Hi,
I'm facing a timing jitter which happens periodically due to some interrupt,
which is causing the task to be delayed. Since this happens periodically, it
is indeed a slow varying timing jitter, for example every 400ms, the timing
deviation is in the order of few ms ( 4-6ms). This is not getting filtered
out by DLL, and results in a slow varying oscillations which never dies.
Any way to identify and limit these timing jitters and not to take any
action on drift correction by DLL ?
-ben
--
Sent from: http://linux-audio.4202.n7.nabble.com/linux-audio-dev-f58952.html
spectmorph-0.3.4 has been released.
Overview of Changes in spectmorph-0.3.4:
----------------------------------------
* Added optional ADSR Envelope
* Make LV2 and VST plugin stereo to allow supporting stereo in the future
* LV2 plugin description fixes
* Added about dialog to plugin/smjack UI
* Remove BEAST plugin (plugin code will be moved to BEAST)
* Fixed compilation for newer g++ >= 6 (std::fabs)
* Get rid of some malloc() calls in linear morphing
What is SpectMorph?
-------------------
SpectMorph is a free software project which allows to analyze samples of
musical instruments, and to combine them (morphing). It can be used to
construct hybrid sounds, for instance a sound between a trumpet and a flute; or
smooth transitions, for instance a sound that starts as a trumpet and then
gradually changes to a flute.
SpectMorph ships with many ready-to-use instruments which can be combined using
morphing.
SpectMorph is implemented in C++ and licensed under the GNU LGPL version 3
Integrating SpectMorph into your Work
-------------------------------------
In order to make music that contains SpectMorph, you currently need to use
Linux. There are four ways of integrating SpectMorph sounds into music you
create:
- LV2 Plugin, for any sequencer that supports it.
- VST Plugin, especially for proprietary solutions that don't support LV2.
- JACK Client.
- BEAST Module, integrating into BEASTs modular environment.
Note that at this point, we may still change the way sound synthesis works, so
newer versions of SpectMorph may sound (slightly) different than the current
version.
Links:
------
Website: http://www.spectmorph.org
Download: http://www.spectmorph.org/downloads/spectmorph-0.3.4.tar.bz2
There are many audio demos on the website, which demonstrate morphing between
instruments.
--
Stefan Westerfeld, Hamburg/Germany, http://space.twc.de/~stefan
People,
For playing from multiple sources, ALSA plus the version of .asoundrc
below has been rock-solid for a couple of years now:
I now want to use things like "recordmydesktop" which work fine with the
mic but they do not record sound from audio or video clips that are
playing eg from YouTube or local mpv etc. So, yet again, I started to
try and make sense of JACK because I was pretty sure that it could solve
this problem - but I have found sound on linux has always been a major
pain and messing around with configs and rebooting etc on my big
workstation is tedious so I think I will try to get something working on
my laptop to begin with.
Hopefully some jack gurus will have some suggestions about how to do
what I want to do? I presume it is possible?
Thanks,
Phil.
pcm.!default {
type plug
slave.pcm "asymed"
}
# This is the audio output:
pcm.dmixer {
type dmix
ipc_key 1024
slave {
pcm "hw:1,0"
period_time 0
period_size 1024
buffer_size 4096
rate 44100
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
# This is the microphone
pcm.dsnooped {
ipc_key 1027
type dsnoop
slave.pcm "hw:1,0"
}
# This makes both channels work together.
pcm.asymed {
type asym
playback.pcm "dmixer"
capture.pcm "dsnooped"
}
--
Philip Rhoades
PO Box 896
Cowra NSW 2794
Australia
E-mail: phil(a)pricom.com.au
Hi,
I'm using JACK version jackdmp 1.9.11-RC1, whenever I start the JACK server
i get the subject line error message. I execute the following command.
$sudo modprobe snd-seq-midi
Is there a way to set this permanently and no need to execute every time i
start JACK ?
-ben
--
Sent from: http://linux-audio.4202.n7.nabble.com/linux-audio-dev-f58952.html
I want to know the optimal buffering which i can use for designing my
application.
My use case is as follows, I receive digital radio signals through a tuner
and does the channel and audio decoding in separate threads.
Finally the audio is send to jack callback and played out.
How much of buffering is enough for real time streaming between threads.
I want keep the optimal buffering between these threads.
Please suggest guidelines for using with Jack.
-ben
--
Sent from: http://linux-audio.4202.n7.nabble.com/linux-audio-dev-f58952.html
Hi All,
This is my first post to Linux Audio. I had a look at alsa_in/out programs, and
the man page says it performs drift compensation for drift between the two clocks.
I would like to know more about the implementation details such as the drift
compensation using PI controller. Any paper/presentation documents available
other than the C code. Please share me the details.
-ben
Anyone else having problems pushing commits there? It was fine yesterday, but
today I get an instant refusal.
No problems with github.
--
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.
Does anyone know of software that can log these without significantly adding to
the load itself?
--
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.