Linux-audio-dev

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7513 discussions

by Richard Caldwell

Im trying to get this working between two LAN connected machines running Suse8.1. It appears to compile and run OK, both Client and Server applications on the two machines but Im not getting any audio across the network. My speakers and mic are working OK on both machines as Ive tested them with with alsamixer. Im not getting any errors so I dont know where the problem is. I run the server by running TC__talk_srvr and I run the client by running TCP_talk_clnt xxx.xxx.xxx.xxx (where xxx.xxx.xxx.xxx is the IP address of the server machine. Id greatly appreciate any help. Alternatively Id be glad to try any other application that will give me 2 way audio/voice across a LAN. R.C.

22 years, 8 months

[linux-audio-dev] [ANN] JACK Rack 1.1.1

by Bob Ham

Be gone foul bugs! * fixed control output ports segfault * fixed desktop installation prefix stuff * fixed bug dealing with duplicate plugin ids * now quits when you close the window * added a "New" option to clear the rack * rack is now automatically cleared when you load a file http://pkl.net/~node/jack-rack.html Bob

22 years, 8 months

[linux-audio-dev] more on XAP Virtual Voice ID system

by robbins jacob

The main difficulty I see with the VVID system is how to initialize parameters on a new voice before the voice is activated with a VoiceOn event. The MIDI standard deals with this by allowing 2 particular parameters, pitch and velocity, to be privileged as voice-on initializers. These parameters are included inside the MIDI Voice On message so that the instrument always receives them when they are needed: right when a voice is being activated. It has been the general feeling in LAD discussions that XAP should not put 2 particular parameters into the Voice On event but, instead, should allow any parameter-set events to be intializers. How to achieve this? In XAP, it is not sufficient to consider all parameter setting events received before a Voice On message to be initializers. Consider that, before an actual voice is attached to a VVID, all events sent to that VVID are discarded. For instance, if a voice is activated and then reappropriated halfway through a note, before the sequencer has stopped using the VVID, the remainder of the events which the sequencer sends on the voiceless VVID are simply discarded. Furthermore, any system which requires initializers to be timestamped before the voice-on event forces at least a 1 timestamp delay into voice activation which is undesireable. We could instead put the voice-on event at the same timestamp as the initializers and require that the plugin read ALL events for a particular time position before discarding ANY events, but this makes it more complex for plugins to read their events. If the plugin found a voice-on event at the end of a queue section for a particular timestamp then it would need to loop back to the first event in the queue matching that timestamp and read again for any initializers. Alternately, we could require that event ordering has 2 criterion: -first- order on timestamps -second- put voice-on ahead of all other event types. A little ungainly, but effective as it frees the plugins to assume that they'll get voice-on first but must consider all other events on that timestamp as arguments to the initialization of the voice. Of course this only applies to events having a particular VVID on a particular channel of a particular plugin instance. I admit that introducing another criteria to event ordering is inelegant, but it is the best approach to voice activation that I can think of. I think that introducing any sort of mandatory 1 or 2 timestamp delay is far worse. -jacob robbins.... current projects: soundtank.......<BR>.......... _________________________________________________________________ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail

22 years, 8 months

[linux-audio-dev] Re: Sidechains, was Re: [linux-audio-announce] [ANN] swh-plugins 0.3.4

by Steve Harris

On Fri, Jan 10, 2003 at 11:12:21 +0100, Robert Jonsson wrote: > >Ahh, theres a misundersantding about what sidechains do here, they really > >are just an audio stream. > > Hmmm, how do you recon? A sidechain for a compressor may be an audio > stream, but it's the "control" behaviour of that stream that is of > interest, right? That should mean that it would be possible to extract > just the behaviour and only distribute "that". Not in practice, each compressor will want to extract different features from the audio stream. > >There is an auto-phaser (sweeping allpass filter), but no auto wah > >(sweeping bandpass), or auto panner, true. I tend to build things like > >that out of SSM or pd. But there should be complete plugins. > > Ya, working mostly in MusE I'm somewhat limited in the number of > solutions I can try. Currently MusE only supports basic features (pretty > much modeled after Cubase). OK, I'm not familar with MusE, I should try it again. - Steve

22 years, 8 months

[linux-audio-dev] // a 'vsound' for alsa ? //

by delire

is there any application at all like vsound for alsa? i notice that it can't write a buffer from the alsa driver.. i have a half-duplex card on a lappie so i cannot record and play at the same time, hence vsound is very useful to me. cheers, julian oliver melbourne, australia

22 years, 8 months

[linux-audio-dev] [ANN] swh-plugins 0.3.6 *cough*

by Steve Harris

* Really fixes the UID clashes. * Includes changes to make it build on ia64 and mips, thanks to Anand Kumria. I will now be walking around with a paper bag over my head for the next few days. - Steve

22 years, 8 months

[linux-audio-dev] Thanks

by Levi Burton

I've been using Linux for quite some time now and finally got around to using it for audio and MIDI. I must say i'm very impressed with all the applications i've been using. Thanks to all (developers and users) for making Linux audio/music development a reality. Levi

22 years, 8 months

[linux-audio-dev] *tap, tap*

by David Olofson

"Is this thing on...?" (Seems like at least one message was lost, and it's been *very* silent for a while...) //David Olofson - Programmer, Composer, Open Source Advocate .- The Return of Audiality! --------------------------------. | Free/Open Source Audio Engine for use in Games or Studio. | | RT and off-line synth. Scripting. Sample accurate timing. | `---------------------------> http://olofson.net/audiality -' --- http://olofson.net --- http://www.reologica.se ---

22 years, 8 months

[linux-audio-dev] new member on the list, UAP and Compression

by remy muller

hello to evey body, i'm new to this list. I'd like to know more about Unified Audio PLugin Architecture. I will begin a practical on this topic in mars. where can I register to a mailing-list or forum or... I' also looking for papers about audio dynamic compression/expansion/limiting. Does anybody have some good links on this topic? (I'm not looking for "compression for the newbies" but some more in-depth considerations. like analog compressor modeling.... aliasing due to non linear processing.....) thanx remy muller http://membres.lycos/mdsp/

22 years, 8 months

[linux-audio-dev] question re: optical spdif/in cards

by Thomas Vander Stichele

Hi, for a project at our radio station I need to simultaneously record two signals coming from an Amek DMS mixer. The signals are digital optical S/PDIF (which I assume is the same as TOSLINK ?). I am looking for a card or a combination of two cards that would allow me to record both these signals at the same time under linux. Of course I'd like to go for the cheapest stable solution possible :) I've been looking at specs of the Hammerfall Light, it seems to feature more than one ADAT input port, which from the docs looks to be the same type of connector as optical spdif. I'm just not very sure from the docs if this will really work; ie if one Hamerfall Light can record two spdif signals simultaneously. One Hammerfall is about in the right price range, two is probably too expensive :) So, basically my questions are: a) will one hammerfall record both optical spdif signals at the same time ? b) what other, cheaper, card is there that can record an optical spdif input ? (I looked at some obvious choices, the hercules cards aren't supported well enough and the sblive cards only take coaxial optical) Thanks in advance, Thomas The Dave/Dina Project : future TV today ! - http://davedina.apestaart.org/ <-*- thomas (dot) apestaart (dot) org -*-> I feel like I'm at something you always say you need more time Well I'll stay right here and I'll wait for good until I find a love worth mine <-*- thomas (at) apestaart (dot) org -*-> URGent, the best radio on the Internet - 24/7 ! - http://urgent.rug.ac.be/

22 years, 8 months

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Linux-audio-dev