Linux-audio-dev

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7520 discussions

[linux-audio-dev] Jackd and cards with multiple PCM interfaces

by Jonathan Woithe

Hi all I am currently working on an ALSA driver enhancement for a card which actually contains two seemingly sample-locked ADCs. I'm making the second ADC accessable via a second PCM interface attached to the card. In general things work fine - under OSS emulaion I can open both (one as /dev/dsp, the other as /dev/adsp). However, jackd doesn't seem to know or care about the second pcm interface on the card and I can't find any obvious way to tell jack about it. It seems suitable entries in .asoundrc might help but it's not clear to me that this will work without xruns (see http://www.alsa-project.org/alsa-doc/doc-php/asoundrc.php). Can jackd work with multiple PCMs from the same card, and if so, what's the best way? Regards jonathan

20 years

[linux-audio-dev] playing ascii?

by Aaron

Hi I need help with a conceptual and practical problem. How do you get sync between- audio/midi and ascii text plus notation displayed in a graphical program written in gtk+2? Here is the problem: I have notation is lilypond (www.lilypond.org) in a text editor, either vim emacs jedit or a custom text editor if need be. the audio of that text as midi or wav lets say the same notation displayed Graphically in Denemo (www.denemo.sourceforge.net) The result needed is: 1. to add text in the editor and have them show up graphically, and to add notes graphically and have them show up as text. Also the cursors in the applications should stay in sync. 2. Playback: To have audio playback of the sound either in midi or wav where the cursor in Denemo and the text editor stay in synce, and scroll karoke style. 2a. Have a scrub feature where a specific measure or set of notes can be played either from the ascii text or the graphic notation in Denemo. ------------------------------------------- So first conceptually how can this be acheived? Can I use a preexisting text editor such as VIM, Emacs or Jedit or must I create my own to acheive this? What libs scripts apps would help ease the implementation of the above. Is this a reasonable feature to hope to implement? And if it isn't, how could it be done anyways. thanks Aaron

20 years

[linux-audio-dev] ALSA MIDI Sync

by Peter Brinkmann

Hi, I'm trying to figure out how to build an ALSA MIDI sequencer that'll sync with other sequencers. I've looked around but didn't see any recent information. (The closest hits were a five-year-old page of Takashi Iwai's, a three-year-old LAD thread with the subject line "MIDI sync and ALSA sequencer", and a LAU thread from last year, none of which sounded entirely encouraging.) So, my question is, what's the recommended way of synchronizing two MIDI sequencers these days, using ALSA? Is there any protocol (MTC, MMC...) that you would recommend so as to be compatible with a large number of other software packages? Any hints would be appreciated. Pointers to sample implementations of MIDI sync with ALSA would be great. Best, Peter

20 years

[linux-audio-dev] Merging digital sampling sessions

by David Kastrup

Hi, I have a minidisc player with S/PDIF output and an ALSA supported soundcard (details if the card are not really important for the problem at hand). I sample recordings via arecord -D spdif -f cd Ok, and here is the rub: playback is not perfect, and neither seems to be the sampling. Probably at times error interpolation sets in, probably at times samples get lost or replicated. Not much, perhaps once per minute or so. This leads to audible clicks, of course. Then there are leading and trailing zeros. Now what I want to do is just sample the stuff three times and do a three-way merge. Never mind clever click detecting and interpolation schemes: just use redundancy. I have not found a tool for that. Anybody have a clue how to go about that? If things come to worst, I'll have to write a merging tool. Bonus points would be if the thing worked on-line, taking one recording, sampling new samples and just writing the diff out. Run this three times, then run a merge program on the first complete sampling as well as all diffs. That would save quite a bit of disk space. However, it is not really mandatory: just merging three full size files would be fine. Anybody have an idea whether there is some suitable starting point or utility for this kind of thing? -- David Kastrup, Kriemhildstr. 15, 44793 Bochum

20 years

Re: [linux-audio-dev] dealing with complex numbers

by Taybin Rutkin

ISO C99 supports complex as a variable type and the GNU compiler supports that. Taybin -----Original Message----- From: Erik de Castro Lopo <mle+la(a)mega-nerd.com> Sent: Dec 13, 2005 4:05 PM To: The Linux Audio Developers' Mailing List <linux-audio-dev(a)music.columbia.edu> Subject: Re: [linux-audio-dev] dealing with complex numbers Artemiy Pavlov wrote: > Hey everybody! > > This may be a little bit off-topic, but can anyone suggest me any reading on > how to use complex numbers in C or C++? Is there any library or ++ classes > for such computations? In C++: #include <complex> which is a template clase. You can then do: typedef std::complex<double> dcomplex ; Erik -- +-----------------------------------------------------------+ Erik de Castro Lopo +-----------------------------------------------------------+ Open Source and Free Software means that you never sacrifice quality of the code for meeting deadlines set up by people not participating directly in the software development process.

20 years

Re: [linux-audio-dev] dealing with complex numbers

by Taybin Rutkin

And by 'nando, I meant Eric. :/ Taybin -----Original Message----- From: Taybin Rutkin <taybin(a)earthlink.net> Sent: Dec 13, 2005 4:30 PM To: The Linux Audio Developers' Mailing List <linux-audio-dev(a)music.columbia.edu>, The Linux Audio Developers' Mailing List <linux-audio-dev(a)music.columbia.edu> Subject: Re: [linux-audio-dev] dealing with complex numbers Sorry, meant to include this link: http://developer.apple.com/documentation/DeveloperTools/gcc-4.0.1/gcc/Compl… http://www.gnu.org/software/libc/manual/html_node/Complex-Numbers.html 'nando's C++ usage looked a bit cleaner to me, IMHO. Taybin -----Original Message----- From: Artemiy Pavlov <artemio(a)kdemail.net> Sent: Dec 13, 2005 3:22 PM To: The Linux Audio Developers' Mailing List <linux-audio-dev(a)music.columbia.edu> Subject: [linux-audio-dev] dealing with complex numbers Hey everybody! This may be a little bit off-topic, but can anyone suggest me any reading on how to use complex numbers in C or C++? Is there any library or ++ classes for such computations? I'd appreciate any help. Thanks! With respect, Artemiy.

20 years

Re: [linux-audio-dev] dealing with complex numbers

by Taybin Rutkin

Sorry, meant to include this link: http://developer.apple.com/documentation/DeveloperTools/gcc-4.0.1/gcc/Compl… http://www.gnu.org/software/libc/manual/html_node/Complex-Numbers.html 'nando's C++ usage looked a bit cleaner to me, IMHO. Taybin -----Original Message----- From: Artemiy Pavlov <artemio(a)kdemail.net> Sent: Dec 13, 2005 3:22 PM To: The Linux Audio Developers' Mailing List <linux-audio-dev(a)music.columbia.edu> Subject: [linux-audio-dev] dealing with complex numbers Hey everybody! This may be a little bit off-topic, but can anyone suggest me any reading on how to use complex numbers in C or C++? Is there any library or ++ classes for such computations? I'd appreciate any help. Thanks! With respect, Artemiy.

20 years

[linux-audio-dev] dealing with complex numbers

by Artemiy Pavlov

Hey everybody! This may be a little bit off-topic, but can anyone suggest me any reading on how to use complex numbers in C or C++? Is there any library or ++ classes for such computations? I'd appreciate any help. Thanks! With respect, Artemiy.

20 years

[linux-audio-dev] NPTL jack+ardour: large memlock required (was: Jack and NPTL (again?))

by Jonathan Woithe

Hi On Tues Nov 22 I wrote: > I've recently moved a system over to Slackware 10.2 which utilises NPTL and > I'm aware of the issues NPTL has raised in the past. Based on a comment on > the Jack website though I sort of assumed that things were now in hand and > that Jack had a workaround in place for the issue. > > Despite this I have found that jackd itself (when run using set_rtlimits) > gives an error (-11, EAGAIN) when "creating realtime thread". I have since discovered (thanks to strace) that the problem lies in an mmap2() call which requests a size of around 8MB. It appears to be part of the NPTL pthread_create() function. The error returned by mmap2() (EAGAIN) indicates either a locked file or "too much locked memory" (according to the manpage). Because this is an anonymous map the problem must have been the latter. Increasing the "locked memory" limit from 20MB to 40MB made jackd start without having to resort to the LD_ASSUME_KERNEL hack. Another problem then surfaced: with jackd running under NPTL, jack applications using NPTL (eg: ardour) suddenly started to require all of their memory to be lockable. For ardour with a large session this can get quite large (I saw 130MB at one point last night). So, with jackd running with LD_ASSUME_KERNEL = 2.4.19 and a memlock limit of 20MB everything worked and ardour (with NPTL - ie: without LD_ASSUME_KERNEL) was obviously not trying to lock excessive amounts of memory. With jackd running with NPTL and a large enough memlock limit things work but jack clients appear to require much more memory be locked. One thing I will try next is recompiling ardour; perhaps there's something funny there. In any case though, does any of this ring a bell with anyone? Regards jonathan

20 years

Re: Re: [linux-audio-dev] Merging digital sampling sessions

by Jonathan Woithe

David Kastrup writes: > On another note: it would seem like a simple enough task to let the > recording start with the first non-zero sample. ... I'd also like to prune > trailing zeros at some point of time. Although it is not a native ALSA app (due mainly to me not having the time to port it yet), batchrec does this. It still works with ALSA through the OSS emulation layer; adding native ALSA support is on my todo list for the next few months. You can grab it from http://www.physics.adelaide.edu.au/~jwoithe/batchrec-1.2.0.tgz It's pretty raw (no configure script yet) but shouldn't be too hard to compile on most systems. Batchrec can be requested to wait for signal above a certain amplitude before it records. It will also stop recording after the same threshold has been present for a user-specified time (if desired); at that point recording will stop, the file will be closed and the program will then wait for signal above the threshold again. I've used it extensively with analog inputs (hense the use of a threshold rather than exactly 0). The only fly in the ointment will be whether the SPDIF input on your soundcard can be mapped in the OSS emulation layer in such a way that OSS applications can use it. Regards jonathan

20 years

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Linux-audio-dev