Hi
you are my last resource.
I'm trying to start fluidsynth in daemon mode (-s) at the startup. At
first I tried to adapt a guide for Arch but I realized that in Debian
(sid) the configuration is different:
https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=760210
I'm using systemd. I've tried several times to write a proper
fluidsynth.service but I always failed.
Here's my last try:
--------------->----------------------------->---------------------------->
[Unit]
Description=FluidSynth Synthesizer
After=syslog.target
Before=sound.target
Requires=dbus.socket
[Service]
Type = forking
ExecStart=/usr/bin/fluidsynth -s -a alsa
/usr/share/sounds/sf2/FluidR3_GM.sf2
Environment=DISPLAY=:0
#Restart=always
#RestartSec=5
[Install]
WantedBy=default.target
#WantedBy=multi-user.target
----------------->-------------------------->----------------------------->
This is the error:
# systemctl status -l fluidsynth.service
● fluidsynth.service - FluidSynth Synthesizer
Loaded: loaded (/etc/systemd/system/fluidsynth.service; disabled)
Active: inactive (dead)
ott 02 19:56:00 fede-xps systemd[1]: Started FluidSynth Synthesizer.
ott 02 20:13:51 fede-xps systemd[1]: Stopped FluidSynth Synthesizer.
ott 02 22:09:05 fede-xps fluidsynth[24181]: fluidsynth: warning:
Requested a period size of 64, got 940 instead
ott 02 22:09:05 fede-xps fluidsynth[24181]: fluidsynth: warning:
Requested 16 periods, got 8 instead
ott 02 22:09:05 fede-xps fluidsynth[24181]: FluidSynth version 1.1.6
ott 02 22:09:05 fede-xps fluidsynth[24181]: Copyright (C) 2000-2012
Peter Hanappe and others.
ott 02 22:09:05 fede-xps fluidsynth[24181]: Distributed under the LGPL
license.
ott 02 22:09:05 fede-xps fluidsynth[24181]: SoundFont(R) is a
registered trademark of E-mu Systems, Inc.
ott 02 22:09:05 fede-xps fluidsynth[24181]: Type 'help' for help topics.
ott 02 22:09:05 fede-xps fluidsynth[24181]: >
The error about the period size can be avoided setting "-z 940", but I
don't understand why this error appears only when fluidsynth is started
as a systemd service and not when I run that command in a terminal. I
guess that it may be related to when the command is launched in systemd
levels. I think that I should play with the values of After, Before and
Requires in the Unit section.
But I hope that someone in this list can help me, I don't have time
currently to study systemd (and it would be probably too complex for
me).
Thank you in advance.
Federico
Yes it's up there now :)
The most noticeable change from a user point of view is that individual part
outputs (corresponding to channel sends on a hardware mixer) are no longer
affected by the main volume control. The panel window has also been updated to
reflect this situation, and red 'clip' bars have been added.
The accuracy of peak and clip indication has been improved and decay times
increased to give a smoother, clearer response.
There are the usual crop of small refinements and bugfixes under the hood.
At the moment Sourceforge is still showing 1.2.3 as the latest version,
although 1.2.4 is there at the top of the 'files' list.
This happened once before and can't remember what the solution was. Has anyone
else had this happen?
http://sourceforge.net/projects/yoshimi/
--
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.
Hey all!
After quite a long period of "no time or no space to make music", I've
finally had time and peace of mind enough to sit down and make some new
music. I've just published a new track to my SoundCloud, which features
vocals from the user "snowflake" of ccmixter.org (a great resource for
finding acapella/other remix material, which I sincerely recommend). It's a
bit of hiphop/glitch mixed in there, and I hope you'll enjoy it. Please let
me know what you think! :)
https://soundcloud.com/zthmusic/never-stop-raising-the-bar
If there's anyone who can't download from/listen on SoundCloud, just let me
know and I'll arrange for an extra download through the website. The song
is licensed CC-BY-SA-NC.
Thanks a bunch for listening and for any feedback I get!
Cheers,
Gabriel/zth
The Guitarix developers proudly present
Guitarix release 0.31.0
For the uninitiated, 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).
Here is the " Ultimate Guide to Getting Started With Guitarix
<http://libremusicproduction.com/articles/ultimate-guide-getting-started-gui…>"
This release fix a bug in the preset naming schema ( vowel mutation in
preset names will crash guitarix) and introduce some new LV2 plugs:
* GxRoomSimulator
* GxDigitalDelay
* GxLiveLooper
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>
The Guitarix project never accepted Donations, and still wouldn't do.
But, if you ever wished to donate the project, I would kindly ask you to
back the MOD Kickstarter campaign here:
https://www.kickstarter.com/projects/modduo/mod-duo-the-limitless-multi-eff…
to reach this Goal:
>
> If the campaign reaches U$100.000 the MOD Duo will offer an Audio
> Interface from it's USB connection. This means that when you plug the
> MOD Duo to your computer you will be presented with a 4 input audio
> device (two pre-processed + two post-processed) that can be used for
> recording the MOD's audio directly to your favorite software.
>
> Add this to the quality of our analog circuit and you'll have, as a
> free bonus, a professional grade audio interface that if bought alone
> would cost the price of a MOD Duo.
>
[*]Here is a list of all included plugs:
Guitarix tube emulations
========================
12ax7
12AU7
12AT7
6DJ8
6C16
6V6
12ax7 feedback
12AU7 feedback
12AT7 feedback
6DJ8 feedback
pre 12ax7/ master 6V6
pre 12AU7/ master 6V6
pre 12AT7/ master 6V6
pre 6DJ8/ master 6V6
pre 12ax7/ push-pull 6V6
pre 12AU7/ push-pull 6V6
pre 12AT7/ push pull 6V6
pre 6DJ8/ push-pull 6V6
noamp
Guitarix Tonestacks
===================
default
bassman
twin
princeton
jcm800
jcm2000
mlead
m2199
ac30
soldano
mesa
jtm45
ac15
peavey
ibanez
roland
ampeg
ampeg_rev
sovtek
bogner
groove
crunch
fender_blues
fender_default
fender_deville
gibsen
engl
Guitarix Cabinets
===================
4x12
2x12
1x12
4x10
2x10
HighGain
Twin
Bassman
Marshall
AC-30
Princeton
A2
1x15
Mesa Boogie
Briliant
Vitalize
Charisma
Guitarix internal mono plugins
===============================
Mono : Distortion : JCM 800 Preamp
Mono : Distortion : MultiBand Distortion
Mono : Distortion : Multi Band Distortion
Mono : Distortion : Overdrive
Mono : Distortion : Tube Screamer
Mono : Echo / Delay : Delay
Mono : Echo / Delay : Digital Delay
Mono : Echo / Delay : Dubber
Mono : Echo / Delay : Duck Delay
Mono : Echo / Delay : Echo
Mono : Echo / Delay : MultiBand Delay
Mono : Echo / Delay : MultiBand Echo
Mono : Echo / Delay : ReverseDelay
Mono : Guitar Effects : Compressor
Mono : Guitar Effects : Crybaby
Mono : Guitar Effects : Expander
Mono : Guitar Effects : GCB 95
Mono : Guitar Effects : Multi Band Compressor
Mono : Misc : abGate
Mono : Misc : Detune
Mono : Misc : Oscilloscope
Mono : Misc : Recorder
Mono : Modulation : Chorus Mono
Mono : Modulation : Flanger GX
Mono : Modulation : Flanger Mono
Mono : Modulation : MultiBand Chorus
Mono : Modulation : Parametric pitch shifter
Mono : Modulation : Phaser Mono
Mono : Modulation : Ring Modulator Mono
Mono : Modulation : Tremolo
Mono : Modulation : Vibe Mono
Mono : Reverb : Convolver
Mono : Reverb : Freeverb
Mono : Tone control : Amp impulse
Mono : Tone control : Baxandall
Mono : Tone control : BiQuad Filter
Mono : Tone control : Cabinet
Mono : Tone control : Feedback
Mono : Tone control : Fender 6G7
Mono : Tone control : Graphic EQ
Mono : Tone control : ImpulseResponse
Mono : Tone control : low high pass
Mono : Tone control : moonlight
Mono : Tone control : Peak EQ
Mono : Tone control : Scaleable EQ
Mono : Tone control : Tonestack
Mono : Tone control : Treble boost
Mono : Tone control : Volume
Guitarix internal stereo plugins
=================================
Stereo : Distortion : Postamp
Stereo : Echo / Delay : Digital Stereo Delay
Stereo : Echo / Delay : Duck Delay St
Stereo : Echo / Delay : Stereo Delay
Stereo : Echo / Delay : Stereo Echo
Stereo : Guitar Effects : Multi Band Compressor stereo
Stereo : Misc : Bass Enhancer
Stereo : Misc : Panoram enhancer
Stereo : Misc : Stereo Recorder
Stereo : Modulation : Chorus
Stereo : Modulation : Flanger
Stereo : Modulation : Phaser
Stereo : Modulation : Ring Modulator
Stereo : Modulation : Vibe
Stereo : Reverb : Convolver
Stereo : Reverb : Plate reverb
Stereo : Reverb : Stereo Verb
Stereo : Reverb : Zita Rev1
Stereo : Tone control : 3 Band EQ
Stereo : Tone control : Moog Filter
Guitarix LV 2 plugins
======================
GxAmplifier-X
GxAmplifier-Stereo-X
GxBarkGraphicEQ
GxChorus-Stereo
GxCompressor
GxDelay-Stereo
Gxdetune
Gxdigital_delay
Gxdigital_delay_st
Gxduck_delay
Gxduck_delay_st
GxEcho-Stereo
GxExpander
GxFlanger
GxFuzz
GxGraphicEQ
Gxlivelooper
GxMultiBandCompressor
GxMultiBandDelay
GxMultiBandDistortion
GxMultiBandEcho
GxPhaser
GxRedeye Big Chump
GxRedeye Chump
GxRedeye Vibro Chump
GxReverb-Stereo
Gxroom_simulator
Gxshimmizita
Gxstereoecho
Gx Alembic Mono
Gx Studio Preamp Stereo
Gxswitched_tremolo
GxTremolo
Gxvocoder
GxZita_rev1-Stereo
GxAutoWah
GxWah
GxBooster
GxEchoCat
GxMetalAmp
GxMetalHead
GxTiltTone
GxTubeScreamer
GxTubeDelay
GxTubeTremelo
GxTubeVibrato
GxTuner
Hey hey everyone,
is there a tool to extract a single cycle wave from sampled audio
automatically without control by sight? I want to extract such a wave from a
voice sample. It won't be singing but talking. That way finding pitch and
experimenting looks difficult for me.
If at all possible, I'm looking for a non graphical utility.
Thank you for suggestions!
Ta-ta
----
Ffanci
* Internet: http://freeshell.de/~silvain
On Wednesday 01 October 2014 09:14:22 Paul Davis did opine
And Gene did reply:
> On Wed, Oct 1, 2014 at 7:27 AM, Gene Heskett <gheskett(a)wdtv.com> wrote:
> > Your ears are probably the best tool. Some hear well, and some do
> > not. I am amazed at the number of people who cannot tell if mp3 has
> > ever been in the mix. To me its obvious, when your ears get tired of
> > it, and want to "change the station" in just a minute or so, its
> > been an mp3 at some point.
>
> For crying out loud, stop this nonsense!
>
> It is established without any shadow of a doubt that the overwhelming
> majority of the population CANNOT tell the difference between a
> reasonable bit-rate encoding in mp3 format and the original PCM data.
> This isn't up for debate.
>
> Reasonable bit-rate means 256kbps; by the time you reach 320kbps even
> expert listeners have a very very hard time differentiating the mp3
> from the PCM in an ABX test.
256 kilobit mp3? I don't know if I have one that is so little compressed.
The most I've picked up here & there was 96 and below, usually with an un-
known history. Even 128kbit mp3 can be detected in a few seconds. Here I
have done a/b testing against some well produced cd's I've ripped, and
with ogg running at Q7, which gives a filesize similar to an mp3 at 128kb,
and cannot tell the difference. Back these ears up 55 years to before I
started wearing out rifle barrels, and I expect they might detect the
diff.
> By all means talk about low bit rate encodings and how they are no
> good, but please folks - double blind testing doesn't lie, and the
> double blind tests are close to unambiguous at this point.
Cheers, Gene Heskett
--
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS
Last Sunday I set up my laptop for a 3.5-hour recording session, Audacity crashed after 2.5 hours. Anyone know why this might have occurred, what to do about it, whether Audacity is in fact best for simple recording?
Jonathan E. Brickman
Ponderworthy Music | jeb(a)ponderworthy.com<mailto:jeb@ponderworthy.com> | (785)233-9977 | http://ponderworthy.com
On 09/28/2014 07:06 PM, Paul Davis wrote:
> patch from svn diff attached
>
i see that your patch only reads the pretty-names (metadata) if exists
at runtime and uses them as the "official" port-names instead.
so, who or what does the pretty-name setup? that is, who or what sets
the pretty-names of each port in the first place?
i was thinking about the user using the qjackctl client/port aliases
functionality already in place and use it eg. to run jack_set_property()
on each renamed client/port alias on jack server start (as read from
Qjackctl.conf file, on a per preset basis).
what you think?
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
On Fri, 26 Sep 2014, Philipp Überbacher wrote:
> Alles klar. Well, mostly. I guess that alsa_midi only makes sense
> when alsa is used as a backend, so I don't quite see why it is a server
> option instead of a backend option. Anyway, using the -X alsa_midi as
> server option works.
That is not true at all, Audio and MIDI are separate and so it is very
possible (even likely) that the backend for audio may not be relevant to MIDI.
MIDI is handled by ALSA, but is not treated as a part of the same audio
interface as the sound that goes with it. So the user who has a FW audio IF
with no MIDI would not be able to use a USB MIDI along with it if this was in
the backend only. Along the same lines is the FW user who does have MIDI in
their FW IF wants to use a lot of the older SW that creates ALSA MIDI ports
they are again stuck if it is only available in the ALSA back end.
The only thing I would like to see different in a2jmidi is the naming. (I am
not sure how jack1 works) a2j opens as one client called a2j that when expanded
has the alsa client name followed by the actual port name. This is hard to read
and use. It is also confusing to new users who ask where the port is just
because a2j has not been expanded. I would guess the best thing (from a user
POV) would be for a2j to open a new jack client for each ALSA client with the
ALSA name. I am not sure what other consequences this would have though :)
--
Len Ovens
www.ovenwerks.net
Hi,
Some of you might be interested in a new project I have been working on:
http://jackdub.channellinux.com/
It is a fully realtime automated music playback system built with various
FLOSS tools. The system runs entirely in the cloud and doesn't even have a
sound card.
It's a work in progress so YMMV...
Cheers
--
Patrick Shirkey
Boost Hardware Ltd