Linux-audio-dev July 2010

linux-audio-dev@lists.linuxaudio.org

69 participants
72 discussions

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Faust MIDI?

by Carlo Capocasa

Hi, is it possible to send and receive MIDI with a FAUST program? Carlo

12 years, 4 months

Port Argotlunar VST to LV2!

by Louigi Verona

Hey guys! Will anyone be interested in porting this GPLed VST to LV2? http://argotlunar.info/ Louigi Verona.

13 years, 10 months

Re: [LAD] FOSS Ethernet Soundcard - IETF Ethernet AVB

by errordeveloper＠gmail.com

Hi everybody, Fallowing up the long discussion i'm trying to sort of give the information you all seem to be missing. there was a meantion of this, but not too many of you have paid any attention: here you can find an article about the current status of the protocol mess: http://prosoundnewseurope.com/pdf/PSNLive/PSNLive_2009.pdf (page 28) obviously eas50 is good to go, but Ethernet AVB is right thing really. the only thing that it's still work in progress, but many of the proprietary vendors, which already have their own networking solution (like Harman with HiQ-net, the one i can name of top of my head) are involved in AVB stadard deveelopment. The idea of AVB is to bypass the IP layer, which is right thing really. you don't need to assign IPs to your audio nodes, really! in avb you'd just have to select channels that nodes whant to listen to. there is a fair bit of documentation on the ietf.org AVB group's page. but XMOS is looking to be the best point of refference: http://www.xmos.com/news/15-jun-2009/xmos-simplifies-ethernet-avb-implement… is think we should forget everything else and crack on with the XS1 AVB implementation! their XS1 chips seem to be really great, their are basically every innovative and open-source minded. the official toolchain is LLVM-GCC based. you can use C, C++ or their own XC. XC is basically C with some stuff omited (like goto and floats) and XMOS IO stuff added, don't just say WTF, look at it first! you should also watch the videos here: http://www.xmoslinkers.org/conference-online-wf especialy the two about the "XMOS Architecture" and the AVB presentation. some dev-kits are quite expencive, but that's due to low-volume really ;) there is alos a nice USB Audio kit! plus there is alittle board that is cheap and has two RJ45's on it already :) I'm myself studying the XC book at the moment. And geting familiar with the tool set :)~~ looks very exciting, cause these are the invovative chips! ok, may be an FPU is really missing on XCore, but how many DSPs have it anyway? well quite a few, but there was no FPU on dsps for ages! :)) also XC or C/C++ are so much more obvious then the bloody "menthal american military engeneers non-sense" called HDL-whatever! Cheers Everyone, Hope you will appreciate my excitment :)~ (l0l) -- ilya .d

13 years, 10 months

any pointers for RT kernel for ARM?

by nescivi

Hiho, I managed to get SuperCollider and JACK running on my IGEP [1], on the pre- configured Ubuntu on the SD, but of course the audio is still bumpy. So... I'm looking for a RT kernel for this little machine... Anyone have any pointers? I did find this one: http://beagleboard.org/project/omap-rt-patch/ but the project doesn't show much recent activity. and... I also noticed that JACK didn't want to run with alsa as backend; oss as backend works though (but gives the bumpy sound). Anyone willing to share their experiences doing linux audio on ARM processors? sincerely, Marije [1] http://www.igep-platform.com/

14 years

what does "full range" mean with regard to ambisonics and speakers?

by "Bearcat M. Şandor"

Folks, I'm a recovering audiophile. When i was reading the magazines and reading about over priced (in my now opinion) speakers, the words "full range" tended to mean that a speaker was reasonably flat from 20 Hz to 20 kHz. Granted those were unusual. I have read that speakers in an ambisonic set up should be "full range". I'd like to set up a ambisonic speaker system (8 channel to start), and the prospect of 8 full range channels is daunting. Since it seems they would be stand or wall mounted (at least some of them) that means monitors and subwoofers. Since all channels must be the same, that means 8 subwoofers...somewhere in the room. So what does "full range" mean usually and what does it mean in terms of talk on this list? Thanks, Bearcat

14 years, 1 month

announcing envy24control, mudita (*) edition.

by Niels Mayer

Summary of updates from envy24control 0.6.0 (GIT HEAD) to "1.0.0": (0) After a decade, incremented version to 1.0.0 (**) (1) Implemented missing "Peak Hold" functionality in meters and reimplemented meters for increased efficiency and lower X resource usage. (see http://www.linuxaudio.org/mailarchive/lad/2010/7/12/171535 & https://bugzilla.redhat.com/show_bug.cgi?id=602903 ) (2) All volumes are represented as decibels, including the 0 to -48dB range of the hardware peak-meters, the 0 -to- -144dB attenuation for all inputs to the digital mixer, the 0 -to- -63dB attenuation of the analog DAC, and the +18 -to- -63dB attenuation/amplification of the analog ADC. (3) All gtk "scale" widgets have dB legends; the "PageUp" "PageDown" keys allow rapid movement between the marked levels. (4) Got rid of myriad compile warnings and other minor fixes across codebase. Some screenshots: http://nielsmayer.com/envy24control/Screenshot-Envy24Control-AnalogVolume.p… http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorInputs.… http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorPCM.png ------------ To the ALSA project: please consider this patch to alsa-tools' envy24control (**): http://nielsmayer.com/envy24control/envy24control-0.6-to-1.0.patch (patch to 'envy24control' from GIT trunk/head of alsa-tools) http://nielsmayer.com/envy24control/envy24control-1.0.README (summary of changes from 0.6.0 to 1.0.0) ------------ Those wanting to compile directly, or run a 64 bit linux binary I've built: http://nielsmayer.com/envy24control/envy24control-1.0.tar.gz (full directory, just follow README directions to build/install) http://nielsmayer.com/envy24control/envy24control-1.0-fc12-x86_64.tar.gz (x86_64 binary that should work on fedora12 and equivalent OpenSuse release) I'd appreciate any testing results or comments on this "1.0.0" release. In particular, I'd like some assurance that the dB markings on sliders in "Analog Volume" panel are correct (compared to values reported by 'alsamixer'). I'm looking for testing with following devices (as I think my testing covers code for M-Audio Delta 44 & Delta 66, Terratec Dmx6fire & EWX2496) specifically: M-Audio Delta 1010, M-Audio Audiophile 2496, M-Audio Delta 1010LT TerraTec EWS 88MT, TerraTec EWS 88D, TerraTec Phase 88, Hoontech SoundTrack DSP 24 (all variants). Thanks, Niels http://nielsmayer.com PS: (*) http://en.wikipedia.org/wiki/Envy#In_philosophy :: "mudita, taking joy in the good fortune of another. This virtue is considered the antidote to envy and the opposite of schadenfreude." (**) http://git.alsa-project.org/?p=alsa-tools.git;a=tree;f=envy24control;h=d5a5…

14 years, 2 months

Document on using Jack Apps

by Arvind Venkatasubramanian

Hello: I am very much new to Linux and Linux audio. I am trying to measure audio IO latency for my system. Jdelay seems to be the right tool but when I run it on the terminal, I am getting message "Signal below threshold..." which probably might be because Jack is trying to capture audio and ends up getting the low noise floor because I do not have meaningful signal source connected. But then I tried to patch the this App to the qJACKctl app but the settings console is not straight forward to interpret. as it involved many parameters. I guess there are # of frames per period that may eventually be used to calculate the target latency but *is there a step-by step document description for Jdelay and any other JACK tools and also using JACK audio server in an effective manner*? Also, qJACKctl console does not offer options for very low sampling rates like 8 KHz. With ALSA, this should be possible but may be this particular tool does not support it. Can anyone help here? Thanks, -F

14 years, 2 months

Re: [LAD] [LAA] announcing envy24control, mudita (*) edition.

by Niels Mayer

[trimmed down reply-to linux-audio-dev(a)lists.linuxaudio.org] On Tue, Jul 27, 2010 at 8:25 AM, john ffitch <jpff> wrote: > Any chance of an option to envy24control to allow colour blind people > to see three zones? I had been using it for years before I was told > that there is a red section at the top and green below > ==John ffitch John -- Thank you for writing and you'll be happy to know that the new "mutida24" (to avoid confusion with the old and overly colorful envy24contol) does not use coloration in the same way, and yes, I'd imagine the old choice of colors would be particularly bad for http://en.wikipedia.org/wiki/Color_blindness#Red-green_color_blindness . Although there's been some recent minor changes, this is the look of the new meters: http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorInputs.… http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorPCM.png Hopefully, at least for any colors at issue, there is enough contrast against the background to make it properly visible. Please let me know of any specific issues w/ the simplified, and more X-efficient meters in this version, which try to be less eye-candy, and more useful for making a visual level determination of inputs, outputs, or digital mixer inputs. The peak metering features a green/white/orange/red progression of the peak color (the thin band that stays fixed while the meter below bounces around). But in addition, and more helpful to you, are he peak-level readouts in dBFS from -48 to 0dBFS for all inputs and outputs, including in the "Analog Volume" panel. The 0dBFS label is red with white background, whereas the other levels remain black with white background. That should provide enough redundancy in the metering to not hide important information from you. However, I'm not sure of the perceptibility, of the red "0dBFS" marking. On the other hand, it does say "0dBFS" (in red) which is quite different-looking than, say "-5.02" (in black) In http://nielsmayer.com/envy24control/Screenshot-Envy24Control-AnalogVolume.p… the green (at 0dB) and red (+6 +12 +18) markings on the ADC would not be very useful for you; fortunately, the "+" should be visually distinctive from all the other "-" markings in blue. Other alternatives would be Bolding amplification values, and underlining unity gain. Thanks, Niels http://nielsmayer.com

14 years, 2 months

Re: [LAD] what does "full range" mean with regard to ambisonics and speakers?

by Martin Leese

Jörn Nettingsmeier wrote: ... >> it's no problem (and in fact customary) to use subwoofers with >> ambisonics for low-range extension. you can either forego localisation >> in that band and just use a mono sub (perfectly ok), use 2 and have some >> l/r signal in ithem, or you can use 4 subs in the corners of the room >> and drive them with a special horizontal decode. For subwoofers (but not main speakers) you can also use three. To see the various options discussed in more detail, visit: http://www.ambisonia.com/wiki/index.php/Bass_management Regards, Martin -- Martin J Leese E-mail: martin.leese stanfordalumni.org Web: http://members.tripod.com/martin_leese/

14 years, 2 months

(Off-topic) Audigy 4 and Surround

by JohnLM

Hi! This might be a complete offtopic here, but there hardly is another place to ask, where there are serious audio-tech savvy people. :) So I have Creative Audigy 4 sound card on WinXP x64. It's been a pain to set-up a working 5.1 surround system. In fact I've never done it. Has anyone got it right? I had no luck with Linux' ALSA either. I currently run it with the bloody Creative's CMSS, which is pretty good for stereo music recordings and most movies. I knew my 5.1 wasn't right, but until I tried ambisonics I was quite oblivious to the fact. So the actual problem. There is no real surround. The back speakers actually put out the same thing as front, with, what appears to be, some phase offset. This is most likely CMSS doing, but If I disable CMSS I get ouput only of two front speakers. Regardless of CMSS setting anything that is recorded as in "directly behind" doesn't play at all. With Linux and ALSA, there is a bit different problem. ALSA appears to cap the maximum amplitude below capability of the sound card, so I get "clipping artefacts" at *much* lower volumes than WinXP. So neither ALSA driver, nor music players were satisfactory to me and I never went though pains to get surround on it. I had not tried my luck with OSS4 yet. If it was up to me I'd use stripped down OSS4, which would provide direct access to 6-channel output. On top of it I'd put JACK and then route what I want, where I want. This is no small task however, and I'm not even sure if it can be done at all. I ask of you to share any experiences and ideas you have with Audigy, on any of these platforms. Thanks in advance! -- John

14 years, 2 months

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Linux-audio-dev July 2010