Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7513 discussions

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.

14 years, 4 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 | |-------------------------------------------------------------------------|

14 years, 4 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

14 years, 4 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

14 years, 4 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

14 years, 4 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 ----

14 years, 4 months

LAC: Anyone driving to Dublin after the Linux Sound Night(s)?

by rosea.grammostola

Hi, I might be staying in Dublin during the LAC. The last train on friday and saturday is leaving on 23:10. The concerts might not be ended at that time. Is there someone who is driving from Maynooth to Dublin after the sound night and who can take someone with him? Regards, \r

14 years, 4 months

LAC Accomodation

by Harry Van Haaren

Hey guys, I was just looking at rates for acccomodation for the LAC, I was about to book a single person in the GlenRoyal hotel, but if somebody doesn't mind sharing with me and has booked I can chip in 1/2 the cost? I'm told I don't snore :) Cheers, -Harry

14 years, 4 months

Realtime Audio on Android

by Patrick Shirkey

Hi, I'm working on a cross platform mobile OS media player and doing some research into Realtime audio playback on Android at the moment. It seems there is a fairly small amount of info online and a lot less fully fledged code examples than available for the iPhone platform. For example I have already implemented my demo app in objective-C but it looks like it is going to be significantly more effort to do the same for Android. I have found a couple of interesting posts and articles about the current state things. Maybe people round here would like to comment on the situation? Here's what I have found so far: 1: There are three main API's - mediaPlayer, SoundPool, AutdioTrack with openSL ES in the works for release sometime in the near future. 2: All of the Android API's do not provide true low latency support and have varying bugs that need to be worked around with native code. The pd team have made some progress on the workarounds based on jack style callback wrapper classes. 3: There is no standard library for decoding audio file formats at the lower level. Although several options exist including native java implementations for mp3 (JLayer), and custom code from developers like Ivan Memruk from mindtherobot.com 4: There is no native midi or OSC support and it looks like there is no LV2 or ladspa port at this point either. There are very few open source demo apps available which provide a complete solution for media playback and recording at low latency. It seems that the pd implementation is the only one. Overview: http://www.wiseandroid.com/post/2010/07/13/Intro-to-the-three-Android-Audio… http://mindtherobot.com/blog/555/android-audio-problems-hidden-limitations-… <http://mindtherobot.com/blog/555/android-audio-problems-hidden-limitations-…>pensl-es/ <http://mindtherobot.com/blog/555/android-audio-problems-hidden-limitations-…> Sample code: http://mobiledeviceprogramming.blogspot.com/2010/10/fun-with-android-audiot… http://mindtherobot.com/blog/580/android-audio-play-a-wav-file-on-an-audiot… http://mindtherobot.com/blog/624/android-audio-play-an-mp3-file-on-an-audio… http://blog.pocketjourney.com/2008/04/04/tutorial-custom-media-streaming-fo… Workarounds / Middleware: http://gitorious.org/pdlib http://www.typhon4android.org/ Cheers. -- Patrick Shirkey Boost Hardware Ltd.

14 years, 4 months

Re: [LAD] [ann] CAPS 0.4.5

by Jeff McClintock

> An informal experiment shows that jack-rack 1.4.7 and ardour 2.8.11 > both do exactly the right thing when a plugin referenced by a saved > session gains a port: they set the new port to its default value and > happily carry on working. > > (Both also cope with the removal of a port. These are the only > publicly available hosts I have tested.) What happens when you modify version-1 of your plugin and remove a port (making Version2), then later re-add a new (unrelated) port with different semantics? (Version 3)... Then load a project created with version 1. Does the host in THIS situation set the new port to it's default value, I doubt it. More likely it 'restores' it to some invalid setting. Do you want a fragile, crash-prone, plugin ecosystem?, or a robust one? I Agree with Paul on this one. Best Regards, Jeff

14 years, 4 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev