to be activated on left click and reset when the button was released... as there is no 'null' point on a mouse, it would be hard to use it as a pitch bend/mod without leaving an offset when you want both to return to zero.

18 years, 7 months

[linux-audio-user] snd-usb-usx2y on amd64?

by Arnold Krille

Hi all, sorry for cross-posting. I just wanted to know wether there are any known problems with the snd-usb-usx2y on amd64? 'Cause I got such a laptop now and experience some trouble with my tascam. :-( If I use it at non-realtime and high-realtime is works (at least for the first start of jackd) but realtime and/or low-latency just freezes the system. Maybe there is something I can do to increase the verbosity of the driver while crashing? The data of my machine: kernel is gentoo-2.6.13-r2 with the bundled alsa-1.0.9b, the usb-driver is ohci-hcd. The machine is a turion64 and I have acpi turned on... Thanks in advance, Arnold -- visit http://dillenburg.dyndns.org/~arnold/ --- Wenn man mit Raubkopien Bands wie Brosis oder Britney Spears wirklich verhindern könnte, würde ich mir noch heute einen Stapel Brenner und einen Sack Rohlinge kaufen.

18 years, 7 months

Re: [linux-audio-user] Software for Sine Wave Generation

by Kjetil Svalastog Matheussen

Douglas Ward: > What program is suggested for replacing this toolchain? I think Csound can > accomplish this. (Can this be accomplished with Audacity+Nyquist, too? Is > there a plug-in that will play multiple frequencies at once?) Are there any > favourite command line applications that can be used to produce multiple sine > waves at once? Csound seems overly complex for this application, but I do not > know of alternatives. Is there anything other than Csound? I just want to > make sure I'm not overlooking a great specialized program before digging into > Csound. This should be in a FAQ... Theres a lot of programs that let you do this. Yes, it can be done with Nyquist or CSound. Other alternatives are pd, supercollider, clm, snd, galan, ecasound. http://twiki.org/cgi-bin/view/Wikilearn/LA_SynthesisTools But there are many others, Octave for example. For the job you want, a graphical program such as pd, galan or spiralsynthmodular might be the quickest solution. (others?) --

18 years, 7 months

[linux-audio-user] snd_intel8x0 Channels count non available

by Jody Noury

Hi, I have two "cards": an onboard sis7012 (snd_intel8x0) and a pci Ensoniq ES1371 (snd_ens1371). cat /proc/asound/cards 0 [SI7012 ]: ICH - SiS SI7012 SiS SI7012 with CMI9738 at 0xdc00, irq 10 1 [AudioPCI ]: ENS1371 - Ensoniq AudioPCI Ensoniq AudioPCI ENS1371 at 0xc800, irq 10 Both works. I'm trying to make a ~/.asoundrc , first test is to give a "comprehensive" name: pcm.hwEnsoniq { type hw; card 1; device 0; } pcm.hwSis { type hw; card 0; device 0; } I can use hwEnsoniq: aplay /usr/share/sounds/KDE_Window_UnSticky.wav -D hwEnsoniq But: aplay /usr/share/sounds/KDE_Window_UnSticky.wav -D hwSis Playing WAVE '/usr/share/sounds/KDE_Window_UnSticky.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono aplay: set_params:887: Channels count non available I can only use Sis chip with a parameter like: pcm.plugSis { type plug slave { pcm "hw:0,0" } Then: aplay /usr/share/sounds/KDE_Window_UnSticky.wav -D plugSis works: Playing WAVE '/usr/share/sounds/KDE_Window_UnSticky.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono Does someone could explain why I could not set up Sis as hw type ? My aim is to be able to use this two cards as output without time syncro (need not jack) but as predict It's a little hard to do for someone like me :) Thanks, Jody

18 years, 7 months

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Linux-audio-user October 2005