Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

4 participants
7538 discussions

by Ben Loftis

On Monday 21 March 2005 12:06 pm, linux-audio-dev-request(a)music.columbia.edu wrote: > jack.udp for me means many underrun in this configuration: > > x86 pc > jackd -d alsa -d hw:0 > jack.udp recv > ^ > | (FastEthernet network (100mbps)) > | > | > powerpc pc > jackd -d dummy > jack.udp -r IP send In order for this to work, the sending side needs to know "how many" samples it needs to send. In other words, if the x86 side needs exactly 44,100 samples per second, there is no way in your existing setup to make sure that the powerpc doesn't send 44,101 or 44,099 samples per second. You'll need to run an actual alsa client on the sending side, and use wordclock or some other mechanism to keep the two soundcards in sync. You might be able to hack something into the receiving side which tells the sender each time it consumes a buffer, but that's probably not going to get you where you want to be. -Ben Loftis

20 years, 11 months

[linux-audio-dev] OSC-Question

by Arnold Krille

Hi all, after some of the latest discussions about audio-apps without gui, my head is filled with giving JackMix[1] OSC-Support and perhaps splitting it into a text-based / osc-based server doing the mixing and a gui... So my question arises: Which OSC-implementation to use? I had a look into Steve Harris' liblo and libOSC++. The later seems more appealing to me since I am a C++-Guy. What do you folks think? What do you propose? What are you using? Arnold [1] http://roederberg.dyndns.org/~arnold/jackmix/ -- There is a theory which states that if ever anyone discovers exactly what the Universe is for and why it is here, it will instantly disappear and be replaced by something even more bizarre and inexplicable. There is another theory which states that this has already happened. -- Douglas Adams, The Restaurant at the End of the Universe

20 years, 11 months

Re: [linux-audio-dev] jack transport

by Luis Garrido

> Syncing video playback to Ardour would be a great example of the > usefullness of jack-transport. Unfortunately, i guess the fact > is that the design of mplayer means it can never be a real jack > client. Perhaps a jack client could control mplayer via its slave > mode, for use with non-keyframe based video formats, but thats a > bit of a hack. I needed a jack driven video player to allow me to compose the soundtrack for a videoclip in MusE. Since there was none available, I made my own one. I think it is a not so great but adequate example of the usefullness of jack-transport ;-D One of the solutions I thought of was that suggested by you: mplayer slave mode. I don't remember well why I dropped it. Finally I ended up glueing some libraries to make a very quick hack that did the job. Don't expect good video performance, nor support of keyframed video formats, I only spent a couple of days and got what I needed. I use it successfully with MusE and ardour. You can check the project page and a screenshot: http://sourceforge.net/projects/xjadeo/ http://sourceforge.net/project/screenshots.php?group_id=131926 Regards, Luis

20 years, 11 months

[linux-audio-dev] appropriate LADSPA plugin for rejection?

by Andrew Gaydenko

Hi! The aim is to investigate some signal which consists of main harmonic and others with rather low level. I'd like to reject main harmonic _and_do_not_affect_ to other harmonics. What LADSPA plugin is the most sutable for such work? Thanks! Andrew

20 years, 11 months

[linux-audio-dev] Quick question regarding raw midi access via alsa sequencer

by Ivica Ico Bukvic

Hi all, I did a bit of hacking on my app trying to make it alsa sequencer compatible but did not want to do too much changing in terms of how it deals with raw MIDI data. From looking at the API reference it seems that I have 2 choices: 1) Use raw midi option and specify "virtual" name which makes my app's MIDI I/O appear in the alsa sequencer, *but* does not give me an option of changing the node names which obviously is very important when it comes to working with a lot of apps concurrently. So therefore here's my first question: Is there a way to specify a "virtual" port so that I can receive raw MIDI data, have ports show in the alsa sequencer and on top of that be able to *rename* the port as necessary? 2) The other option is obviously to use alsa-sequencer API but in that case is there a way to simply convert the stream of received MIDI data into raw midi format so that I can use my built-in raw MIDI parsing engine for parsing the messages? Any help is greatly appreciated! Best wishes, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/ -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.3 - Release Date: 3/15/2005

20 years, 11 months

[linux-audio-dev] Re: Jack compatible LAME streamers (oddcast, ices2, darkice?)

by ben racher

So I messed around for a little bit and found something that kinda works. I was routing alsa-player in ardour (all this with jack of course), my sound card input also into ardour, mixing the two channels in ardour, and outputting the master out of ardour into oddcast, which then sent to icecast server, which I connected to with xmms, which sent the audio to soundcard output. And here was the trick. I increased the buffer size in Jack from 1024 to 4096 which consumed far less cpu resources therefore decreasing ringbufferfull errors from oddcast, and I also used jack.plumbing to do whatever it does between oddcast and ardour, which kept the stream from crashing. So now whenever I get the ringbuffer full error, the stream just jitters as long as the error persists, and seemed to return back to normal as soon as cpu resources became available again. So... sort of a solution. I'm hoping that in actual practice, I'll be able to divide some of these tasks between multiple machines, probably have oddcast running on its own machine, and pumping in an audio stream with jack.udp I'll try to go check on this later today, and will report back if its stable at all. Pretty impressive though that my g4 1.5ghz could handle all that. All this would probably work better on an intel system though? - Ben > I've been trying to get oddcast to work the past couple days, but keep > running into this problem > > ringbuffer full, tried to write 4096, but wrote 0 > > its usually in response to something distracting the processor for a > second, like dragging a window across the screen. Have tried on a 1.5ghz > ppc processor and 700mhz pIII and problem persists. Have searched for > other users reporting similar problems, and found one hint on the > oddsock site > > .... but unfortunately their site is in a period of transition, plug > above line into google and you'll find a couple similar complaints. > Apparently, the code for oddsock was scrapped from ices2-jack, which > would explain why I have no problems when trying to do this with Ogg, > but maybe Ogg is less processor intensive? I'm only having problems when > trying to encode with LAME. > > If anybody has any ideas, I would love to get this working for our radio > station. But if not oddcast, does anybody know if there is any other gnu > mp3 encoders out there (darkice?) that are jack compatible. Seems like > the darkice guy has been working on livesupport, which I can't wait to > see in action. > > - Ben Racher >

20 years, 11 months

[linux-audio-dev] Jack compatible LAME streamers (oddcast, ices2, darkice?)

by ben racher

I've been trying to get oddcast to work the past couple days, but keep running into this problem ringbuffer full, tried to write 4096, but wrote 0 its usually in response to something distracting the processor for a second, like dragging a window across the screen. Have tried on a 1.5ghz ppc processor and 700mhz pIII and problem persists. Have searched for other users reporting similar problems, and found one hint on the oddsock site .... but unfortunately their site is in a period of transition, plug above line into google and you'll find a couple similar complaints. Apparently, the code for oddsock was scrapped from ices2-jack, which would explain why I have no problems when trying to do this with Ogg, but maybe Ogg is less processor intensive? I'm only having problems when trying to encode with LAME. If anybody has any ideas, I would love to get this working for our radio station. But if not oddcast, does anybody know if there is any other gnu mp3 encoders out there (darkice?) that are jack compatible. Seems like the darkice guy has been working on livesupport, which I can't wait to see in action. - Ben Racher

20 years, 11 months

[linux-audio-dev] Fwd: [ardour-users] no sound when SC1 is activated

by Wolfgang Woehl

Steve, this is from ardour-users. I know the problem too: SC1 to SC4 more or less always stop working after some twiddling. Meaning the audio gets through only when SC* are bypassed or removed. I suppose you don't know the problem but maybe you can tell me what exactly you need to know to debug this. Wolfgang

20 years, 11 months

[linux-audio-dev] Amount of analog channels with nonuncompressing demux on ENVY24HT-S based cards

by Hans Davidson

Hello ! I want to setup a multichannel audio environment for use together with software that synthesize the waveforms for each channel. My newest audio card is a "Hollywood@home 7.1" card based on the ENVY24HT-S chip. I have managed to use the card to output mono and/or stereo sound under Slackware 10.1, but still want to find out if the card supports the conversion of uncompressed multichannel audio (for example 6 or 8 16-bit (or 24-bit) samples per frame from a FIFO through operating system and hardware) into analog signals ? If this is not possible with the internal DACs of the card, then I hope that it at least will be possible through ALSA with external DACs connected to the card's optical output, that seems to be of a high speed kind and expected to support ADAT (a standard established by Alesis, I think). I am not so interested in encoding the audio in DTS or some Dolby Multichannel format or simmilar, since i want the channels to be handled independently. For example I do not want to label a speaker as "left" or "right" and then select the speaker by assosiating a direction to the sound. Instead I want to be able to assign D/A channels to MIDI file tracks or MIDI channels and then placing the speakers there I want them (not there any multichannel decoder expect them to be). If I try to use several cards, then a lot of synchronization problems need to be solved, to avoid divergation in FIFO usage between cards and to maintain phase coherency. In addition to arrangements, like pathing cards to use a common chrystal oscilator and/or reserving some DAC-channels for synchronizaion purposes, I also need to convince the motherboard to accept several sound cards without causing problems during resource allocation (for example around IRQ mapping) or distributing the cards between several PCs connected together with maybe 100Mbps network cards. When I have solved these questions, then I may have to think more about the best algorithms to synthesize sound, by looking at existing open source programs and/or implementing additive synthesis with or without the use of FFT or some other optimized code. I am new to this list, so please excuse me if the answers to my questions are hidden in any recent messages in the archive. Thanks in advance for feedback/answers ! Hans Davidson _______________________________________________________ Skicka gratis SMS! http://www.passagen.se

20 years, 11 months

[linux-audio-dev] Linuxtag 2005 audio booth participation

by Christoph Eckert

Hi, first of all, sorry for the cross posting, but I thought the topic is worth it. Again, this year the Linuxtag will happen in Karlsruhe, Germany. It's Europe's most important event concerning free, especially Linux based software. During the last years, Frank Neumann has politely organized an audio booth there. Unfortunately, it can happen that he'll not be able to do all management stuff this year again. So, I'll try my very best to help out this year, otherwise we wouldn't have a booth. We're still looking for people who are interested in helping, mainly being at the booth, answering questions and demoing software. If you like to contribute, please let me know. For detailed information about the Linuxtag see http://www.linuxtag.org/2005/en/home.html or contact me via personal mail. Best regards ce

20 years, 11 months

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Linux-audio-dev