Linux-audio-user November 2004

linux-audio-user@lists.linuxaudio.org

188 participants
181 discussions

by Ivica Ico Bukvic

Hi all, I just wanted to share with you a quick report from the ICMC. The conference was much fun although many events were overlapping and therefore it was impossible to attend everything. A number of new open-source audio-oriented software was demoed many of which are also available for the Linux platform. My Linux demo went relatively well (apart from the 2 consecutive X-server crashes, courtesy of the crappy binary-only ATI's driver, and an odd issue where disconnect of hdsp pcmcia did not yield acknowledgment from the 2.6.7 kernel and therefore subsequent reconnect did not recognize the card's presence). There were ~20+ people in the audience at any given time throughout the presentation (some of them were there only for a portion of the presentation, so people kept coming and going). Stuff I demoed (in no particular order): *Jack *Freqtweak *Jamin *Pd/Gem *Latency tests *Ardour *Sweep *Rezound *Audacity *gAlan *Rosegarden *Qsynth/fluidsynth *ZynAddSubFX *VKeybd *Cinelerra *Blender *Xine/DVD playback *Spiral Synth Modular *KDE productivity Stuff I wanted to demo but ran out of time: *Hdsp latency/performance *RTcmix synthesis language *Supercollider *Csound Etc. Stuff I observed preparing and doing this demo (please understand that I have no idea whether these problems are result of my own setup or are justifiable bugs -- nonetheless I am including them here in hope they may shed some light towards their resolution): *ZynAddSubFx crashes when a lot of polyphony is created even though the cpu utilization is not topped-off *Jamin crashes when pushing limiters to the extereme (not consistently) *fluidsynth has some weird looping problems (I am trying to track this one down as apparently this is specific to my setup) *Ardour's real-time preview via sliding the timing bar does not work (I do have one release prior to the latest, though) *Spiral Synth Modular crackles a bit even with large buffer sizes when using OSS (haven't had the time to do exhaustive tests nor did I try using Jack object as of yet) *Audacity has some instabilities when used in conjunction with Jack *There is a real need for Jack to be able to restart (should a need for its restarting ever arise) without losing all of the connections Stuff that really rocked: *I was able to run Jack on my via82xx laptop soundcard using rt mode with 64x4 buffers (5.33ms at 44.1KHz) *almost* rock solid (there is still some issues due to crappy ATI driver and obviously 2.6.7 kernel that is still sub-par to the 2.4.x kernel series performance, but that mainly amounted to perhaps a 1 xrun/minute). *I was able (although after the demo) to run hdsp with 64x2 (2.9ms at 44.1KHz) *almost* rock solid (random occasional xruns but no xruns when adding new connections and/or apps to the Jack session). *I tested the PD with the hdsp's 64x2 buffer size and used the latency test patch by connecting audio out with audio in on the multiface and got ~6ms :-). The general reaction from the audience was quite positive as many of the users who even were familiar with the Linux Audio scene were impressed by the diversity and quality of the software offering as well as the flexibility of the Jack's framework and the overall user-friendliness of the UI/desktop environment. I would like to therefore use this opportunity to once more thank all of you for being such active and generous contributors to this great community. Without you none of this would have been possible! ---------------------- The panel on the "Standards from the Computer Music Community" was very much interesting as it covered many of the important facets of today's computer music scene, but also more importantly revealed some of the greatest strengths of Linux, including (but not limited to): *openness of the standards and therefore ability to generate umbrella meta-standards (i.e. LASH) *longevity *ability for a new standard to supersede the reigning old standard solely based on its merit (i.e. Alsa vs. OSS), and not due to its commercial PR and/or widespread use *Minimization of the misrepresentation of the standard's features and/or abilities Having had this wonderful opportunity to be a part of such panel (many thanks to Matt Wright for inviting me!) has truly reinforced and clearly defined the advantages as well as strong reasons for being a part of this community. Many thanks go to all of you who have generously offered your insight in these issues. Time permitting, I will post a more in-depth list of the ideas I've covered during the panel. Matt Wright should also have slides ready on his site sometime soon. Best wishes, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/ --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 11/1/2004

20 years, 6 months

RE: [linux-audio-user] recording bats?

by Cornell III, Howard M

Um, so in one case the bat signal was given as 40-100 kHz and in another case as 10-120 kHz. So BW would be somewhere between 60-110 kHz. Since some people can hear the bat sonar, it probably does extend down to 10 or 15 kHz. Would you say a 192 kHz recording could be made to cover a BW of 15-111 kHz? That might have reasonable "fidelity". -----Original Message----- From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio-user-bounces@music.columbia.edu] On Behalf Of Dan Mills Sent: Monday, November 08, 2004 5:34 PM To: linux-audio-user(a)music.columbia.edu Subject: Re: [linux-audio-user] recording bats? On Monday 08 November 2004 21:01, Cornell III, Howard M wrote: > Remember, you have to sample at a rate least twice the highest > frequency of interest. The data suggests you need the at least 192 > kHz rate to record 96kHz fundamental signal. Plus the special microphone. A minor niggle but in some circumstances (and this might be one of them) it can matter, you only need twice the Bandwidth of the signal being recorded, NOT Twice the highest frequency! For all practical purposes in audio applications these are equivalent, but there are cases where subsampling a signal is interesting. For example if I have recorded a ultrasonic source using a 192Khz SR and I know that the source is between say 50 & 60 Khz then I can bandpass filter in the DSP and sub sample the output of the BPF at just above 20Khz (say 24K) which reduces the amount of data I need to store WITHOUT losing any information that was present in the original signal. This is kind of cool for playing with sonar where the DSP requirements for signal correlation get really silly if you work at the input rate. Regards, Dan.

20 years, 6 months

RE: [linux-audio-user] recording bats?

by Cornell III, Howard M

Remember, you have to sample at a rate least twice the highest frequency of interest. The data suggests you need the at least 192 kHz rate to record 96kHz fundamental signal. Plus the special microphone. Howard -----Original Message----- From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio-user-bounces@music.columbia.edu] On Behalf Of Chris Cannam Sent: Monday, November 08, 2004 2:35 PM To: linux-audio-user(a)music.columbia.edu Subject: Re: [linux-audio-user] recording bats? On Monday 08 Nov 2004 20:17, Eric Dantan Rzewnicki wrote: > I just had a crazy idea ... Sorry if this is off topic a bit. Does > anyone know what frequency ranges bats use? Would a 96KHz 24bit card > be able to capture anything useful from their sounds? Depends on the bats, but generally yes. Some of them are on the edge of the human hearing range (I used to be able to hear the bats at my parents' house, although my hearing is no longer quite good enough). First Google hit for "frequency range of bats" is a bit less optimistic than I am: http://hypertextbook.com/facts/1998/JuanCancel.shtml Either way, wouldn't the microphone be more of a limiting factor than the soundcard? Chris

20 years, 6 months

[linux-audio-user] Libsamplerate support for ecasound

by ed orphan

Got the file from its home site and a packaged file from another. I got the packaged file installed and ecasound detects it during ./configure, so all is well. I just wasn't sure if ecasound wanted a file called Libsamplerate or if there was something that had to be configured. Anyway, everything is fine now and ecasound compiled and installed correctly.

20 years, 6 months

[linux-audio-user] Re: linux-audio-user Digest, Vol 14, Issue 22

by Michele Spinolo

> > > I have a audio workstation running with 3 Terratec EWS88D cards, > > which provides only digital I/O. > > > > I'm thinking to upgrade my PC adding an analog capable card, which > > will be used primarly for D/A conversion. I'm thinking about > > Terratec EWS88MT which could be found for a quite cheap price both > > new or used, but I understand it's a quite old card. > > Buy it! It's a very good high end card, should work with ALSA and it's > not that old. With Linux, very new hardware often isn't supported yet, > so buying about one year in the past is always recommended. And the > Terratec card is still very up to date. > > Ciao > -- > Frank Barknecht _ ______footils.org__ Good news here!:-) It will be used as the output stage of a real-time convolution machine, so Hi-Fi quality DAC would be needed. I noticed declared S/N ratio is 110db, I was wondering if this performance is enough for hi-fi audio quality. Thanx! Michele

20 years, 6 months

[linux-audio-user] Cannot boot linux-2.6.10-rc1-mm3-RT-V0.7.18 on x86_64

by thewade

Hello again! I am reinstalling linux because of a foolish choice I made earlier that forced me too. My machine is an AMD64 laptop: (http://www.aproximation.org/application/AMD64laptop.html) Im trying to build and install 2.6.10-rc1-mm3-RT-V0.7.18: (http://www.aproximation.org/application/kernel.config-2.6.10-rc1-mm3-RT-V0.… And it builds fine (with warnings of course), but on boot I get a stacktrace that flys by too fast to deduce where the problem originates. And I cannot find the error in /var/log/message! One possible candidate for trouble is from dmesg: ehci_hcd 0000:00:03.3: init 0000:00:03.3 fail, -95 ehci_hcd: probe of 0000:00:03.3 failed with error -95 but again I see no kernel panic message. Can anyone tell me how to find the error? Perhaps a kernel flag that writes what is being displayed to the screen, to disk? Thanks for all the past and future help! -thewade

20 years, 6 months

[linux-audio-user] Terratec EWS88MT and BruteFIR

by Michele Spinolo

I was wondering if someone on this list is using BruteFir with this card, in positive could he be so kind to post the BruteFir configuration file he is using? Thanks! Michele

20 years, 6 months

[linux-audio-user] terratec Phase 88 and Sekd Arc 88 supported under linux?

by Marius Janßen

Hello all, does anybody of you know if the phase 88 by terratec or the SEKD Arc 88 is supported with ALSA etc. ? Any comments welcome Thanks in advance Marius

20 years, 6 months

[linux-audio-user] success/failures with M Audio / Edirol Firewire cards ?

by Olivier Kaloudoff

Hi, I plan to buy one of the following external mobile sound interface, to record live with linux; - Edirol FA 101 (Firewire, XLR, spdif, MIDI) - M Audio Firewire 410 (Firewire, XLR, spdif, DTS) is there someone out there using this kind of device already with linux ? the manufacturers were unable to give me informations about linux compatibility, so I wonder if I'll be able to use all the features with the standard linux tools, or only a subset. would be really happy to ear feedback from you. Best Regards, Olivier Kaloudoff LUG Linux Azur http://www.linux-azur.org http://www.edirol.com/products/info/fa101.html http://www.m-audio.com/products/en_us/FireWire410-main.html

20 years, 6 months

[linux-audio-user] Soundcard with good-performing D/A

by Michele Spinolo

Hi guys, I have a audio workstation running with 3 Terratec EWS88D cards, which provides only digital I/O. I'm thinking to upgrade my PC adding an analog capable card, which will be used primarly for D/A conversion. I'm thinking about Terratec EWS88MT which could be found for a quite cheap price both new or used, but I understand it's a quite old card. So any ideas for a new card for my setup? Thanks! Michele

20 years, 6 months

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Linux-audio-user November 2004