Linux-audio-dev January 2003

linux-audio-dev@lists.linuxaudio.org

99 participants
111 discussions

by Kay Pruefer

Hi, Cow Outputs Waves is a waveform editor. You draw graphs for amplitude and frequency, cow synthesizes a wavfile. The GUI uses Qt. The package includes also a program to play the cow-sounds and also wavfiles with a midi-keyboard (or any other thing that can output note on/off commands) and several tools. You might be interested in this program if you are using a tracker and need cool sounds or intent to play accords and find it ugly that the wavfiles are getting shorter or longer depending on the tone pitch. Hrm. Maybe youre also interested if you dont use a tracker :) http://kuh.sourceforge.net for pre-compiled binarys for Mandrake, Redhat and SuSE have a look at: http://apps.kde.com/na/2/info/id/1457 HAND, Kay.

23 years, 1 month

[linux-audio-dev] rave + rant

by Paul Davis

i just want to stop for a moment and reflect on the power of the open source software model. for a long time, a rather glaring defect in ardour was the inability to record at a higher frame rate (say, 48kHz or 96kHz) and then easily produce an audio file of the piece at 44.1kHz, the standard for redbook cd audio. i avoided trying to add the code because i knew that resampling was complex and had some significant depth to it - i was busy enough with other things. then erik comes along with his libsamplerate library, and it took me less than 15 minutes to add the capability to the backend of ardour (plus another 45 minutes of work on the GUI to control it). now, The Unix Way calls for the use of tools like sox to do this work: small, independent applications that do one thing and do it well. Ok, that's not exactly a good description of sox, but you get the point. There is nothing wrong with this model for some purposes, especially for people who want to play around with many different possibilities. but when rob pike wrote "the unix programming environment", he was clear (at least to me) that the main strength of The Unix Way was in providing a really productive environment for *prototyping*. it turns out that its a really nice environment for getting many kinds of real work done too, especially for many experimental music folks, and i wouldn't want to see the end of the toolset that sox is just one aspect of. but ... but ... i am just glowing with the way libraries like libsamplerate and libsndfile provide the same simple "just plug it together" functionality that the unix shell and all our pipe-connected utilities do. this time, its not to being offered (directly) to command line users, but to people writing GUI-based software and thus ultimately to their users. it really makes me feel good to be able to turn around and explain to my disbelieving kids "yes, somebody wrote this really useful software and they just made it available so that other people could use it. that made it easy for me to make my software (with which i do the same) be so much more useful for many people". thanks to erik, thanks to RMS for the GPL and thanks to everybody on this list and elsewhere who is making this revolution possible. --p

23 years, 1 month

[linux-audio-dev] Compilers

by Levi Burton

Greetings, I was curious what compiler versions people have had the best luck with. I am currently using: gcc version 2.96 20000731 (Red Hat Linux 7.3 2.96-113) I am considering upgrading via RPM to what appears to be 3.2 (gcc-3.2-7.i386.rpm). Good idea? Bad Idea? Thanks! Levi

23 years, 1 month

[linux-audio-dev] [ANN] ALSA Patch Bay 0.4.1

by node＠users.sourceforge.net

Hi all, Another day.. another release :) * fixed plugin stuff to work with 2.9x C++ symbols * fixed fltk compile * added an alsa-patch-bay.desktop * fixed fltk segfault http://pkl.net/~node/alsa-patch-bay.html Bob

23 years, 1 month

[linux-audio-dev] LADSPA proposal for named values

by Pascal Haakmat

Hi all, It appears as though LADSPA at the moment lacks a means to provide names for control values (this topic has probably crossed the list before so I'll try to keep it short). The common solution for this problem (e.g. as used in Steve Harris' plugin collection) is to use an integer control port to encode for the different options that the port can have, then add some text to the PortName to describe those options (e.g. "1=sin, 2=saw, ..."). This has a few drawbacks though. First, it doesn't really solve the original problem; the user is still inputting ordered numeric values, even though the relationship that is being modeled can be totally arbitrary. Second, it makes for very long PortNames, which causes layout troubles. Therefore (if such a thing does not already exist) I propose to extend the LADSPA specification with a convention: if a PortName matches the following regular expression: .+ \((\d+=.+,)*(\d+=.+)+\) (in words: any sequence of characters following by a '(', followed by one or more comma-separated 'value'='name' pairs, followed by a ')'), then the host may assume that the given values map onto the given names and display them as such. In addition, the host may suppress displaying the '(...)' part of the PortName. This proposal follows current best practices, requires no modifications to the LADSPA protocol or data structures, is easy to implement, and highly unlikely to adversely affect any existing plugins/hosts. Any comments? Pascal. -- The GNOME U sound editor: http://awacs.dhs.org/software/gnusound

23 years, 1 month

[linux-audio-dev] 2.4.20-rc1 + lowlat + preempt + alsa + jack = dead computer

by Fernando Pablo Lopez-Lezcano

[sorry for the wide distribution] Is anyone out there trying out this combination?: kernel 2.4.20-rc1 capabilities patch low latency patch preemptible kernel patch almost current alsa cvs [20021028.170432 usa pacific time] If I start jackd and then use the freqtweak jack client I get a completely dead machine in a very short time (from a few seconds to 10 or 20 seconds). Same alsa driver, but running on 2.4.19 (up to 20-pre4) seems to be fine and I can play around with freqtweak for as long as I want :-) No clues are left behind... something in the kernel is deadlocking, I guess. I tried removing first the preemptible kernel patch with the same result. Just a moment ago I tried again after removing the low latency patch and I still get the same result... that pretty much leaves alsa and the kernel. Got the same result on a laptop intel810 and an ice1712 card. Interestingly enough just the jack server running is not enough to kill the machine. Adding a jack client (just tried alsaplayer, same problem as freqtweak) kills the machine really fast. Maybe there is a problem with the scheduler when there are several SCHED_FIFO tasks competing for the processor? -- Fernando

23 years, 1 month

[linux-audio-dev] API for playing wav file samples

by Vandana

Hi, I am building a sound editor and need an API for playing 'wav' file samples. I tried the SOX API but the quality is very bad. Is there some other API, I can try. Kindly advice.

23 years, 1 month

Re: [Jackit-devel] Re: [linux-audio-dev] 2.4.20 + lowlat +preempt + alsa + jack = dead computer

by Fernando Pablo Lopez-Lezcano

> >I browsed the Kernel Source and there is only one mark_inode_dirty in > >pipe_write (in fs/pipe.c). So we know where it is hanging... > > > >And in __mark_inode_dirty (in fs/inode.c) there is one > > spin_lock(&inode_lock) > >call, and I guess that is where the whole thing is hanging. So something > >is holding that lock... how do I find out who is doing that? Apparently > >the handling of inode_lock is confined to inode.c. I'll keep reading. > > > >Maybe the pipe in question is one of the pipes that jack uses for ipc? > > seems *damn* likely ... sorry to just be chiming in with a useless comment! Do you think the problem might be jack? Some race condition that only happens (or is much more likely to happen) in 2.4.20? It does sound more like a kernel bug, but I have not seen a hung machine without having jack and clients running... -- Fernando

23 years, 1 month

RE: [linux-audio-dev] 1.5 ms latency possible?

by David Seymour

Actually what I'm talking about is simply monitoring. Tons of people have trouble monitoring when recording digitally simply becayse that can't monitor off of the A/D in real time. The RME DSP and the Lynx cards take care of this with a seperate digital router that basically takes the digital signal off of the A/D and shoots it right back to the D/A to send out to the console or headphone system for monitoring. You can then toss on any FX you desire. I honestly set up a distant mic for my reverb stuff and put digiverb on it if the singer needs it. David Seymour Benchmark Media Systems 800-262-4675 www.benchmarkmedia.com -----Original Message----- From: David Olofson [mailto:david@olofson.net] Sent: Thursday, January 02, 2003 10:25 AM To: linux-audio-dev(a)music.columbia.edu Subject: Re: [linux-audio-dev] 1.5 ms latency possible? On Thursday 02 January 2003 16.21, David Seymour wrote: > Would it be safe to say that trying to lower latency within the > system for monitoring purposes is simply a waste of horsepower > given the availablilty of I/O cards with routing DSP build into > them? I know LYNX and RME have cards that will do this. Depends on what you mean by monitoring. These audio interfaces generally have no DSP "effects" whatsoever, so as soon as you need basic EQ or whatever, you're out of luck. (AFAIK, this goes for the the cards you mentioned, as well as all Echo cards, and all Envy24 based cards.) So, if you're using external hardware for all effects, you're probably fine with ZLM. The audio interface will basically work like a patch panel in your analog system. If you do everything virtual, you can't even set up basic monitor sound, since that usually requires at least a simple reverb. (Dry signals and headphones don't mix...) //David Olofson - Programmer, Composer, Open Source Advocate .- The Return of Audiality! --------------------------------. | Free/Open Source Audio Engine for use in Games or Studio. | | RT and off-line synth. Scripting. Sample accurate timing. | `---------------------------> http://olofson.net/audiality -' --- http://olofson.net --- http://www.reologica.se ---

23 years, 1 month

RE: [linux-audio-dev] 1.5 ms latency possible?

by David Seymour

Would it be safe to say that trying to lower latency within the system for monitoring purposes is simply a waste of horsepower given the availablilty of I/O cards with routing DSP build into them? I know LYNX and RME have cards that will do this. David Seymour Benchmark Media Systems 800-262-4675 www.benchmarkmedia.com -----Original Message----- From: David Olofson [mailto:david@olofson.net] Sent: Tuesday, December 31, 2002 5:35 PM To: linux-audio-dev(a)music.columbia.edu Subject: Re: [linux-audio-dev] 1.5 ms latency possible? On Tuesday 31 December 2002 23.24, Anders Torger wrote: [...] > I've thought about RTLinux actually, but it is simply too much > work, porting drivers to start with... You can probably get away with using normal drivers (OSS or ALSA) in mmap() mode, and hooking the interrupt with an RTLinux or RTAI ISR. Start up the I/O from user space as usual, and then divide the IRQ period into the number of fragments you need, using an RTL or RTAI periodic thread. Just keep in mind that PCI cards generally have a 64 byte PCI burst size limit, so going below that won't work. //David Olofson - Programmer, Composer, Open Source Advocate .- The Return of Audiality! --------------------------------. | Free/Open Source Audio Engine for use in Games or Studio. | | RT and off-line synth. Scripting. Sample accurate timing. | `---------------------------> http://olofson.net/audiality -' --- http://olofson.net --- http://www.reologica.se ---

23 years, 1 month

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Linux-audio-dev January 2003