Linux-audio-dev October 2003

linux-audio-dev@lists.linuxaudio.org

60 participants
67 discussions

by Tim Hockin

I'm potentially in the market for a new box before end of year. Has anyone found anything to show whether Athlon64 (or the high-end FX line) are worth it for audio? Assume some legacy audio code under legacy OS :-/ Tim

20 years, 5 months

[linux-audio-dev] Fixed vs Floating point comparison.

by Juan Linietsky

Benno Sennoner and I were discussing today on IRC about the usual fixed point vs floating point (regarding to some resampling code) We developed some tests and ran them on a variety of computers. It would be interesting if ladders here could run them on different computers (and specially non x86, like amd64. or the Gx processors) so we can see what performance can we expect on each , and how things seem to be shaping for the future. It will also be a key factor on how our projects will develop in the future. The code is available at: http://reduz.dyndns.org/resamp_fixp.c // fixed point version http://reduz.dyndns.org/resamp_float.c // floating point version, portable http://reduz.dyndns.org/resamp_float_fistl.c // X86 VERSION ONLY!! Uses fistl instruction The results dont mean just int vs float performance. They also test for float->int conversion, which is common in most algorithms that work with buffers. It is a linearly interpolated resampling with volume control. please use GCC options - -O3 -ffast-math -march=<yourcpu> so it's a fair comparison results from other compilers is also appreciated! Here's some results for reference **************************************************************** vendor_id : AuthenticAMD model name : AMD-K6(tm) 3D processor cpu MHz : 412.508 cache size : 64 KB flags : fpu vme de pse tsc msr mce cx8 pge mmx syscall 3dnow k6_mtrr bogomips : 822.47 resamp_fixp - 0m8.460s resamp_float_fistl - 0m27.390s **************************************************************** vendor_id : AuthenticAMD model name : AMD Duron(tm) Processor cpu MHz : 951.701 cache size : 64 KB flags : fpu vme de pse tsc msr pae mce cx8 sep mtrr pge mca cmov pat pse36 mmx fxsr syscall mmxext 3dnowext 3dnow bogomips : 1900.54 resamp_float - 0m11.180s resamp_float_fistl - 0m5.810s resamp_fixp_optimized - 0m2.790s ************************************************ Benno gave me some results: Intel p4 1800 celeron resamp_float_fistl - 4.00user resamp_fixp - 4.51user (float faster?) -------------------------------- VIA nehemiah 1GHz resamp_float_fistl - 0m21.079s resamp_fixp - 0m7.129s ************************* Some results on SPARC: 2x 125MHz HyperSPARC resamp_fixp - user 2m59.010s resamp_float - user 0m44.290s (float faster! :) **************************** Intel 2.4 GHZ P4: resamp_float - 0m3.440s resamp_float_fistl - 0m2.960s resamp_fixp - 0m1.450s ************************ 1.25GHz G4 (laptop) resamp_float - 0m11.170s resamp_fixp - 0m2.130s ----------------------------------- Conclusions SO FAR. My own conclusions about the subject is that the float -> int conversion is STILL the biggest bottleneck on most common architectures. And until this is sorted out, fixed point is still the best solution for some specific cases, and I dont see any problem mixing it with floating point code. If you look at that algorithm closely in the source, you could replace "counter/ increment" for purely fixed point values, and do the rest (managing the samples) in float. This will undoubly speed it up.. Cheers! Juan Linietsky

20 years, 5 months

[linux-audio-dev] JACK tutorial

by james＠dis-dot-dat.net

Hi, I've recently been learning to use JACK, and I had a look around for some kind of introductory article. I couldn't find one, so I wrote the tutorial I would have wanted, as I learnt. The tutorial is here: http://dis-dot-dat.net/jacktuts/starting/index.html Please have a look, make suggestions. Flames are fine, too. Let me know if I've made some huge error. Or maybe I'm not doing things the best way? Whatever, let me know. If this is well received, I'll write about more advanced topics as I get to them - mixing, the transport, threading without blocking the process callback, etc. I'll probably be asking a lot of questions later on, especially about best practise. Thanks peeps, James

20 years, 5 months

[linux-audio-dev] Oss & threading -- is it possible

by hexe_2003＠directbox.com

Hello ! I've read somewhere that it is not possible that threads write different data to /dev/dsp. But how does Cheesetracker, for example, make it then? Sascha Retzki __________________________________________________ Verpassen Sie keine eBay-Auktion und bieten Sie bequem und schnell über das Telefon mit http://www.telefonbieten.de Ihre eMails auf dem Handy lesen - ohne Zeitverlust - 24h/Tag eMail, FAX, SMS, VoiceMail mit http://www.directbox.com

20 years, 5 months

[linux-audio-dev] last CFP: LCA2004 audio miniconf

by Conrad Parker

Hi, This is the last reminder for anyone interested in presenting at the audio mini-conference at Linux.Conf.Au, Monday January 12 2004. Submissions are due at the end of this week (Fri 31 Oct), details at: http://www.metadecks.org/events/lca2004/ It's turning out to be an excellent day of LAD hacking and jamming, with a full day of presentations lined up and an evening's playing at a local bar in the planning. The full schedule will be released next week. A mailing list has been set up for general discussions and info about the mini-conference, info for subscribing is at: http://lists.linux.org.au/listinfo/lca-audioconf To register for this mini-conference you MUST register for the main Linux.Conf.Au conference. Unfortunately audio-miniconf-only registrations are not available, sorry. Nevertheless you will not be disappointed: the main conference has four parallel tracks of tutorials and paper presentations running from Wednesday to Saturday, covering everything from programming VR applications to writing user-level device drivers; check it out and drool: http://lca2004.linux.org.au/programme.cgi Lastly ... if Tim Mayberry or Nick Mainsbridge could please drop me an email, or if anyone could give me current contact details for either of them, that would be much appreciated :) cheers, Conrad.

20 years, 5 months

[linux-audio-dev] [ann] guitar preamp plugin, v2

by Tim Goetze

hi all, a new, completely redone version of my valve-inspired preamp plugin is up. improvements over the predecessor include: * smoother distortion; clipping kicks in veeery gently * a better 'gain' knob model and balance between input and output loudness * initial hi-pass filtering now part of the unit * gentle mid-range frequency boost * closer matching of the original circuit's harmonic distortion characteristics * 4x oversampling at 44-48 kHz sample rates more info and the plugin sources at http://quitte.de/dsp/preamp.html enjoy, tim

20 years, 5 months

[linux-audio-dev] Softsynth latency control [Jack?]

by Florian Schmidt

Hi, i was thinking about latencies of softsynths and how cubase vst handles this when playing back recorded midi tracks as opposed to playing the softsynth directly. I'm talking about the vst-softsynths here. not the ones that get controlled via usual midi ports. It seems that during playback of a prerecorded midi track cubase vst knows the latency of the softsynth and and arranges for that so that the softsynth is really tight [no offset to other recorded audio material]. This is, of course, only possible because the midi signal if routed vst-internally is handled different than midi that gets sent out. I suppose that todays available audio/midi-sequencer support midi in a form that it is synchronized to the latency of the audio signal that is sent out. So, if the audio signal takes 2*128/44100s [2 buffers a 128 frames] to reach the ear of the user, then the prerecorded midi signal is delayed this exact amount of time, because midi signals is supposed to be of zero latency. And for external hardware synthesizers this is pretty much true. This way prerecorded audio and midi tracks are nicly synchronised during playback. Now, for the case of a sofstsynth this scenario doesn't fit, because the synth is not zero latency. This means that a "tight" midi track is audible with the softsynths output latency. Now, Jack is really an audio server but it could also be used to communicate the latency of the softsynth to the Audio/Midi Sequencer. I know there's calls to get the output latency of physical output ports. But it would be nice to have calls to ask for the physical output latency of clients [this value can be different for different clients - maybe with different output devices]. This way, the audio/midi-sequencer could ask jack for a list of clients and offer this choice to the user who then can select which midi tracks correspond to which softsynth. The Sequencer could then send the midi events a tad bit earlier [the output latency of the softsynth]. What do you think? I have no insight into the implementation details of jack, so i don't know how possible this scenario is and if it fits with design decisions. Regards Florian Schmidt

20 years, 5 months

[linux-audio-dev] CheeseTracker 0.9.0 released!

by Juan Linietsky

Cheesetracker is a portable Impulse Tracker clone. It supports all the main Impulse Tracker and FastTracker/SoundTracker features, plus many more. It is licensed under the GNU Public License. It runs under Linux/BSDs, MacOSX(QT/Mac or Qt/X11), or Win32(Cygwin). It can be obtained at http://cheesetronic.sf.net For those unfamiliar with trackers, this is basically an all-in-one sequencer/sampler/sample editor/mixer/fx processor bundle, wich provides fast and flexible means for professional grade music composing. Including in the release are tutorials and documentation. Volunteers to help to better documment it would be highly appreciated. The ChangeLog for this release follows: v0.9.0 ------ -Removed sample mode (Scream Tracker 3 mode) as It's obsolete and not needed for backwards compatibility. -Instruments are now layered and can perform up to 4 simultaneous voices with individual parameters each. -Added an effect buffers system. Instruments are now routed to custom buffers (each with individual effect chains), which can also re-route to other buffers. This allows to create very complex effect routes for realtime processing. -Effect buffers are "process on demand", which means they are smart enough to notice when they are doing nothing, thus disabling themselves. -Added a few internal effects: Amplifier, Clipping/Distortion, Recursive Delay Line, Stereo Enhancer, Chorus and Reverb. -Added a LADSPA effect source plugin. LADSPA plugins can be added to the chains. -Created new file formats that save all the new features: .CT (CheeseTracker Module) .CI (Cheesetracker Instrument) and .CS (CheeseTracker Sample) -Added preview to the sample file selection box, just hilite a file and use your keyboard to play notes (/ and * work in there too). -Readded JACK Driver (Kasper Souren) -Added RTAUDIO driver, allows for porting to Win32/ASIO and OSX/CoreAudio -Fixed some big endian compatibility issues. CheeseTracker should work fine again on big endian machines. -MacOSX port and build system/build fixes courtesy of Benjamin Reed -Fixed tons and tons of bugs. AND NOW PLEASE READ: How fast CheeseTracker reaches version 1.0 depends on YOU. The focus of this version is STABILITY. Because of this, I need to receive as many bug reports as I can, both program and build system. If you find a bug, I'd be enormously grateful if you submit it. Even if it is an obvious bug to you, chances that other people will find and report the same bug are much smaller than what you may think. If you dont report a bug that annoys you, the chances of it reappearing in the next version will allways be higher. Planned for 1.0.0: -=-=-=-=-=-=-=-=- -Rock Solid stability -WAV exporting -A hopefully working Windows port. This depends mainly on the Qt-Win32 project. If you are a good Windows programmer and would like to see CheeseTracker working in there sooner, please give those guys a hand! Enjoy!! Juan Linietsky

20 years, 6 months

[linux-audio-dev] Re: Correct sustain pedal emulation (midi hold controller) ?

by vanDongen/Gilcher

On Monday 27 October 2003 15:08, Benno Senoner wrote: >Assume I press C2 with velocity 50 pedal up, the C2-pedalup (associated >to velocity 50) sample sounds. >Now I press the sustain pedal and press C2 with velocity 100. >What should the sampler do ? Quickly fade out the C2-pedalup >(velocity-50) note and trigger the >C2-pedaldown (velocity 100) note ? >And of course when you release the pedal all sustained notes will get a >note-off. I am a piano player. I would expect the C2 note to be replaced with the next attack on that note. If the pedal is pressed, the damper stays up, so the next attack will be the hammer hitting the strings, and the sounding note is not faded out before. So I think you should fade out the previous note after or on the attack of the next one. Maybe even a lot later. There is an adding resonance effect if you keep hitting the same note with the pedal down, but that is probably the sympathetic vibrations from the other strings. If the pedal is not pressed the damper returns to the string as the key is released to play the next attack. You will get a note off message then anyway. Note that releasing the pedal should only send note-off to those notes that are sounding, but whose key is not pressed. If you have keys pressed down, the dampers will be up, regardless of the position of the sustain-pedal. So these should continue. I have no idea if digital piano's actually do this correctly, but it is the way a accoustic piano works. Gerard electronic & acoustic musics-- http://www.xs4all.nl/~gml

20 years, 6 months

[linux-audio-dev] LADCCA mailing list

by Bob Ham

Hi all, There's now a mailing list specifically for LADCCA discussion, ladcca-devel. You can subscribe from here: http://mail.nongnu.org/mailman/listinfo/ladcca-devel Bob -- Bob Ham <rah(a)bash.sh> "At some point, keystroke recorders got installed on several machines at Valve. Our speculation is that these were done via a buffer overflow in Outlook's preview pane." -- Gabe Newell on the Half-Life 2 source leak

20 years, 6 months

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Linux-audio-dev October 2003