hello
can someone please help me with this?
i can find all the mp3s in a directory and its sub-directories with this:
find -name "*.mp3"
i can also ffmpeg a mp3 to m4a with this:
ffmpeg -i INPUT.mp3 -y -acodec libfaac -ab 192k OUTPUT.m4a
now, i would like to pipe all that find finds into ffmpeg.
thanks
guerrier
In the tradition of releasing (or at least announcing)buggy, mostly
untested software, here's:
js_wrap - a simple wrapper for non Jack-Session aware apps whose state
can be fully specified by their commandline
Use
js_wrap -- gnome-terminal
if you want to get a gnome terminal in your jack_session.. Or
js_wrap -- a2jmidid
if you want to have the alsa to midi bridge in your jack_session..
Get the software via copy and paste of the following commands:
git clone https://github.com/fps/js_wrap
cd js_wrap
mkdir bld
cd bld
cmake ..
cd ..
make -C bld
sudo make install
and get to fixing bugs ;D
In my experiments it worked just fine with gnome-terminal though there
are some errors and the code is pretty much copy-pasta'd from
lash_wrap... So expect breakage and panic and hilarity which will ensue..
Always use condoms,
Flo
guitarix/gx_head, a guitar mono tube amplifier simulation for jack
a new release (0.18.0) is available.
please refer to our project page for more information:
http://guitarix.sourceforge.net/
new features in short:
* add tube model 12AT7
* fix runtime issues when build with g++ > 4.5
* add presence level controller
* add bass booster level controller
* switch to function pointer based engine
* fix issues with GtkBuilder > 2.14 and Glade files
* fix correct use of included zita-resampler source
* add factory settings by autoandimat
download site:
http://sourceforge.net/projects/guitarix/
have fun
Many thanks to Fons Adriaensen for lead me to find and fix a bogus bug
related to correct object bindings in C++
kudos to Fons
_________________________________________________________________________
As well, a new release (1.4) of gxtuner is available,
gxtuner comes now with jack_session support.
Read the README for details.
http://sourceforge.net/projects/guitarix/files/gxtuner/
regards
guitarix development team
I'm just in the process of porting my plugin to a platform where the GUI
runs on Windows, but the Audio processing runs on an real-time optimized
Linux box.
With the proliferation of iPads etc, I see this approach of mixing a cool
portable GUI with a Linux 'powerhouse' audio processor become more common in
studios.
Surly passing functors (which are pointers?) to audio code running in a
separate address space can not work?
Jeff McClintock
> From: Florian Paul Schmidt <mista.tapas(a)gmx.net>
> Subject: [LAD] RT-Safe UI/Engine Decoupling using Functional
> Programming and Reference Counted SmartPointers
> To: linux-audio-dev(a)lists.linuxaudio.org
> Message-ID: <4E3CA5A4.7060208(a)gmx.net>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi,
>
> during the process of writing a new small jack sampler which fits my
> workflow I came up with this little scheme to solve the UI/engine
> decoupling problem. For the purpose of spreading the idea or
> alternatively getting answers about how it's broken and sucks I decided
> to write a little article describing it..
>
> http://178.63.2.231/~tapas/wordpress/?page_id=45
>
> The (largely unfinished and unusable) sampler project is here:
>
> https://github.com/fps/jass
>
> Let me have it..
>
> Regards,
> Flo
Thank you, I apologize. That is now fixed.
Jeremy
On Sun, Aug 7, 2011 at 12:09 PM, Alvaro Paz da Silva <alanpasi(a)gmail.com>wrote:
> Hi
> File not found... :(
> https://nodeload.github.com/jeremysalwen/kn0ck0ut-LV2/tarball/1.0
>
> Alanpasi
>
> On Sun, Aug 7, 2011 at 6:58 AM, Jeremy Salwen <jeremysalwen(a)gmail.com>wrote:
>
>> Hello all,
>>
>> I am pleased to announce that the first release of Kn0ck0ut-LV2<https://github.com/jeremysalwen/kn0ck0ut-lv2>is ready. Kn0ck0ut-LV2 is a port of the very popular VST "Spectral
>> Subtractor/Vocal Remover" plugin Kn0ck0ut<http://www.freewebs.com/st3pan0va/>,
>> with added features and improvements.
>>
>> Kn0ck0ut-LV2 is primarily useful for either isolating or removing the
>> "center channel" of stereo recordings. If you have center panned vocals,
>> you can use this to get the classic effects of "karaoke" or "vocal
>> extraction". However, if you turn the controls to the extremes and/or feed
>> it two unrelated signals, you can also get a variety of unique sounds.
>>
>> On top of Kn0ck0ut's functionality, I offer these improvements in
>> Kn0ck0ut-LV2:
>>
>> * Improved performance through use of FFTW and more efficent buffering code
>> * Completely variable FFT size and overlap amount controls.
>>
>>
>>
>>
>> * Experimental "Phase Compensation" option, which will perhaps preserve
>> additional fidelity in certain cases.
>> * Restored Low-cut filter which was removed in later releases.
>>
>> If this all sounds great, a direct download link for the sources is here<https://github.com/jeremysalwen/kn0ck0ut-LV2/tarball/1.0>.
>> In order to compile, you need lv2core, fftw3, and lv2-c++-tools
>>
>> A special thanks to St3pan0va, the original author of kn0ck0ut.
>>
>> Comments/questions are welcome.
>>
>> Jeremy Salwen
>>
>> _______________________________________________
>> Linux-audio-user mailing list
>> Linux-audio-user(a)lists.linuxaudio.org
>> http://lists.linuxaudio.org/listinfo/linux-audio-user
>>
>>
>
Hello all,
I am pleased to announce that the first release of
Kn0ck0ut-LV2<https://github.com/jeremysalwen/kn0ck0ut-lv2>is ready.
Kn0ck0ut-LV2 is a port of the very popular VST "Spectral
Subtractor/Vocal Remover" plugin Kn0ck0ut<http://www.freewebs.com/st3pan0va/>,
with added features and improvements.
Kn0ck0ut-LV2 is primarily useful for either isolating or removing the
"center channel" of stereo recordings. If you have center panned vocals,
you can use this to get the classic effects of "karaoke" or "vocal
extraction". However, if you turn the controls to the extremes and/or feed
it two unrelated signals, you can also get a variety of unique sounds.
On top of Kn0ck0ut's functionality, I offer these improvements in
Kn0ck0ut-LV2:
* Improved performance through use of FFTW and more efficent buffering code
* Completely variable FFT size and overlap amount controls.
* Experimental "Phase Compensation" option, which will perhaps preserve
additional fidelity in certain cases.
* Restored Low-cut filter which was removed in later releases.
If this all sounds great, a direct download link for the sources is
here<https://github.com/jeremysalwen/kn0ck0ut-LV2/tarball/1.0>.
In order to compile, you need lv2core, fftw3, and lv2-c++-tools
A special thanks to St3pan0va, the original author of kn0ck0ut.
Comments/questions are welcome.
Jeremy Salwen
Hi, very noob C question. I do this:
renato@acerarch /usr/include/alsa $ grep snd_seq_open *
seq.h:int snd_seq_open(snd_seq_t **handle, const char *name, int streams, int mode);
seq.h:int snd_seq_open_lconf(snd_seq_t **handle, const char *name, int streams, int mode, snd_config_t *lconf);
so I see that the function snd_seq_open has its prototype declared in seq.h... but
where is the actual function definition?
cheers
renato
hi *!
sorry for the slightly off-topic post, but since spatial audio has been
a frequent topic lately, i think some people here might be interested.
linux or FLOSS won't be exactly in the limelight, but yours truly will
make sure there are at least 2-3 boxes with your favourite OS and audio
tools humming along in various places. oh, and you might come early and
watch a few high-end mixing consoles boot - the startup screen will
bring tears to your eyes (as will the price tag, unfortunately :)
unfortunately, there will have to be an admission fee, which we haven't
decided on yet. but we're trying to keep it reasonable. don't shout at
me when it turns out to be a bit more costly than LAC, though...
jörn
*.*
ICSA 2011 - International Conference on Spatial Audio
November 10 - 13, Hochschule für Musik, Detmold
Organizers:
Verband Deutscher Tonmeister (VDT), in cooperation with
Deutsche Gesellschaft für Akustik e.V. (DEGA), and
European Acoustics Association (EAA).
Contact/Chair:
Prof. Dr.-Ing. Malte Kob
Erich-Thienhaus-Institut
Neustadt 22, 52756 Detmold
Mail: icsa2011attonmeister.de
Phone: +49-(0)5231-975-644
Fax: +49-(0)5231-975-689
Summary:
The International Conference on Spatial Audio 2011 takes place from
November 10 to 13 at Detmold University of Music.
This expert‘s summit will examine current systems for multichannel audio
reproduction and complementing recording techniques, and discuss their
respective strengths and weaknesses.
Wavefield synthesis systems, a higher-order Ambisonics array, as well as
5.1/7.1 installations in diverse acoustic environments will be available
for comparative listening tests during the conference.
Structured plenary talks, paper and poster sessions will revisit
fundamentals and present latest research.
A series of workshops will be dedicated to practical implementations of
spatial sound capture and playback methods, and their esthetic and
psychoacoustical implications for music perception.
Concerts that include music specially arranged for the conference will
let you experience various spatial sound systems in "live" conditions.
Call for papers and music:
Your contributions are welcome, either as presentations, posters, or
workshops. Submissions will undergo a review process, and accepted
contributions will be published in the conference proceedings.
The conference language is English.
We are planning structured sessions on the following topics:
* Multichannel stereo
* Wave field synthesis
* Higher-order Ambisonics / spherical acoustics
* 3D systems
* Binaural techniques
An additional session will be dedicated to related miscellaneous
contributions, such as hybrid systems and perception/evaluation of
spatial music reproduction.
Hi,
I am trying to get my own ALSA plug-in to work with some real time controls. ( Is this the right place to ask?)
I am successful with the PCM part of it. What I mean is from:
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html
External Plugin: Filter-Type Plugin
static snd_pcm_sframes_t
myplg_transfer(snd_pcm_extplug_t *ext,
          const snd_pcm_channel_area_t *dst_areas,
          snd_pcm_uframes_t dst_offset,
          const snd_pcm_channel_area_t *src_areas,
          snd_pcm_uframes_t src_offset,
          snd_pcm_uframes_t size)
{
// my PCM processing works
// I want to add a control parameter that can be set in real time by an app
}
SND_PCM_PLUGIN_DEFINE_FUNC(myplg)
{
// This all works for PCM processing just like the examples
...
err = snd_pcm_extplug_create(&mix->ext, name, root, sconf, stream, mode);
   if (err < 0) {
       free(mix);
       return err;
   }
...
}
SND_PCM_PLUGIN_SYMBOL(myplg);
Now I want to add a simple real time adjustment, an integer value that can be sent by an application to adjust the sound (PCM samples) in real time. I tried some ways of doing it it but without success, I am not understanding the basics.
I first looked at doing a ctl. I was able get separately (without PCM processing) get a control to work but that looks like ts only for hardware control? I can't connect it to my PCM processing.
http://www.alsa-project.org/alsa-doc/alsa-lib/ctl_external_plugins.html
I looked at LADSPA but I'm not sure where that is going to take me.
I am now looking at
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html
External Plugin: I/O Plugin
But, I am confused because I also see pcm type callbacks.
So, I do not have a specific coding question (yet) but I just need a general direction. ...
-Should I use ctl_external_plugins and figure out how to use it with my PCM?
-Should I use a LADSPA example?
-Should I go wth External Plugin: I/O Plugin?
-Something else?
I hope I am clear enough about my question and thanks for any pointers you can provide.
Bob