Linux-audio-dev

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7447 discussions

by Egor Sanin

Hello LACers! I've got myself into a mixup. I've booked a hostel in Dublin for the Conference, but now I want to get out to Maynooth. I've been in touch with Kilcar B&B and Annette has me covered for Friday night, however, after that she's full. I remember someone here was asking about teaming up for a place, so perhaps there's a chance someone over there is willing to split a room and costs. If you don't mind a roommate, please email or call me. I'm in Dublin right now at Avalon house. I've got a mobile here: 085 7836111. I will be walking around a lot today, so phone is better. Thanks gang, I'm looking forward to meeting everyone!

13 years, 5 months

Re: [LAD] Linux-audio-dev Digest, Vol 51, Issue 5

by Stéphane Letz

> > > Message: 14 > Date: Wed, 4 May 2011 15:10:55 +0000 > From: Fons Adriaensen <fons(a)linuxaudio.org> > Subject: Re: [LAD] Looping audio to oneself via Jack > To: linux-audio-dev(a)lists.linuxaudio.org > Message-ID: <20110504151055.GE10034(a)linuxaudio.org> > Content-Type: text/plain; charset=us-ascii > > On Wed, May 04, 2011 at 04:04:37PM +0100, Rui Nuno Capela wrote: > >> 2. client3 loopbacks to itself and then it all applies as "the bug". >> question is: is it jack's bug? maybe not. the other half/part of the >> problem still aplies, as it depends on the client3's process code >> flow--which buffer port is read/written first? ins or outs? and in what >> order? hmm... > > If one signal (the looped back one) cas disappears because you > disconnect _another_ one, I'd call that a bug. 100%. > > Note that fixing this does not imply you can't get your own > undelayed output anymore. That just depends on the order of > your computations and jack_port_get_buffer() calls. > > Ciao, > > -- > FA > I suggest three things : - move this discussion on jack-dev list... - write a test client to show the problem. - test it with jack1 and jack2 to see what happens. Stéphane

13 years, 5 months

Re: [LAD] [LAU] FW: Frequency Space Editors in Linux

by Kjetil S. Matheussen

Jeremy Salwen: > > I just recently discovered ceres3: > http://www.music.columbia.edu/~stanko/About_Ceres3.html > > have you heard of it? > Ceres is larger and more updated than Ceres3. http://archive.notam02.no/arkiv/src/ceres-0.57.tar.gz (Ceres3 has a few more options for some of the transforms, but lacks lots of other stuff.)

13 years, 5 months

[LAA] Release of aj-snapshot-0.9.3

by Lieven Moors

Hi everyone, This is a small update to aj-snapshot. Aj-snapshot is a small command-line utility that can be used to store/restore ALSA and JACK connections to/from an XML file. This release fixes a small and a big bug: - Correction to the license headers which referred to the wrong program... (copy/paste error) - Make the -i flag ignore clients in both directions. (in the previous release, only connections FROM the ignored clients were ignored) The website: http://aj-snapshot.sourceforge.net/ Git repository: git clone git://aj-snapshot.git.sourceforge.net/gitroot/aj-snapshot/aj-snapshot Please report bugs on the sourceforge project page (don't be shy ;-) You will find direct links for bug reports or feature requests on the main website. greetings, lievenmoors

13 years, 5 months

Strange gettimeofday behaviour - out of order times.

by Tim E. Real

I discovered gettimeofday occasionally gives me a time value which is less than a previous time value. The value is actually 'out of order' and really belongs ahead of a few others, according to examined printouts. So I tried clock_gettime(CLOCK_MONOTONIC, ..), same result. Then I tried clock_gettime(CLOCK_PROCESS_CPUTIME_ID, ..) and it appears to be OK so far, giving me what I want - truly linear time. Still checking some things, maybe there's a reason, the app's fault. From what I've read gettimeofday is thread safe, but the incorrect time values are being read by the same thread always. Any advice here, anyone seen this? Also, when using Jack, is it advisable to choose the same clock method as Jack is using (cycle, hpet, system)? Thanks. Tim.

13 years, 5 months

Looping audio to oneself via Jack

by Fred Gleason

Howdy Folks: Here's a seemingly trivial requirement that has me stymied: I have a Jack client (external, built against 0.102.20) where it occasionally makes sense to connect an output port to an input on the same client. QJackCtl happily makes the connection; the process callback delivers data as expected, but it's all zeros for the connected input port! I can route the same client output port to a physical output (ALSA backend) and hear the expected audio. Likewise, a physical capture port connected to the input on the client receives data from the card as expected. It's only the "loopback" within the same client that doesn't want to go. At first I suspected that the JackPortIsTerminal flag might have some bearing on this, but changing this value seems to have no effect. Am I missing something blatantly obvious here, or is this something that is just not within the scope of Jack to allow? Any pointers greatly appreciated. Cheers! |-------------------------------------------------------------------------| | Frederick F. Gleason, Jr. | Chief Developer | | | Paravel Systems | |-------------------------------------------------------------------------| | ...one of the main causes of the fall of the Roman Empire was that, | | lacking zero, they had no way to indicate successful termination of | | their C programs. | | -- Robert Firth | |-------------------------------------------------------------------------|

13 years, 5 months

Warming up for LAC2011

by Fons Adriaensen

Hello all, LAC2011 is due to start in a few days ! Anyone else staying at the Kilcar B&B ? I'll arrive thursday at around 18:00 - will there be the traditional pre-conference pizza gathering ? (or Irish beers, or some stronger liquids) ? Ciao, -- FA

13 years, 5 months

No audible MIDI jitter anymore for hw MIDI out

by Ralf Mardorf

Thank you coders :) now I can use Linux to make music :). Regarding to 'nobody does report success, just bugs are reported' I guess NO MIDI JITTER is worth a crossposting. $ uname -a Linux edubuntu 2.6.33.9-rt31 #1 SMP PREEMPT RT Tue Apr 26 15:05:31 CEST 2011 i686 GNU/Linux jackd 1.9.8 I guess svn 4290 or earlier Qtractor 0.4.8.69 svn 1978 Ardour2 1:2.8.11-1ubuntu1 a2jmidi_bridge 6-0ubuntu3 jackd --sync -Xalsarawmidi -dalsa -r48000 -p256 snd-hrtimer loaded and used as ALSA seq timer source, unfortunately there aren't enough timers to be used with JACK clients, when using the "-c" switch. With Ardour (synced to Qtractor by JACK transport) I recorded - Yoshimi, JACK MIDI - Hydrogen (as sample player only, not using it's pattern sequencer)and DSSI plugins hosted by Qtractor, ALSA MIDI - via PCI Envy24 MPU an Oberheim Matrix-1000, Yamaha DX7, Yamaha TG33 connected by a MIDI thru chain I didn't record all instruments at the same time, but one after the other and sync is perfect, no audible jitter. Unfortunately I couldn't use -Xalsarawmidi for recording MIDI events on MIDI tracks, because there's no bridge. FWIW, Ardours latency compensation is ok too. Ardour just isn't restoring the SMPTE offset, which isn't a big deal, because I don't use it now and manually setting it does work. My machine is a $ hwinfo --cpu Model: 15.107.2 "AMD Athlon(tm) X2 Dual Core Processor BE-2350" Clock: 2100 MHz $ hwinfo --memory Memory Size: 4 GB When building the kernel I set 'echo "CONFIG_HIGHMEM4G is not set" >> .config' and 'echo "CONFIG_HIGHMEM64G=y" >> .config' and btw. 'echo "CONFIG_MK8=y" >> .config' too. The distro is Edubuntu 10.10 Maverick 32-bit + Ubuntu Studio meta packages. Using the FLOSS nv driver for a NVIDIA GeForce 7200 GS on an ASUS M2A-VM HDMI motherboard. The two PCI sound cards are not used as one virtual sound card. Both are active, but I only did use one of the cards. I didn't use any USB device, but e.g. a PS/2 mouse and keyboard. PPPoE and screen saver were enabled, CPU frequency scaling was set to performance. Still one issue left. Any recommendations for a sound card with good audio qualitiy ;)? I guess I could spend around 700,- EUR and I need minimal 2 or better 4 analog IOs, mic pre amps etc. aren't needed, I'm using a mixing console. At the moment I 've got two TerraTec EWX 24/96 PCI cards, which cause a muddy bass sound. Thank you very much for the good work again, Ralf

13 years, 5 months

FW: Frequency Space Editors in Linux

by Ivica Ico Bukvic

Hi Mike, Forwarding this to the list where users and developers might be able to assist. Hope this helps! Best wishes, Ico -----Original Message----- From: Michael Van [mailto:m_vanwagner@yahoo.com] Sent: Saturday, April 09, 2011 12:44 PM To: ico(a)linuxaudio.org Subject: Frequency Space Editors in Linux Hello Linux Audio, I just wanted to find out if anyone know of any Linux programs that do sophisticated noise removal from recordings, like the frequency space editing process of Windows programs, Adobe Audition or Cool Edit. I wondered if there is a plugin for Audacity that might do it. I need to use something other than standard noise sample removal plugins because the crackle is only present when the music is playing, not present during quiet stretches. Thanks, Mike

13 years, 5 months

Decode and Info about media

by Alfs Kurmis

Hello Experts. Is here library with C #include headers for so what ? How can i gotta info about Wav, mp3, ogg, m4a, wma [wmv] ... files SampleRate, Channels, PlayLen( in samples ), bytesPerSample , bitrate, VBR ... , and after i gotta this info , decode em to RAW stream ? At least i need everything about Wav, mp3, ogg. Info about sound file should not be printf()'ed to stdout , but must be filled in info structure. Decoder library should have functions - loadFile, Play, SeekToSample , Stop. At least possibility - play file from sample or second.part . Or should be used 2 libs, one for gotta info and one for decoding ? Any C example and pointer welcomed. Tnx in advance @ all. Alfs Kurmis ----

13 years, 5 months

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Linux-audio-dev