Linux-audio-user April 2013

linux-audio-user@lists.linuxaudio.org

111 participants
126 discussions

Re: [LAU] S/PDIF in with ESI Juli@ and JACK

by Ralf Mardorf

-------- Forwarded Message -------- From: Ralf Mardorf To: linux-audio-user Subject: Re: [LAU] S/PDIF in with ESI Juli@ and JACK Date: Wed, 03 Apr 2013 22:20:50 +0200 On Wed, 2013-04-03 at 13:44 -0600, Daniel Worth wrote: > > > My analogue ins and outs work fine, but only 2 ins show up in > JACK, even though I'd imagine I'd get one more for my S/PDIF > in. > > > > > S/PDIF on these cards is only two channels in/out. These are not ADAT > connections. Yes but the OP at least should get the 2 analog and the 2 SPDIF channels, assumed the card is capable of using analog and SPDIF at the same time ;). Is the driver for this card ok? For my ICE1712 ^^ stereo cards jack does show 12 capture and 10 playback channels.

11 years, 1 month

Claudia and Ardour3

by Yvonnick Noel

Hello all, I am giving Ardour 3 a try. This is a wonderful piece of software. I just note the following problem: I create a studio under Claudia, add Ardour as an app, and then just try to record midi data from my keyboard on a midi track named 'Synth'. The clip does appear and I can see the notes getting drawed on the track, but in the end I get the following message: [ERROR]: cannot open MIDI file /home/user/ladish-projects/Ardour3_7174/interchange/Ardour3_7174/midifiles/Synth-1.mid for write and the clip disappears. The midifile directory does exist but the correct path is: /home/user/ladish-projects/Ardour3_7174/interchange/Ardour3/midifiles/ I don't know is the directory is created by Ardour, or Claudia or Ladish. But it may also be that I am doing something wrong. I tried to redefine the file locations in the Session>Properties but this did not solve the problem. Any idea? Thanks in advance, Yvonnick

11 years, 1 month

ABX tester

by Raffaele Morelli

Hi, what abx tester are you using? (if any) I am trying to build abx tester from http://phintsan.kapsi.fi/abx.html with no success. debian testing here, amd64 release. /r

11 years, 1 month

Any HW and SW based backing-band solutions?

by Jostein Chr. Andersen

Hi, This is a 2-part question: I'm playing in a little duo and we are using a Boss BR-80 which provides us with the backing-band duties on stage. The problem here is that it's only one output (Line/Phones - 22 ohms) that gives way to low output for line level, are someone aware of a decent pre amp (DIY is ok) that can be plugged into line input level in a mixer? The second question is about a backing-band software: In addition to the main output, it's nice when everybody can have his own mix (mono is ok). I'm not aware of a good SW based backing-band solution for Linux, are you? Right now, I can only think about something like playing a multi-track from the command line, IE. SOX (one track for each monitoring plus one or two for mixer input). Programs like Audacity and Ardour can't be used here, they take to long time to load and are also a gigantic overkill for this usage. The song need to be triggered (IE. with help from a script) from the space bar by a finger or foot, and the next song must be ready for playing when the current one is finished. I will not have any problem to fix the command line/script solution by myself. Any thoughts or solutions? Jostein

11 years, 1 month

Which audio interface and hardware controller should I get?

by Rusty Perez

Hi folks! I'm a new Linux audio convert. I'm in the process of building my system right now! I'd like some suggestions about usb or firewire audio interfaces. First of all, I'll tell you what my ideal setup would be. I'd like more than two simultaneous analog inputs. I'd like midi input, and even hardware controller capabilities. I know that some units have this all built in to one. I'm very intrigued by the Tascam 1884, an older firewire unit which has 8 analog inputs with preamps. It also offers midi i/o and works as a midi control surface in a Mackie Emulation mode. I don't know if it has any linux support. This would be the dream, but I have to start somewhere. I have available to me a Tascam us122 which I could start with if it is supported. I have looked at a few pages detailing supported audio devices, but they seemed a little older so I thought I should come to the list to get up to date info! In any case, your suggestions are welcome! Thanks! Rusty

11 years, 1 month

Re: [LAU] What is the best MP3 encoder?

by Len Ovens

On Tue, April 2, 2013 7:06 pm, Ralf Mardorf wrote: > On Tue, 2013-04-02 at 18:16 -0700, Len Ovens wrote: >> RME has no choice really. If they want their gear to be used to make >> bluray sound tracks, they have to support 192k. This is the >> certification >> needed by equipment for that use. True, the sound is not any better than >> if the studio used 48k and resampled the finished product to 192k (maybe >> worse). but this is not about sound quality or RME doing marketing... it >> is Hollywood doing the marketing... > > People are convinced it does sound better, "People" are convinced of many things. The list is rather long and probably includes most of what is wrong in this world these days... or for the past 10000 years for that matter. (the 10000 number is drawn from thin air and is in no way related to any understanding I have of our species origin) > http://forum.dvdtalk.com/hd-talk/578983-more-96-192khz-blurays-please.html > , I don't think they are all wrong. Perhaps the players really do sound > better at 192 KHz, because of better matching filters or what ever, just I believe the new DVDs include other audio differences besides just sample rate/depth. Multi-channel (surround) for one.. combining that to stereo would have some sonic difference as well even if A/Bing on stereo only. Setting the Bluray players up to play blueray disks 1 or 2 db louder would be enough to make them sound better... Just remixing for surround is a remix and the mix will probably be "just different". That difference is "Fresh" and new to the listener. People who buy a new system also have a new set of speakers generally, to handle the extra channels if nothing else. Too many new/different things for a good test. Testing using a blueray player would not be the way to go, rather testing each of the different tech changes from one to the other with known program material at known levels. -- Len Ovens www.OvenWerks.net

11 years, 1 month

Re: [LAU] What is the best MP3 encoder?

by Len Ovens

On Tue, April 2, 2013 7:36 am, Ralf Mardorf wrote: > On Tue, 2013-04-02 at 16:33 +0200, Ralf Mardorf wrote: >> On Tue, 2013-04-02 at 15:34 +0200, Peder Hedlund wrote: >> > Your car can probably do 140 mph even though you never go that fast. >> > Being able to use the card in 192 kHz probably doesn't cost that much >> > extra for the manufacturer and I guess the marketing department really >> > loves being able to use it in the advertising. >> >> Yes, it's the second sentence :D >> http://www.rme-audio.de/en_products_hdspe_aio.php . RME has no choice really. If they want their gear to be used to make bluray sound tracks, they have to support 192k. This is the certification needed by equipment for that use. True, the sound is not any better than if the studio used 48k and resampled the finished product to 192k (maybe worse). but this is not about sound quality or RME doing marketing... it is Hollywood doing the marketing... -- Len Ovens www.OvenWerks.net

11 years, 1 month

Looking: cheap analog in to ADAT out converter

by Julien Claassen

Hello everyone! I know, no longer strictly on topic. but the wealth of information is too much, so I'm asking those who know. I'm looking for an A/D converter, more precisely a converter from analog ins to ADAT out. I've seen the Q-ADAT, which sounds almost reasonable. I've also seen the ADA8000. But frankly both seem a bit much. I mean that literally "a bit". any cheaper alternative known or anyone here, wants to seel their old analog to ADAT converter? Thanks for any advise! Warm regards Julien ---------------------------------------- http://juliencoder.de/nama/music.html

11 years, 1 month

Creating a very long audio file?

by Arve Barsnes

I have a series of wave files that I want to combine into one long mp3 and flac. I have checked each file individually, and I'm convinced that I've confirmed my previous notion that these files combined last a total of just above 12 hours. Now, normally I've been using sox to combine the wave files, and using the result in lame and flac to create the result files, but in this instance I'm getting wrong results. A normal run creates a wave file (and resultant mp3) of just above 5 hours. I figured this might be because of a limitation in the wave headers, so I made sox output to a w64 file instead. Now both the wave and result mp3 turned out at 18+ hours. Any idea why sox is getting a wrong result here? Do I need to tell it that the input files are regular wave? When I tried to pipe the output, like this: sox 01.wav 02.wav -t wav - | lame - result.mp3 (or something like that) the file turned out at only 2+ hours, while sox 01.wav 02.wav -t raw - | lame -r -s 44.1 - result.mp3 turned out a 56 hour file. Nevertheless, I think I'm on to something with the last command, but I might have misunderstood some sox or lame documentation. Tips? Arve

11 years, 1 month

rtirq update - 2013 edition (was: rtirq as systemd service?)

by Rui Nuno Capela

(apologies for the x post again, now with url typo corrected) > On 03/30/2013 11:50 AM, Simon Lewis wrote: > > Since fc17 upwards uses systemd and not sysvinit, should rtirq be made a > systemd service? > rtirq has been updated for both old sysvinit and the new systemd support. upstream packages available: http://www.rncbc.org/archive/#rtirq http://www.rncbc.org/archive/rtirq-20130402.tar.gz http://www.rncbc.org/archive/rtirq-20130402-31.src.rpm http://www.rncbc.org/archive/rtirq-20130402-31.noarch.rpm cheers && thanks -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

11 years, 1 month

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Linux-audio-user April 2013