Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7520 discussions

by Mike Cookson

JACK NetSource GUI is renamed to JACK Network Manager (jack-netmanager-gtk), since jack_netsource is only a command line front-end for jack netmanager module. Changes: - Name changed from jack-netsource-gui to jack-netmanager-gtk (humany name - JACK Network Manager), since jack_netsource is just a wrapper, controlling jack "netmanager" module. - Presets support, deprecating creation of script with tray support, due to ability to reuse ordinary scripts by JACK Network Manager itself. During of preset saving name is requested and approriate script is stored in presets directory. - Ability to start sources at LADISH rooms. - ladish_launch is deprecated. ladish_control snewapp and rnewapp commands used instead. - Improved desktop launcher - New system tray icons; added application icon, unwantedly removed from 0.2.2 - Fixed help text - Options for jack_netsource from various jack versions are stored in separated file now - Internationalization of bash scripts through gettext - Build system improvements: all text files are configured - e.g., instalation prefix now is working. Configured files are stored into separate build directory, which is autonomous and can be distributed. Also, uninstalation is available. - Internationalization improvements: added update-locales script, which updates template with translations at once, and localization of bash scripts through gettext. Note for distribution maintainers: if you place menu items for special audio software into extra submenus inside standart Audio menu, it would be nice to have JACK Network Manager item in one place with QJackNet (at least) and QJackCtl. And create dummy jack-netsource-gui package for smooth upgrade. Page at GTK-Apps: http://gtk-apps.org/content/show.php/JACK+Network+Manager?content=122327 Latest screenshot is third.

15 years, 5 months

May I asked something OT?

by Ralf Mardorf

A lot of kids wish to have a kill switch for their guitars. A kill switch is a short circuit, to 'stop' the audio signal. I'm not fine with this solution, but the kids argue, that e.g. an interruption does cause unwanted noise, especially for over drive sounds. IMO even using opto-electronics won't solve the issue, because the noise of the transistor overdrive effect still would be hearable, while for a short circuit there is silence. Has anybody an idea to solve this without a short circuit? I'm really not a fan of short circuits. Note, it's not possible to do an interrupt all the times behind the latest noise generator and even an interrupt could cause noise itself, while a short circuit indeed is a good way to cancel sound.

15 years, 5 months

Routing signals in mixer (Ardour, Qtractor)

by Louigi Verona

Hello! Both Ardour and Qtractor are capable sequencers. Ardour is getting midi soon, Qtractor is getting automation. All of this is good news. I was wondering if the devs are interested in implementing routing signals in a mixer as the next important priority. This would allow not only effects chains which is usually very useful, but also usage of vocoders. At the moment it is impossible to use a vocoder plugin inside Ardour or Qtractor. The beauty of vocoders is that they can be used not only on voice, but as electronic instruments as well. I am often using them to create beautiful textures, but in order for the result to be good, I need to automate parameters of the vocoder. LADSPA Vocoder, for instance, is capable of generating very beautiful sounds, but only if you can change the bands in real time, fade the in and out. Of course, sometimes it can be recorded live, but in many cases it needs to be automated and stored in a project. Tell me what you think. -- Louigi Verona http://www.louigiverona.ru/

15 years, 5 months

Re: [LAD] Floating point processing and high dynamic range audio

by lieven moors

On 07/22/2010 05:31 AM, Philipp Überbacher wrote: > Excerpts from Philipp Überbacher's message of 2010-07-22 03:16:00 +0200: > > > Excerpts from fons's message of 2010-07-22 02:24:04 +0200: > > > On Thu, Jul 22, 2010 at 01:05:01AM +0200, Philipp Überbacher wrote: > > > > > > > I think the word loudness is a problem here. Afaik it usually > refers to > > > > how it is perceived, and twice the amplitude doesn't mean twice the > > > > perceived loudness. It may mean twice the sound pressure level, > energy, > > > > or intensity (if we ignore analogue anomalies, as you wrote in > some other > > > > answer). > > > > > > Subjective loudness is a very complex thing, depending on the > > > spectrum, duration, and other aspects of the sound, and also > > > on circumstances not related to the sound itself. > > > > > > For mid frequencies and a duraion of one second, the average > > > subjective impression of 'twice as loud' seems to correspond > > > to an SPL difference of around +10 dB. > > > > I had a brief look at the section about loudness in musimathics and it > > mentions 10 dB based on the work of Stevens, S.S. 1956, > > "Calculation of the Loudness of Complex Noise" and 6 dB based on > > Warren, R. M. 1970, > > "Elimination of Biases in Loudness Judgments for Tones.". > > I think I've encountered the 6 dB more often in texts, which doesn't > > mean it's closer to the truth, if that's possible at all. > > Knowing a 'correct' number would be nice for artists and sound > > engineers, but if it varies wildly from person to person, as Gareth Loy > > suggests (no idea where he bases this on) then this simply isn't > > possible. Picking any number within or around this range is probably as > > good as any other. > > > > > I often wondered what criterion we use to determine which > > > objective SPL difference sounds as 'twice as loud'. We don't > > > have any conscious numerical value (there may be unconscious > > > ones such as the amount of auditory nerve pulses, or the amount > > > of neural activity), so what it this impression based on ? > > > > > > The only thing I could imagine is some link with the subjective > > > impression of a variable number of identical sources. For example > > > two people talking could be considered to be 'twice as loud' as > > > one. But that is not the case, the results don't fit at all (it > > > would mean 3 dB instead of 10). > > > > I never thought about that to be honest. It's immensely complex. It > > might have to do with each persons hearing capabilities, for example > the > > bandwidth of loudness perception or the smallest discernible loudness > > difference. If it really is very different from person to person, then > > an explanation that takes the different hearing capabilities into > > account could be sensible, don't you think? > > I did find some more approaches to the problem, but those are just > ideas. From my personal experience I have to say that I have a very hard > time saying when something is twice as loud. A musically well trained > person might have an easier time, I wouldn't know, but for me twice as > loud is something that is very vague. This might already explain the > large deviation between subjects as described in musimathics. It lead me > to another idea though, the evolutionary perspective. Evolutionary it > likely never was important whether a sound is twice as loud. The only > situation I can imagine where judging loudness probably was important > is judging distances. How far is the animal I can't see, be it prey or > predator, away from me? We know that this takes more than the SPL into > account, and 'twice as loud' doesn't have relevance in this context. So > maybe the loudness perception is linked with spatialization. I think this is a very interesting idea. Could this be linked to some kind of avarage SPL of all the sounds human beings are exposed to (and this variable changes throughout history). Because when we try to judge the distance of a barking dog, our brain would use the knowledge of all other dogs we heard barking before, to estimate the distance of that dog. If we never heard a dog before, maybe we would use the sounds of other animals as a reference, and so on... greetings, Lieven > > My other ideas are rather stupid, just ways to get the right numbers for > your two person idea. > I simply used ln instead of log and got 7, but that's not even Neper and > has no relevance. > > The other idea of that kind is to assume a field quantity, which would > result in 6 dB. I'm still easily confused about 10*log and 20*log, but I > think 20*log is usually used for sound pressure, but maybe not for > psychoacoustic effects. > -- > Regards, > Philipp > > -- > "Wir stehen selbst enttäuscht und sehn betroffen / Den Vorhang zu und > alle Fragen offen." Bertolt Brecht, Der gute Mensch von Sezuan > > _______________________________________________ > Linux-audio-dev mailing list > Linux-audio-dev@... > http://lists.linuxaudio.org/listinfo/linux-audio-dev

15 years, 5 months

"El-Cheapo" software-only equivalent

by rom

Hi all, i'm new to this list. I'd like to ask some advice about a small multitrack recorder program i wrote, and have been using for some time. Basically, what it does is to: - simultaneously capture sound from several consumer-grade soundcards. - use libsamplerate to stretch the audio streams, re-syncing them to the one chosen as "master". The stretch ratio is continuously re-calculated to make the overall frame count of the stretched stream match the overall frame count of the master. - write the "corrected" streams plus the "master" stream to parallel .wav files using libsndfile. The purpose is the same as the quite famous "El-Cheapo Howto" ( http://quicktoots.linuxaudio.org/toots/el-cheapo ), just with no soldering involved :-) Of course, i know the solution is far from perfect, but i use it to record some friends of mine who play in a blues/punk band, and the result is not that bad. Now, the question is: do you think this piece of code can be of any interest for someone out there? Do you think i should i publish it on an open source repository ? Or maybe there's already some other software i'm not aware of, that does the same thing? thanks for your patience, please excuse my bad english. bye alberto

15 years, 5 months

[Fwd: Re: No nagging, a serious question]

by Ralf Mardorf

On Sun, 2010-07-04 at 21:52 +0100, Dan Mills wrote: > Trying again, I accidentally sent this off list the first time.... So I can add, I anyway will test to use two PCI cards, at least for MIDI, for audio would be nice too. -------- Forwarded Message -------- From: Ralf Mardorf To: Dan Mills Subject: Re: [LAD] No nagging, a serious question Date: Sun, 04 Jul 2010 22:47:47 +0200 It was off-list?! On Sun, 2010-07-04 at 21:35 +0100, Dan Mills wrote: > If you are working in a world where you know the available hardware in > detail LOTS of things become easy, for example I can time stamp an > incoming MIDI byte with the sample number of reported as current > position by the sound card (There is only one), instant way to get > effectively zero latency jitter. > This doesn't work nearly so well when there is more then one sound card Good to read about this issue. I always disable the on-board audio devices, but I would add a second PCI card to my PC and sync it with the already installed sound card, so I better don't do it. It at least would be nice to have several MIDI IO by simply using some cheap Envy24 cards. Unfortunately those cheap cards seems to use just one of the two MPUs supported by the Envy24. Cheers! Ralf

15 years, 5 months

MVerb for LADSPA/LV2/???

by Dave Phillips

Greetings, Martin Eastwood has posted the code for his MVerb: http://martineastwood.com/ Open-source, GPL3'd free software. Maybe someone could whip up a plugin or standalone app from this code ? PS: If you download the zipfile note that it does not include a top-level directory, i.e. it'll dump its contents into the current directory. Best, dp

15 years, 5 months

Re: [LAD] [LAU] New modular synthesis library w/ ruby bindings

by Harry Van Haaren

On Mon, Jul 19, 2010 at 8:06 PM, Bernardo Barros <bernardobarros2(a)gmail.com>wrote: > At least for me it would be easier to think as c = 0 > > c = 0 > c' = 12 > c, = -12 > c'' = 24 > c,, = -24 > > because most of the notes in a regular score is mostly like to happen > in the middle and things looks simpler that way. UNLESS you're already > familiar with all kind of notes with MIDI number notation. > Hey, Yes I was referring to the MIDI standard there.. Should have mentioned that.. There's no "rule" telling you you should do it differently, I'm expressing my opinion that for me it would seem more logical to have middle C at 60. -Harry

15 years, 5 months

Re: [LAD] [LAU] Free Software Support Re: Reverb to use with LinuxSampler/piano?

by drew Roberts

On Monday 19 July 2010 15:13:54 Bernardo Barros wrote: > Yes, but a lot of stuff is not presently appealing to industry. > Like... ,music and multimedia stuff: LilyPond, SoundCard drivers, > InkScape, GIMP, Ardour, Qtractor and so on. But yes, even though I'm a > socialist I agree with you that big corporations would benefit with > Free Software a lot. I wouldn't be too sure about that. You might just need to think of which industry. And A lot of the groups I mentioned are normally small businesses around here. There are a decent number of one man operations wrt loclsmithing. Barber shops go from one barber to several but they are not big corporations by any means. The associations may be bigger, but if the members wanted the software written for them to be Free, they could use their association to coordinate their efforts. > > 2010/7/19 drew Roberts <zotz(a)100jamz.com>: > > On Monday 19 July 2010 12:12:54 you wrote: > >> Maybe they always overlap somehow, software needs support. Really good > >> and complex software need more support than one person can do just > >> with his free time. I am not sure from where this support could come > >> from. > > > > I keep thinking that industry groups would be wise to support the > > development of Free software for their industries. > > > > For instance: > > > > Locksmiths > > http://www.aloa.org/ > > http://www.uklocksmithsassociation.co.uk/ > > http://www.masterlocksmiths.com.au/home.aspx > > > > Barbers, Beauty Salons, Furniture Stores, Ocean Engineers, Corner Grocery > > Stores, what have you. > > > > all the best, > > > > drew > > > >> From universities, donations, companies, governments? The Linux > >> kernel, for example, is very complex and needs a lot of work all the > >> time since new hardware and software need it , and it is well > >> supported by really big companies like google, novell and red har. > >> Other areas this is not so easy to get support from companies, so I > >> now just can think of the other options: donations from people that > >> make money or work seriouly with the software, universities and > >> government programmes. The last one seems unlikely to happen but here > >> in Brazil the present government is officially supporting Linux and > >> some software projects. Donations from professionals and Universities > >> can help Free Software a lot too. > >> > >> > >> 2010/7/19 drew Roberts <zotz(a)100jamz.com> > >> > >> > On Sunday 18 July 2010 11:36:42 you wrote: > >> > > On Sun, Jul 18, 2010 at 10:04 AM, drew Roberts <zotz(a)100jamz.com> wrote: > >> > > > Commercial and Free are not opposites. > >> > > > >> > > no, just siblings who fight with each other like crazy for 18 years > >> > > before developing a deep and abiding love for each other :) > >> > > >> > Free and Commercial can intersect / overlap. Venn wise. > > > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user(a)lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user

15 years, 5 months

Re: [LAD] May I asked something OT?

by Sakari Bergen

On July 18, 2010 03:57:06 pm Ralf Mardorf wrote: > A lot of kids wish to have a kill switch for their guitars. > A kill switch is a short circuit, to 'stop' the audio signal. > > I'm not fine with this solution, but the kids argue, that e.g. an > interruption does cause unwanted noise, especially for over drive > sounds. IMO even using opto-electronics won't solve the issue, because > the noise of the transistor overdrive effect still would be hearable, > while for a short circuit there is silence. > > Has anybody an idea to solve this without a short circuit? > > I'm really not a fan of short circuits. Note, it's not possible to do an > interrupt all the times behind the latest noise generator and even an > interrupt could cause noise itself, while a short circuit indeed is a > good way to cancel sound. If you don't like short circuiting the mics, just switch the output jack hot lead between ground and the volume pot(s). Connecting the output to ground is the same as turning the volume down. There is no need to short the mics... On 19.07.2010 08:30, Tim E. Real wrote: > If you play a Gibson you can set the neck pickup volume to zero and > the bridge pickup volume to full and then toggle the pickup switch, > rapidly if desired, like Eddie van Halen on You Really Got Me. > Tim. Tom Morello gets real creative with his guirar, and uses this technique alot. He also unplugs his guitar to make the pedals and amps make noise (oscillate), and controls the noise with his wah pedal and by touching the tip of the plug to the guitar bridge (which is grounded). Good demonstration of how just cutting the signal lead can cause lots of noise, while a short will be more or less silent :) -Sakari-

15 years, 5 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev