Linux-audio-user January 2010

linux-audio-user@lists.linuxaudio.org

166 participants
153 discussions

by Mark Rufino

> * On newly created sessions, the mute buttons > affect the meters but not the signal. I cannot reproduce this one. Mute works as it should. > Right click on mute and choose main outs. There are different points in the signal path where you can mute the signal. -- http://www.reverbnation.com/jeffreybrayne www.jeffreybrayne.com http://www.last.fm/music/Jeffrey+Brayne www.totalsoul.com www.bigjeffmusic.com

15 years, 4 months

Ardour - change colors of strips in mixer?

by Ectropic Harmony

Is it possible to change the colors in the strips/channels within the mixer window of Ardour? I have multiple groups of strips/channels that I'd like to color-code.

15 years, 4 months

about gvim.

by Philippe Hezaine

Hi, Sorry. A bit [OT]. Am I right if I say you could install gvim *without* gnome? At least it's like that on my Gentoo. In this case g means GUI and its dependencies are only GTK+ or even Motif. I need a solid confirmation. I can't find this information anywhere. -- Phil.

15 years, 4 months

you guys saw this digital, stringless electric guitar?

by david

http://www.misadigital.com/ OSS and Linux powered! -- David gnome(a)hawaii.rr.com authenticity, honesty, community

15 years, 4 months

Audio Levels with your delta 1010

by Mark Rufino

> On Mon, Jan 25, 2010 at 12:22:24AM -0600, Ectropic Harmony wrote: > >> I haven't had a chance to test the entire set-up of multiple >> connections just yet (see message below for what I'm thinking about >> trying, I don't know if it'll work) but I have managed to get a basic >> recording going with mics into mixer into delta interface into >> computer. >> >> One issue right now - >> >> Levels. >> >> Really low levels. >> > I'm not sure if, from that, I can determine whether or not I need a separate mic preamps type device. Any ideas? Hopefully it's just a matter of me incorrectly setting levels. :-\ But if I need a preamp device I'll definitely get one. When you saw low levels, how low? If you are using a mixer into your delta 1010 then you have mic preamps. If you are plugging a mic into your delta 1010 (which I assume you're not- that is only 1/4" inputs, right?, then those are line inputs and your mic needs amplification. Also, for your condenser mics you do have phantom power, right? Just checking, without that they will not make much, if any, sound. However, assuming things are working correctly, what level are you shooting for? digital 0 and analog 0 are not the same thing, different scales, so shooting for 0 on your mackie then shooting for 0 in ardour is not going to work- ardour ideally should be peaking around -18, maybe a hair higher once in a while. That is called 0dbfs (0 decibels below full scale). "Full scale" is calibrated to something analog, and -18 is one standard (people calibrate to different things depending on the situation), so 0dbfs in that case would be +18dbv (I think dbv is the scale, maybe dbvu? doesn't necessarily matter though). So if you are recording at +18 analog, that means your mackie is red lighting everything, and a mackie most definitely is not going to sound good at +18, and your delta 1010 inputs aren't designed for that either, so you're going to get some ugly sound. Just something to think about, it may or may not be your issue. So when you are gain staging, you want each piece in your signal chain to operate where it is meant to operate efficiently. In your mixer, if you max out your channel gain then lower the fader, that's not really what you want to be doing- the gain is not a "volume control" really, it is meant to bring the operating level of your mic to its correct spot. Same thing with your channel faders, you don't want them all up high then the master fader down. However, inside ardour, this is different, you can have your faders up and your master down, to a degree. Ardour's internal headroom is much much higher than any analog component, so you're not clipping ardour inside, but then as you send sound out too loud you're clipping your converters, which can be done with some converters, but not something on the m-audio level (me either, I have a phonic mixer, probably made in the same factory as the mackie). So keep your levels conservative and you'll get a better sound. If there are inaccuracies to what I've said please correct, but for the most part I think my info is correct. -- http://www.reverbnation.com/jeffreybrayne www.jeffreybrayne.com http://www.last.fm/music/Jeffrey+Brayne www.totalsoul.com www.bigjeffmusic.com

15 years, 4 months

[ANN] jack_capture V0.9.40

by Kjetil S. Matheussen

(I don't agree about the linux-audio-* mailing list announcement policy, so from now on, program announcements will only be posted to linux-audio-announce(a)lists.linuxaudio.org !) Download from ============= http://archive.notam02.no/arkiv/src/?C=M;O=D jack_capture ============ jack_capture is a program for recording soundfiles with jack. The default operation of the program is executed by writing "jack_capture" in the terminal without any extra command line options: $ jack_capture ...which will record what you hear in your loudspeakers into a stereo wav file. Most important new features since 0.9.36 ======================================== * Direct support for mp3 using liblame. * Console cleanup. Terminal should not be messy when quitting jack_capture. * Better buffering schemes. * Less used memory. Features ======== * Autogenerated filenames are unique and humanly readable. * The 4GB size barrier for wav files is handled by continuing writing to new files when reaching 4GB. * Supports all soundfile formats supported by sndfile. (wav, aiff, ogg, flac, wavex, au, etc.) (option: -f <format>) * Supports mp3 by using liblame. (option: -mp3) * Option for writing raw 16 bit data to stdout. (option: -ws) * Built-in console meter, plus option for automatically starting and stopping the Meterbridge jack meter program. Port connections to Meterbridge are done automatically, and on the fly, by jack_capture. * jack_capture can connect to any input or output jack port. When "connecting" to a jack input port (i.e. a writable port), jack_capture constantly monitors which jack ports which are connected to the input port, and make sure jack_capture is always connected to the same ports. In other words, jack_capture will reconnect its ports automatically during recording to match the connections of the ports. This is for instance convenient when recording the playback ports since jack_capture can be started first, and then other programs can start and stop at any moment while all sound still should be recorded. * No limit on the number of jack ports jack_capture can connect to. (I.e. the --port argument can be specified more than once, plus that it accepts wildcard arguments. For instance, jack_capture --port "*" will connect to all current jack ports, both input and output ports, except jack_capture's own ports.) * Buffers are automatically increased during runtime to prevent underruns and to avoid wasting memory by preallocating too much. (handled by using lockless atomic fifo/lifo queues to store temporary sound data instead of ringbuffers) * The disk thread is automatically reniced to a higher priority when using more than half of the buffer. * Significantly better recording performance than Ardour. (probably because jack_capture writes all channels into only one file and that it is not creating peak files). (tested on athlonXP) * No problem writing at least 256 channels of 32 bit wav at once to a local hard drive. (tested on icore7)

15 years, 4 months

linuxaudio.org downtime

by Robin Gareus

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 - From Sunday evening (UTC) until now we had a server outage of ~20 hours in which all linuxaudio.org services were not accessible. We apologize for the inconvenience, the cause of which was hardware failure of the server (cooling failed and the system halted due to overheating). It took some time to get access to the hardware, fix it and subsequently perform a complete file-systems check.. Thanks to Ico who maintains the hardware, the services are up an running again and there has been no data and email loss. Sorry for this boring message. You may rest assured that this outage did not include any major animal involvement and no one was harmed during the process. The three spiders are alive and well. Cheers, robin -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAktd4G4ACgkQeVUk8U+VK0IH3QCgkSNGtNAoM24gpN80XD01YZm9 QncAn38ig9CDxT1bW1TVwh/DD8lLIGPn =yyzu -----END PGP SIGNATURE-----

15 years, 4 months

[OT] about gvim

by Philippe Hezaine

Am I right if I say you could install gvim *without* gnome? At least its like that on my Gentoo. In this case g means GUI and his dependencies are GTK+ or Motif. I need a confirmation. -- Phil.

15 years, 4 months

Jretune - message for Joel White

by fons＠kokkinizita.net

Hello, I did send you the jretune sources but your ISP (comcast) seems to think I'm a spammer. Would you have an alternative address ? Ciao, -- FA O tu, che porte, correndo si ? E guerra e morte !

15 years, 4 months

usb and jack

by rosea grammostola

Hi, I found this info: "USB and jack The USB interrupt period is 1 msec. To be able to get lower latency with jack when using it with an USB device, you have to use a setting as 48kHz and 3 period. It will makes the buffer time a multiple of 1 msec and you will get a much lower latency as with the default 2 period. Additionaly, loading the snd-usb-audio with the parameter "nrpacks=1" will give you a much lower latency (for this to work take care that CONFIG_USB_BANDWIDTH is not set and CONFIG_USB_DYNAMIC_MINORS is not set in your running kernel)." http://proaudio.tuxfamily.org/wiki/index.php?title=Howto_RT_Kernel#USB_and_… 1) is this info still up-to-date? 2) how do I exactly take care of this: "Additionaly, loading the snd-usb-audio with the parameter "nrpacks=1" will give you a much lower latency (for this to work take care that CONFIG_USB_BANDWIDTH is not set and CONFIG_USB_DYNAMIC_MINORS is not set in your running kernel)" (Debian (based) systems) \r

15 years, 4 months

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Linux-audio-user January 2010