Linux-audio-dev

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7513 discussions

[linux-audio-dev] [ANN] GMIDImonitor 3.0 released

by Nedko Arnaudov

GMIDImonitor is GTK+ application that shows MIDI events. New in this release: * Switch to autotools. * LASH is now optional. * JACK MIDI support (both old 0.102.20 and new one 0.103.0). * LASH, JACK MIDI and ALSA MIDI can be force disabled by passing option to configure script. * Shortcut for clearing list (ctrl-x). Project site: https://gmidimonitor.bountysource.com/ Screenshot: https://gmidimonitor.bountysource.com/FileDownload?file_id=50&inline=yes Source tarball can be downloaded from project site, "Downloads" section -- Nedko Arnaudov <GnuPG KeyID: DE1716B0>

18 years, 6 months

[linux-audio-dev] denormals or ... ?

by Dave Phillips

Greetings: I have a problem with a piece I'm working on in Rosegarden 1.5. I've appended the text I sent to Chris Cannam, along with his response (I hope he doesn't mind). Btw, the machine is based on an AMD64 3200+, with 2G RAM and an 80G hard drive. Sound runs through an M-Audio Delta 66. Distro is 64Studio 2.0. My questions lead, Chris's replies follow : >> 2) I've written a piece consisting of six tracks with these specs: >> >> Tr.1 (audio) drum loop, repeated, 1 plugin (CAPS plate reverb 2x2) >> Tr.2 (MIDI) bass part, repeated, uses patch from 8mbgmsfx.sf2 >> soundfont, no fx (part is rendered via QSynth which does have its >> reverb active) >> Tr.3 (audio) rhythm guitar loop, repeated, 3 fx (dj EQ mono, CAPS >> compress, CAPS plate 2x2) >> Tr.4 (audio) lead guitar, no repeats or copy, 4 fx (CAPS plate 2x2, >> AmpIV, AM pitch shifter, multiband EQ) >> Tr.5 (audio) riff loop, copied, 3 fx (dj EQ mono, CAPS compress, >> CAPS plate 2x2) >> Tr.6 (audio) another percussion loop, copied, no fx >> >> At 120 BPM the piece is 200+ measures long. It plays along >> swimmingly, but at m. 74 (about two minutes into the piece) the sound >> cuts out entirely. RG continues to run, but it pops up a message >> telling me that there's not enough CPU for realtime processing. Up to >> that point JACK reports approximately 33% CPU usage and no errors, >> but then it reports 0(1024) for xruns. I don't know what the second >> number (1024) means to JACK, but it keeps rising (with no sound) >> until I halt RG. > > That sounds a bit like a plugin denormal problem. Although denormals are usually less of a problem on AMD64. The 0(1024) in qjackctl I think means that JACK has reported 1024 xruns via the reporting API but they haven't been mentioned in the message log. I don't actually know what causes that. Does CPU usage (as reported by a plain old CPU usage reporting program) actually peak at that point? >> So my question is: Given my machine, should my CPU be topping out in >> this scenario ? > > Probably not, or at least I wouldn't expect it to suddenly peak if usage has previously been on the low side. The only thing RG does that may affect CPU usage drastically during playback is that it doesn't start running plugins until they actually have something to work on, and it stops running them if they've fallen silent for a certain period and have no more input coming up (this is in contrast to e.g. Ardour which runs plugins all the time during playback, taking a more strictly correct view). So, if anyone has anything to add to Chris's assessment, I'd like to know about it. If the problem is indeed related to denormals, is there a way to fix it ? Comments and suggestions are most welcome. Best, dp

18 years, 6 months

[linux-audio-dev] libtool test modules

by Stefano D'Angelo

Hi all, I have a little problem that maybe some of you already faced. I'm developing a test suite for my library and I wrote some libtool test modules which I wish to build with automake. The problem is that using check_LTLIBRARIES or noinst_LTLIBRARIES dynamic modules are not built, but only static libraries instead (while with any other prefix it works). How can I overcome this? Here the Makefile.am: MODFLAGS = "-module" INCLUDES = -I$(top_srcdir)/include @INCLTDL@ AM_CPPFLAGS = -DTESTSDIR=\"`pwd`\" TESTS = lists dynarrays dl check_PROGRAMS = $(TESTS) lists_LDADD = ../lib/libnasprocore.la dynarrays_LDADD = ../lib/libnasprocore.la dl_LDADD = ../lib/libnasprocore.la @LIBLTDL@ noinst_LTLIBRARIES = famod1.la goodmod.la famod1_la_SOURCES = famod1.c famod1_la_LDFLAGS = $(MODFLAGS) goodmod_la_SOURCES = goodmod.c goodmod_la_LDFLAGS = $(MODFLAGS) Thanks in advance, Stefano

18 years, 6 months

[linux-audio-dev] [PATCH] jack-keyboard-1.5 - add MIDI channel parameter

by Nedko Arnaudov

I've made little patch that adds optional command-line parameter to jack-keyboard for chaning output channel. For example to use channel 10 instead of channel 1 (default), run jack-keyboard with: jack-keyboard -c 10 -- Nedko Arnaudov <GnuPG KeyID: DE1716B0>

18 years, 6 months

[linux-audio-dev] Re: [CLAM] CLAM at the 2007 Google Summer of Code

by Xavier Amatriain

You better start by doing the application tutorials linked from the CLAM documentation page and then head into the smaller examples in the CLAM repository. You might want to start with the longer tutorial but the whole thing is only if you want to dive into the realms of spectral processing (besides, this tutorial is a little bit out of sync with the current release). XA Rahul Murmuria wrote: > I am interested in contributing to CLAM as part of GSoC. Although I am > new to audio-related projects, I am very excited with the prospect. I > need some initiation to setup CLAM before I can think of sending in a > proposal, but the IRC Channel is dormant for some time now! > > the svn checkout gave me Annotator, CLAM, NetworkEditor, SMSTools and > Voice2MIDI sources. I am trying to understand the application. Is it > recommended to take that 40 hour tutorial now?!

18 years, 6 months

[linux-audio-dev] Getting out of the software game

by Gordon JC Pearce

After a few days of careful consideration, I've decided that I no longer want to be involved in developing software for Linux. It's been a difficult decision to make, having used Linux as my main desktop OS for around 10 years now, but I feel that the community as a whole is going in a direction that is not compatible with my moral compass. To that end, I'm pulling everything I've written under the GPL or a GPL-compatible licence. If there are copies out there, great, feel free. Anything I'm interested in will be rewritten from the ground up under a BSD-style licence, which to be honest I've always preferred. Part of the reason for this is the increasing difficulty of using binary drivers with Linux. I know a lot of people don't like them, but I like to have things like accelerated video *and* custom kernels without all the buggering about involved in getting it working. In particular the Debian-based distributions seem to be intentionally hamstrung when comes to supporting binary-only drivers, which makes running the custom kernel required for low-latency work *and* the binary nVidia driver almost impossible. I don't want to be associated with this nonsense any more. It's not what Free Software is about. Gordonjcp

18 years, 6 months

[linux-audio-dev] CLAM at the 2007 Google Summer of Code

by Pau Arumi

STUDENT: Audio and Music software is your thing? Need a part-time job of $4500 for this summer? Want to have fun developing free software? We are happy to announce that the CLAM project is participating, for the first time, in the 2007 edition of Google's Summer of Code. CLAM is not the only audio-linux project in GSoC, Ardour and Mixxx also did it. See [1] for the whole list APPLICATION DEADLINE: 24 March 2007 Find all the information in the CLAM web: http://clam.iua.upf.edu and its SoC wiki page [2] We are very excited to offer a number of ideas that would benefit CLAM now that it is about to reach its 1.0 release. We also encourage you to propose new ideas if you feel none of the ones offered by the CLAM team suits your profile or interests. Looking forward to working with you… The CLAM team 1. http://code.google.com/soc 2. http://iua-share.upf.edu/wikis/clam/index.php/GSoC_2007 -- Google Summer of Code is a program that offers student developers stipends to write code for various open source projects. Google will be working with a several open source, free software and technology-related groups to identify and fund several projects over a three month period. Historically, the program has brought together over 1,000 students with over 100 open source projects, to create hundreds of thousands of lines of code. The program, which kicked off in 2005, is now in its third year, following on from a very successful 2006.

18 years, 6 months

[linux-audio-dev] TMP directory problem

by Dave Phillips

Greetings: Okay, I'm stumped by this one. I'm trying to compile a small app on my old machine, it's running Dynebolic with the required devel packages. I've built other apps in this environment with no problem, but when I compile this one app I keep receiving this kind of error : /tmp/cc30Bs2G.o : No space left on the device So how do I get more space in the /tmp directory ? I tried setting TMP and TMPDIR, got no joy. I'll gladly supply more details if needed, but that's basically what's happening. Any suggestions from the gurus ? Best, dp

18 years, 6 months

[linux-audio-dev] Open Sound System v4.0 released

by Hannu Savolainen

FOR IMMEDIATE RELEASE Open Sound System v4.0 Released. CULVER CITY, CA, March 15, 2007: 4Front Technologies is announcing the availability of Open Sound System (OSS) version 4.0 for Linux, Solaris, FreeBSD, Open Server6 and UnixWare7. Open Sound System is a cross platform audio architecture that provides drivers for most consumer and professional audio devices and comes with an API that allows applications to be simply recompiled on any of the supported operating systems. New Features: o New and improved transparent Virtual Mixer engine - Allows up to 16 applications to share the same "real" audio device. - Supports recording and full duplex in addition to playback. - Ability to mix stereo and multichannel audio streams up to 7.1/192Khz/32bit. - Supports full 24 bit range without loss of precision during internal computations. - mmap() support for games like DoomIII and Quake4. - Each application has its own independant volume controls. - Supports loop back recording. o Full Solaris Audio Device Architecture (SADA) emulation on Solaris so that legacy Solaris audio apps can run on Open Sound System drivers. o Advanced Linux Sound Architecture (ALSA) Library emulation support so that popular ALSA apps (the ones that use the ALSA library interface) can run on Open Sound System. o 100% backwards compatibility for Open Sound System (OSS) v3 API. o 64bit internal processing guarantees audio fidelity and precision if the audio data needs to be converted. o New device enumeration and mixer API makes it very easy to manage devices programatically. o Uses up to date native kernel interfaces and installation methods will enable OSS to keep up with changes to the operating systems for the forseeable future. o Updated drivers for all devices supported in the older OSS v3.9x versions. Support for obsolete ISA bus devices has been finally withdrawn from Open Sound System v4.0. The virtual mixer in Open Sound v4.0 will give the user multiple virtual full duplex multichannel audio streams. It is possible to run a full duplex VOIP session, view a DVD in full 5.1 surround sound and play popular video game all at the same time without any complicated device setup and configuration. Open Sound v4.0 brings unmatched device and system management capabilities that make it ideal and easy to setup in virtualized environments. For more information and to download a free-for-personal-use copy of the software, visit 4Front's WWW site at http://www.opensound.com. --- xxx --- All trademarks and copyrights belong to their respective owners. Open Sound System is a trademark of 4Front Technologies. Copyright (C) 1996-2007, 4Front Technologies, All Rights Reserved. Contact: Dev Mazumdar 4Front Technologies 4035 Lafayette Place, Unit F Culver City, CA 90232 USA. Tel: (310) 202 8530 E-mail: info(a)opensound.com Fax: (310) 202 0486 Web: http://www.opensound.com

18 years, 6 months

[linux-audio-dev] Hi, is it possible to record multi channel speech audio and down sample it 2 stereo channels BUT with horizontal audio positioning of each audio channel ie. like dolby surround effect. I'm not looking to make multi channel podcast, just a plain stere

by Steve Harris

On 13 Mar 2007, at 09:15, Valent Turkovic wrote: > Hi, > is it possible to record multi channel speech audio and down sample it > 2 stereo channels BUT with horizontal audio positioning of each audio > channel ie. like dolby surround effect. Sure, but you might be disappointed by the results, Pro Logic is not very sophisticated (eg. there's no separate rear left and right channel, and the bandwidth is limited). Also the only free software implementation that I know of is not state of the art, partly due to patent problems, and partly due to lack of interest. There's a LADSPA plugin: http://plugin.org.uk/ladspa-swh/docs/ladspa- swh.html#id1401 which does Pro Logic matrix encoding, you will need some other software to mix your independent streams into LCR channels (you can do it by ear with a 3+ variable send bus mixer), but once you have that you can feed it to the plugin and get a pro logic compatible stream out. PS ignore the documentation, it has been tested, and it does work, just not brilliantly. You will get better results if you EQ the mics a bit to cut the level of the highs and lows a bit, especially in the rear channel. You would get much better results with an ambisonic stream, but I don't /think/ you can make stereo-compatible ambisonic streams, and not many people have ambisonic decoders. Other people on this list know a lot more about ambisonics though. - Steve

18 years, 6 months

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Linux-audio-dev