Linux-audio-dev March 2005

linux-audio-dev@lists.linuxaudio.org

68 participants
63 discussions

by Jordan

Hello all. I don't really have any business asking, but I am more and more interested in digital HD recording, and I have spent many hours recently studying hardware, software, techniques, et cetera. I don't really have the funds to create such a system, but it is fun to plan it out, in case my church or someone else would be interested in such a system in the future. The recent discussion of jack over networks has gotten me wondering a few things. Here is my current fantasy rack setup: 1U: UPS 1U: KVM 2U: RAID 2U: Master/DAW 1U: Slave Recorder/Node 1U: Slave Recorder/Node 1U: Slave Recorder/Node 1U: Slave Recorder/Node 1U: Gigabit ethernet switch (all computers connected through it) Basically, all slaves would boot off of the RAID server (for easier maintenance) into a cut-down Linux kernel, and will boot into text-only mode, starting only the programs absolutely required to run Ecasound. The slave node would begin recording when the master tells it to, and all audio data would be sent to the RAID server. I am thinking that each slave would be able to handle 8 channels of mono input. All nodes would be equipped with the OpenMosix software, so that they can assist the Master when they are not busy. The master would boot into GUI mode, so that the operator wouldn't be intimidated by learning Linux command line. The master would be able to see the status of all slaves, and what each channel is doing. The master would able to assign friendly names to each of the input channels (ie: vocal 1, vocal 2, drums 1, bass, etc.). When the operator checks a box at each channel, to indicate whether it should record or not (no point in recording if nothing is plugged into it). The operator can also enter a friendly name for the project. When the operator issues forth the "BEGIN RECORDING!!!!" command, a directory is created on the server, and all nodes begin recording. The all data is sent directly to the server, then opened on the DAW from there. The DAW operator would be able to record a live mix into a JACK-capable recorder, such a ReZound, then burn it to CD. The operator would also be able to back-up the project (ie: all of the raw audio) to a data DVD. So, I guess the question is, would this work? -jordan

19 years, 9 months

[linux-audio-dev] jack.udp issues

by Antonio "Willy" Malara

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 the first thing that came in my mind was some network troubles like mtu.. further investigation (like involving two x86 hosts, or working directly without the switch, working with 802.11b networks...) made me think that jack.udp uses the audiocard driver for some kind of timing. now, i've some troubles running an audiocard with jack in the powerpc (my main computer) there is a way to make this system usable with dummy driver? i've noticed that this dummy driver has a "delay in microsecs" parameter, now i think if i can calculate the right number the problem will fade away, but i dont know how to do the math :( some one has an idea on this? or has solved a similar trouble? another question out of this trouble: someone knows how to raise the IRQ priority of the USB controller to an usable value? possibly not involving the use of the realtime patch, inusable on ppc-powermac, thanks for attention and eventual replies, and keep the good work on this audio software :) willy

19 years, 9 months

[linux-audio-dev] jack.udp issues

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

19 years, 9 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

19 years, 9 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

19 years, 9 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

19 years, 9 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

19 years, 9 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 >

19 years, 9 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

19 years, 9 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

19 years, 9 months

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Linux-audio-dev March 2005