Linux-audio-dev August 2010

linux-audio-dev@lists.linuxaudio.org

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Faust MIDI?

by Carlo Capocasa

Hi, is it possible to send and receive MIDI with a FAUST program? Carlo

11 years, 6 months

Port Argotlunar VST to LV2!

by Louigi Verona

Hey guys! Will anyone be interested in porting this GPLed VST to LV2? http://argotlunar.info/ Louigi Verona.

13 years

Re: [LAD] FOSS Ethernet Soundcard - IETF Ethernet AVB

by errordeveloper＠gmail.com

Hi everybody, Fallowing up the long discussion i'm trying to sort of give the information you all seem to be missing. there was a meantion of this, but not too many of you have paid any attention: here you can find an article about the current status of the protocol mess: http://prosoundnewseurope.com/pdf/PSNLive/PSNLive_2009.pdf (page 28) obviously eas50 is good to go, but Ethernet AVB is right thing really. the only thing that it's still work in progress, but many of the proprietary vendors, which already have their own networking solution (like Harman with HiQ-net, the one i can name of top of my head) are involved in AVB stadard deveelopment. The idea of AVB is to bypass the IP layer, which is right thing really. you don't need to assign IPs to your audio nodes, really! in avb you'd just have to select channels that nodes whant to listen to. there is a fair bit of documentation on the ietf.org AVB group's page. but XMOS is looking to be the best point of refference: http://www.xmos.com/news/15-jun-2009/xmos-simplifies-ethernet-avb-implement… is think we should forget everything else and crack on with the XS1 AVB implementation! their XS1 chips seem to be really great, their are basically every innovative and open-source minded. the official toolchain is LLVM-GCC based. you can use C, C++ or their own XC. XC is basically C with some stuff omited (like goto and floats) and XMOS IO stuff added, don't just say WTF, look at it first! you should also watch the videos here: http://www.xmoslinkers.org/conference-online-wf especialy the two about the "XMOS Architecture" and the AVB presentation. some dev-kits are quite expencive, but that's due to low-volume really ;) there is alos a nice USB Audio kit! plus there is alittle board that is cheap and has two RJ45's on it already :) I'm myself studying the XC book at the moment. And geting familiar with the tool set :)~~ looks very exciting, cause these are the invovative chips! ok, may be an FPU is really missing on XCore, but how many DSPs have it anyway? well quite a few, but there was no FPU on dsps for ages! :)) also XC or C/C++ are so much more obvious then the bloody "menthal american military engeneers non-sense" called HDL-whatever! Cheers Everyone, Hope you will appreciate my excitment :)~ (l0l) -- ilya .d

13 years

any pointers for RT kernel for ARM?

by nescivi

Hiho, I managed to get SuperCollider and JACK running on my IGEP [1], on the pre- configured Ubuntu on the SD, but of course the audio is still bumpy. So... I'm looking for a RT kernel for this little machine... Anyone have any pointers? I did find this one: http://beagleboard.org/project/omap-rt-patch/ but the project doesn't show much recent activity. and... I also noticed that JACK didn't want to run with alsa as backend; oss as backend works though (but gives the bumpy sound). Anyone willing to share their experiences doing linux audio on ARM processors? sincerely, Marije [1] http://www.igep-platform.com/

13 years, 2 months

Best way to insert DSP module in audio chain in Linux

by Kim Therkelsen

Hi, I want to make is a DSP module that improves the sound quality of the sound coming from the laptop builtin speakers by applying DSP (FIR/IIR filters). I want to route all audio played back (from for instance an offline WAV or MP3 file, from a movie, or streamed from the internet) to pass through the DSP plugin whenever headphones are NOT connected. When headphones are connected no processing should be applied. I want to place the DSP module as close to the hardware as possible to make sure all audio is really routed through the DSP plugin so I always can enjoy the DSP improvements made to the sound. Where should I place this module/DSP code? - As a plugin to PulseAudio? - As a plugin to ALSA? - As its own virtual sound card? - In the audio driver for my built-in sound card? - As a kernel module? I have tried loading the LADSPA-module using the module-ladspa-sink in PulseAudio but I am not sure this is the best solution. I need to remove some limitations in module-ladspa-sink to get it working properly ( multichannel audio, only DSP processing for internal speakers). My requirements: 1) Must be able to detect if headphones are connected 2) Must be able to process stereo and multichannel (5.1/7.1) formats. I need the multichannel formats to perform binaural downmixing to stereo. So it is important that I can receive multichannel audio and mix it down to stereo and output it to a stereo soundcard. /Kim

13 years, 3 months

Re: [LAD] mixer

by fons＠kokkinizita.net

Alex, I hope you don't mind quoting part of your message on the list, as the subject is of general interest. Alex stone wrote offlist: > Fons, i've read in the mailing list in the thread concerning mixers, > that you have something you're working on. > ... > More than any other facet of working in software for audio/midi/vid, > i've been frustrated and tired out by poor navigation options, and i > don't think i'm on my own. > The mouse is insufficent for ease, speed, and pain free day in day out > use, and i ask you to at least consider (or not, as you decide) > alternatives such as the one above, to make a difference in the way > users commonly interact with this type of software. I more than agree with this. Defining the way a user interacts with a large software mixer is 90% of the work of writing one. Common GUI methods just don't work for this sort of thing. The thing that is slowly taking some form here will observe some GUI guidelines quite strictly, in particular there will be *NO* repeat *NO* resizing, scrolling or moving of windows, menus will be used sparingly and be small, and popup dialogs will be used only for operations that can be assumed to be infrequent (mainly setting up things, selecting and connecting plugins, etc.). When setting up a new session, you define the resources: - number of channel strips - number of real groups - number of control groups - number of visible sections - number of strips per section - number of jack/alsa inputs - number of jack/alsa outputs These are more or less fixed afterwards, in the sense that you can't remove any, but probably I can allow to increase these numbers (the point here is that all snapshots taken for such a configuration must remain valid without requiring changes to the resources). Anyway there is no penalty (apart from a small amount of memory used) for providing more than you need. A _section_ is a group of strips, typically 8, 10 or 12. Normally you should select the section size so you can have two of them side by side, as this facilitates many operations. For each visible section there is a control strip, which contains e.g. buttons to select which strips are shown. You will have default sections, e.g. inputs 1-8, 9-16, 17-24, etc, same for real groups and control groups. There are also user-defined sections: these can contain copies of any strip you want in any order you want. For example you could group all inputs used for drums in a secion, or have a 'master' section that contains mainly groups and maybe some solo mics. There is some 'intelligence' in the section selection buttons, for example, if you select to see inputs 17-24 in the right section, you click that button (or use a kb shortcut), then clicking again restores the previous view. Or selecting a section that is already visible will swap the two, etc.etc. Each visible section is vertically divided in two, the bottom half showing the faders, channel ID, mute, solo etc. most of the time, while the top half can show parts of the strips, e.g. EQ, sends or dynamics. Again this layout is configurable to some extent, you can e.g. select to have some aux sends visible all the time. The top part of each visible section can also be used for plugins, metering, 'visual' panning, or whatever extra functions that would need to be displayed. All Jack/Alsa ports are generic, each of them can be assigned to any function inside the mixer. That means e.g. that Jack connections can remain fixed no matter how you 'rewire' the mixer, and this in turn means that nothing is lost if you connect directly to a multichannel Alsa device, e.g. a MADI card. The mixer will have its own plugin interface, and it will not accept any of the existing standards. If someone wants to use a LADSPA or LV2 I will consider porting it. Condition for this will be the *quality* of the plugin, and nothing else. Ciao from very hot Crete (I'll be cooling down 20m below the water surface in a short time). -- FA There are three of them, and Alleline.

13 years, 3 months

Fwd: Claim: Microsoft Pressured Digidesign to Deny Support of GNU/Linux

by Niels Mayer

Comments anybody :-) http://techrights.org/2010/08/27/throwing-a-wrench-at-protools/ ........................... I have heard from a reliable source, inside Digidesign, that they actually have Protools running on Linux, and that the port from OSX isn’t that hard, but are under contract obligation to Microsoft to not release a Linux port. Otherwise Microsoft can revoke their access to the Windows SDK. .......................... Niels http://nielsmayer.com

13 years, 3 months

Re: [LAD] Memory corruption&coredump on Exit in my first vala program spekle

by Niels Mayer

On Sat, Aug 28, 2010 at 6:44 PM, Alexander Kojevnikov <alexander(a)kojevnikov.com> wrote: > You can try running spekle under valgrind [1]. After testing it on a > flac file I already see that spek_spectrogram_put_pixel() is used with > wrong arguments causing an out-of-bounds memory write. Thank you for the clue. I'm especially thankful that I won't have to run valgrind for hours on end with my CPU fans whining at top-speed just to decode a video or mp3 :-) ... I will just go straight to debugging spek_spectrogram_put_pixel(). I've added your suggestion to my issue: http://code.google.com/p/spekle/issues/detail?id=1#c3 > On a side note, feel free to submit patches adding batch support to > Spek, there's even an open issue for it [ http://code.google.com/p/spek/issues/detail?id=6 ] I added the following comment on your issue: http://code.google.com/p/spek/issues/detail?id=6#c5 ........... http://spekle.googlecode.com is a fork of Spek 0.6 specifically designed for batch use and for usage without an X server or GUI. It would be great if the changes for spekle could be incorporated back into Spek. Although down the road, some functionality I plan to add in spekle ( http://vamp-plugins.org ) may be out-of-scope for Spek. It's not something that'll be an obvious patch since I replaced spek.vala with http://spekle.googlecode.com/svn/trunk/spekle/src/spekle.vala and and spek-window.vala turned into http://spekle.googlecode.com/svn/trunk/spekle/src/spek-object.vala and there's a different set of build targets as well http://spekle.googlecode.com/svn/trunk/spekle/src/Makefile.am ... ........... Other code-mangling includes needing to add an extra callback parameter to the Pipeline() constructor: http://spekle.googlecode.com/svn/trunk/spekle/src/spek-pipeline.vala and lots of small changes to http://spekle.googlecode.com/svn/trunk/spekle/src/spek-spectrogram.vala . To recognize what changed, comment-outs of your original 'spek-0.6' code are done with "//" at the beginning of a line, and most of my changes are marked with an "NPM" comment. Since I just got 'spekle' running, and my current changes are a first-pass hacking of your original spek sources, it probably makes sense for me to continue doing development and debugging on 'spekle' and at some point in time in the future, submitting a more definitive patch for http://code.google.com/p/spek/issues/detail?id=6 . In case you're interested in collaborating on spekle, I added you as a committer (akojevnikov). It may also make sense to test out batch-mode ideas in spekle first and then move them back to 'spek' when they've settled down. Let me know how you wish to proceed, and thanks for making Spek available and GPL! -- Niels http://nielsmayer.com

13 years, 3 months

Memory corruption&coredump on Exit in my first vala program spekle

by Niels Mayer

Anybody have any ideas on http://code.google.com/p/spekle/wiki/EvidenceOfMemoryCorruptionOnExit (warning contains large images, http://spekle.googlecode.com/svn/wiki/evidence-of-memory-corruption-on-exit… for error text only). issue: http://code.google.com/p/spekle/issues/detail?id=1 Summary: after successfully writing a spectrogram image in my first Vala programming http://spekle.googlecode.com/ (a displayless-version of Alexander Kojevnikov's http://www.spek-project.org/ ) I get the following error, despite not explicitly calling free() out of my Vala code: Spekle: saving 'j_s_bach-d_minor_invention.ogg.png'... DONE! *** glibc detected *** spekle: free(): invalid pointer: 0x00007fa7e05ca010 *** ======= Backtrace: ========= /lib64/libc.so.6[0x3ac0874ac6] /usr/lib64/libpixman-1.so.0(pixman_image_unref+0x83)[0x3ac9e161b3] /usr/lib64/libcairo.so.2[0x359c016b85] /usr/lib64/libcairo.so.2(cairo_surface_finish+0x36)[0x359c02c9d6] /usr/lib64/libcairo.so.2(cairo_surface_destroy+0x55)[0x359c02ca45] spekle[0x406b1e] /lib64/libgobject-2.0.so.0(g_object_unref+0x15f)[0x3ac380da1f] spekle[0x407d3d] /lib64/libgobject-2.0.so.0(g_object_unref+0x15f)[0x3ac380da1f] spekle[0x403857] /lib64/libc.so.6(__libc_start_main+0xfd)[0x3ac081eb1d] spekle[0x403439] The errors seems to occur intermittently, and more often when using media that is ogg. Retrieving via HTTP seems to reduce the error potential, but it still happens that way too... The relevant libraries and compiler I'm using: vala-0.9.3-1.fc12.x86_64 vala-tools-0.9.3-1.fc12.x86_64 vala-devel-0.9.3-1.fc12.x86_64 libpng-devel-1.2.44-1.fc12.x86_64 libpng-1.2.44-1.fc12.x86_64 gtk2-devel-2.18.9-3.fc12.x86_64 gtk2-2.18.9-3.fc12.x86_64 ffmpeg-0.6-2.fc12.x86_64 ffmpeg-libs-0.6-2.fc12.x86_64 ffmpeg-devel-0.6-2.fc12.x86_64 See http://spekle.googlecode.com/svn/trunk/README for instructions on obtaining by 'svn' and building on Linux. Niels http://nielsmayer.com

13 years, 3 months

mudita24 peaks & peak reset

by David Santamauro

Hi, As I was checking out the new mudita24, I noticed that the peak line on the meters doesn't go to zero after the signal is 'silence'. I have to manually click "reset peaks". Is this on purpose? David PS: nice overhaul, great work, thanks.

13 years, 3 months

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Linux-audio-dev August 2010