Linux-audio-dev

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3 participants
7535 discussions

[LAD] Yet another "io GNU/Linux" iso released

by Manu Kebab

Hi, A new 64bit iso is up ;) io GNU/Linux is a Live DVD/USB based on Debian Sid and focused on multimedia. Kernel 3.14.4, Jack2 as default sound server, e18 as desktop environment and a big collection of installed software... Full persistence for USB install (with encryption) and more cool stuff... A great nomade studio :) For more infos: manual, packages list, screenshots, video etc... Check: -> http://manu.kebab.free.fr/iognulinux.html -> https://sourceforge.net/projects/io-gnu-linux/ Feedbacks welcome, enjoy :) MK

11 years, 8 months

[LAD] LV2 plugin for all applications

by Martin Lohberger

Hey Guys, I'm new with LV2 and have some troubles: I'm familiar with LADSPA and I normaly link my plugins with my .asoundrc to alsa. I want to apply this plugin not only to one application, I want to use it for all aplikctaions like aplay or anything else. Is there a possibility for LV2 Plugins, to use the system-wide? Does anyone have experience? Thanks in advance, Martin

11 years, 8 months

Re: [LAD] AES67 + JACK

by Jamie Jessup

Hey All... Again, This has indeed caught my interest... After doing some (most likely not enough) reading, I am left a little curious for how some things happen. In terms of connection management and SIP under unicast routing, I can see a little bit how something like the Dante controller could happen. I am guessing a third party (ie the Dante Controller like thing) could tell two user agents to coneect with a REFER method. However when it comes to instructing an existing session to stop, I am at a loss. Does anybody know if there is a particular SIP method to get an existing SIP session to stop? An interesting thing that cought my eye about the standard is that it seems what discovery method is chosen is left open to the implementor. Avahi/Bonjour seems to be a natural choice, but I am not quite sure how "interoperable" all these Audio oIP standards would be if they all use different discovery methods. Just rambling... On Sun, May 25, 2014 at 4:11 PM, Jamie Jessup <jessup.jamie(a)gmail.com> wrote: > > *There is a proprietary ALSA driver for (at least) some Dante Cards, > butyou have to ask your reseller specifically for it and pay an additional > fee.* > > > Good to know, have you had any experience with the proprietary stuff? > Whats the experience like in comparison to the Mac/Windows equivalents? > > > On Fri, May 16, 2014 at 11:59 AM, Markus Seeber < > markus.seeber(a)spectralbird.de> wrote: > >> On 05/16/2014 05:37 PM, Jamie Jessup wrote: >> > I would love to see this happen in the world of LA. I was recently about >> > to embark in a HW project for an AES50 implementation, however after a >> > quick skim of the standard this sounds a little more exciting :). >> > >> > A Dante "Virtual Soundcard" style thing for ALSA/JACK would be amazing. >> >> There is a proprietary ALSA driver for (at least) some Dante Cards, but >> you have to ask your reseller specifically for it and pay an additional >> fee. >> >> -M >> _______________________________________________ >> Linux-audio-dev mailing list >> Linux-audio-dev(a)lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-dev >> > > > > -- > Jamie Jessup > > http://jamiejessup.com > -- Jamie Jessup http://jamiejessup.com

11 years, 8 months

[LAD] Standalone resources browser - mockup, near of publishing, need discussion

by Zlobin Nikita

Goal of this tool is to allow to search for audio plugins and for other resources types (sampler banks, instrument definitions, probably more) independently of application, needing it. I'm still thinking about final name for it, before to publish (since it should support not only plugins, but also instruments), and unsure even where to create home page for it (will describe in bottom). As for plugins: currently each audio application has own plugin browser. In some cases it is very handy (ingen, carla), sometimes awful (audacity, may be more). My hope is to make it external with filemanager-like workflow. Currently there is one implemented way to select plugins: dragging. Drag object is some resource URL, which should be enough for drag receiver to exactly find pointed resource. But unlike file managers, it is threated as text - text editors just paste it instead of try to open and konsole - same, instead of to ask, what to do, as it always does for dropped files. Of course, applications are free, how to threat such resources, and apps could check for content before to decide, what to do with it - i.e., check, is it url, etc. Dragging to jack patchbay canvas may be used to load plugin in single host (not sure, what about instrument, since there is usually separate sampler engine, usually managing all instruments). Proposals about drag object format and url types for each resources are welcomed. I have idea, how to implement for ladspa/dssi, but unsure for lv2. Currently there are following formats: LADSPA: ladspa:/<path to library>/<label> DSSI: same, but begining with "dssi:/" LV2: lv2 plugin url (i don't know, is it possible to load plugin from specific library, and they recommend to identify plugins only by URL - at least remember URL's for database, used across sessions). Not implemented: Just have to learn VAMP and somehow try VST - for some completeness. For soundfonts, gig banks, and instrument definitions there may be same way as for ladspa/dssi. Sampler banks mode should be handy for linuxsampler/fluidsynth gui, not providing any instrument browser (qsampler doesn't have it, and linuxsampler database supports only gig format). Screenshot: http://wstaw.org/m/2014/06/05/plasma-desktopau6238.png About publishing. I have account on savannah, and want to publish it there. But i'm going to add vst support, and i'm not sure, is it possible (even if licensing is kept, how about fact itself, that it supports this feature)?

11 years, 8 months

[LAD] Open Source Audio Meeting in Cologne, Germany (User Group)

by Nils

Dear musicians, programmers and normal people, I plan to create a user group, a regular meeting, in my home town Cologne, in Germany. The first meeting ever will be already on June 18th, 19:00. After that every two month or so. There are more dates on the website (see below). Here is a brief website with all the necessary information. http://cologne.linuxaudio.org/ (Any language is welcome but the chances are that most people will be from the area and therefore speak German. So the page is in German) If you intend to come you can put your name on this etherpad, but this is not required. Anybody can show up. http://yourpart.eu/p/linuxaudio-cologne Topics will be unorganized Q&A, showing off programs and music, sharing knowledge and tips and hopefully one day shorter or longer presentations, tutorials, workshops etc. I expect most people to use Linux but any OS is welcome, therefore I named it just "Open Source Audio" and not Linux Audio. So if you are in the area please join us! If you are not in the area but know people in the area, please tell them. Greetings, Nils http://cologne.linuxaudio.org/ http://www.nilsgey.de P.S. Despite the domain saying linuxaudio.org this is an independently and privately organized event. It is not intended to replace or get in conflict with the Linux Audio Conference.

11 years, 8 months

[LAD] FFADO 2.2.0 has been released

by Jonathan Woithe

The FFADO project (www.ffado.org) is pleased to announce the release of FFADO 2.2, the userspace framework supporting Firewire audio interfaces under Linux. While the focus of this version has been bug fixes and improvements to existing drivers, support for some additional devices has been added. FFADO is brought to you thanks to the work of Daniel Wagner, Pieter Palmers, Philippe Carriere, Adrian Knoth, Arnold Krille, Takashi Sakamoto, Jonathan Woithe and users who have tested FFADO against their devices, provided patches and given suggestions. Changes and additions in FFADO 2.2: * Many bugs fixed * Mixer, router and monitor support for Saffire Pro 14 * Mixer and router support for Presonus Firestudio Tube, Presonus Firestudio Project, and M-Audio Profire 2626 * Support Echo Audio AudioFire12 with firmware versions 5.0 and later (addresses ticket 360) * Echo Audio device mixer additions: - hide SPDIF mode switch on AudioFire12 - digital interface switch on AudioFire8 and Pre8 - phantom power switch for AudioFire4 (addresses ticket 364) - playback routing for AudioFire2/4 (addresses ticket 335) * Presonus device mixer additions: - better support the FP10 (formerly known as the Firepod) - implement support for the Firebox and Inspire1394 * M-Audio mixer additions: - improve support for the Ozonic - add support for Firewire Solo (ticket 336), Audiophile, 410, 1814 and ProjectMix I/O * Device-specific mixer for Yamaha GO44 and GO46 interfaces added * RME Fireface 400/800 improvements: - mixer and device settings can now be saved to and restored from device flash - clock source selection made more consistent * DICE EAP / RME Matrix mixer enhancements: - "per output" view with mono/stereo control - saving of mixer settings to file - more consistent cooperation with jackd when sample rate is changed * Audio streaming support added or refined for additional MOTU interfaces: - Traveller mk 3 - Ultralite mk 3 - Ultralite hybrid (using firewire interface only) - 4pre Known issues: * Saffire Pro 40 at 96 kHz fails to start most of the time (ticket 326) * Saffire Pro 24 and Pro 40 MIDI problems (tickets 372 and 375) * Saffire Pro 24 and Pro 40 lack an ADAT/SPDIF switch * Only mixer control is supported on M-Audio 1814 and M-Audio, ProjectMix * M-Audio Audiophile, 410 and 1814 require startup workaround. See http://sourceforge.net/p/ffado/mailman/message/30807938 * M-Audio Audiophile, 410 and 1814 will only work with FFADO when loaded with the latest firmware

11 years, 8 months

[LAD] Openframeworks

by Bart Brouns

Hi list, I just discovered openframeworks http://www.openframeworks.cc/ And it almost looks too good to be true. For example, have you seen this?: http://vimeo.com/59653952 Why is ofx not so prolific in lad land? I searched the lad mailing list and found just a handful of mentions over the years. Who's using it? Can you recommend it? Would you have used it if it was available at the start of your project? Best, Bart. studio magnetophon Bart Brouns Biesenwal 3 6211 GZ Maastricht 0031-6-22919561 bart(a)magnetophon.nl www.magnetophon.nl

11 years, 8 months

[LAD] kernel module load error

by Patrick Shirkey

Hi, Does anyone here know why I am seeing this error while loading a kernel module: Unknown symbol __stack_chk_guard (err 0) Unknown symbol __stack_chk_fail (err 0) I can't find anything online that explains the cause of the error in relation to loading kernel modules. -- Patrick Shirkey Boost Hardware Ltd

11 years, 8 months

[LAD] Ring buffers again

by Will Godfrey

As well as the jack ring buffer, I've looked at several others now, and their example code. The most significant thing that seems to be different about them (from a usage point of view) is the way they handle overflows. However, if the buffer size is defined as an exact multiple of the data type/structure and only complete structures are pushed or popped, would I be right in thinking that you would only need to check on an all/none basis? Have I missed something that could cause a partial data transfer? -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

11 years, 8 months

Re: [LAD] Call for testing (final): ALSA driver for some firewire devices

by Takashi Sakamoto

Hi Daniel, I notice that I cannot help you without enough logs with which I can realize what happens. At lease, 'verbose' options for aplay/arecord, '#' nodes in /proc/asound/cardX and dmesg. Additionally, would you please add 'LANG=C' when you get output from software. I cannot read Spanish language... > hi, it does not work. Try in following steps: 1. $ arecord /tmp/test.wav -D hw:UFX1604 -c 16 (I expected you receive 'Available formats' output) 2. $ arecord /tmp/test.wav -D hw:UFX1604 -c 16 -f (here the format, maybe S32_LE) (I expected you receive 'rate is not accurate' output) 3. $ arecord /tmp/test.wav -D hw:UFX1604 -c 16 -f (the same above) -r (got rate) (I expected it will run correctly.) I hope you to do these test when killing pulseaudio. The way is here: https://wiki.ubuntu.com/PulseAudio/Log Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

11 years, 8 months

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Linux-audio-dev