Linux-audio-dev

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7513 discussions

Re: [linux-audio-dev] Realtime audio and IO schedulers

by Florian Schmidt

On Sun, 23 Jan 2005 10:44:48 +1100 Cal <cal(a)graggrag.com> wrote: > Indeed it does, so let's not stir it up again unnecessarily! Sorry, i did not see a reply in that thread. Flo -- Palimm Palimm! http://affenbande.org/~tapas/

20 years, 7 months

Re: [linux-audio-dev] Re: Fixing Freeverb and Denormals

by Neil Klepeis

Hi, I compiled the CMT plugins with the g++ options -msse2 and -march=i686 and the freeverb denormals problem (seen with ardour on a dual Xeon machine) seems to be _almost_ entirely fixed. Also, I don't see any weird problems with the ardour host (yet :). But, I still see the DSP load increasing from around 5% to about 15%, although now it is _much_ farther along in the tails of the reverb than before and I get _zero_ xruns. [before, the DSP shot up to 60-90% with tons of xruns and usually the zombification of ardour]. Much cooler. --Neil PS: I also am using the new undenormalize inline function described in an earlier message. On Sat, Jan 22, 2005 at 03:17:28PM +0100, Christian Schoenebeck wrote: >> I think for such an important piece like freeverb it might be worth to add >> architecture dependant assembly optimizations. E.g. with SSE(1) you simply >> have to change the SSE control register to get rid of the denormal problem. >> With that no denormal fixes for individual sample points are necessary >> anymore which definitely pays off in performance. Agreed, but I'm not sure how doing that will affect the host - ie does -mfpumath=SSE (or whatever the option is) always leave the FPU stack in a sane state when control goes back to the host? I dont think it would normally return the FPU to 387 mode, which might do bad things to the host occasionally. OTOH it might be fine, I dont understand the interaction with the 387 and SSE. - Steve

20 years, 7 months

[linux-audio-dev] Fixing Freeverb and Denormals

by Neil Klepeis

Hi all, The denormal problem with freeverb/ardour on P-4's is killing me. Since CMT plugin development seems to be a little stagnant, I decided to take it into my own hands a bit, but I might need a little help from a kind someone out there. Is there anyone who can give me a tip on proper syntax for changing the denormals.h file in freeverb (assuming this is where the problem lies)? Details: On this list and/or posts elsewhere I found that the denormals.h file for freeverb doesn't work too well with some newer gcc compilers; one recommendation was to exchange the first statement below with the second definition: #define undenormalise(sample) if(((*(unsigned int*)&sample)&0x7f800000)==0) sample=0.0f static inline float undenormalise(volatile float s) { s += 9.8607615E-32f; return s - 9.8607615E-32f; } I tried changing it out one-for-one, but.. no good. Ardour still zombifies after reverb tails die out. Maybe there is somewhere else that needs changing wrt denormals, or maybe I am just clueless :). Anyway, I then tried using something like the following....but still no good, [ardour now doesn't zombify, it's just that freeverb sounds like @$!&]. #define static inline float undenormalise(volatile float s) { s += 9.8607615E-32f; return s - 9.8607615E-32f; } You can probably see that I have no idea what I'm doing, but this seems like it should be so _simple_. Maybe now is the time to go read up on C++. Thanks, Neil

20 years, 7 months

Re: [linux-audio-dev] Re: Fixing Freeverb and Denormals

by Neil Klepeis

Steve, OK. I put your denormal-zeroing function w/SSE into the freeverb.cpp code, and when compiling with the SSE option and calling set_denormal_flags() in the activateFreeverb3 function (see below), I get this error wherever asm is called: freeverb/freeverb.cpp: In function `void set_denormal_flags()': freeverb/freeverb.cpp:48: error: can't find a register in class `BREG' while reloading `asm' Do you have an idea of what might be wrong? Here is where I put the call to your function. Does this make sense? void activateFreeverb3(LADSPA_Handle Instance) { Freeverb3 * poFreeverb = (Freeverb3 *)Instance; poFreeverb->mute(); /* call the Steve Harris denormal zeroing function w/SSE flag */ set_denormal_flags(); } --Neil PS: Maybe we should move this discussion to just linux-audio-dev for now PPS: For reference, this is your function for zeroing denormals w/SSE that I am using: #ifdef __SSE__ #include <xmmintrin.h> #endif void set_denormal_flags() { unsigned long a, b, c, d; #ifdef __SSE__ asm("cpuid": "=a" (a), "=b" (b), "=c" (c), "=d" (d) : "a" (1)); if (d & 1<<25) { /* It has SSE support */ _MM_SET_FLUSH_ZERO_MODE(_MM_FLUSH_ZERO_ON); asm("cpuid": "=a" (a), "=b" (b), "=c" (c), "=d" (d) : "a" (0)); if (b == 0x756e6547) { /* It's an Intel */ int stepping, model, family, extfamily; family = (a >> 8) & 0xf; extfamily = (a >> 20) & 0xff; model = (a >> 4) & 0xf; stepping = a & 0xf; if (family == 15 && extfamily == 0 && model == 0 && stepping < 7) { return; } } asm("cpuid": "=a" (a), "=b" (b), "=c" (c), "=d" (d) : "a" (1)); if (d & 1<<26) { /* bit 26, SSE2 support */ _mm_setcsr(_mm_getcsr() | 0x40); } } else { fprintf(stderr, "This code has been built with SSE support, but your processor does not support\nthe SSE instruction set.\nexiting\n"); exit(1); } #endif } -------------Steve Harris wrote:--------------- For something like a reverb the best option is going to be to build with SSE maths and call the function I posed here a week ago or so. There are so many places where denormals can occur in reverbs that youre going to burn a load of CPU trying to kill them. OTOH I'm not sure how that function would interact with the host when loaded as a plugin

20 years, 7 months

[linux-audio-dev] [PATCH]: ALSA 1.07 support for Novation X-Station/ReMOTE25/Speedio

by Charles C. Bennett, Jr.

Hi All - Thanks to the assistance of Nick Dowell at Focusrite I have working MIDI over USB for Novation ReMOTE 25 Audio AKA X-Station. The attached patch builds and runs cleanly using the latest Fedora Core 3 update kernel (2.6.10 #741). I don't have the MIDI only device or the Speedio to test with but if they operate as Nick describes these should also do MIDI over USB with my patch. I don't think I broke anything but if anybody can check out this patch with other devices, particularly Roland and MIDIMAN devices I'd be much obliged. Thanks, ccb

20 years, 7 months

[linux-audio-dev] MuSE or Glame Bandpass Analog Filter

by Dubphil

Hello all, After having struggle a lot in order to make ecasound and ardour working on my Gentoo (Christopher is it working well with your's ?) I have decided to use another strategy to setup my Dub Soundsystem. I have choosen : always TerminatorX (Tx) to play the tracks and MuSE 0.7.1 to do the rest. The problem is that MuSE does a strange thing. Let me explain : I have 2 stereo tracks track 1 connected to Tx without effect track 2 connected to Tx with the Glame Bandpass Analog Filter So, if I activate the plugin of the track 2 the track 1 looks mono and I can hear that the effect is also acting on it. is this a bug in MuSE or it is the Glame Bandpass Analog Filter to claim the fault ? Best regards Philippe PS : it would be nice if the interface window of a plugin closes when we remove the plugin from the track :)

20 years, 7 months

[linux-audio-dev] linuxsampler cvs-snapshot from 13.06.2004 controlling it

by Julien Claassen

Hi! I tried to use the cvs snapshot from the 13th of June last year to do some work. But it wouldn't accept commands. I did: $ linuxsampler It told me that everything was well. Then I connected to it with telnet: $ telnet localhost 8888 That succeeded, too. Then I tried just writing a comment: # this is a comment Still worked, but when I tried to put any real command to it, it just said: ERR:0:Unknown command. Or something like it. I took the commands from the draft. But the sampler even wouldn't accept "quit". I tried quit, Quit and QUIT and the same with single and double quotes. Any fix for this? I need to use the text-mode (telnet way). Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net - the Linux TextBased Studio guide

20 years, 8 months

[linux-audio-dev] [ANN] Snd-ls V0.9.2.0

by Kjetil Svalastog Matheussen

http://www.notam02.no/arkiv/src/snd/ Snd-ls v0.9.2.0 --------------- Released 21.1.2005 Contains -------- Snd v7.10 from 20.1.2005 About ----- Snd-ls is a distribution of the sound editor Snd, made by Bill Schottstaedt. Its target is people that don't know scheme very well, and don't want to spend too much time configuring Snd. It can also serve as a quick introduction to Snd and how it can be set up. Install ------- 1. Edit the file config.scm, for configuration settings. 2. Run ./build 3. Run ./install as root. It is not necessarry to uninstall any previously installed versions of Snd. Snd-ls should not interfere with already installed versions. After installing, the name of the executable is "snd-ls". To uninstall, run ./uninstall Required packages ----------------- guile gtk-2 jack libsamplerate ladspa liblrdf fftw3 libxt-dev (guile-devel, gtk-2-devel, liblrdf-devel and gcc is also needed at runtime) Changes ------- 0.9.2-beta1 -> 0.9.2.0 -Updated SND to v7.10 from 20.1.2005. Many important changes. 0.9.1.5 -> 0.9.2-beta1 -Updated SND to v7.8 from 12.8.2004. Many important changes. 0.9.1.4 -> 0.9.1.5: -Made the apply-button always apply to selection if there is a selection. 0.9.1.3 -> 0.9.1.4: -Removed some options from the edit-menu that interfered with the way snd_conffile works. 0.9.1.2 -> 0.9.1.3: -Made the insert-option in the edit-menu insert monofiles into the currently selected channel. -Added libxt-dev to the list of required packages. (Thanks to robin) -Made mono-files play in both left and right channel when using Jack. 0.9.1.1 -> 0.9.1.2: -Short fix for the nodeline-class. 0.9.1 -> 0.9.1.1: -Fixed Append File edit-menu option. -Workaround for trouble with ladspa default settings. 0.9.0 -> 0.9.1: -Official announced. -Upgraded SND from 20.7.2004 to 2.8.2004 -Huge amount of testing at Notam by 14 unexperienced guinea pigs for a whole week; many bugs fixed. Links ----- Snd: http://ccrma.stanford.edu/software/snd/ Guile: http://www.gnu.org/software/guile/guile.html Credits ------- Snd is made by Bill Schottstaedt. This small package is put together by Kjetil Matheussen / Notam, with consulting help from Bill Schottstaedt. --

20 years, 8 months

[linux-audio-dev] Re: Plans/Roadmaps for 2005?

by Juhana Sadeharju

>From: Mario Lang <mlang(a)delysid.org> > >[1] http://delysid.org/tuneit.html First, I have one more project: -Overview file format for libsndfile; e.g., (min,max,rms) per block; the overview file would be used, e.g., for fast waveform drawing Then, thanks. I will check the pitch detectors you have. I have collected papers on pitch detectors if you want take a look at them -- I will check too, of course. Do you know how your code could be changed to the formant detector? It must be a harder project. Two first formants should be tracked. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

20 years, 8 months

[linux-audio-dev] strange error from linuxsampler

by Fons Adriaensen

Hello, qsampler gives me this strange error: Loading gig file 'NON_MODAL '/home/fons/FreePiano.gig''...gig::Engine error: Failed to load instrument, cause: Can't open "NON_MODAL '/home/fons/FreePiano.gig'" I've no idea where the 'NON_MODAL' thing is coming from, or how to get rid of it... -- FA

20 years, 8 months

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Linux-audio-dev