Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7516 discussions

by Mario Sottile

System: Ubuntu Mate 14.04. Hi, there. With some problems that I could resolve, I compiled with success: - zita-resampler-1.3.0 - zita-njbridge-0.1.1 I did this in two computers. Now, in one computer, I run zita-j2n with success. In Patchage, I can see the Jack server, it is there. I connect PureData to it. Now, in the other computer, I try to run zita-n2j and it says: /mario@circo3d:~$ zita-j2n --jname zita --chan 2 192.168.1.41 8800// //Cannot read socket fd = 6 err = Success// //CheckRes error// //JackSocketClientChannel read fail// //Fatal error condition, terminating.// //Server is not running// //Server is not running// / ... but Jack is running. I use QjackCtl, but I tried with jackd from command line and it tells me the same error. In this computer, zita-n2j runs well. What's happening with zita-j2n? Is Kokkini Zita in this list? Thanks in advance.

9 years, 1 month

Will I can get the configuration I need...?

by Mario Sottile

For some days, I'm testing some ways to send/receive audio to/from two computers connected with WiFi to a router. I use Jack in both machines and I need to use WiFi, no wires, because of a necessarily comfort issue. What I tested: - netsend~/netreceive~ from http://www.nullmedium.de/dev/netsend~/. It works, but it makes a lot of click/pops: unacceptable. - I compiled zita-njbridge and, finally, I made it work: also unaceptable because of similar noises and silences. What I didn't test yet: - netsend~/netreceive~ from http://www.remu.fr/sound-delta/netsend~/, because it is not compiled for Linux and I don't know how to do it. - I tried to use jack.trip but, because an error, I should compile 1.1 version. I didn't do it, because I have to install Qt5. I'm afraid of doing a mess... - NetJack1: I like the idea of compressing audio, but I don't like the master-slave design because I have to use both soundcards speakers and microphones. - NetJack2: also it has master-slave system. I don't know if I will can make the configuration I need. Also, I have to send from one computer to the other, a screencast (I would use VNC or "X over SSH", I have to test them). It doesn't matter if it is not perfect, it is only for monitoring. Now, the question: *Will I can get what I need? Will I can send audio to/from both computers with no clicks/pops/silences and also send video from one to another? *If the answer is "NO", I'm ending right here my research. Thanks in advance.

9 years, 1 month

Music made with LMMS

by Yann Collette

Hello, Here is a new "album" made with LMMS: https://www.jamendo.com/album/160826/extreme-limite-de-la-sagesse I used version 1.1.3 with some patches to enable the Carla rack under LMMS. I used samples from: - https://archive.org/ - http://freesound.org/ - http://wiki.laptop.org/go/Free_sound_samples (percussion samples) I feel LMMS is a great tool to play with sounds and I am glad to see that there are still some work done for a new release ... There are some encoding problem on Jamendo. I send them several messages to highlight these problems, but still no answer. So, if the sound feel a little bit strange from time to time, it's maybe a jamendo problem (or this is maybe due to the song :) ). Best regards, Yann

9 years, 1 month

LV2 mono instruments recommended?

by Stefan Westerfeld

Hi! I'm currently implementing a LV2 plugin for SpectMorph, and so far I only defined one audio port in the ttl file/source, as the output of the morphing algorithm is just mono. I have tested with Ardour and Qtractor. Ardour does what I'd expect: it plays my instrument output on both channels. Qtractor however just plays my mono data on one channel, the other is silent. Which is not what I want. So I am considering making my LV2 instrument always stereo, and just copy the output buffer from left to right - this is cheap, and probably not much more expensive than what the host would need to do anyway. Does this sound like the reasonable way to go? Cu... Stefan -- Stefan Westerfeld, http://space.twc.de/~stefan

9 years, 1 month

Portaudio or something else?

by Bengt Gördén

Hi, I think I need some advice about portability. I haven't been doing any programming for years on audio so I'm fairly rusty. My son is about to start a programming project (C++) in school (secondary school). It's about active noise reduction/cancellation. The program will be used on both Windows and Linux. Is Portaudio the best choice for portability or is there some other option that might be better? regards, -- /bengan

9 years, 2 months

Mixed boolean & numbers

by Will Godfrey

I'm finding quite a lot of occasions where variables defined as 'bool' are sometimes being set with true or false and other times 0 or 1. On one occasion there is something like x = n + {boolean variable} This last one seems quite unsafe to me as I didn't think the actual value of true and false was guaranteed. Am I being overly cautious or should I change them all to one form or the other? -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

9 years, 2 months

Audio plugins: Streamable audio ports?

by Tim E. Real

Are there any plugin architectures that allow input data length different than the output length such that the 'run' function can ask for more or less input data, for example via some kind of stream? Instead of passing 'run' a block of data, host would pass these streams so that 'run' can pull and push whatever lengths it needs. There would be compatibility information on each stream so that other streams could accommodate. I thought I read of an LV2 extension or something... Or am I imagining something like Pulse? Thanks. Tim.

9 years, 2 months

Tascam US-16x08 0644:8047

by Greg

Greetings, I am fairly new to USB dev (in linux in particular, but also in general), but I would very much like to try to get support for the above device working in snd-usb-audio. - Is this an appropriate place to discuss snd-usb-audio? - Are there any recommended reading pointers for behavior of the quirk table? I patched parse_audio_format_rates_v2(), get_sample_rate_v2(), and set_sample_rate_v2(), and through some sort of beginner luck was able to get aplay audio out of the first two channels. That was incomplete hackery though (eg fixed sample rate), and I would like to learn how to properly add quirk support. There have been other reports that this device worked OOTB, but I fail to see how! I've also been examining the traffic to the device with wireshark and a win7 vm, but the learning curve for USB is a bit steep, so I am digesting. (: If anyone can provide suggestions on lsusb output alone, here's what I have: http://pastebin.com/pA9MLQet cheers, Greg [x-post from alsa-devel due to empty thread - see: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094682.html]

9 years, 2 months

Advanced Gtk+ Sequencer v0.7.54 - minor version increases soon

by Joël Krähemann

Hi all I'd like to get some feedback on GSequencer v0.7.54. Many things have changed so far. Good real-time is still a pain especially as doing much GUI interaction. How-ever I have coded a strategy to counter the issue. Here are some new features listed that should work: * automation editor to automate ports, currently only LADSPA, DSSI and Lv2 * Configuration in place of device, pcm-channels, samplerate, buffer size and format * Virtual MIDI mapping to route from GSequencer to DSSI and Lv2 * Notation editor to edit notes of DSSI and Lv2 plugins Things I'm not sure if still works: * auto-scroll of notation editor on playback * export to WAV files with different samplerate and format All existing features for sure: * sequencer editors with copy and later for paste in notation editor * reallocate audio channels and pads * destroy machines * link lines Believed to be broken: * different threading models than super-thread with channel scope are believed to be broken Currently unmaintained: * Original file format temporally replaced by a light-weight one Note: having 128 channels on DSSI or Lv2 synths is a bit over-helming. So you might want to adjust the channels and do MIDI mapping. Cheers, Joël Krähemann

9 years, 2 months

FFADO 2.3.0 released

by Jonathan Woithe

The FFADO developers are pleased to announce FFADO version 2.3.0, a package of userspace drivers for firewire audio interfaces. While there are no significant new features in this version compared to the last, FFADO 2.3.0 contains a large number of incremental improvements. Users of FFADO are encouraged to upgrade. This source-only release can be downloaded from the ffado.org website at http://ffado.org or via the direct link: http://ffado.org/files/libffado-2.3.0.tgz Notable changes include: * Configuration entries added for additional devices which work with the generic support layer (PreSonus Studiolive 32.4.2, Presonus StudioLive 16.0.2, ICON FireXon, Onyx Blackbird and the new Onyx 1640i, among others). * Support added for the newer Focusrite Saffire Pro 26. * Improved build support for various downstream consumers. * Better routing for selected Saffire devices and the Firestudio Mobile. * Significant cleanup and refinement of the M-Audio and Yamaha driver. * Compilation fixes for recent versions of gcc. * Recover from dead streams without causing jackd to shut down. Thanks go out to the developers and users who contributed code and information which went into this release: Kristian Amlie, Melanie Bernkopf, David Binderman, Philippe Carriere, Yves Grenier, Florian Hofmann, Hector Martin, Mathieu Picot, Philippe Ploquin, Stefan Richter, Takashi Sakamoto, Jano Svitok, Karl Swisher, Steven Tonge and Jonathan Woithe.

9 years, 3 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