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.
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.
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
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
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
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.
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.
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]
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
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.