Linux-audio-dev

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7520 discussions

[linux-audio-dev] a long winded way to say "model/view/controller"

by Paul Davis

http://industryclick.com/magazinearticle.asp?magazineid=141&releaseid=11131… well, OK, its a little bit more than that, but not a whole lot. nice idea, anyway.

22 years, 8 months

[linux-audio-dev] [OT] Linux Music & Sound Software site updated

by Dave Phillips

Greetings: Once again I've updated the Linux soundapps sites. All sites are current and can be accessed via these URLs : http://linux-sound.org (USA) http://www.linuxsound.at (Europe) http://linuxsound.jp/ (Japan) Many thanks to Frank Barknecht for his assistance with linuxsound.at. Many thanks also to all my site providers: the mirrors have been donated by their respective owners as a service to the community, for which I am most grateful. Enjoy ! Best regards, == Dave Phillips The Book Of Linux Music & Sound at http://www.nostarch.com/lms.htm The Linux Soundapps Site at http://linux-sound.org

22 years, 8 months

Re: [linux-audio-dev] Synchronizing two Delta 1010 with ALSA ?

by Benno Senoner

Thanks alot Martin and Paul for the useful hints, My questions: Basically the impossibility to do sync-start on multiple delta 1010 cards is because the hardware lacks the ability to do so ? de m-audio site/manual talks about "sample accurate sync between multiple cards" but I guess this means only that once started the frequencies do not drift but there could be a small offset between the single channels. Assuming there is a small offset when starting the cards, one could assume too that (if you run the audio init code SCHED_FIFO) it would be quite small (in order of a few samples). This means that if you use this approach: while(1) { snd_pcm_write_to_card_1(); snd_pcm_write_to_card_2(); } the small start offset would mean that one of the two card audio buffers are a bit less filled than the other one and like Martin has said, as long as you do not need sample accurate sync start it would work ok. Paul: sorry I did not know that ALSA allows you to treat two cards as a single logical card. As said it has been about two years I last used the ALSA API and many things have changed since then. Are there some online resources available that describe how to do this card linking ? Are the cards then started in sync (or near in sync for cards that do not support sync start) ? BTW: what kind of cards support multicard sync start ? RME ? Regarding the SPDIF cable between the two cards can I use a common cinch mono cable or must a SPDIF cable be shielded and/or have a precise resistance ? Note: In my case I do not need sample accurate sync but I was just curious if it would be possible to do that and/or if a single threaded audio app could experience problems in presence of playback start offsets. I searched the net for postings or notes about sync start with the delta 1010 but I haven't found any except this (article from year 2000) ".... Midiman UK told me that the current drivers can already keep four cards in perfect sync, but there are some fixed offsets between them; this will be overcome in a future driver release. ...." see here: http://www.sospubs.co.uk/sos/jan00/articles/midiman1010.htm So I was wondering what "overcome" would mean in this case: starting the two cards as close as possible in order to minimize offsets or using some adaptive algorithm that (assuming that the hw allows it) measure the dma ptr offsets and adjust them accordingly. again thanks for your useful infos. cheers, Benno ------------------------------------------------- This mail sent through http://www.gardena.net

22 years, 8 months

[linux-audio-dev] Quad Chorus for LADSPA plugin

by Nick

Hi, If someone would like to take a quad chorus I made and make it a LADSPA plugin feel free. The source code is @ www.sourceforge.net/projects/audiostar I had an implementaion of StereoPhonic-Quadraphonic Matrix processor, but I don't think the phase is correct, and nor is is 4-channel either. If you had four channels you would calculate the left rear channel as 90 degrees phase shifted, and then sum it into the formula. The quadraphonic processor still does make an interesting surround sound effect even though it is not technically correct. The algorithm is from an electronics book, the formulas were given as part of the circuit design. If someone is interested I can post those formulas to you for implementation, but really the quadraphonicprocessor.cpp already does it, just the left rear channel is phase shifted 90 degrees and mixed into the stereophonic stream. Since it is operating on only stereo channels, I keep a copy of the previous left channel about n samples behind, and slightly mix it to achieve something like a phase effect. It does actually sound like surround sound still. The Quad Chorus is just a stereo chorus but it has two additional taps. This is the ultimate in fatting up analog waveforms, but the implementation suffers from swishing if you set the LFO rate too high. It also does a deep flange if you set rate all the way down and put feedback up. I had it go through the quadraphonic, but the way I do stereo on my new synth, it doesn't sound proper, so I removed it. I'd make it a LADSPA plugin but I don't know yet how to make them. I have another chorus design but it is not real time. It is a six tap chorus, passed through a phaser, and then put through an all pass with a variable delay length setting. This can make echo effects. It was actually really cool sounding. I also had an FFT chorus that mixed pitch shifted copies of the signal onto the stream like suggest in The Computer Music Tutorial. I am working on getting it working with FFTW. Hrmm, oh, I also had one funny chorus, that was also not real well with real-time.I took the idea of the QuadraFuzz pedal, split the stream into four (or more) bandpass (with resonance) and then put them through a chorus effect. Then I remixed them through comb filters to get a reverb feel to it. That also sounded pretty cool, but I had too of a hard time getting it to work in real-time. IF someone could take those designs and make them real-time they were REALLY funky and people would like them. Just some ideas you can experiment with. -- Nick <nicktsocanos(a)charter.net>

22 years, 8 months

Re: [linux-audio-dev] [OT] linuxsound.at site problem

by CK

I read: > Any assistance will be vastly appreciated. I'm just trying to contact the people I know of that are still involved in vbs/atnet I hope someone will soonish contact you. regards, x -- chris(a)lo-res.org Postmodernism is german romanticism with better http://pilot.fm/ special effects. (Jeff Keuss / via ctheory.com)

22 years, 8 months

[linux-audio-dev] [OT] linuxsound.at site problem

by Dave Phillips

Greetings: I'm writing to the lists in the hope of finding someone who can advise me on contacting someone in charge of www.linuxsound.at in Austria. The former contact was Georg Hitsch, I've written to him but as yet have received no reply. The site is still on-line, but for some reason I can longer log in to update it. Thus only the US and Japanese sites have been updated recently. Any assistance will be vastly appreciated. Best regards, == Dave Phillips The Book Of Linux Music & Sound at http://www.nostarch.com/lms.htm The Linux Soundapps Site at http://linux-sound.org Currently listening to: King Sunny Ade, "Ja Funmi"

22 years, 8 months

[linux-audio-dev] LADSPA Instruments

by robbins jacob

While reading over the ladspa v1.1 header, i noticed that hint number 0x2C0 is defined as follows: /* This default hint indicates that the Hz frequency of `concert A' should be used. This will be 440 unless the host uses an unusual tuning convention, in which case it may be within a few Hz. */ #define LADSPA_HINT_DEFAULT_440 0x2C0 These "LADSPA_HINT_DEFAULT_..." hints tell a host how to set the values for a port when the user doesn't supply any values. But then it hit me, "what kind of port could have a default defined as 440 HZ, concert A?" Clearly, the only reasonable use for a port of this type is to control the pitch of an instrument-type object. Any ladpsa plugin that has a port with hint DEFAULT_440 set, be it a true synthesis-engine with no audio input ports, or some sort of tonal filter, can be used to generate coherent melodic output (ie tunes,songs,melodies,pitched-vocalizations...). This is important because it means that we can implement _right_now_ ladspa instruments that can be used without the user having to hook up control A to port B or generally know what the hell they're doing. In fact, the sine oscillator plugin, provided by Richard Furse as an example plugin for the ladspa sdk, is a reference example that host programmers can code off of _right_now_. Just remember that each ladspa plugin instance is a single voice, so polyphony must be implemented in the host. Fortunately, this is not that hard(*). The DEFAULT_440 hint took me by surprise because there aren't many ladspa virtual synths out there. With the use of this hint, coding a synth is about as simple as we would expect from ladspa: very. Many projects with integrated softsynths could put some of their core dsp routines into ladspa plugins and start seeing immediate benefits because the widespread use of the ladspa format facilitates sharing plugins between apps. Once again, this is all completely possible _right_now_ with the current version of ladspa. What's even better is that ladspa instruments will be automatically ported to the new XAP plugin format when it becomes available. This is because one of the first XAP plugins will be a plugin to "wrap" LADSPA plugins inside an XAP plugin. Plugin developers will then be able to choose whether to fully move to the new plugin architecture, continue using ladspa, or, (isn't linux great:) both. For the record, the XAP standard has settled on using linear pitch instead of pitch in HZ. linear pitch is defined as a floating point value, with 0 = 440 hz, and scale 1.0/octave. (example 440 hz = 0 lin pitch, 220 hz = -1 lin pitch, 110 hz = -2 lin pitch, 55 hz = -3 lin pitch). As you can see, the mapping between linear pitch and HZ is an isomorphism, so translation between the two is not much of an issue. Remember that each ladspa plugin instance is only ONE voice in a possibly polyphonous instrument. This means that plugin authors do not need to worry about the MIDI standard (voice_on, voice_off) at all if they don't want to. Translating from midi to ladspa is one of the host's burdens. I have placed a simple scheme for doing this at http://soundtank.sourceforge.net/input_maps ---jacob.................. (*) to make a host use multiple ladspa plugin instances as a single polyphonous instrument do the following: -make a linked list of all the plugin instances as you activate them -make a linked list called active_instances and one called inactive_instances -add all plugin instances to the inactive_instances list -have your realtime thread move instances from inactive_instances to active_instances when it activates them to handle a MIDI voice_on event, and back when a MIDI voice_off arrives saying that the note is finished. I have placed a simple scheme for handling input events at http://soundtank.sourceforge.net/input_maps _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963

22 years, 8 months

[linux-audio-dev] gwc-lib released on sourceforge

by Jeff Welty

Hi all, I've posted gwc-lib on: http://gwc.sourceforge.net/ This is an EXTREMELY crude release from the standpoint of documentation at least, (and it took *forever* to get this far), and ..., but I want to allow for input into the API. It so far only includes the denoising algorithm. I do plan to include the declicking and decrackling algorithms. What got me motivated was 3 things: 1) Conrad Parker's intitial query into getting the algorithms into sweep, and subsequent private emails helping me understand many issues. 2) Erik de Castro Lopo's release of Libsamplerate 3) Paul Davis' great post (spurred by libsamplerate) on linux-audio-dev about how sharing these basic algorithms in a library format has really made it a better world for all of us: http://eca.cx/lad/2003/01/0006.html In theory, there are only 2 files you need to look at to start playing around - gwc_lib.h which describes the API (sorta), and denoise.c, which is an example of how to use the algorithm. The basic idea is you initialize the algorithm with parameters, feed it noise samples via buffers, then start using it by writing and reading buffers. There appears to be a small performance hit by using the buffers on the order of 10%. If you've a mind, please grab it and let me know how it goes. Fire back ideas about the API. And, no, I'm not gonna LADSPA-ize it. Cheers, Jeff Welty

22 years, 8 months

[linux-audio-dev] Compiling with libsndfile

by Rohan Parkes

I'm trying John Hall's book Programming Linux Games, which recommends using libsndfile to load sound files. Unfortunately, the API has changed since the book as written, and I can't seem to get projects to compile with it now. There don't appear to be any instructions on the library's homepage about linking. Hall just says to compile the project with the -lsndfile flag. This doesn't seem to work now - the resulting project doesn't link the object in. The library does seem to be correctly installed I found a message from the library's author http://www.eca.cx/lad/2002/Mar/0311.html which says to link the program with gcc `sndfile-config --libs` file1.o file2.o -o program. I don't seem to have the sndfile-config utility, which was suppoed to get installed along with the library. Running ./configure created makefiles in some of the subfolders in the installation folder. I'm mainly interested in the examples folder, but I get errors when I try to compile them with the makefile, assuming this is what I am supposed to do. I'd appreciate any advice. Rohan Parkes Melbourne Australia

22 years, 8 months

[linux-audio-dev] lock-free ring buffer code

by Tim Hockin

> > i remember i have read a statement about a lock free ringbuffer > > implemented in C somewhere. > > courtesy of paul davis: > > you should use a lock free ringbuffer. we will be adding example code > to the example-clients directory soon. existing code is in ardour's > source base (for C++). the example code will be in > example-clients/capture_client.c. > > where ardour is ardour.sf.net. (i doubt there is anything hugely non-C in > the ringbuffer code proper). maybe it's just me, but I can't find said file...?

22 years, 8 months

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Linux-audio-dev