Linux-audio-user August 2004

linux-audio-user@lists.linuxaudio.org

148 participants
163 discussions

Re: [linux-audio-user] some thoughts about Linux audio software documentation

by jjbenham＠chicagoguitar.com

> -----Original Message----- > From: Dave Phillips [mailto:dlphilp@bright.net] > Sent: Thursday, August 12, 2004 01:13 PM > To: 'LAU Mail' I always wanted to write documentation for audio projects. The problem is I cannot figure out how to do that. I don't know how to learn a certain aspect of a program without reading through the source code. jcsound is an example. It compiles smoothly. I run it with the orc and sco included. I have been reading through the source for days trying to figure out what is going wrong. I am learning more alot about c programming and the csound api in the process. For me at the moment jcsound will load csound as an alsa seq client. I write to the port using aplaymidi --port 128:0 and I connect the jack client to alsa_playblack blah blah. I just don't get any sound. I experienced the same kind of thing when I was trying to learn how to apply the ladspa plugins. The only one I was able to get to work with applypluggin was the one you documented in the orielly article Delay_5s. Once I get more experience at c++ I will check it out again. I noticed most ladspa plugins were written in c++. Of course all these plugins work in a host program like snd or ardour etc ... Is there any other way to learn how to use a program other than just reading through all the code? I could ask the developers to explain it to me but after that the might as well have wrote the docs themselves. Jeremiah

19 years, 8 months

Re: [linux-audio-user] New install issues (Was: Trouble compiling JACK 0.98.1)

by Sampo Savolainen

The ice1712 chip is used in a huge amount of audio hardware. The chip always works in a certain way, regardless of which of it's abilities are used by the card it is soldered to. I use the same configuration in asoundrc for my delta 44, and I see 10/12 channels in jackd. Of course, there isn't any point in using channels beyond the 4+4 channels available in the break out box. Also, that asoundrc entry isn't really needed. You can use hw:0 (replace 0 with the card number of your 1010). If you have more specific needs, then you need an asoundrc. I have in my asoundrc entries "channel1" and "channel2" which represent two stereo channels I can use with non hw interfacing alsa programs. With jackd, you really shouldn't have to care. Just use hw:X Sampo Quoting robin fell <robin.fell(a)ntlworld.com>: > On Mon, 2004-08-16 at 04:51, Russell Hanaghan wrote: > > On Sun, 2004-08-15 at 20:22, Jos Laake wrote: > > > My '.asoundrc' file looks like this: > > > ----- > > > pcm.ice1712 { > > > type hw > > > card 0 > > > } > > > > > > ctl.ice1712 { > > > type hw > > > card 0 > > > } > > > Go to the alsa pages and locate the page for the ice1712 chip. Down > the > > bottom there is an .asoundrc file posted for a "hoontech" card. Create > a > > new .asoundrc from that post. When you run Qjackctl, set the capture > and > > playback channels to "0". See what happens. > > Out of curiosity, I followed the ALSA card matrix for Hoontech to a page > which has an _identical_ .asoundrc to that posted by Jos. > > Is this not the correct one? Do you have a URL, or if small enough > could it be posted? > > If you'd like, I'll accept the .asoundrc and any instructions/notes you > have and add them to the ALSA wiki for the Delta1010. > > cheers > R > >

19 years, 8 months

[linux-audio-user] [ANN] PHAT version 0.2.0

by Pete Bessman

Once again, I'm cross posting this to the users list in case anybody is interested in fooling around with the demo apps. I'm pleased to announce the release of PHAT, the Phat Audio Toolkit, version 0.2.0. PHAT is a collection of GTK+ widgets that may prove useful to audio applications. New to this release is a shameless ripoff of Blender's "sliderbutton" widgets. I like 'em more than regular GtkSpinButtons. Also, there have been sundry improvements to the fansliders, and all widgets are now mousewheelable and focusable, and they respond to sensitivity changes (i.e. they're well behaved). After installing, you'll have two demo apps at your disposal: phatsliderbutton phatfanslider (A few) more details are available at www.gazuga.net/phat.php Enjoy. -- Pete <http://www.gazuga.net> "Nothing great was ever achieved by being realistic!" --Tom Venuto

19 years, 8 months

[linux-audio-user] XMMS will not play off website through jack.

by David Baron

Complains about the sample rate. I am set to -r44100 which is best the sound card will do. The error message says XMMS can compensate. If I turn of jack and select the regular alsa output plugin, plays just fine.

19 years, 8 months

[linux-audio-user] Re: linux-audio-user Digest, Vol 11, Issue 71

by David L. Craig

On Sat, Aug 14, 2004 at 10:05:19AM -0400, linux-audio-user-request(a)music.columbia.edu wrote: > > I'm considering purchasing one of these cards but would like to know > if anyone here uses one. I know that some of you use the 1010 but this > one is the LT version. Any comments ? I can get a good price via eBay, > and I really need a pro-audio card, so all comments will be much > appreciated. > I picked up a full 1010 on clearance for $400 at Guitar Center a month or so ago. The salesman thought a new, improved version was imminent. You might want to shop the 1010. -- May the LORD God bless you exceedingly abundantly! Dave Craig - - - - - - - - - - - - - - - - - - - - "'So the universe is not quite as you thought it was. You'd better rearrange your beliefs, then. Because you certainly can't rearrange the universe.'" --from _Nightfall_ by Asimov/Silverberg

19 years, 8 months

Re: [linux-audio-user] Green's Function in Audio --- Demo Song

by davidrclark＠earthlink.net

Hi Erik, > I'm curious, is there anything that your downsampler does that couldn't > be done with SecretRabbitCode..." It's actually a resampler, not merely a downsampler. I haven't read your code, nor have I used it other than in programs written by others which utilize your library, so I don't know. (And I **THANK YOU** for writing that library --- many use it.) It was easier for me to write my own than to try and determine whether or not your resampler was suitable for my particular situation. However: >From your web page, it *appears* that you are using the sinc method developed by Julius Smith of Stanford. Smith's method is very fast, but not as accurate as FFT/overlap with large windows. Without Kaiser windowing, it wouldn't have seen the light of day due to the truncation effects. I need something that preserves the phase and other information as accurately as possible between the channels, not a small-windows approximation. I need a guarantee of accuracy, and I simply didn't have the time to fully investigate the sinc method with Kaiser and other windows. I should also say that the FFT-overlap method is entirely *inappropriate* for other situations, especially where the sample rate changes a lot and/or very quickly. Here the sinc method is *very good* to *excellent*. (Actually I have my own variant of that too, which is why I chose not to use it for my own straightforward, fixed 24/96 to 16/44.1 conversions.) Why I wrote my *own* FFT-overlap resampler: A feature of all programs I write using my own class libraries is that they automatically understand stdin/out, sockets, FIFO's, and files. One program can accept data with a WAV-style header from any/all of these IPC methods, and the same program (all programs) can put out WAV data using any of these methods. So all these programs are scriptable rather than part of a library. That is, the resampler can be put into any script (bash, ksh, Python, PERL, awk) that uses those IPC mechanisms, connecting it easily with any other program that was written in the same manner or with standard UNIX utilities including the "lowly" head, tail, grep, etc. I developed this technique to allow me to rapidly create prototype programs, the downsampler script being but one of many such scripts that utilize programs that communicate via these IPC mechanisms. This technique also allows me to execute "pipelines" from the command line, which is the origin of many of my scripts. That was probably a longer answer than you anticipated, but I hope I answered your question. Again, I thank you for writing that library. I'm sure many programmers would have had a tough time without it. Again, the sinc method is really the only way to go for many (most?) programs that require resampling. Best regards, Dave.

19 years, 8 months

[linux-audio-user] question about M-Audio 1010 LT

by Dave Phillips

Greetings: I'm considering purchasing one of these cards but would like to know if anyone here uses one. I know that some of you use the 1010 but this one is the LT version. Any comments ? I can get a good price via eBay, and I really need a pro-audio card, so all comments will be much appreciated. Best, dp

19 years, 8 months

[linux-audio-user] Green's Function in Audio --- Demo Song

by davidrclark＠earthlink.net

Hello everyone. Rather than participate in the recent interesting discussions, I decided to record/process a song and fix my software, especially since my favorite window manager is none at all. I prefer running scripts while having a beer then taking a nap or watching a tape of Joe Shmoe 2. (I've never cried so much from laughing, but I'm easily amused.) And rather than another demo, I decided to post a complete song which utilizes the Green's function technique I have mentioned previously. I'd like to say that this song was created exclusively with Linux, or perhaps more appropriately, GNU audio software, but I used Windows sequencers. I used my own command-line software to process the mix, created from monophonic recordings of each musical part or sub-part. I then applied JAMin to the final mix, using another piece of my own software, jackd, and qarecord to capture the result. I then downsampled it with my own overlapping FFT program, then used lame to produce the mp3 file. I think that's all, but I could easily have left something out. As a matter of fact, I had to fix some bug in my stuff with "head --bytes=...." http://home.earthlink.net/~davidrclark/latest_mp3.html ---------------------------------- For the person who may have read the Green's function notes without falling asleep: The Green's function technique that was used was for a *point* source in three dimensions in a very large room (~12,000 m^3) except for the overhead kick drum for which I used a *volumetric* source. In the simulated concert hall, the overhead location is very close to the center of the room in one axis, so the timing for echoes is bad. Using the volumetric source diffuses this effect a little, just as rough surfaces diffuse sound in a concert hall (otherwise you'd hear it in a concert hall, also). This concert hall has about 3 billion modes and they're all represented. ---------------------------------- Regarding the piece itself, I've noticed that people like to comment about the actual music. Well, this is intended to be a demo, so it's "very-high-quality elevator music." The sound sources are a Korg N264 and a Roland XV-3080. My focus is on the mathematical physics first, then on the application (reverb, stereo separation, echo, filtering, etc.), then on the music --- until it's finished. Then I like to just listen.... This is the only area that I've ever worked in where I am my own customer, and it's really nice to have it that way. Damn nice. Thanks very much to the developers of GNU software, the Linux kernel people, and the developers of JAMin, jackd, qarecord, and lame. And a special thanks to the developer of the "head --bytes=..." utility. Regards to all, Dave.

19 years, 8 months

[linux-audio-user] 2.6 kernel, Alsa, and a good /etc/modprobe.conf file

by Rick B

Hi all, I need the /etc/modprobe.conf file from a Fedora Core 2 system that has Alsa working correctly on the 2.6 kernel. If someone could just copy and post theirs it would be much appreciated. Oddly enough there's some on the internet that are broken, but I couldn't find one example of one that did work. I'm trying to help someone out with their new install, but I still use 2.4.26. Thanks in advance. Rick B

19 years, 8 months

[linux-audio-user] I'm tired of being a moron

by Mark Wilson

I'm relatively new to the audio on Linux scene, and admittedly don't have near the best equipment to do anything serious, but I'd at least like to hear some sounds of out of the synth and/or MIDI sequencing software on my Agnula distro. I don't have an external keyboard, so I'm limited to writing the score with rosegarden or lilypond. I'd like to edit the MIDI sequences I produce with something like muse, but I can't figure out how to hear what changes I make *from within muse*. I don't have an on-board sequencer (/dev/sequencer is never recognized or accessible, which is no surprise, really) on my card. I thought I'd try using timidity with the -iA option, and when I do that, then try to play a midi file from within muse, muse crashes. I just wanna' figure out how to do two things: 1) tweak my midi sequences for timing, rubato, expression, using somethink like muse or rosegarden, and be able to hear what I've done by clicking the play button(s) provided 2) create my own timbres using some of the cool-looking synth software, that I can't figure out how to hear whether I've made any changes or created any new patches. The subject line says it all, really. ;-) Thanks for any help you can give the clueless... Mark ===== -- Seek professional help! Ask a librarian. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

19 years, 8 months

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Linux-audio-user August 2004