Linux-audio-user February 2005

linux-audio-user@lists.linuxaudio.org

173 participants
197 discussions

[linux-audio-user] Ardour in linux Journal....AGAIN...

by Timothy Alan Gorman

Just got my march issue.. ARDOUR on the cover.. YEAH BABY YEAH.... but there is some confusion.. the author writes that the m0audio delta66 has "12" input channels.. ? (page 65 under the qjackctl image) I don't get it.. are they digital inputs? I thought them-audio had 4x4 ...?

19 years, 3 months

[linux-audio-user] soundfile player question

by derek holzer

Dear list, I've got a question about using various soundfile players with Mozilla Thunderbird. I've got a webpage on exhibit in a festival right now, with a selection of MP3s to listen to. I thought it would be a simple thing, but lusers always find ways of making simple things break ;-) The problem is that, each time a user clicks on a link, it opens a new instance of each soundplayer. Of course, most people don't bother to close the other player, so pretty soon there are too many instances and the sound just stops. I've tried XMMS and VLC in particular, and have resorted to using MPG321 because it will wait for the first soundfile to finish before playing the next one. But the users can't see the app playing the sound, and more than that, they can't interrupt one sound with another because there's no interface to stop playback. So what I'd like to know is which app will support enqueuing files for playback in a single instance? I've tried all the XMMS and Thunderbird options available to me and can't come up with the answer. And I can't stand around all day instructing people on how to keep the computer from crashing ;-) Suggestions welcome! best wishes, Derek current project: http://berlin.soundscape-fm.net -- derek holzer ::: http://www.umatic.nl ---Oblique Strategy # 36: "Consult other sources -promising -unpromising"

19 years, 3 months

[linux-audio-user] [ANN] QjackCtl 0.2.15 released!

by Rui Nuno Capela

Hi, QjackCtl 0.2.15 has been released. As a major new feature you are now allowed to rename (alias) the JACK/ALSA connections client/port names to something intelligible. Another nice one is about actual ALSA hardware device names which are now presented for selection as a pull-down menu on the setup dialog. Grab it from: http://qjackctl.sourceforge.net As taken from the change log: - JACK/ALSA client and port name aliasing (renaming) is now an optional feature for the connections window; all client/port aliases are saved on a per preset basis (as proposed for Lionstracs' Mediastation). - Server state now shown (back gain) on the system tray icon tooltip; speaking of which, tooltips are now also featured on connections, status and patchbay windows. - New actual hardware device selection menu featured on setup dialog; these new button menus are only available for the ALSA driver settings. - Server path factory default to jackd instead of jackstart; preset setup button icons are back. - Fixed rare connection port item removal/disconnection dangling pointer bug. Have fun. -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

19 years, 3 months

[linux-audio-user] New to list

by Noah Roberts

I just joined this list and figured I would introduce myself. I have been using Linux for ~10 years or more but never really for the purpose of audio work. I recently became reinterested in playing music and have since found that Linux audio has taken a few leaps since I started using it. I will probably be asking for help getting my midi keyboard working and such (I just bought the connectors and have never done this before), but for now I have already used Linux to record several "songs" with ardour, hydrogen, and jack. My soundclick site is http://www.soundclick.com/noahroberts/ I use Linux exclusively for everything but playing Tomb Raider and have for several years. Lately I have had to use Windows for printing out powertabs though; maybe that will change soon. -- Our enemies are innovative and resourceful - and so are we They never stop thinking about new ways to harm our country and our people - and neither do we. -- George W. Bush

19 years, 3 months

[linux-audio-user] the bandwidth of each harmonic

by Paul

...a simple, but powerfull paradigm that helps to make simple algorithms that produces very beautiful sounds. Hi. I want to open a topic about what I call "bandwidth of each harmonic", to say some ideas that helped me to make beautifull sounds. I am Paul, the author of ZynAddSubFX software synthsizer ( http://zynaddsubfx.sourceforge.net ) and I wish to share some ideas ;) Referring to musical sounds,usually the harmonics are considered simple sine function. Of course, the reality shows different thing, and so, the harmonics was started to be considered as sine functions modulated by a lower frequencies (I saw some time ago a link that describes that the choir sound is beautifull because of the "micromodulations",etc...). A very good thing is to "have a look" in a frequency domains. Let's take for example a choir that sing same note (like A=440Hz). Because all are human, even they are very well trained singer, they will not sing exactely the same note; for example one will sing at 435 Hz or other, at 443 Hz, and so on. Now, the first harmonic (the fundamental note) is not longer a sine of 440 Hz, but a narrow band signal that has a certain bandwith. For example, let's take a very simple case: if all sing at same loudness, and they sing from 435 Hz and the 445 Hz. In this case, the bandwidth will be 10 Hz. Of course, in real choir the frequency distribution of the harmonic will be not flat, but usually will be a curve that looks like a normal (gaussian) curve(found this after I did some research= very fine frequency analysis with very long FFTs). Now, let's go to the second harmonic: if you multiply 435 and 445 with 2, the difference will be 20 Hz. So, the bandwidth of the second harmonic is 20 Hz. So here a important rule of real instruments (esp. ensembles): the bandwith of each harmonic is proportional to the it's frequency; so in this example the first harmonic has bandwidth 10Hz, the second 20Hz, the third 30Hz, and so on. Here I made a very fine frequency analisys of a synthsized sound(a real orchestra sound will give similar results): [img]http://zynaddsubfx.sourceforge.net/doc/paul1.png[/img] You see, that the harmonic's bandwith increase according to their frequency. If you don't increase the bandwidth of higher harmonics, the resulting sound will be unpleasant especially on higher bandwiths of the first harmonic. So, not all quasi-periodic sounds are good, usually only if you increase the bandwidth of the harmonics. Now, what's happens if there are a lot of harmonics or the pitch of the sound is enough low? Let's see: [img]http://zynaddsubfx.sourceforge.net/doc/paul2.png[/img] The upper harmonics will merge to a single frequency band that will sound like a hiss, that is pleasent to the ear(eg. choir). I found that bandwidth of each harmonics can go even greater than 50 cents (quater of a tone) and still sound musical - I can give you some example wavs. Please note that this a different situation than 2 notes detuned by 50 cents (that sound very dissonand). Also, the best is that the phases are random. In real life this is ensured by the reverberation of the hall, for example if a instrument plays with a vibratto (a flute). Perhaps you noticed that always if more instruments plays the same note(as a enseble), the sound is very pleasant. I consider that this is the cause: the bandwidth of each harmonic. Unfortunately, there is very little on the internet about this stuff, because, I think, that are used more complicated paradigms (like some statistics on how sine harmonics modulates, or so). Now I want to give examples on how you can synthesize sounds, using this ideea. You can use zynaddsubfx or another synthesizer in most cases. Of course: I don't claim that I invented the bandwidth of each harmonic (or frequency distribution of each harmonic), because the ensembles, the choirs, the reverberations, exists since thousands of years (or more :) ). I just consider this a very important fact on how sounds are beautifull and it helped me to synthsize good sounds even since I was on highschool :D. You can make bandwidth of each harmonics by: 1) make more oscillators and detune them a bit (and a slight vibratto helps alot). Most synth use this and allows this, and perhaps it is known by you. This is one of the simplest method. This is implemented in ZynAddSubFX as "ADDsynth" module 2) generate white noise and filter each harmonics with a bandpass filter and mix the results. Be carefull to make higher harmonics to have higher bandwidth. This is implememnted in zynaddsubfx as "SUBsynth" module. 3) You see that above graphs. You can represent them as numbers in the frequency domain that represent the amplitudes of the frequencies, add random phases and do a SINGLE IFFT and voila! A very beautifull sound will "born". This new ideea is implemented in ZynAddSubFX as "PADsynth" 4) you can do other things, like a vibratto on a periodic oscillator and do a FFT to all sound, put random phases, and do a IFFT. For more information, I made few years ago a page that describes above things at: http://zynaddsubfx.sourceforge.net/doc_0.html Good luck. Paul __________________________________ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com

19 years, 3 months

[linux-audio-user] Florin - thanks for the pointers

by dwight

to the Linux Journal articles on MIDI. Just what the doctor ordered.... http://www.linuxjournal.com/article/7773 http://www.linuxjournal.com/article/7912 http://www.linuxjournal.com/article/7918 http://www.linuxjournal.com/article/8050 seasoned Win/MFC/C++ developer, musician. new to Linux. now have Mandrake 10.1 on two boxes. making some progress...

19 years, 3 months

Re: [linux-audio-user] Re: Decent reverb

by davidrclark＠earthlink.net

Bob, You recently posted: > I use the ' less is more ' when it comes to reverb! > Also using just one reverb as an Aux send and sending all tracks to it > with varying amounts really helps. This is what many hardware synths do for a "performance" or "program." All of the instruments go through the same reverb (and chorus). The amount of reverb can be varied for each channel. With care, this can be made to sound ~OK. Electronic instruments are recorded directly, so there are no room acoustics. Other instruments are recorded in rooms with noticeable acoustic properties. When these instruments sampled under different conditions are combined, the room acoustics are imbalanced. In order to correct this problem, one can add reverb to those instruments which have little to none while adding none to those instruments which were recorded in some sort of room. What is really being done here is that the user is attempting to recreate the room that some of those instruments were sampled in, which may sound OK. If one then adds additional nonphysical reverb to the mix, the sound begins to deviate from that of a real room --- so "less is more" --- that is, less sounds more like a real room. Any rooms that samples are recorded in is usually small, so normally there isn't much overall reverb with this approach. I can't say and am not saying that this is what you are doing, but it may at least partially explain why you say "less is more." Personaly, I don't like the sound that my hardware synths produce when I attempt to do what you described. Part of it has to do with the reverb, but it also has to do with the artificial stereo separation for most instruments. It never sounds good with headphones with this approach no matter what I do. Even those genres which use little reverb don't sound good. The room acoustic models, such as they are, and listening models are physics-deficient.

19 years, 3 months

Re: [linux-audio-user] Re: Decent reverb

by davidrclark＠earthlink.net

Hi Tim, > Over-the-top reverbs have their uses, however, they do have to be pleasing > to ear. Absolutely. I've created some electronic music tunes and appreciate this use of reverb. But it's better in my experience to listen to these over-the-top reverbs (which essentially create a new instrument from an existing one) in good rooms or alternatively with a good room acoustics model. When I refer to "reverb," I'm actually talking about room acoustics, including reverb. There is also a phenomenon referred to as binaural listening, which you may have heard of. In other posts, I've referred to this as "stereo separation." Many people familiar with binaural listening, including myself, will tell you that this is the best "stereo" or "surround sound" that they've ever heard, bar none. Although in the past, binaural recordings have not been successful, I suspect this is because at the time it was tried, headphones were very heavy and rather expensive on top of (not instead of) the cost of speakers. They may be commercially viable in the future with all the inexpensive, lightweight headphones around. One of the advantages of calculated impulse response functions is that binaural images are easily obtained from any monophonic recording --- and as accurately as you want. Regards, Dave.

19 years, 3 months

[linux-audio-user] Audigy2/emu10k1 ALSA JACK audio capture problem

by Rick Wright

Hi all, This is my first post to the list and I'm still pretty much a newb, so go gently... :-) I have an Audigy2 Platinum Pro ZS card that I would like to use for 6 channel input and output work. I am using JACK (0.99.49) via the QJackCtl (0.2.14) frontend. I've installed ALSA 1.0.8 with Lee's emu10k1 multichannel v0008. I am running a pretty much stock, up to date FC3 distro using kernel 2.6.10 with the realtime-lsm kernel module installed so I have realtime privileges for jackd. /proc/asound/devices: 4: [0- 0]: hardware dependent 9: [0- 1]: raw midi 8: [0- 0]: raw midi 19: [0- 3]: digital audio playback 18: [0- 2]: digital audio playback 26: [0- 2]: digital audio capture 25: [0- 1]: digital audio capture 16: [0- 0]: digital audio playback 24: [0- 0]: digital audio capture 0: [0- 0]: ctl 1: : sequencer 6: [0- 2]: hardware dependent 10: [0- 2]: raw midi 11: [0- 3]: raw midi 33: : timer All software seems to be working properly. My problem is that I cannot figure out how to configure my 6 desired capture channels so that I can read signals on the L & R channels of my 3 Line In inputs. I just think I have a configuration problem Any and all help greatly appreciated. Here is the PCM device definition from my .asoundrc file (which is a modification of the example file found on the included reference webpage): ################################################################################ # # 5.1 Channel Surround Sound # # Reference information: # http://alsa.opensrc.org/index.php?page=SurroundSound ctl.jack51 { type hw card 0 } pcm.jack51 { # "asym" allows for different handling of in/out devices type asym playback.pcm { # # route for mmap workaround type route slave.pcm surround51 # # Had to switch all L and R Channels to conform playback # # channels within JACK to standard 5.1 channel mapping. # # The trailing 1's indicate unity gain (valid values are 0.0-1.0) ttable.0.1 1 # routes 0 to 1 (playback_1 [0] to output channel 1 [1]) ttable.1.0 1 # routes 1 to 0 (playback_2 [1] to output channel 0 [0]) ttable.2.3 1 # routes 2 to 3 (playback_3 [2] to output channel 3 [3]) ttable.3.2 1 # routes 3 to 2 (playback_4 [3] to output channel 2 [2]) ttable.4.5 1 # routes 4 to 5 (playback_5 [4] to output channel 5 [5]) ttable.5.4 1 # routes 5 to 4 (playback_6 [5] to output channel 4 [4]) } capture.pcm { # # 2 channels only type hw card 0 } } ################################################################################ Using this "pcm.jack51" from within QJackCtl (equivalent to. jackd -R -dalsa -d jack51 -S) I get 2 Capture channels (which both appear to contain the R channel of Line In #1) and 6 playback channels (which are correct). BTW, I can use my jack51 PCM from within QJackCtl because I edited the "Interfaces" menu via the qjackctlrc file so that I have a "jack51" menu option. Once JACK is started, channel interconnectivity works correctly. How do I write the "capture.pcm" part of my .asoundrc file so that all 6 inputs become capture channels in JACK? I've done loads of reading on customizing a .asoundrc file and have tried many permutations in attemting to configure my Audigy2's 6 input channels, but no luck so far... There seems to be a good deal of information for configuring ALSA for *playing* audio, but not so much for *capturing* audio! Does anyone have any experience with this that I could draw on and would be willing to help? Other questions: Is this supported for my hardware yet? Any recommendations for multichannel (at least 6 in & out) cards that are currently well supported? Thanks in advance for any assistance. I can post more detailed info as needed. -Rick

19 years, 3 months

Re: [linux-audio-user] head phones processor

by davidrclark＠earthlink.net

A couple of days ago, Andrew asked the following: > Is there some kind of (software) head phones processor with the aim > to eliminate an impression a sound is inside a head? I wasn't sure whether he was asking about fixing existing material or avoiding the problem in the first place, so I asked him. He replied that he was interested in fixing existing material. He finds, as I do, that the "sound inside the head" problem tires him out as he listens. There are headphone amplifiers that help fix this problem by by signal processing means. There may be software, but I'm unaware of a any specific software. I have experimented with some techniques, but none were satisfactory compared to preventing the problem in the first place (for example, the engineered IR's I've produced or simply making binaural recordings at the start). Is anyone else interested in a software solution to this problem of fixing existing material? Does any software exist that emulates these binaural headphone amplifiers? If there is no Linux software yet interest, I may spend some time going back to working on this problem, time permitting. I would think that there should be a lot of interest, but apparently very few people in general are familiar with the improved sound of binaural audio images. Again, a fix is going to be far short of doing it right in the first place. It also colors the sound. Thanks for replies and regards to all, Dave.

19 years, 3 months

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Linux-audio-user February 2005