Hi All,
I've written up a blog post on some recent changes to the JAudioLibs'
AudioServer API [1]. This is a Java callback audio API loosely
inspired by PortAudio, and the recommended approach for adding JACK
support to a Java application with JNAJack. The AudioServer API makes
it easy to switch between JACK support and JavaSound support without
requiring code changes.
The recent code additions provide for better runtime service discovery
and optional extension features. For example, it is now possible to
more easily control JACK connections, server autostart and client ID,
as well as for the first time directly access the JackClient if
necessary. More info in the blog post.
The source code on GitHub [2] is now up-to-date for testing, though a
new binary download is not yet available.
Other changes include fixes to JNAJack to build against JNA 3.5+
(binary downloads already work), and minor improvements to the
JavaSound server performance, particularly on Linux (ALSA /
PulseAudio).
Comments and feedback welcomed.
Thanks and best wishes,
Neil
[1] http://praxisintermedia.wordpress.com/2013/11/06/jaudiolibs-audioservers-a-…
[2] https://github.com/jaudiolibs/
--
Neil C Smith
Artist : Technologist : Adviser
http://neilcsmith.net
Praxis LIVE - open-source intermedia development - www.praxislive.org
Digital Prisoners - interactive spaces and projections -
www.digitalprisoners.co.uk
OpenEye - the web, managed - www.openeye.info
Greetings,
Noticed on the Audacity list :
https://code.google.com/p/r8brain-free-src/
Sample-rate conversion from Voxengo, "free" as defined under the MIT
license, not the GLP/LGPL.
Best,
dp
hi all
does anyone know if qmidiroute is still alive, and if so, who is currently
maintaining it ?
i found this git repo : https://github.com/royvegard/qmidiroute
but i cant find a way to contact the owner of this repo (Roy Vegard Ovesen)
grtz
Thijs
--
follow me on my Audio & Linux blog <http://audio-and-linux.blogspot.com/> !
Hi
I would save a internal audio buffer to file on exit, to reuse it after
e new start.
Currently I use stdio fopen/fwrite/fread and save the binary data from
the array.
That works well.
Now I've started to play with libsndfile, first I use SF_FORMAT_WAV |
SF_FORMAT_FLOAT which work as nice as stdio. The file size is the same
then with plain binary data.
To save some bytes on the disk I tried SF_FORMAT_FLAC | SF_FORMAT_PCM_24
which reduce the size nearly to the half. The drawback is that the
floats in the internal buffer could go out of the range from -1.0 <->
1.0 which leads to crackles when the buffer get refiled. As long the
values are in range, flac works very well.
I tried it with the sf_command (sndfile, SFC_SET_NORM_FLOAT, NULL, SF_TRUE)
for write and read, but that didn't help.
Yes, I use sf_write_float to write/read the flac file, even if flac
didn't support floats. When I understand the libsndfile api right,
libsndfile will handle that.
So my quetion will be, which is the common format to write flac file
with libsndfile, and is there a way to write values out of range into a
flac file?
regards
hermann
Re-posting. Maybe I can get some responses here :)
---------- Forwarded message ----------
From: Rafael Vega <email.rafa(a)gmail.com>
Date: Sun, Nov 3, 2013 at 3:31 PM
Subject: Measuring phase and frequency response of a filter.
To: linux-audio-user <linux-audio-user(a)lists.linuxaudio.org>
Hi.
I'm building a little ear training application for eqing for which I'm
building some filter banks in puredata. Can someone point me to a practical
way of measuring and plotting my filter's frequency and phase responses?
Jack apps, pd patches or code should be fine.
Thanks!
--
Rafael Vega
email.rafa(a)gmail.com
--
Rafael Vega
email.rafa(a)gmail.com
Hi all,
Due to excessive posts - in particular on the linux-audio-user email
list, we are going to rate-limit the number of emails to
13 emails per user per 24 hours
with direct follows-up replies to self counting twice.
It is not unreasonable to assume that a single user posting more than
12 emails per day is either involved in a heated flame-war or otherwise
disrupting reasonable discussion. Neither of which is acceptable on the
LA email lists.
We understand that some communication requires rapid exchange of many
small snippets and recommend users who need to regularly post short
messages many times per day to rather use services such as twitter, IRC
or simply private email. The LA email lists are mainly intended for
discussion, questions & answers, sharing and preservation of knowledge.
Note that there are currently over 3000 persons subscribing to the LA
lists, every single email sent to the list will result in 3000+ forwards
and be archived in various places on the web.
Offenders will be notified with a warning message with the 10th email
within a day and after the 13th email temporarily banned for 6 hours
from further posting to the given email list.
Feel free to comment on this email here on this list, but official
inquires should be addressed to members of the consortium. See
http://linuxaudio.org/contact
The system will be activated on 31/Oct/2013 if not vetoed. The actual
rate-limit may also be adjusted over time to reflect list-behaviour.
yours truly,
robin
Hi all.
Ive been working on a linux-port of IRCAMs OpenMusic lately, and think
its approaching a useful state now.
I would very much appreciate it if some of the brave ones out there did
some testing and sent me feedback:
http://forge.ircam.fr/p/OM6/downloads/label/421/
The tarball you'll find there includes the OM-6.7 image, source code, a
README and the various resources it uses (.so's, libs, fonts, icons,
help-files, tutorials, other).
This beta is not prepared to be installed anywhere, only to be run from
inside the directory where you unpack it (type ./OM_6.7_beta_4 ... ).
Plan is to make an installable version once the dust settles some.
Theres one thing which needs installing to work however, the music-fonts
used in the various editors and factories etc. Theres an .rpm here:
http://forge.ircam.fr/p/OM6/downloads/get/omfonts-1.0-1.noarch.rpm.zip
- which hopefully takes care of the fonts (theyre also available for
manual install in the tarball). The .rpm installs them (4 .ttf-files)
inside its own folder /usr/share/fonts/omfonts/*
The present version uses JACK for audio + midi I/O, and will try to
start a JACK server if its not already running.
To get any sound from the usual midi-ports, the linux-port uses
libfluidsynth, and attempts to load one of the standard soundfonts into
it. However you should be able to route midi-output to any client you
prefer (with jack-midi support).
Thanks for any feedback.
Cheers,
-anders
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hi all,
what's the proper way to list all alsa-devices, including those
(plugin, slave,...) defined in asoundrc?
e.g. on my desktop i have
$ cat ~/.asoundrc
pcm.jack {
type jack
playback_ports {
0
1
}
capture_ports {
0
1
}
}
$ cat /proc/asound/cards
0 [MID ]: HDA-Intel - HDA Intel MID
HDA Intel MID at 0xf4ff8000 irq 42
1 [Bt878 ]: Bt87x - Brooktree Bt878
Brooktree Bt878 at 0xf3fff000, irq 17
but aplay will show me:
<snip>
$ LANG=C aplay -L -l
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
sysdefault:CARD=MID
HDA Intel MID, VT1828S Analog
Default Audio Device
front:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
Front speakers
surround40:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=MID,DEV=0
HDA Intel MID, VT1828S Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
hdmi:CARD=MID,DEV=0
HDA Intel MID, VT1828S Digital
HDMI Audio Output
**** List of PLAYBACK Hardware Devices ****
card 0: MID [HDA Intel MID], device 0: VT1828S Analog [VT1828S Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: MID [HDA Intel MID], device 2: VT1828S Alt Analog [VT1828S Alt
Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: MID [HDA Intel MID], device 3: VT1828S Digital [VT1828S Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
</snip>
there is no mention of my 'jack' pcm, *and* there are devices that i
don't know where they come from: namely 'null' and 'pulse' (pulseaudio
is installed but not running, as confirmed by
<snip>
$ pgrep pulse
$
</snip>
so i figure that the extra devices come from system-configuration
files in /usr/share/alsa/alsa.conf(.d/*.conf)
but why does my "jack" PCM not show up?
i know that i can extend my asoundrc to:
<snip>
pcm.jack {
type jack
playback_ports {
0
1
}
capture_ports {
0
1
}
hint {
description "use jack as alsa-device"
}
}
</snip>
but i don't fully understand why i have to manually add the "hint" (of
which i haven't found *any* documentation) in order to make the device
show up with
char**hints=NULL;
snd_device_name_hint(-1, "pcm", (void***)&hints);
any clues?
fgasmdr
IOhannes
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Hi guys,
As those interested know the raspberry pi lacks an audio input. Now, I
came across this some days ago:
http://www.noiseisgood.co.nz/?p=365
Apparently it is possible to wire an i2c adc/dac to the raspberry pi
and utilize some already available alsa driver.
Just thought I would share and ask if anyone has tried anything similar?
Regards,
Robert
