Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7439 discussions

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

21 years, 8 months

[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

21 years, 8 months

[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

21 years, 8 months

[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

21 years, 8 months

[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.

21 years, 8 months

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

21 years, 8 months

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 ---

21 years, 8 months

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 ---

21 years, 8 months

[linux-audio-dev] [Fwd: ams segmentation fault]

by Eric Dantan Rzewnicki

I don't know whether this is a bug in ams, or a problem with my system...I suspect the later, but can't figure out what's wrong. But, I am unable to get ams to run. It appears to compile OK. But, as noted below, running it results in a "Segmentation fault". I have installed qt3 and fftw. I edited the top of make_ams to match my system: box1:~/audio_code/ams-1.5.5$ head -5 make_ams QT_BASE_DIR=/usr/lib/qt3 QT_LIB_DIR=$(QT_BASE_DIR)/plugins-mt/styles QT_BIN_DIR=/usr/bin QT_INCLUDE_DIR=/usr/include/qt X11_LIB_DIR=/usr/X11R6/lib My ditribution is Debian stable (woody). My system is a PII 400Mhz w/768MB RAM. The version of g++ is 2.95.4. I have a working install of jackd 0.44.0 from cvs. I'm starting jackd like so: # jackd -v -R -d alsa -d ymfpci -r 44100 -zt -n 3 -p 512 I can start up multiple instances of jack_metro, connect them to the alsa ports and hear the beeps. So, jack appears to be working. But, if I start ams nothing is reported in the jackd output about a new client. ams does attempt to connect to the X server when started. I know this because if I 'su -' to get to root and then start ams: box1:/home/eric/audio_code/ams-1.5.5# ./ams ams: cannot connect to X server but, if I su without the dash so that root inherits my user environment ams gets past the point of connecting to the X server (assuming that comes first) and exits with a segmentation fault. I'm not sure what other information would be useful to you in helping me to sort this out. If there is anything else I should tell you, please let me know. Thanks in advance, Eric Rz. -------- Original Message -------- Subject: ams segmentation fault Date: Sat, 28 Dec 2002 21:47:34 -0500 From: Eric Dantan Rzewnicki <eric(a)zhevny.com> Organization: zhevnycom To: Eric Dantan Rzewnicki <rzewnickie(a)rfa.org> CC: linux-audio-user(a)music.columbia.edu References: <3E0BCBEC.9DB82093(a)zhevny.com> <20021227103148.GH681(a)ecs.soton.ac.uk> <1041005858.4921.17.camel@eviltwin> <20021227181307.GD19938(a)rfa.org> Eric Dantan Rzewnicki wrote: > I can now start jackd as root and > then start ams -j. Sorry I didn't think of this earlier. :-\ > > Adding /usr/local/lib to /etc/ld.so.conf and running ldconfig as Steve > suggested allows ams to find libjack.so.0 (I really should have > remembered that piece, too). > > Now that I have it running I'll go finish reading the ams documentation > and figure out how to use it. :) hmm... I perhaps spoke too soon. I had ams working yesterday on one box (at the office)... But, now I can't seem to get it working at home. It seems to compile fine, but when i try to start it: box1:/home/eric/audio_code/ams-1.5.5# ./ams Segmentation fault box1:/home/eric/audio_code/ams-1.5.5# ./ams -j Segmentation fault It doesn't matter if jackd is running or not. Either way it seg faults... What could cause this? What information should I provide to help sort this out? Thanks, Eric Rz.

21 years, 8 months

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

by Anders Torger

I have made a program which is to be run on a dedicated computer, equipped only with a multichannel soundcard (hammerfall), a small LCD and a remote control. This forms a dedicated surround processor (if interested in the software, look at http://www.ludd.luth.se/users/torger/almusvcu.html) for use in multichannel HiFi systems. In some configurations, a 64 sample processing block is relevant, leaving a total IO-delay of 128 samples, which in 44.1 kHz is about 3 ms. With such a configuration, I have got the system going for a few hours at most before underflow. Of course the relevant processes are realtime scheduled, I have even used tricks with sched_yield to get the processes yield in the right order (I will probably have to pay for that sooner or later though) for the most safe operation. So, my question to this list is if someone has run a similar system for days without underflows, and if there is some guide or tips on how to put together the for the moment best low-latency kernel, if there are kernel features that should be avoided, some bad driver or filesystem perhaps, and if there are some hardware setting tricks (apart from giving the sound card the highest interrupt priority). Perhaps tips on which user space processes that should not run (I have no syslogd or cron, but sshd is running) Is it even possible today to build an embedded system with 1.5 ms processing block based on Linux, which runs until hardware or power failure? Oh, my software does not necessarily need to access any filesystem while it is processing, although it would be good if it could (to continuously save volume settings and similar if changed through the remote control). So it can be seen as a strict soundcard-to-the-cpu-and-back problem. /Anders Torger

21 years, 8 months

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Linux-audio-dev