Linux-audio-dev

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3 participants
7535 discussions

by jani＠lasipalatsi.fi

Hi, I've written a small program (Leevi is its name) to drive Lexicon MX300, similar console what Lexicon ships for Windows operating system along with their devices. Leevi supports Linux and BSD, and needs either libusb-0 or libusb-1. It does not require any user interface libraries, as everything is built on libX11, so if you can boot to runlevel 5, you can run Leevi. Please note: As Lexicon wasn't very keen to tell me how to talk with their devices, USB protocol is reverse engineered by me and there are myriad of things that I haven't yet revealed. Although Leevi is proven to work without breaking anything, there is always a small change that something goes wrong. Therefore I don't take any responsibility, you use Leevi at your own risk. Check BSD License in Leevi's homepage. Leevi is yet in development stage, changing effect in stereo mode and changing routing to/from stereo does not yet work. All other features should be fully functional. http://leevi.sourceforge.net - Jani Salonen

14 years, 11 months

[LAD] Writing an IEEE 802.1BA (AVB) compliant network backend for Jack2

by Christoph Kuhr

Hi list, i want to write a jack2 network backend, like netjack2, for my bachelor thesis. There is a new network standard comming up specialized for audio and video transmission, called AVB (Audio Video Bridging -> IEEE 802.1AS, 802.1Qat, 802.1Qav,...). I want do integrate this standard into jack2. I want to choose AVB (not ALSA or NET) from the driver dropdown box. does anyone know where i can find this implementation in the source code?

14 years, 11 months

Re: [LAD] [ANN] IR: LV2 Convolution Reverb

by drew Roberts

On Thursday 24 February 2011 12:09:19 David Robillard wrote: > This is, of course, a big problem in terms of our greater mission > to provide software that caters to the needs of precisely nobody while > irritating everybody else. > > To resolve this situation, we now have an exciting new Clippy inspired > assistant that hops around your screen begging you to add MIDI tracks > constantly. > > If, after 20 minutes, you have still not created a MIDI track, Ardour > will overwrite every .waf file found in your home directory with a 4/4 > electronic kick drum loop, then shut down. > > -dr Dude! This is just what I have been wanting but am too much of a newb to have even thought of undertaking such a complex project on my own! Hence my starting my soundwall project which I newbily thought was going to be a simple little app. I have some code inspired by some "anon" silence that could perhaps be included in this project. wwnnsnmsnm. cage-ily, drew

14 years, 11 months

[LAD] the future of display/drawing models (Was: Re: [ANN] IR: LV2 Convolution Reverb)

by Paul Davis

On Tue, Feb 22, 2011 at 3:49 PM, Philipp Überbacher <hollunder(a)lavabit.com> wrote: > The rest sounds nice, and it might well be that X has become old, but I > don't see the big improvement coming up. Windows are called surfaces > now, can have different shapes and are more flexible, compositing, > transformations, I got that bit, but I don't see the UI improvement. > I've seen the demos with shapes flying around the desktop, I've seen > the conventional compositing window managers and wayland will probably > do all that and more, but I don't see the improvement in User > Interfaces. what its going to do,i think, is two-fold: 1) promote more and more toolkit design that makes everything just a compositing stack. GTK has already moved significantly in this direction, but could go a lot further. Qt is in a similar position. the more this happens, the easier it is to reason and create new GUI widgets that do cool things, easily and simply, because its all part of a very simple model: you draw to your surface, it will be composited onto the screen in ways that you don't have to worry about. sounds a bit like X ... except that X is explicitly *not* a compositing model. for a simpler explanation of the kind of thing i mean, consider the difference in ardour between the main "tracks" area of the editing window and all the widgets around it. its fundamentally impossible to implement the tracks with widgets - it uses a "canvas" object instead which embodies idea like z-axis stacking, transparency and so forth. but likewise at present it would be a lot of work to implement all the widgets as canvas "items". now fast forward a few years, and find a spot where the drawing model for the canvas, the button widgets, the tree/listviews, for everything *inside* the program is the same as the model for everything *outside* the program. drawing a particular "thing" on any other thing becomes identical, whether the other thing is a "window", a "button", a cell of a listview, etc, etc. 2) more and more apps able to take advantage of v-blank sync to reduce computational load due to unnecessary redraws. instead, the whole system will be a lot like a video-framebuffer version of JACK: the vblank interrupt arrives. everything with a surface gets a chance to redraw if it needs to, the surfaces are composited together, and boom, its on the display. no more guessing how often to redraw stuff, no more wierd ass hacks to get smooth animation, etc. if you think this sounds like special effects, i suggest a few minutes playing with a relevant iPod/iPhone/iPad app where these smooth transformations of what is on the screen is a central metaphor in how the UI's work.

14 years, 11 months

[LAD] Call for alpha testers: FLAM

by Luis Garrido

Oh, well, since GUIs are this week's flavor in the list, I might as well throw FLAM's alpha in the mix. I am preparing FLAM's (Front-ends for Linux Audio Modules) first release. Among other things, FLAM intends to allow programmers and non-programmers alike to create their own (external) GUIs for audio plugins. At this moment only Rosegarden as a host and LADSPA as plugin type are supported, but this is hopefully just a first step. Project page: http://vagar.org/code/projects/flam Tutorial with a few screenshots: http://vagar.org/asciidoc/flam/primer/primer.html Source code repository: git clone http://vagar.org/git/flam I'd welcome any feedback, here or in the FLAM forums (registration required): http://vagar.org/code/projects/flam/boards Thanks! Luis

14 years, 11 months

[LAD] do any of the jack client examples show playing a file from disk?

by drew Roberts

do any of the jack client examples show playing a file from disk? if so, which? if not, any links to simple code that does this? c++ or c? thanks in advance for any pointers. drew -- http://freemusicpush.blogspot.com/

14 years, 11 months

[LAD] Dealing with time

by Joel Roth

Hi all, This meditation isn't about gray hair, sagging flesh or receding libido. :-) In debugging Nama's new audio editing functions, I'm noticing that I have more and more code dealing with time. I am concerned seeing a multiplicity of method names like adjusted_region_start_time or unadjusted_mark_time. I'm wondering if perhaps I can centralize or at least systematize this functionality. Nama deals with several kinds of time: Ecasound time. Positions in seconds or samples from the perspective of the Ecasound audio engine WAV time. Displacements in audio files. Track / Region time. Positions in a track or region Mark time. Nama currently has only one type of mark: marks anchored to an absolute project timeline. I think it also needs marks for positions in a tracks WAV files, which the user may trim or offset. Edit / offset-run time. This is what started the entire issue. To record a fix for a note at time T in a WAV file W, I use Ecasound's select object to offset all the WAV files in a project to start at time T. The fix, W', then gets placed at T using playat. MIDI time. There are plenty of references on this, and it's a subject of its own. Nama needs at least to know enough to work across the various systems for bridging between ALSA, JACK and MIDI. So do I need some Big Abstraction(tm), or shall I just continue to work incrementally? How do you think about time? I don't expect a simple answer or an answer at all, but it can help to formulate the question. Regards, Joel -- Joel Roth

14 years, 11 months

[LAD] [ann] nova-simd

by Tim Blechmann

hi all, for some time, i have been working on nova-simd, a cross-platform SIMD library library written in c++. it abstracts the platform-specific APIs and provides a `vec' class, which maps to a SIMD floating-point vector on the target machines and which can be used to formulate more complex algorithms. for some commonly used vector functionality, helper functions are provided, which are built using the vec class. features: - same interface for different SIMD architectures - supported backends: sse-family, avx, altivec, arm/neon - some backends include vectorized implementations of libm functions - extensively used in supercollider - header-only c++ (no runtime dependencies, composable), no dependencies canveats: - little documentations/examples - no release, no tarball, just a git repository [1] and a web interface [2] - header-only c++ (no c support) maybe it is useful for other people as well... cheers, tim [1] git://tim.klingt.org/nova-simd.git [2] http://tim.klingt.org/git?p=nova-simd.git;a=summary -- tim(a)klingt.org http://tim.klingt.org Art is either a complaint or do something else John Cage quoting Jasper Johns

14 years, 12 months

Re: [LAD] Calculate R M S (Alfs Kurmis)

by Jeff McClintock

> From: Fons Adriaensen <fons(a)linuxaudio.org> > On Fri, Feb 18, 2011 at 07:36:44AM +1300, Jeff McClintock wrote: > > > With a RMS VU Meter you measure a 1KHz tone as a reference. > > A contradiction... A VU does not measure RMS, whatever does measure > RMS is not a VU. Isn't a VU Meter a standard root-mean-square function followed by a 300ms integration to give it some 'weight'? ...calibrated against a 1kHz tone?

14 years, 12 months

[LAD] Calculate R M S

by Alfs Kurmis

Hi Experts. I wanna normalize my sound stream by loudness (energy / pressure / intensity) , not by peaks. How i do it ? Is available Jack plugin for so what ? What is (we hear as) "loudness" ? RMS or +(average) or something else ? Is somewhere available examples how to calculate RMS ? Is it done simlpe by :  int i, n;  double sums, rms;  sums=0.0;  n=10;  rms=0;  for(i=0; i<n;i++)  {   sums = sums + ( (double)i * (double)i );  }  rms = sqrt(sums / n);  printf("          rms = %12.12fnn", rms ); Is so sipmple algo good enough for frequencies > 10 kHz ? How to calculate RMS with hi-precision for frequencies > 10 kHz ? With inerpolation or so ? What is reference ( 0dB ) RMS for example for 16bit PCM signal 1024 samples ?   sqrt( 1024*(0x7FFF ^ 2) / 1024 )  ==  sqrt( (0x7FFF ^ 2)  ) == 0x7FFF  Is it simple  0x7FFF (32000 dec) ? How to calc RMS for stereo signal ? So , or somehow else ? for(i=0; i<n;i++)  {  ....    sums = sampleL/2 + sampleR/2;  } In case operating with float point, what should be a bit faster,  / 2  or * 0.5 ? How to gotta RMS value in dB ?    20 x log10(Measured/Reference0_dB)  or 10 x log10(Measured/Reference0_dB)  ?? Im just physics student and im new in DSP, so pleaz no angree about simple or stuppid questions. Any examples and hints welcomed. Many Tnx in advance. Alf ----

14 years, 12 months

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Linux-audio-dev