Linux-audio-user April 2004

linux-audio-user@lists.linuxaudio.org

141 participants
177 discussions

[linux-audio-user] sndfile-info and lots of questions

by R Parker

Hi, As I've gone through my most challanging mastering project, I've developed more questions than answers. There's been reference, on this list, to documentation that explains file formats-- I don't recall the document title or where to find it. Ultimately, I need a resouce that explains things like; the number of available samples for the different bit depths (16bit range from -ABC to +XYZ, 24bit...), DC offset is, peak amplitude is, RMS is, etc. I need to know if samples are syncronous with decibel level, is maximum samples equal to 0db? In the following sndfile-info report, what are Length and Block Align? Version : libsndfile-1.0.6 ======================================== File : guajira-jam.wav Length : 30396228 RIFF : 30396220 WAVE fmt : 16 Format : 0x1 => WAVE_FORMAT_PCM Channels : 2 Sample Rate : 44100 Block Align : 4 Bit Width : 16 Bytes/sec : 176400 data : 30396184 End ---------------------------------------- Sample Rate : 44100 Frames : 7599046 Channels : 2 Format : 0x00010002 Sections : 1 Seekable : TRUE Duration : 00:02:52.313 Signal Max : 25922 Is Signal Max a measurement of used samples? One of the problems I've confronted during mastering is JAMin, Ardour and hardware meters tell me that a track is peaking around -0.5db but the hardware mixer indicates overload. Of course metering balistics being what they are this is understandable. The mixer documentation doesn't tell me at what level the overloads are set to go off at and I haven't found a configuration interface. With this metering and overload discrepancy I'd like to read a sndfile-info report that tells me; of the available samples, this file uses a minimum of X and maximum of X. I wonder if some of qualities any of us should know about our mastered files includes: Min Sample Value Max Sample Value Peak Amplitude Possibly Clipped DC Offset Minimum RMS Power Maximum RMS Power Average RMS Power Total RMS Power Of course another challange is tools like sndfile-info assume that a file exists. This is not always the case and in my situation it's almost never true. I return JAMin output to an Ardour return bus and don't produce a file until the return bus is exported. Printing a track to the file system and then analyzing it is no way to save time. Anyway, I appreciate all the responses to my past questions and am hopeful that someone can look at the current mumbo jumbo and prescribe some effective medications; coffee, sleep, black bear gallbladders, urls to useful documents, etc. ron __________________________________ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html

21 years, 2 months

Re: [linux-audio-user] sndfile-info and lots of questions

by davidrclark＠earthlink.net

Ron, I think Erik gave the most accurate answers to your questions; however, I do have a couple of comments: 1) Erik said that 0db is the maximum sample value. Just a clarification, he means the maximum possible sample value, not the maximum sample value in a particular file. For example, for 16 bits, 32767. It's not a relative measurement, but an absolute number, despite the db label. 2) Erik also said that the Block Align is an internal detail. Well, yes that's correct, but it isn't anything complicated or secret. It is the number of BYTES for a sample (2 for 16 bits, 3 for 24 bits) multiplied by the number of channels. 16-bit stereo should give you 4; 24-bit stereo should give you 6. This information is redundant in WAV headers, and perhaps this is why Erik said not to worry about it. It would be better if it wasn't there because it's primarily an opportunity to screw up a program (or a posting!).

21 years, 2 months

[linux-audio-user] ice1724, routing oss playing back to spdif

by Andrew Gaydenko

Hi, The aim is to route oss-only-playing-back-software to spdif out. Is it possible? I use ice1724 driver. Thanks in advance, Andrew

21 years, 2 months

[linux-audio-user] How achieve smooth playing of multiple instances of linux audio players

by Robert Rozman

Hi, I have HTPC entertainment with home automation control integrated (Based on Misterhouse). I also have couple of sound cards for 2 independent stereo outputs that go to separate rooms. Now I start 2 instances of alsaplayer - playing music to those rooms. Now I have also some other processes on PC (Apache, PVR 350 video recording, media serving to LAN) - so I get occasional hickups on music output. But I'd like to play music smoothly probably at the price of resposivness of other applications. With what settings/configuration/additions could I achieve that. I assume that assigning higher priorities may be not enough ? How do you tweak your machines to get smooth audio play ? Regards, Robert.

21 years, 2 months

[linux-audio-user] 2.4.26 low latency patch

by Andrew Gaydenko

Hi, Does subj exist already? Andrew

21 years, 2 months

[linux-audio-user] alsa setup

by Atte André Jensen

Hi I'm confused about how to setup alsa. Originally I managed to get things working, but after buying an usb sound card, I'm even so confused that I'm not exactly sure what to ask, but I'll try. I run debian/unstable on a laptop with my own 2.6.4 kernel and alsa 1.0. I have the following hardware that I believe to be related to alsa: *Onboard i810 soundcard *USB Edirol UA-1A sound card *4 port usb hub *2 evolution usb-keyboards connected to the hub *I need one virtual midi port for each keyboard, so 2 for now The problem is that right now the cards get assigned a card number that reflects in what order they were inserted. I'd rather have them show up on fixed locations. Below is my childish attempt on a /etc/modutils/alsa file. I'm pretty sure it's not correct to refer to snd-usb-midi, since I don't have a module by that name. Could anyone please give me a hand + some general bckground information? Thanks in advance!!!! alias char-major-116 snd alias char-major-14 soundcore # i810 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss alias snd-card-0 snd-intel8x0 alias sound-slot-0 snd-card-0 # ua-1a alias sound-service-1-0 snd-mixer-oss alias sound-service-1-1 snd-seq-oss alias sound-service-1-3 snd-pcm-oss alias sound-service-1-8 snd-seq-oss alias sound-service-1-12 snd-pcm-oss alias snd-card-1 snd-usb-audio alias sound-slot-1 snd-card-1 # virtual midi 1 alias snd-card-2 snd-virmidi alias sound-slot-2 snd-card-2 # virtual midi 2 alias snd-card-3 snd-virmidi alias sound-slot-3 snd-card-3 #evolution 1 alias snd-card-4 snd-usb-midi alias sound-slot-5 snd-card-5 #evolution 2 alias snd-card-5 snd-usb-midi alias sound-slot-5 snd-card-5 -- peace, love & harmony Atte http://www.atte.dk

21 years, 2 months

[linux-audio-user] [ANN] Radium v0.63

by Kjetil Svalastog Matheussen

http://www.notam02.no/arkiv/src/radium-0.63.tar.bz2 http://www.notam02.no/radium/ (Note, the CVS is very outdated) Radium V0.63 Alpha Linux Port Released 15.4.2004 HOW TO MAKE IT RUN WITHOUT READING THE REST OF THE README FILE make ./start.sh INTRODUCTION This is the second public, and the first official, release of the linux port of the music editor Radium. The Amiga version has been stable since January 2002. Radium is a new type of graphical (currently only) notebased music editor. For more info about the program, check out the online manual placed at http://www.notam02.no/radium/ somewhere. COMPILE The following packages are currently used to build and run Radium: *libgc - http://www.hpl.hp.com/personal/Hans_Boehm/gc/ *Python - www.python.org (at least V2.0 and tkinter) *Python development package *pyqt - http://www.gnu.org/directory/devel/specific/PyQT.html *pygtk - http://www.daa.com.au/~james/software/pygtk/ (pygtk1, use pygtk-0.6.11) *xterm - Yupp. Most of these things are allready included in the radium/ directory (mainly because of compatibility problems between various versions of gtk, pygtk and libglade), so you only need to get hold of "pyqt". "pyqt" is available as package for most linux distributions. CURRENT STATE The biggest problem is that saving does not work. Loading does though. Other things that needs to be fixed is that the playing behaves strange when reaching the end of the blocks, and that the alsa-seq code needs some care (midi/alsaseq/). ABOUT THE CODE The code is a mix between C and Python. The state of some of the code is quite horrible and ununderstandable, but relatively bug-free, eh hopefully. :) CONTACT k.s.matheussen(a)notam02.no http://www.notam02.no/radium/ --

21 years, 2 months

[linux-audio-user] saving qjackctl settings

by derek holzer

Has anyone here been able to save their settings in Qjackctl? If so, what's the trick? It's a bit of a pain to have to redo everything every time! thx, d. -- derek holzer ::: http://www.umatic.nl ---Oblique Strategy # 172: "Use `unqualified' people"

21 years, 2 months

[linux-audio-user] minor update to Linux soundapps pages

by Dave Phillips

Greetings: Some URL errors fixed, Speech section completely updated (thanks to Antti Kaihola), LilyPond entry massively rewritten, and a few more apps in the New Additions... http://www.linuxsound.at (Europe) http://linuxsound.jp (Japan) http://linux-sound.org (US) Best, dp

21 years, 2 months

Re: [linux-audio-user] Non-xrun noise with ALSA, Jack and an USB soundcard

by Jorma R

Hi, Thanks for the replies, all. I get the noise regardless of whether I run Jack from the console or qjackctl so that is not the cause. > Can you record the noise? I was expierincing a similar thing > a few weeks ago and discovered it to be caused by my netowrk > card under heavy traffic. > > --ant I could basically record the noise to an external recorder. As said - the captured audio is always clean. This noise comes in in between the capture and playback. In other words it is either Jack, Alsa or the card. Is there a way I coud get Jack to run without -S and with a -p greater than 1024 with 6 input channels? Error message below. -Jorma [jorma@turpea jorma]$ jackstart -v -R -dalsa -dhw:0 -r44100 -p2048 -n2 back from read, ret = 1 errno == Success getting driver descriptor from /usr/lib/jack/jack_alsa.so getting driver descriptor from /usr/lib/jack/jack_dummy.so jackd 0.94.0 Copyright 2001-2003 Paul Davis and others. jackd 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 registered builtin port type 32 bit float mono audio capabilities: =i cap_setpcap,cap_ipc_lock,cap_sys_nice,cap_sys_resource+ep loading driver .. new client: alsa_pcm, id = 1 type 1 @ 0x8058b40 fd = -1 apparent rate = 44100 creating alsa driver ... hw:0|hw:0|2048|2|44100|0|0|nomon|swmeter|rt|32bit control device hw:0 configuring for 44100Hz, period = 2048 frames, buffer = 2 periods Couldn't open hw:0 for 32bit samples trying 24bit instead ALSA: cannot set period size to 2048 frames for capture ALSA: cannot configure capture channel cannot load driver module alsa __ Sano ei kuukausimaksuille. Hanki Suomen ainoa aidosti kuukausimaksuton GSM-liittymä osoitteesta http://www.saunalahti.fi

21 years, 2 months

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Linux-audio-user April 2004