Just tried to run a mididing [sic?] with Boost 1.60. Using the latest git
of mididings. Got this:
TypeError: No to_python (by-value) converter found for C++ type:
boost::shared_ptr<mididings::backend::BackendBase>
I found many similar reports with other tools, all referencing Boost 1.60,
all talking about reverts to Boost 1.5* as the solution. Anyone have other
ideas? I'd rather not revert Boost if I don't have to do so.
--
*Jonathan E. Brickman jeb(a)ponderworthy.com
<http://login.jsp/?at=02e47df3-a9af-4cd9-b951-1a06d255b48f&mailto=jeb@ponder…>
(785)233-9977*
*Hear us at http://ponderworthy.com <http://ponderworthy.com> -- CDs and
MP3s now available! <http://ponderworthy.com/ad-astra/ad-astra.html>*
*Music of compassion; fire, and life!!!*
(sorry for cross-posting; please distribute)
The IEM – Institute of Electronic Music and Acoustics – in Graz, Austria
is happy to announce its call for the 2017 Artist-in-Residence program.
http://residency.iem.at/
The residency is aimed at individuals wishing to pursue projects in
performance, composition, installation and sound art, development of
tools for art production and related areas. Individuals are asked to
submit a project proposal that is related to the fields of artistic
research of the IEM, as:
* Spatialization/higher-order Ambisonics
* Sonic Interaction Design
* Audio-visuality
* Algorithmic Composition
* Algorithmic Experimentation
* Standard and non-standard Sound Synthesis
* Live Coding
Duration of residency: 5 months
Start date: June 1st 2017 (negotiable)
Monthly salary: approx. EUR 1100 (net)
*APPLICATION DEADLINE: 1st of October 2016*
_The Institute:_
The Institute of Electronic Music and Acoustics is a department of the
University of Music and Performing Arts Graz founded in 1965. It is a
leading institution in its field, with more than 25 staff members of
researchers and artists. IEM offers education to students in composition
and computer music, sound engineering, contemporary music performance
and musicology. It is well connected to the University of Technology,
the University of Graz as well as to the University of Applied Sciences
Joanneum through three joint study programs.
The artwork produced at IEM is released through the Institute's own
OpenCUBE and Signale concert series, as well as through various
collaborations with international artists and institutions.
IEM's main activities are centered around the following research areas
* Computer Music
* Artistic Research
* Signal Processing and Acoustics
_What we expect from applicants:_
* A project proposal that adds new perspectives to the Institute's
activities and resonates well with the interests of IEM.
* Willingness to work on-site in Graz for the most part of the Residency.
* Willingness to exchange and share ideas, knowledge and results with
IEM staff members and students, and engage in scholarly discussions.
* The ability to work independently within the Institute.
* A dissemination strategy as part of the project proposal that
ensures the publication of the work, or documentation thereof, in a
suitable format. This could be achieved for example through the release
of media, journal or conference publication, a project website or other
means that help to preserve the knowledge gained through the Music
Residency and make it available to the public.
* A public presentation as e.g. a concert or installation, which
presents the results of the Artist Residency.
_What we offer:_
* 24/7 access to the facilities of the IEM.
* Exchange with competent and experienced staff members.
* A desk in a shared office space for the entire period and access to
studios including the CUBE which has a 24-channel loudspeaker system and
infrared motion tracking, according to availability.
* During the period from July 1st until end of September the resident
will have extensive access to the studios of the IEM.
* Regular possibilities for contact and exchange with peers from
similar or other disciplines.
* Infrastructure (electroacoustic music studios, icosahedral
loudspeaker array, motion capture technology).
* Concert and presentation facilities (CUBE 24 channel loudspeaker
concert space).
* A monthly salary of approx. EUR 1100 net per month in addition to
health and accident insurance.
_What we cannot offer to the successful applicant:_
* We can not provide any housing.
* We also cannot provide continuous assistance and support, although
the staff is generally willing to help where possible.
* We can not offer any additional financial support for travel or
material expenses.
An application form providing more information is available at
http://residency.iem.at/
Feel free to contact residency(a)iem.at <mailto:residency@iem.at>if you
have any
questions.
Dear all,
Due to maintenance to the network that the linuxaudio.org server is part
of there will be scheduled downtime of all websites and some other
services from Friday June 2 23:00 until Saturday June 3 09:00 EST. We
apologize for any inconvenience this might cause.
Best regards,
Jeremy Jongepier
root(a)linuxaudio.org
Hi,
we have an open position for a student assistant for linux audio driver
development in our group "digital hearing devices" at the university of
Oldenburg, Germany:
http://www.careerservice.uni-oldenburg.de/admin/public/_jobboerse/jobs/1907…
Unfortunately this document is in German only, therefore please find a
google translation below. German language skills are not required, and
remote collaboration could be possible.
Best regards,
Giso
Student assistant for programming Linux audio driver
searched
To connect a multi-channel sound card ( development of the Uni
Hannover ) on the processor board BEAGLEBONE black with Linux
Operating system using standard libraries examined the
AG Auditory signal processing a student assistant ( with or
without bachelor's degree , but without master's degree ) . The scope
is initially 120h (3 months a 40h / month).
Candidate / candidates should programming skills in C and
bring knowledge of Linux . Previous experience with Linux
Kernel development is helpful but not required.
If interested, please contact Prof. Dr. V. Hohmann
(volker.hohmann(a)uni-oldenburg.de).
> I've been knocking my head against a wall for more than a year trying to
> figure out how to correctly mix two streams of audio while using
> libsndfile for input and libao for output. My main requirement is that
> I cannot assume anything about the output drivers -- that is, I cannot
> depend on the output driver (ALSA, OSS, Sun, etc) being able to do the
> mixing for me. Many of my target platforms lack any sort of mixing
> services. I need to do this myself. I tried starting a mixer/player
> thread that would work in a producer/consumer relationship with one or
> two audio file decoder threads. I can play one sound at a time just
> fine. When I try to do both, I get distortion followed by a segfault.
Hi,
not sure if I understood correctly: do you just want to mix N files?
Like you I'm learning libsndfile and libao so this is my attempt to mix
some audio files:
http://pastebin.com/dm7z8b3Z
HTH,
Andrea
P.S.
Can someone explain line 88 (I already read the sndfile FAQ)?
On Mon, May 16, 2016 at 11:30 PM, <
linux-audio-dev-owner(a)lists.linuxaudio.org> wrote:
> You are not allowed to post to this mailing list, and your message has
> been automatically rejected. If you think that your messages are
> being rejected in error, contact the mailing list owner at
> linux-audio-dev-owner(a)lists.linuxaudio.org.
>
>
>
> ---------- Forwarded message ----------
> From: Andrea Del Signore <sejerpz(a)gmail.com>
> To: linux-audio-dev(a)lists.linuxaudio.org
> Cc:
> Date: Mon, 16 May 2016 21:26:40 +0000 (UTC)
> Subject: Re: [LAD] mixing while using libao and libsndfile
> On Sun, 15 May 2016 16:34:34 +0000, David Griffith wrote:
>
> > I've been knocking my head against a wall for more than a year trying to
> > figure out how to correctly mix two streams of audio while using
> > libsndfile for input and libao for output. My main requirement is that
> > I cannot assume anything about the output drivers -- that is, I cannot
> > depend on the output driver (ALSA, OSS, Sun, etc) being able to do the
> > mixing for me. Many of my target platforms lack any sort of mixing
> > services. I need to do this myself. I tried starting a mixer/player
> > thread that would work in a producer/consumer relationship with one or
> > two audio file decoder threads. I can play one sound at a time just
> > fine. When I try to do both, I get distortion followed by a segfault.
> >
> > So, I'm back to a demo program. What must I do to this program to cause
> > it to start playing one audio file, then play another N seconds later?
> >
> > David Griffith dave(a)661.org
> >
>
> Hi,
>
> not sure if I understood correctly: do you just want to mix N files?
>
> I'm a noob with both libsndfile and libao :)
>
> Here my code: http://pastebin.com/dm7z8b3Z
>
> HTH,
> Andrea
>
> P.S.
> Can someone explain line 88 (I already read the sndfile FAQ)?
>
>
>
I've been knocking my head against a wall for more than a year trying to
figure out how to correctly mix two streams of audio while using
libsndfile for input and libao for output. My main requirement is that I
cannot assume anything about the output drivers -- that is, I cannot
depend on the output driver (ALSA, OSS, Sun, etc) being able to do the
mixing for me. Many of my target platforms lack any sort of mixing
services. I need to do this myself. I tried starting a mixer/player
thread that would work in a producer/consumer relationship with one or two
audio file decoder threads. I can play one sound at a time just fine.
When I try to do both, I get distortion followed by a segfault.
So, I'm back to a demo program. What must I do to this program to cause
it to start playing one audio file, then play another N seconds later?
David Griffith
dave(a)661.org
===begin code===
/*
* gcc -o mixer mixer.c -lao -lsndfile
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <string.h>
#include <ao/ao.h>
#include <sndfile.h>
#define BUFFSIZE 512
int playfile(FILE *);
int main(int argc, char *argv[])
{
FILE *fp1, *fp2;
if (argc < 2) {
printf("usage: %s <filename>.ogg <filename>.aiff\n", argv[0]);
exit(1);
}
fp1 = fopen(argv[1], "rb");
if (fp1 == NULL) {
printf("Cannot open %s.\n", argv[1]);
exit(2);
}
fp2 = fopen(argv[1], "rb");
if (fp2 == NULL) {
printf("Cannot open %s.\n", argv[1]);
exit(3);
}
ao_initialize();
playfile(fp1);
playfile(fp2);
ao_shutdown();
return 0;
}
int playfile(FILE *fp)
{
int default_driver;
ao_device *device;
ao_sample_format format;
SNDFILE *sndfile;
SF_INFO sf_info;
short *shortbuffer;
int64_t toread;
int64_t frames_read;
int64_t count;
sndfile = sf_open_fd(fileno(fp), SFM_READ, &sf_info, 1);
memset(&format, 0, sizeof(ao_sample_format));
shortbuffer = malloc(BUFFSIZE * sf_info.channels * sizeof(short));
frames_read = 0;
toread = sf_info.frames * sf_info.channels;
count = 0;
default_driver = ao_default_driver_id();
memset(&format, 0, sizeof(ao_sample_format));
format.byte_format = AO_FMT_NATIVE;
format.bits = 16;
format.channels = sf_info.channels;
format.rate = sf_info.samplerate;
device = ao_open_live(default_driver, &format, NULL);
if (device == NULL) {
printf("Error opening sound device.\n");
exit(4);
}
while (count < toread) {
frames_read = sf_read_short(sndfile, shortbuffer, BUFFSIZE);
count += frames_read;
ao_play(device, (char *)shortbuffer, frames_read * sizeof(short));
}
ao_close(device);
sf_close(sndfile);
}
===end code===
--
David Griffith
dave(a)661.org
I'm not sure if this has been covered before...
While I understand that generally you can't be certain of writing or reading
all bytes in a block of data in one call, what about the specific case where
you *always* read and write the same number of bytes and the buffer is an exact
multiple of this size.
e.g data block is 5 bytes & buffer size is 75 bytes.
No I'm not intending to use such an example, I just want to cover worst case :)
If that doesn't work, what about the case when you are always working in
powers of 2?
e.g data block is 16 bytes & buffer size is 1024 bytes.
--
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.
The Guitarix developers proudly present
Guitarix release 0.35.0
Guitarix is a tube amplifier simulation for
jack (Linux), with an additional mono and a stereo effect rack.
Guitarix includes a large list of plugins[*] and support LADSPA / LV2
plugs as well.
The guitarix engine is designed for LIVE usage, and feature ultra fast,
glitch and click free, preset switching, full Midi and/or remote
controllable (Web UI not included in the distributed tar ball).
This release introduce the new GUI design by Markus Schmidt aka. boomshop
Beside that, it comes with a couple of fixes and some new plugins.
Also included be the MOD UI's for the LV2 plugins used by the MOD[*]
For all changes, please check out the changelog.
Please refer to our project page for more information:
http://guitarix.sourceforge.net/
Download Site:
http://sourceforge.net/projects/guitarix/
Forum:
http://guitarix.sourceforge.net/forum/
Please consider visiting our forum or leaving a message on
guitarix-developer(a)lists.sourceforge.net
<mailto:guitarix-developer@lists.sourceforge.net>
regards
hermann
[*] http://moddevices.com/