Linux-audio-dev March 2005

linux-audio-dev@lists.linuxaudio.org

68 participants
63 discussions

Re: [linux-audio-dev] Looking for simple and good multichannel audio soluution.

by Hans Davidson

Thank you Shane and Wolfgang for your answers ! That was valuable input for me ! I visited "Jam", probably the best Music Equipment Shop in Stockholm one or two days ago know to check the price levels on the ADAT related hardware. ADA8000 from Behringer seems to be a nice pice of hardware. The price (about 279 EURO) is not very high, maybe too high this month, though. If I buy it, then I also need a PCI to ADAT adapter, and eventually even with a ADAT input port to be able to use the analog inputs on the ADA8000. I realize that experienced Linux developers and users often prefer to use cards of the RME (or RME Hammerfall) brand to not wasting time or money on cards that are not supported or do not have Linux drivers (or maybe drivers that only supports parts of the card's functionality) and/or by one or more other good reasons. The problem in my case is that the PCI adapters seem to be more expensive than the ADA8000, but may be a good investment anyway if you later want to add another ADAT based box into the system. What may be good to have (build, buy or borrow), is some kind of test equipment that could be used together with a ALSA base test application to verify that the kernel module, the configured sound system and the interface card that you (in my case: I) already have is capable of outputing ADAT signals on the optical connection, and then if that test is positive, it will be safer and cheaper to buy only the ADAT based D/A conversion equipment. I have not looked at the specifications and do not exactly know how much electronics, like shift registers, PIC micro- controllers and/or other things that is needed to build that kind of test equipment. I also find the "El Cheapo" article interresting. Of course the instructions, and problems observed, may be based on one type of card, but indicates an obviously working way, it seems, to use existing onboard crystal oscilators to feed the next bord with the clock signal and convert that boards oscilator cirquit into an amplifier for the clock signal and making all cards running at same speed (of course the cards must be of the same type or at least using the same chrystal frequency). My experience is that after inserting 3 soundcards in a single motherboard, then I may have serious troubles to even have a video signal during the power on self test of the computer and/or entering the BIOS setup, those cards was not modified, so they where probably not broken. To bring down the costs, I am still thinking of using a fast master computer to feed a couple of slave boxes containing a couple of synchronized cards each. It should be possible to synchronize all four cards, but you may need to use buffer amplifiers and maybe a differential (ballanced) transmittion line for the clock signal between the boxes. All three boxes should use 100 Mbps Ethernet through a switch using full duplex mode if possible. If I avoid to build a virtual bigger sound card of the both cards in a box, and instead use a couple of handles for the pair of cards in the application process in that box, then I expect to be able to balance the both queues for that slave box and on the other side of the Ethernet switch, all four flows at the master box and bring down the latancy to an acceptable level without observing underrun errors. To complete the intercard-synchronasation, I also want to reserve one output channel from each card, for example a pair of left channels and a pair of right channels, and feed them using four resistors back into a reserved stereo input on the first slave box to make it easy to compute the inter-card timing properties after starting each card as a four channel output device. I have one interesting question here. Maybe I have been too lacy to look at the soundcard drivers and/or ALSA source code ... : If I configure the .asoundrc file to make a device name available for the four possible independent output channels of the physical card, is then always (when using that device name) all four channels started simultaneously or is there any practical risc that the card may start (or be instructed to start) the two pairs of channels (we can call them "front" and "rear") with some random missallignment in time ? If I am lucky, assuming that the audio cards are supporting full duplex, then I will have 12 free output channels and 6 free input channels by using three computers. Hopefully I can find a couple of old 300MHz Pentium II boxes or something somewhere for use as the slave computers. If I for some reason want to play preproduced digital audio streams with all 12 channels alligned in into same phase (with much better resolution than 1s/44100), then I may have to resample/interpolate the digital audio streams. But I think that this is PROBABLY UNIMPORTANT except when feeding the channels into an analog mixer (bad idea I think) or when having a very well defined setup of the speakers in the room and at the same time collecting (analyzing or recording) sound with a microphone with a fixed position and expecting to have the identical interferense patterns through several starts of the sound system. Best regards ! /Hans Davidson _______________________________________________________ Skicka gratis SMS! http://www.passagen.se

19 years, 6 months

[linux-audio-dev] Re: OSC-Question

by Juhana Sadeharju

>From: Jan Depner <eviltwin69(a)cableone.net> > >> >No, imho one of the main advantages is Qt's Signal/Slot mechanism > >> sigc++ How to implement signal/slot mechanism in simplest terms with C? In my opinion, sometimes it is unnecessary to link to a massive code libraries if only one feature is needed. AlsaModularSynth uses Qt's signals and slots in audio processing, and thus requires the whole Qt and mixes GUI toolkit to audio side. It could be wise to use sigc++ or minimal S/S code in audio processing and Qt only in GUI. But because Qt comes with every Linux distribution, maybe bad dependencies can be allowed. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 6 months

Re: [linux-audio-dev] Re: OSC-Question

by Taybin Rutkin

Sorry, I meant GObject. http://developer.gnome.org/doc/API/2.0/gobject/gobject-Signals.html Taybin -----Original Message----- From: Juhana Sadeharju <kouhia(a)nic.funet.fi> Sent: Mar 29, 2005 11:38 AM To: linux-audio-dev(a)music.columbia.edu Subject: [linux-audio-dev] Re: OSC-Question >From: Jan Depner <eviltwin69(a)cableone.net> > >> >No, imho one of the main advantages is Qt's Signal/Slot mechanism > >> sigc++ How to implement signal/slot mechanism in simplest terms with C? In my opinion, sometimes it is unnecessary to link to a massive code libraries if only one feature is needed. AlsaModularSynth uses Qt's signals and slots in audio processing, and thus requires the whole Qt and mixes GUI toolkit to audio side. It could be wise to use sigc++ or minimal S/S code in audio processing and Qt only in GUI. But because Qt comes with every Linux distribution, maybe bad dependencies can be allowed. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 6 months

Re: [linux-audio-dev] Re: OSC-Question

by Taybin Rutkin

How to implement signal/slot in C? I'd look at glib-2. I think their event mechanism might come close. Taybin -----Original Message----- From: Juhana Sadeharju <kouhia(a)nic.funet.fi> Sent: Mar 29, 2005 11:38 AM To: linux-audio-dev(a)music.columbia.edu Subject: [linux-audio-dev] Re: OSC-Question >From: Jan Depner <eviltwin69(a)cableone.net> > >> >No, imho one of the main advantages is Qt's Signal/Slot mechanism > >> sigc++ How to implement signal/slot mechanism in simplest terms with C? In my opinion, sometimes it is unnecessary to link to a massive code libraries if only one feature is needed. AlsaModularSynth uses Qt's signals and slots in audio processing, and thus requires the whole Qt and mixes GUI toolkit to audio side. It could be wise to use sigc++ or minimal S/S code in audio processing and Qt only in GUI. But because Qt comes with every Linux distribution, maybe bad dependencies can be allowed. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 6 months

[linux-audio-dev] Re: [OSC_dev] [PATCH] liblo & pattern matching callback

by Steve Harris

On Sun, Mar 27, 2005 at 09:39:31AM -0800, Matt Wright wrote: > When we wrote that part of the OSC Spec, we were thinking > of the case in which an OSC Method doesn't need to know > the address through which it was invoked, i.e., "usual" > cases like setting a parameter. That's why the spec > doesn't mention sending either the expanded or unexpanded > OSC address to a handler --- sorry about that. > > Why not simply always send both? That seems more general > and easier to understand than a special case, at least for > me. Well, that would require changing the API, which is a Bad Thing, and there is a user_data parameter than can encode that kind of contextual information when its needed. Also, the method callback functions are too complicated allready :) - Steve

19 years, 6 months

[linux-audio-dev] Looking for simple and good multichannel audio soluution.

by Hans Davidson

_______________________________________________________ Skicka gratis SMS! http://www.passagen.se

19 years, 6 months

[linux-audio-dev] Fwd: Open instruments mailing list

by Christian Henz

(Forwarding to LAD) cheers, Christian -------------------------------------- I've created a mailing list for the discussion of defining an open instrument standard. So far the agreement seems to be to create an XML standard which references external audio files. The use of FLAC has also been mentioned. All who are interested may join at the following link: http://resonance.org/mailman/listinfo/open-instruments The address to post is: open-instruments at resonance.org Archives: http://resonance.org/pipermail/open-instruments/ If you feel another email list should be notified, please send this information on. Perhaps CC the new list so that others may check which lists have already been notified (check the archives). At this point I have sent this to: swami-devel linuxsampler-devel fluid-dev (FluidSynth devel list) Best regards, Josh Green

19 years, 6 months

[linux-audio-dev] Fakiir 0.4 - an RPC code generator

by jaromil

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 re all, i'm writing a software that can read a C++ header and generate code to expose the public functions in classes found: the generated code can be compiled together in the original app, which should also link to a library, then with 4 new lines OSC and XMLRPC servers will be active accepting remote calls to the functions. in fakiir, i make use of liblo-0.18 also published on this list. the software it's in a early phase of development, but since today is able to accept concurrent OSC and XMLRPC calls in the testclass.cpp application that is bundled with the source. http://fakiir.dyne.org i'm happy to hear comments, suggestions or any criticism that comes ciao - -- jaromil, dyne.org rasta coder, http://rastasoft.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Cryptographically signed mail, see http://gnupg.org iD8DBQFCQvU3WLCC1ltubZcRAsP3AJwKkEKeXc4GrWoyTXOqkPUuStKLHwCfReBE NN7TpeyfrNxTIC7dNNN4BPE= =uqyK -----END PGP SIGNATURE-----

19 years, 6 months

[linux-audio-dev] Jack and sample rate conversion : realtime or not ?

by Olivier Guilyardi

Hi, If playing a sound file that has a different framerate from Jack, using libsamplerate, should I : - convert in real time, in the process callback ? - convert the whole file into memory when loading it ? Actually, I've already coded the second option, but just discovered the jack_set_samplerate_callback() method, which seems to require the first option... Is it important to support framerate changes while rolling ? -- og

19 years, 6 months

[linux-audio-dev] Re: jack-udp

by Jordan

Thanks for the replies everyone! :) I was afraid that the computers would not be able to handle more than 8 channels of recording which is why I was limiting them that way. How many channels can one computer handle? What kind of specs (ie: processor/RAM) would that computer need? What interfaces can handle more than 8 analog mics? I would need to record about 32 mic inputs. Also, the OpenMosix would be disabled while recording, and would only be active when in a non-recording mode. Also, this network would be an island. The gigabit ethernet switch would be dedicated to just the audio stuff. -jordan

19 years, 6 months

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Linux-audio-dev March 2005