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 ...?

20 years, 2 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 … [View More]

20 years, 2 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 … [View More]

20 years, 2 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 "… [View More]

20 years, 2 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 … [View More]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 [View Less]

20 years, 2 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...

20 years, 2 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 … [View More]

20 years, 2 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 … [View More]

20 years, 2 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. … [View More]/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 [View Less]

20 years, 2 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 … [View More]

20 years, 2 months

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Linux-audio-user February 2005