Linux-audio-user

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[linux-audio-user] [ANN] Ecasound 2.3.5 released, website addr. changed

by Kai Vehmanen

Hello all, here's version 2.3.5. Not too many changes, but still enough to warrant a new release. I also want to use this opportunity for the following: <*****IMPORTANT NOTICE*****> Ecasound website address has changed. The new address is http://www.eca.cx/ecasound This replaces the old (but still functional) website at http://www.wakkanet.fi/~kaiv/ecasound/. Note that also all the subpages have been modified. I've added some redirect pages, but the old site will only be available for a limited time. Please update any links that point to the old website. And also, my email address has changed (see from). </*****IMPORTANT NOTICE******> CVS is tagged with 'v2_3_5' as usual, and here are the full details: --- 1. Summary of changes Various Mac OS X specific build issues have been resolved. Minor changes have been made to the OSS soundcard support to avoid limitations of certain OSS drivers. Many minor updates have been made to user documentation and to build system scripts. --- 2. What is Ecasound? Ecasound is a software package designed for multitrack audio processing. It can be used for simple tasks like audio playback, recording and format conversions, as well as for multitrack effect processing, mixing, recording and signal recycling. Ecasound supports a wide range of audio inputs, outputs and effect algorithms. Effects and audio objects can be combined in various ways, and their parameters can be controlled by operator objects like oscillators and MIDI-CCs. A versatile console mode user-interface is included in the package. Primary platform for running Ecasound is GNU/Linux. Ecasound can also be run on many UNIX-derived systems such as FreeBSD, Mac OS X and Solaris. Limited support for Windows is available through Cygwin. Ecasound is licensed under the GPL. The Ecasound Control Interface (ECI) is licensed under the LGPL. --- 3. Changes since last release * With this version, Ecasound can now really be compiled for MacOS X. This has been tested on various 10.3.x systems. Also JACK support is reported to work under OS X. * Website address has changed: http://www.eca.cx/ecasound replaces the old http://www.wakkanet.fi/~kaiv/ecasound address. Please update any links that point to the old address. Full list of changes is available at <http://www.eca.cx/ecasound/history.php>. --- 4. Interface and configuration file changes None. --- 5. Contributors Patches - Accepted code, documentation and build system changes Berndtgen Manfred (1) -- ecasignalview build error Kai Vehmanen () -- various Bug Hunting - Reports that led to bugfixes (items closed) jcw (1) -- MacOS-X build errors. kito -at- gentoo-org (1) -- MacOS-X build errors. Stéphane Letz (1) -- Builds errors related to JACK and MacOS-X. Paul Marquardt (1) -- Mac OS X build errors. Raul Megelas (1) -- OSS-devices with no GETBLKSIZE support. --- 6. Links and files Web sites: http://www.eca.cx (fi) http://ecasound.seul.org (us) http://ecasound.sourceforge.net (us) Source packages: http://ecasound.seul.org/download http://ecasound.seul.org/download/ecasound-2.3.5.tar.gz md5sum: bbc0c4d12c1d21a7e71fc1bdb9fb0e2b Distributions with maintained Ecasound support: See http://www.eca.cx/download.php -- http://www.eca.cx Audio software for Linux!

21 years

[linux-audio-user] asterisk open source linux pbx

by Eric Dantan Rzewnicki

I've recently been conscripted into the role of backup to our primary phone system admin with an eye towards leveraging my network admin background in our coming VoIP deployment. Due to this I've been cramming as much telephony information as I can into my brain over the past 3-4 weeks. Our PBX (private branch exchange, i.e. the switch for our office phone system for those unfamiliar with the term) is a proprietary system from NEC. Having to deal with systems like this is contrary to my nature. To sooth the irritation of being rubbed the wrong way by this proprietary technology I've signed on to the asterisk users and dev lists with a few aims. First, as I expected, I've found the members of the asterisk community to really know their stuff when it comes to the standards and protocols that make phone systems work. In a few days of reading the lists I've already learned much that I haven't gotten out of the NEC manuals, but that helps me to better understand their closed system. Second, I'm hoping that down the road I can get an asterisk system into this operation to provide additional services and functionality that would be more expensive to purchase from NEC. I'm writing to LAU to get some feedback on a number of possiblities that come to mind for cross fertilization between this community and the asterisk community. Also I'm hoping there are some here who have experiences with asterisk they would be willing to share. As I understand it asterisk can use ALSA supported full-duplex cards to provide voice i/o. An asterisk server with a number of connections to the phone network and several RME HDSP or other such high channel count multi-channel cards would seem to be a very useful, cost effective and high quality solution for supporting call-in shows and telephone interviews for a radio station. Such a settup could also provide a nice platform for an intercom system for a business or even a home. Is anyone here doing such things? Apropos my recent inquiry regarding bats, telephony has traditionally saved bandwidth by limiting the frequencies transmitted to a roughly 4kHz band since the information important to intelligible speech can be conveyed without the sounds outside that band. Are the concepts used to capture bandlimitted audio for speech the same, or similar to, what would be used to capture the interesting information from sounds produced by animals who hear above the human hearing range? There are a variety of audio data compression and synthesis/resynthesis schemes in use in the telephony world. Have any of these been repurposed for use as effects, perhaps wrapped up in LADSPA plugins? Are there similarities between jack and asterisk in what they need to do to provide audio routing and scheduling? Perhaps this has already occured or perhaps their needs are too different, but could the two projects benefit from sharing ideas or even code? If these are naive questions and the two domains are orthogonal I'm interested in knowing why. Hmm ... as I write I realize a big difference is that many phone conversations happen at once and have no need for synchronization. jack's typical application space involves keeping many channels of audio in sync. So, I guess I've largely answered this one for myself. But, still input from the system programming gurus like Paul and Jack would be most welcome and surely enlightening and educational. I had a few other ideas and questions, but they've slipped away from me. Anyway, this has gotten long enough. Thanks in advance for reading and for any feedback. -Eric Rz.

21 years

[linux-audio-user] Re: linux-audio-user Digest, Vol 14, Issue 28

by Brian Redfern

Well, as far as info about audio goes, for newbies their best bet is to use Suse Linux, it comes with support, so you have a live human to call if you have problems.

21 years

[linux-audio-user] "PDAudio-CF on Laptop with Linux" Adventures

by Daniel Zuckerman

Greetings listers. I thought I'd share some things I discovered last night. After spending an inordinate amount of time (not all at one sitting) trying to figure out how to solve this problem... ~/test-alsa/ arecord -D pdaudiocf -f cd foo.wav Recording WAVE 'foo.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo arecord: pcm_read:1196: read error: Input/output error ...I made some discoveries. This started happening to me after coming fresh to the laptop one evening, ready to transfer an old concert tape. The PDAudio-CF having worked like a champ before, I was starting to get nervous that something odd had happened. Turns out, after going to the #alsa group on freenode and getting a huge amount of attention from one kind soul, who eventually pointed me to one of the alsa list archives, combined with something I seemed to recall reading somewhere, that the tape was at 48k, and the PDAudio-CF doesn't do hardware sample rate conversion. So why I am I telling you this? Well, for one thing, if my notes ever get deleted, I can always check this archive. :) But I hope to save someone else the problem. WHAT TO DO: So, if you get this error, and it seems out of the blue (like the thing was *just* working!!!), Check what the card thinks the incoming sample rate is. You can do this: amixer -c <card#> contents. In there somewhere you'll see something about "IEC958 External Rate", and the next line will be the incoming sample rate (in samples/sec). you'll want to make sure that the incoming sample rate and outgoing sample rate are the same. So in my case for last night, I needed to do the following with the 48k tape: arecord -c hw:1,0 -r 48000 -f dat foo.wav BUT WAIT, THERE'S MORE: Before I got to that solution, I found this nifty shell script in the alsa archives that let's you specify optical/coax input. It automagically figures out which hardware slot the PDAudio-CF occupies. Here's a link the alsa archives message containing the script: http://sourceforge.net/mailarchive/message.php?msg_id=7502478 That's it for today. I'm looking forward to recording my first live convert this weekend using this device (backed up with DAT just in case...)! Regards, Daniel Zuckerman

21 years

[linux-audio-user] Call for audio submissions - Linux Open Source Sound CD

by JakeHarries＠aol.com

Hi I think some people may be interested in this best Jake ------------------------------------------------------- CALL FOR AUDIO SUBMISSIONS Linux Open Source Sound CD (L.O.S.S.) [Planned release date - April 2005] Deadline for submissions: 07-Jan-05 Access Space, Sheffield's lowtech digital arts organisation, is currently calling for submissions for a CD of audio produced with open source software, and the Linux operating system. There is no specific theme for the curated works, as the concept behind the project is freedom of all elements of music manufacture, encapsulating style, production software and distribution techniques. We hope to receive submissions covering a broad and eclectic range of styles, to represent the dynamic nature of contemporary open source audio culture. Therefore, contributions are invited from musicians of all types, programmers, sound artists or artists who use sound. The LOSS CD is to be released under a Creative Commons 'Sampling Plus' license, so as well as being produced with free software, the CD will also extend the ethos of the open source movement into its method of distribution. For more information about this license, please visit http://creativecommons.org. Please do not submit tracks if you are not willing to release your work in this manner. The LOSS project will develop not only through the CD release, but also through a website, aimed at being an ongoing portal for producers of open source music to showcase their work. This will also offer the works for redistribution under the Creative Commons licensing mentioned above. This website will be online later in the year at http://www.access-space.org/loss. How to submit your proposal: [A maximum of 2 tracks per artist, each between 20 seconds and 8 minutes in length.] Send a DATA CD containing the following files: - Your audio track(s) in .wav format, 16bit, 44.1khz in either mono or stereo. - A text document stating your name, contact details (email and mailing address), track title, track length, the software and operating system used for producing the track, and a declaration that your track does not infringe any copyrights or use any unlicensed material. - An optional screenshot (in .jpg or .png format) of your software setup - which may be used for artwork purposes. For more information, or to mail your submission: Linux Open Source Sound CD Access Space 1a Sidney Street Sheffield S1 4RG 0114 2495522 www.access-space.org loss(a)access-space.org Access Space is UK registered charity no: 1103837 Funded by Arts Council England, Yorkshire and Digital South Yorkshire.

21 years

[linux-audio-user] Journals

by Hans Fugal

Are there any affordable journals/magazines out there that people recommend, for computer music in general? I'd love to subscribe to the CMJ but the cost is way out of reach... -- De gustibus non disputandum est.

21 years

[linux-audio-user] How to enable 88.1KHz and above sampling rate on HDSP/Multiface+Jack?

by Ivica Ico Bukvic

Hi all, I guess the title says it all. I was wondering if there was a button or something that would enable this using perhaps a hdspmixer or something similar or perhaps passing appropriate setting via alsa mixer? On my setup jack always fails with anything above 48KHz. Best wishes, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/ --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 11/1/2004

21 years

[linux-audio-user] linux-2.6.10-rc1-mm3-RT-V0.7.18 won\'t boot

by thewade

Ive played around with .config settings, and I get an error I can do something with: rhgb: 1578 Bug: lock held at task time my kernel like from grub.conf looks like: kernel /bzImage-2.6.10-rc1-mm3-RT-V0.7.18 ro root=LABEL=/ rhgb quiet If I remove rhgb as a kernel flag I get another error that I cannot see the origin of, but I catch that it tells me that something like \"deadlock\" has been stoped and that I should report this. I think its \"deadlock\"... What is the problem with my .config? Thanks for the help! I cant wait to get capabilities working! I had 2.6.9-rc3-mm3-VP-T3 working, but without capabilities. -thewade

21 years

[linux-audio-user] connecting soundcards using JACK

by Jeroen Wijnhout

Hi all, Would it be possible to connect my IMic USB soundcard capture port to the onboard AC97 soundcard using JACK? I'm able to connect ports of a single soundcard to each other, however I would like to be able to listen via the onboard card what I'm recording on the external USB soundcard. It seems that I have to start the JACK daemon in such a way that it can talk to both soundcards at the same time. jackd -V:0.94.0 best, Jeroen -- Kile - an Integrated LaTeX Environment for KDE http://kile.sourceforge.net

21 years

[linux-audio-user] Report from the ICMC

by Ivica Ico Bukvic

Hi all, I just wanted to share with you a quick report from the ICMC. The conference was much fun although many events were overlapping and therefore it was impossible to attend everything. A number of new open-source audio-oriented software was demoed many of which are also available for the Linux platform. My Linux demo went relatively well (apart from the 2 consecutive X-server crashes, courtesy of the crappy binary-only ATI's driver, and an odd issue where disconnect of hdsp pcmcia did not yield acknowledgment from the 2.6.7 kernel and therefore subsequent reconnect did not recognize the card's presence). There were ~20+ people in the audience at any given time throughout the presentation (some of them were there only for a portion of the presentation, so people kept coming and going). Stuff I demoed (in no particular order): *Jack *Freqtweak *Jamin *Pd/Gem *Latency tests *Ardour *Sweep *Rezound *Audacity *gAlan *Rosegarden *Qsynth/fluidsynth *ZynAddSubFX *VKeybd *Cinelerra *Blender *Xine/DVD playback *Spiral Synth Modular *KDE productivity Stuff I wanted to demo but ran out of time: *Hdsp latency/performance *RTcmix synthesis language *Supercollider *Csound Etc. Stuff I observed preparing and doing this demo (please understand that I have no idea whether these problems are result of my own setup or are justifiable bugs -- nonetheless I am including them here in hope they may shed some light towards their resolution): *ZynAddSubFx crashes when a lot of polyphony is created even though the cpu utilization is not topped-off *Jamin crashes when pushing limiters to the extereme (not consistently) *fluidsynth has some weird looping problems (I am trying to track this one down as apparently this is specific to my setup) *Ardour's real-time preview via sliding the timing bar does not work (I do have one release prior to the latest, though) *Spiral Synth Modular crackles a bit even with large buffer sizes when using OSS (haven't had the time to do exhaustive tests nor did I try using Jack object as of yet) *Audacity has some instabilities when used in conjunction with Jack *There is a real need for Jack to be able to restart (should a need for its restarting ever arise) without losing all of the connections Stuff that really rocked: *I was able to run Jack on my via82xx laptop soundcard using rt mode with 64x4 buffers (5.33ms at 44.1KHz) *almost* rock solid (there is still some issues due to crappy ATI driver and obviously 2.6.7 kernel that is still sub-par to the 2.4.x kernel series performance, but that mainly amounted to perhaps a 1 xrun/minute). *I was able (although after the demo) to run hdsp with 64x2 (2.9ms at 44.1KHz) *almost* rock solid (random occasional xruns but no xruns when adding new connections and/or apps to the Jack session). *I tested the PD with the hdsp's 64x2 buffer size and used the latency test patch by connecting audio out with audio in on the multiface and got ~6ms :-). The general reaction from the audience was quite positive as many of the users who even were familiar with the Linux Audio scene were impressed by the diversity and quality of the software offering as well as the flexibility of the Jack's framework and the overall user-friendliness of the UI/desktop environment. I would like to therefore use this opportunity to once more thank all of you for being such active and generous contributors to this great community. Without you none of this would have been possible! ---------------------- The panel on the "Standards from the Computer Music Community" was very much interesting as it covered many of the important facets of today's computer music scene, but also more importantly revealed some of the greatest strengths of Linux, including (but not limited to): *openness of the standards and therefore ability to generate umbrella meta-standards (i.e. LASH) *longevity *ability for a new standard to supersede the reigning old standard solely based on its merit (i.e. Alsa vs. OSS), and not due to its commercial PR and/or widespread use *Minimization of the misrepresentation of the standard's features and/or abilities Having had this wonderful opportunity to be a part of such panel (many thanks to Matt Wright for inviting me!) has truly reinforced and clearly defined the advantages as well as strong reasons for being a part of this community. Many thanks go to all of you who have generously offered your insight in these issues. Time permitting, I will post a more in-depth list of the ideas I've covered during the panel. Matt Wright should also have slides ready on his site sometime soon. Best wishes, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/ --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 11/1/2004

21 years

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