Linux-audio-user

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8 participants
20713 discussions

[linux-audio-user] Frequency analysis app?

by Barton Bosch

Does anybody have any suggestions as to which frequency analysis program would be best for looking at field recordings? Thanks, Barton

21 years, 10 months

[linux-audio-user] switching from windows + cubase

by Thomas Pickett

Hi, I'm considering switching for Windows to Linux for my audio stuff. I have a few questions. 1) I have some Cubase projects, and I mostly care about the midi sequencing. If I export them to midi files, will I be able to restore most of the information in Linux? 2) This is my first install. Can you suggest a distro for a 1.6 GHz Athlon machine that would be good and stable for audio yet not be too hard to set up and get working? Preferably something with a 2.6 kernel. I was going to go with Mandrake, but I really don't know. Definitely something like Debian or Gentoo is too involved for me right now. 3) I have a Soundblaster Audigy. Should I look into getting a new soundcard? What is a good internel or external box that is supported well? 4) Can you recommend a set of software that will be enough to get started with? I know that there are many many options and I would just appreciate a little direction so I don't get discouraged. I just want to record vocals, record midi and use my synthesizer, mix them together, and process it all with some plugins. That's all, and thanks for any information. Thomas _________________________________________________________________ MSN Premium: Up to 11 personalized e-mail addresses and 2 months FREE* http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht…

21 years, 10 months

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

by Kjetil Svalastog Matheussen

Mark Knecht: > > http://www.notam02.no/radium/ > > > Kjetil, > Hi. Radium looks interesting, but I'm not sure I understand it from > just a quick look at screen shots. Is it along the lines of a tracker? Its a bit like a tracker, yes. But without a lot of the limitations found in trackers. It takes the best from the tracker, which is fast editing with lots of information using small space; less time used for navigating. The screenshot is a bit outdated though, in >v0.60 you can have a total graphical layout looking more like a vertical pianoroll as an alternative. There is also a new track showing the tempo graphical with gradiently changing colors. > Time seems to be progressing downward. Yes. > What is placed in each column? Its sort of explained here: http://www.notam02.no/radium/docs/radium_Sections/visibleoverview.HTML > Wave files? Loops? Unfortunately just midi data for now. The code is modular build up, though, so its not ~that hard to add sample-support. > I'm guessing the numbers in the left column (headed > with the number 15) indicate how long you play a loop at a given > vertical slot? The track to the right for the track headed with the number 15 shows the sublevel, and the track headed with the number 15 shows the linenumber in the level showed in the sublevel track. Its a flexible way to zoom, and has nothing to do with looping. The number "15" shows the linenumber at sublevel 0. sublevel 0 is colored black sublevel 1 is colored white sublevel 2 is colored brown etc.. Its not that confusing when you actually use it yourself, and you don't have to use it if you don't want to. There are two ways to zoom, the one I have been talking about now, which I have called local zooming, and the other way which is global zooming. The global zooming is just a plain graphical effect. You do a local zoom by pressing left shift+<arrow down/up> and global zooming by pressing left meta+, and left meta+. and left meta+. > Does the relative tempo column effect the pitch of the > files being played? > > Anyway, it looks interesting. > Thanks. :) --

21 years, 10 months

[linux-audio-user] [ANN] Ecamegapedal 0.4.4 released

by Kai Vehmanen

1. A short summary of changes Minor bugs in JACK support have been fixed. Now Ecamegapedal makes sure it won't launch the JACK daemon by accident when probing for available devices on startup. The manual pages have been updated with some new sections. --- 2. What is Ecamegapedal? Ecamegapedal is a real-time effect processor software with a graphical user interface for controlling the effect parameters. It is meant to be used as a virtual guitar-fx or studio effect box. In addition to real-time operation, Ecamegapedal also supports reading from and writing to audio files. All audio object and effect plugin types provided by the Ecasound libraries are supported. This includes ALSA, JACK, OSS, aRts, over 20 file formats, over 30 effect types, LADSPA plugins and multi-operator effect presets. Ecamegapedal's implementation is based on Ecasound and Qt libraries. Ecamegapedal is licensed under the GPL. --- 3. Contributors Patches Kai Vehmanen (various) --- 4. Links and files http://www.eca.cx/ecamegapedal http://ecasound.seul.org/download/ecamegapedal-0.4.4.tar.gz --- http://www.eca.cx Audio software for Linux!

21 years, 10 months

[linux-audio-user] Travel to lad-conf

by Julien Claassen

Hi folks! I'm looking for someone to accompany me to Karlsruhe by train. I'm coming from Paderborn. But I will go through Dortmund, essen, duisburg, Koeln. To those places I can get on my own, from there I need help. For those of you who don't know it: I'm blind. So I get a free companion. That menas half the price. I want to go there on wednesday and leave on sunday. On wednesday Joern can come with me, but the way back is unsure yet. If anyone goes that way: just tell me... Kindest regards Julien http://ltsb.sourceforge.net - the Linux TextBased Studio guide --------------------------------------------------------- SBS C-LAB Fuerstenallee 11 33102 Paderborn Phone: (+49) 5251 60 6060 Fax: (+49) 5251 60 6065 www.c-lab.de/~wegge

21 years, 11 months

[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, 11 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, 11 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, 11 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, 11 months

[linux-audio-user] 2.4.26 low latency patch

by Andrew Gaydenko

Hi, Does subj exist already? Andrew

21 years, 11 months

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