Linux-audio-user February 2005

linux-audio-user@lists.linuxaudio.org

173 participants
197 discussions

[linux-audio-user] Valid download link for Steinberg VST SDK?

by Robert Persson

Does anybody have a valid download link for the Steinberg VST SDK? I need the headers to compile fst, but the Steinberg people seem to be playing sillybuggers with their website and the old links don't work anymore. Many thanks. Robert -- Robert Persson ireneshusband(a)fastmail.fm YahooMess:ireneshusband AIM:shamanicpolice 56% of US Gulf War I veterans have chronic health problems. The disability rate after Vietnam was 10%. For other 20th Century wars it was 5%.

20 years, 1 month

[linux-audio-user] What of "Wired" ??

by Russell Hanaghan

What has become of "Wired", the loudly touted "Next greatest audio app" in Linux? There were several hard and fast releases last November and updates in December then... And should someone from the development team be on the list...nice looking interface...But Puhleeeeeze use Jack for audio interface or add it to the list of already available options. I know it uses Port audio but I have not been able to figure out what the heck Portaudio is let alone how to get it working! R~

20 years, 1 month

[linux-audio-user] echo "csound 3 128 direct" > /proc/asound/card1/pcm0p/oss

by Atte André Jensen

Hi http://www.alsa-project.org/~iwai/OSS-Emulation.html states that echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss should enable direct mode. But when I run echo "csound 3 128 direct" > /proc/asound/card1/pcm0p/oss (which works) what does the numbers "3" and "128" mean and what would be sensible values? I know it's something to do with buffers size, but that's not enough. Also what would such numbers convert to in terms of latency? Thanks in advance... -- peace, love & harmony Atte http://www.atte.dk

20 years, 1 month

[linux-audio-user] [OT] "The short is off the stick" -- What does that mean?

by Wolfgang Woehl

This is from Lamont Dozier's "Fish ain't bitin'" (1973) Can someone explain the expression "The short is off the stick"? Thanks. And meanwhile in DC Tricky Dick is trying to be slick And the short is off the stick Because I'm gonna get it Tricky Dick, please quit There's a story attached: Dozier's record company (ABC) published a letter from Nixon's administration complaining about the single. Boosted the song into top 10. Wolfgang

20 years, 1 month

[linux-audio-user] jackd --realtime eats my memory.

by Jan Holst Jensen

Hi. Being quite a newbie to audio on Linux, I am trying to get rid of xruns when I record using ardour and jackd. I start jackd with jackd -d alsa --device hw:1 --rate 44100 --period 8192 --nperiods 2 but this max. buffer size is still not enough to completely prevent xruns. I see xruns often when I switch between windows and so thought that this is caused by Xfree86 stealing CPU from jackd, since xfree86 runs with nice factor -10. I tried to run jackd as root so I could use the "--realtime" option but my system is then sadly reduced to a thrashing heap of rubble. After loading a recorded session (approx. 45 minutes of audio - 830 MB data) "top" tells me that ardour is using 69.5 %MEM jackd is using 25.6 %MEM and the loading of the session has taken several minutes due to thrashing (kswapd is on heavy work here according to "top"). The system is very unresponsive and I am unable to use ardour for anything useful. If I don't use the "--realtime" option, "top" says ardour is using 20.4 %MEM jackd is using 2.1 %MEM after the session is loaded (in 2-3 seconds), and the system runs quite well (except for the occasional xruns of course). Is "--realtime" forcing jackd and its clients to lock all of their memory in RAM ? I would then need quite a lot of RAM to use the "--realtime" option. Is it really necessary to use _that_ much RAM ? I also tried to renice jackd with -20 so it runs with priority 0. Will the "--realtime" option give me better stability than the renicing ? Cheers -- Jan Holst Jensen, Copenhagen, Denmark ____________________________ System info: Debian Sarge with a custom-built unpatched kernel 2.6.10. Kernel config includes "Pre-emptible kernel" and pure ALSA - no OSS emulation. Compaq Evo N400C laptop. 700 MHz Pentium III with 128 MB RAM. I know this is not top-of-the-pops but I have used it for two years to do stereo recording and 8-track mixing on Windows 2000 quite nicely. jackd version: bonham:~# jackd --version jackd version 0.99.0 tmpdir /dev/shm protocol 13 bonham:~# __________________________________ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail

20 years, 1 month

[linux-audio-user] upsampling wav's - ardour sessions

by MarC

Hi, I'm using ardour to record and I have recorded my songs at 44100 until now. Now I want to switch to 48000. What's the easiest way (always within the best way to do it) to convert my sessions to 48000? I would like something automatic like running a script before starting an ardour session at 44100 or clicking some option at ardour to import it. I have googled this issue and I found: http://sr-convert.sourceforge.net/ but it's not debian-packaged yet and I'm too lazy now... maybe ecasound? but I'm afraid it doesn't consider so much things as sr-convert...

20 years, 1 month

[linux-audio-user] MIDI stopped working

by MarC

I upgraded my debian testing and now MIDI doesn't work anymore. Before I could load soundfonts normally and now I get: marc@marcbcn:~$ sfxload No AWE synth device is found any clue about how to solve it? (google wasn't enough...) thanks in advance, I wouldn't be a debian user without such great mailing lists as LAU and debian-users...

20 years, 1 month

[linux-audio-user] Re: linux-audio-user Digest, Vol 17, Issue 95

by Aaron Trumm

YES!!! YAY YAY!! VICTORY TO THE MASSSESS!!! > I'm using the HDSP 9652 constantly under FC2/Planet and have no > problems what so ever. The card is working well. hdspconf allows me to > set it up as either a master or slave in terms of clocking, and > hdspmixer works great for routing audio and checking levels. > > I have no issues with this card anymore under Linux. > > Cheers, > Mark

20 years, 1 month

[linux-audio-user] emu10k1 multichannel support

by Lee Revell

I posted this to alsa-devel but since my previous post on this list generated a lot of interest, I am just reposting it here. As promised, here's an updated patch to add real multichannel playback support (and improved multichannel capture) to the emu10k1 driver. http://www.alsa-project.org/~rlrevell/emu10k1-multichannel-v001.patch Please test it and report any problems. I am especially interested in any regressions that impact regular PCM playback (the hw:0,0 device). QuickStart: $ jackd -R -v -d alsa -P hw:0,3 -C hw:0,2 -S I tested this and it works well with 16in/16out at 128, 256, 512 frames. 32 and 64 should work too but I can't test as I'm running a stock 2.6.10 kernel for now ;-). You can check that the routing is correct by connecting a JACK client to the playback ports corresponding to the FX buses described in Documentation/Audigy-mixer.txt and Documentation/SB-Live-mixer.txt, and verifying that the output appears on that channel (the FX buses are numbered from 0 but JACK numbers clients from 1). For example (from SB-Live-mixer.txt): name='Music Playback Volume',index=0 This control is used to attenuate samples for left and right MIDI FX-bus accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. The result samples are forwarded to the front DAC PCM slots of the AC97 codec. So "alsaplayer -o jack -d alsa_pcm:playback_5,alsa_pcm:playback_6" should output to FX buses 4 and 5, which you can test by lowering the 'Music' control in alsamixer. With an SBLive, use ports 1 and 2 for the front channels, 3 and 4 for the rear channels. The Audigy uses different channels, see the above docs for more info. In addition to multichannel recording applications, this should also be useful for OpenAL implementations, which are currently restricted to using 21 sources due to the use of an extra voice per stereo PCM. This should allow up to 63 sources. This also adds some new register info including a per channel half loop interrupt that I have discovered by reverse engineering the Windows drivers. Improvements over previous versions: - Routes the 16 channels to the 16 FX buses by default. - Enables the first 16 FX capture outputs by default, required for full duplex operation at latencies lower than 512 frames. - Rewrote the voice allocator to use a more efficient round robin algorithm, eliminating the need to reserve the first 16 voices for the multichannel device. The next free voice is maintained in the card record and the search starts from there. - Use an extra voice for playback timing rather than the EFX capture interrupt. I was only ever able to get that to work at 64 frames. Also there are definite advantages to being able to use the capture and playback devices independently. - Use the newly discovered per-channel half loop interrupt source for the extra voice rather than the channel loop interrupts. For unknown reasons, this works better for multichannel playback, and does not seem to affect regular PCM playback at all. TODO: - Fix the send routing and volume controls for the multichannel device. The current (copy and paste) solution assumes either one or two voices per PCM. So the default settings work fine but changing them with the mixer is likely to have unpredictable effects. - EFX capture should capture output channels 16-32 (mostly unused now) by default, so that we only capture the sources the user has connected to the multichannel recording inputs in the DSP manager. Typically FX buses 0-15 would be connected directly to FX outputs 16-32 so the capture channels would correspond directly to the playback channels. In order for this to work the default DSP configuration has to be changed slightly. Lee

20 years, 1 month

[linux-audio-user] Re: linux-audio-user Digest, Vol 17, Issue 95

by Michele Spinolo

> > I'm going to buy one of these cards, and I was wondering what about the > > differences from one to each other and concernig ALSA's support. > > By ALSA's website seems they are completly supported, but I would like to > > have some users' confimations. > > hey there - i have the HDSP 9652 and love it - it was a bit of a pain to get > it really working at first, because there was a driver issue that we (the > linux audio community in general) were having a hard time ironing out. the > issue didn't exist with the digi9652 card that was almost exactly the same, > just the HDSP 9652 > > in reality though you have to consider what other gear you have - the 9652 > was necessary for me because I needed it to interface via ADAT light pipe to > 24 channels on my digital mixer. But if I didn't have the digital mixer to > do my A/D conversion, the 9652 would've been wrong because it doesn't have a > breakout box with analog inputs like the multiface or something else - the > HDSP 9652 has 24 channels optical i/o and a stereo spdif - that's the only > in and out of that card... > > TotalMix doesn't run under linux, it doesn't matter. Under linux, you'll be > using a program called hdspmixer - it's the exact same thing (am I > remembering that right? yes I think so) > > what is your studio/other gear situation like? Hi Aaron, I will use the HDSP mainly with adat I/O and s/pdif I/O, so I think it's the right choice. Could you please be more detailed regarding HDSP 9652 problem? Thanx! Michele

20 years, 1 month

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Linux-audio-user February 2005