Hi all! I wanted to use Jack.Plumbing with some connect-exclusive rules for when certain apps were up. But it seems that connect-exclusive rules can
create a loop effect where connections are made and remade unnecessarially. Is something wrong with my Jack.plumbing, or am I not understanding something? It seems to be a problem even if I don't have other rules that would go against the exclusive ones, so that just using an exclusive rule at all results in this looping.
Those of you using Jack.Plumbing, are you constantly re-writing the rules, or can't you use exclusive rules to change the setup when other apps run without manually removing all other rules?
Thanks much.
Kevin
Does anybody know why wav files that do not have clipping would clip
when encoding them with lame?
I have a bunch of self-made clips that for various reasons don't have
enough gain. I wrote a wav->wav conversion that determines maximum
gain (as in highest or lowest absolute value) and uses sox' volume
argument to boost. Inspecting it with same gain tools and listening
to it shows the expected gain raise, but not clipping.
Then, when encoding in lame it gets clipped. Lame warns about it and
the values have been boosted to the max. When doing the boost, with
the same factor, directly in lame (feeding in the unboosted wav) I get
the same problem. Very roughly a clip with 3% headroom ends up being
too high by 16%, or about a 20% boost.
What's going on? I assume there is something about mp3 encoding I
don't quite understand.
Is there anything more efficient that I can do about that doing one
lame run to get an appropriate boost value and then doing another
encoding pass?
Here are the stats of a boosted wav file and the resulting lame:
t02c02.tmp.wav_maxgain_tmp.wav:
Samples read: 36900960
Length (seconds): 384.385000
Scaled by: 2147483647.0
Maximum amplitude: 0.969208
Minimum amplitude: -0.970673
Midline amplitude: -0.000732
Mean norm: 0.213396
Mean amplitude: 0.000000
RMS amplitude: 0.274775
Maximum delta: 1.675323
Minimum delta: 0.000000
Mean delta: 0.193639
RMS delta: 0.245325
Rough frequency: 6820
Volume adjustment: 1.030
mp3:
Samples read: 36907776
Length (seconds): 384.456000
Scaled by: 2147483647.0
Maximum amplitude: 1.000000
Minimum amplitude: -1.000000
Midline amplitude: -0.000000
Mean norm: 0.203464
Mean amplitude: 0.000010
RMS amplitude: 0.262007
Maximum delta: 1.636044
Minimum delta: 0.000000
Mean delta: 0.185782
RMS delta: 0.235408
Rough frequency: 6863
Volume adjustment: 1.000
/opt/good-sox/bin/sox WARN sox: `-' input clipped 725 samples
/opt/good-sox/bin/sox WARN sox: `/tmp/cracauer/l.wav' output clipped
360 samples; decrease volume?
Encoding output:
lame -h -b 160 --replaygain-accurate --clipdetect t02c02.tmp.wav_maxgain_tmp.wav t02c02.tmp.wav_maxgain_tmp.mp3_tmp '&&' mv t02c02.tmp.wav_maxgain_tmp.mp3_tmp t02c02.tmp.wav_maxgain_tmp.mp3
+ sh
LAME version 3.96.1 (http://lame.sourceforge.net/)
Using polyphase lowpass filter, transition band: 18000 Hz - 18581 Hz
Encoding t02c02.tmp.wav_maxgain_tmp.wav to t02c02.tmp.wav_maxgain_tmp.mp3_tmp
Encoding as 48 kHz 160 kbps j-stereo MPEG-1 Layer III (9.6x) qval=2
Frame | CPU time/estim | REAL time/estim | play/CPU | ETA
16015/16018 (100%)| 0:22/ 0:22| 0:22/ 0:22| 17.121x| 0:00
average: 160.0 kbps LR: 2551 (15.93%) MS: 13467 (84.07%)
Writing LAME Tag...done
ReplayGain: -9.1dB
WARNING: clipping occurs at the current gain. Set your decoder to decrease
the gain by at least 1.3dB or encode again using --scale <arg>
(For a suggestion on the optimal value of <arg> encode
with --scale 1 first)
--
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
Martin Cracauer <cracauer(a)cons.org> http://www.cons.org/cracauer/
Hi everybody
We are very happy to announce the MOD Pitch Shifter
https://github.com/portalmod/mod-pitchshifter.git
After some unsuccessfull time looking for a nice pitch shifter to offer in
the MOD Cloud we decided to make one ourselves. Kudos for Andre Coutinho
who did most of the coding.
We tried VocProc and Rubberband but none gave satisfactory results, the
first being too complicated and the latter yelding too much latency.
It is very simple to use - a simple semitones shift and a quality lever -
and sounds pretty nice.
The team is working on it to make it even nicer but this first version
sounds pretty good already.
It uses the fftw3 lib which is pretty common and can be installed via the
main linux repositories (ubuntu, debian, arch, etc)
Hope you all enjoy
Kind Regards
Gianfranco
The MOD Team
Hi everybody.
I have been doing some test with netcjack1 and jack2, sending audio between two linux pc. I can send audio from one computer to another, but only with synths and prerecorded audio tracks. When I try to use a live audio input, it doesn't works. I'm doing something wrong or it's because I'm using jack with netone? is there any way to do this if possible?
Thanks!
KMidimon is a MIDI monitor for Linux using ALSA sequencer and KDE4 user
interface.
Changes in 0.7.4
* requires Drumstick >= 0.5
* load and play OVE files (Overture), contributed by Rui Fan
* option to request real-time priority on MIDI input thread
* option to (not) resize columns while recording
* better reporting of file loading errors
* revised universal sysex messages translation
Copyright (C) 2005-2010, Pedro Lopez-Cabanillas
License: GPL v2
More info
http://kmidimon.sourceforge.net
Sources
http://sourceforge.net/projects/kmidimon/files/
Regards,
Pedro
Hi all! I'm wondering why so many of my MIDI ports in Jack are named so similarly, and if they can be changed? I don't remember this being a problem on my previous Linux system, but there are a few differences. I'm avoiding GUIs and need to use CLI tools, so no Qjackctl suggestions, please. I'm using an M-Audio MIDI sport Uno USB midi interface, and the virMIDI device for internal patching, as well as MIDISH. In jack, all of those, along with the main system ports like through are all listed as system midi playback or capture and a number. I've tried both raw and seq jack MIDI modes. I can see which is which with Jack_lsp with the -A or -t options set. But I was hoping to use something like Jack Plumbing for reconnecting, and since the real client names don't show up, I can't get jack plumbing to see them. Since the port numbers aren't always the same, I can't depend on the suffix digits to identify the right port. Then, using MIDISH, where the ports come and go when it starts and stops, its port numbers seem to increment each time as well. Should I perhaps be using A2Jmidid instead of the jack MIDI modes? Is there something else I'm missing? Or is this a difference between Jack 1 and Jack 2? Or differences from Ubuntu 9-10 and 12-10? Sorry to be so wordy, but wanted to be clear. Thanks for any help.
Kevin
Hi,
do session managers take care about the configurations of audio apps,
e.g. do they save and restore ~/.config/rncbc.org/ too?
Can they launch apps by a terminal emulation as I can do it using a
script?
E.g.
xfce4-terminal --maximize -T "♪ jackd" -e "jackd --sync -dalsa -r$sample_rate -p$frames_period"
xfce4-terminal --maximize -T "♪ qtractor" -e qtractor\ $song_path/qtr/$song_name-$song_version_qtr.qtr
Somebody on the Arch general mailing list mentioned to use session
managers, when I ask "How to safe configs to another path than ~" -
https://mailman.archlinux.org/pipermail/arch-general/2013-July/thread.html#…
I'm looking for a solution, not only for audio sessions, but for audio
sessions it's the most important.
Regards,
Ralf
Hello everyone!
I just found, that there were quite a few updates for mod-host in the git
repository. But after having built the software, I discovered a few problems.
When starting mod-host I always get this message:
error: PROTOCOL_MAX_COMMANDS reached (reconfigure it)
In mod-host's interactive shell (mod-host -i), I can get help, but when
typing quite, I get the message:
not found.
The command can even be completed, so a typo is unlikely. :-)
Any ideas or fixes for this?
Warm regards
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
Hi,
I will play one song to my sister's wedding in 14 days :)
the idea is that it starts slow and then every measure goes little faster
(+XX bpm) than previous one. Then it remains few measures in the final
(fast) speed.
Does anybody have experience (with any linux utility) for this?
Thank you.
Milan
