I've not done much of late, busy with other things, but I had cause to
update Linux Mint to
19.3 last week. Just tried testing various things and hit problems with
Jack.
I'm running jackd2 1.9.12~dfsg-2 and get noises with nothing running,
and when I try running
some midi files via Calf Fluidsynth, the noises are nothing like what
they should be.
I then switched to using the latest Yoshimi (1.7.1, thanks Will) and
that also produces
awful noises.
However, if I shut down jack, yoshimi happily switches to direct via
Alsa and sounds perfect.
However, my sequencer (home-grown) relies on Jack, and Calf Fluidsynth
also seems to be
fixed on using Jack.
Any ideas where to start tracking down this problem?
Bill
(getting older... :-( )
--
+----------------------------------------+
| Bill Purvis |
| email: bill(a)billp.org |
+----------------------------------------+
Hi
I find myself in the need of batch normalizing a lot of wav files (44.1kHz,
24bits) to "-1db peak". I don't know much about the technical details,
although I seem to understand that "-1db" is not as simple as that. I
looked to the command line tool normalize-audio and got quite confused. In
reaper I exported files with the rendering meter maxing at -1, I assume
this is the correct target gain. However if I just run normalize-audio on
the resulting file I get this:
atte@pandrup:~/downloads/normalize_test/test$ normalize-audio 0db.wav
Computing levels...
0db.wav 100% done, ETA 00:00:00 (batch 100% done, ETA 00:00:00)
Applying adjustment of 4.02dB to 0db.wav...
0db.wav 100% done, ETA 00:00:00 (batch 100% done, ETA 00:00:00)
So it seems that there's room for 4db of gain, which I find odd, as I
expeced the reaper render to be at -1db. So I looked at the manual, and
found the --amplitude option:
atte@pandrup:~/downloads/normalize_test/test$ normalize-audio
--amplitude=1db 0db.wav
normalize-audio: normalizing to -1.000000 dBFS
Computing levels...
0db.wav 100% done, ETA 00:00:00 (batch 100% done, ETA 00:00:00)
Applying adjustment of 15.02dB to 0db.wav...
0db.wav 100% done, ETA 00:00:00 (batch 100% done, ETA 00:00:00)
Looking at the waveform it now looks clipped. Total confusion here...
What is generally meant by "-1db peak", and is there a way to normalize a
wav to "-1 db peak" on the command line.
NB: First post here after switching to gmail, hope post goes through...
--
Atte
Hi!
Using linux + ladspa should make it easy building signal processors like
the DBX DriveRack or similar devices…
Unfortunately, I don't know professional grade audio devices offering 2
balanced (XLR) input and >4 (preferable 6) balanced (XLR) output ports.
Does anyone know?
Greets!
Mitsch
This was released today.
In brief:
One instantly noticeable feature is a new 'Pan Law' entry in the main window.
There is a new channel based setting for Solo.
Another quite new development is both Channel and Key aftertouch.
Legato performance has been improved and made smoother.
There is a new waveform shape available to AddSynth and PadSynth.
Storing and recovery of GUI window positioning has been improved.
Instrument banks have also had a workover with added proctections.
The underlying code continues to be refined and improved.
More details are in /doc/Yoshimi_1.7.1_features.txt
Yoshimi source code is available from either:
https://sourceforge.net/projects/yoshimi
Or:
https://github.com/Yoshimi/yoshimi
Full build instructions are in 'INSTALL'.
Our list archive is at:
https://www.freelists.org/archive/yoshimi
To post, email to:
yoshimi(a)freelists.org
--
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.
Hi everyone,
I am thinking of upgrading my audio PC/Laptop, and heard a lot of good
things about AMD's Ryzen series, specifically Ryzen 5 3600. But when
regarded pro audio work some people in the usual forums tend not recommend
AMD, because of latency issues.
Are there any Linux users successfully using Ryzen CPUs for low latency
work?
If so what is your recommended setup?
FYI I'm using Manjaro XFCE on my laptop and Manjaro KDE on my desktop.
Audio interfaces are Motu 1248 and RME HDSP9652.
Would be great to hear your experience.
Best regards
Moshe
It seems quite a while since i put any of my efforts up on here. This one is
another of my {cough} extended {cough} developments.
https://soundcloud.com/soft-sounds/reflections-1991-2020
--
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.
On Sun, 19 Apr 2020 12:29:37 +0200 al3xu5 / dotcommon wrote:
> Sat, 18 Apr 2020 09:50:30 +0100 - John Murphy <rosegardener(a)freeode.co.uk>:
>
> > I use a program which plays regions of various .wav files. Its
> > output pipes into aplay like: '| aplay -f FLOAT_LE -c 2 -r 48000'
> >
> > To where can I pipe its output, instead, to make a new .wav file,
> > or make a compressed file? Preferably faster than the region(s)
> > take to play.
> >
> > Or, if not, make aplay write to a file instead of playing?
> >
> > Thank you.
>
>
> Use sox:
>
> sox source.wav -L -c 2 -r 48000 new.wav trim <start_second> <duration_seconds>
>
> where:
>
> -L : little endian
> -c 2 : 2-channels stereo
> -r 48000 : freq (Hz)
> <start_second> : region start (second)
> <duration_seconds> : region duration (seconds)
>
That works a treat. Thank you. sox works so much better when it has
an actual file as input that it works just as well without -L -c -r
As the region lines I get give the start and end positions, I can use
'=' before duration and give it the end position directly like:
REGION "Fri0304.wav" 00:00:01.14181 00:00:05.07359 works in sox like:
sox Fri0304.wav new.wav trim 1.14181 =5.07359
$ soxi new.wav
Input File : 'new.wav'
Channels : 2
Sample Rate : 48000
Precision : 16-bit
Duration : 00:00:03.93 = 188725 samples ~ 294.883 CDDA sectors
File Size : 755k
Bit Rate : 1.54M
Sample Encoding: 16-bit Signed Integer PCM
I should be able to write a script to get the filename and start/end
seconds from the region lines and tell sox what to do.
Many thanks and best regards to all who have responded. I'll be fine
now and I may even try some of the other ways, because they're there. :)
--
John.
I use a program which plays regions of various .wav files. Its
output pipes into aplay like: '| aplay -f FLOAT_LE -c 2 -r 48000'
To where can I pipe its output, instead, to make a new .wav file,
or make a compressed file? Preferably faster than the region(s)
take to play.
Or, if not, make aplay write to a file instead of playing?
Thank you.
--
John.
On Sat, 18 Apr 2020 11:53:35 +0100 Bill Purvis wrote:
> On 18/04/2020 11:05, John Murphy wrote:
> > On Sat, 18 Apr 2020 09:55:12 +0100 Will Godfrey wrote:
> >
> >> On Sat, 18 Apr 2020 09:50:30 +0100
> >> John Murphy <rosegardener(a)freeode.co.uk> wrote:
> >>
> >>> I use a program which plays regions of various .wav files. Its
> >>> output pipes into aplay like: '| aplay -f FLOAT_LE -c 2 -r 48000'
> >>>
> >>> To where can I pipe its output, instead, to make a new .wav file,
> >>> or make a compressed file? Preferably faster than the region(s)
> >>> take to play.
> >>>
> >>> Or, if not, make aplay write to a file instead of playing?
> >>>
> >>> Thank you.
> >>>
> >> Use arecord instead.
> >>
> > Thanks. I saw the -C switch (or arecord), but didn't try it because
[...]
> You should be able to use sox to do that, specify the input format, and
> an output format and it should sort it out. I've not done this for ages,
> but it's the sort of thing sox was created for.
>
> Bill
>
Looks promising, thanks:
SoX can be used in simple pipeline operations by using the special filename
`-' which, if used as an input filename, will cause SoX will read audio data
from `standard input' (stdin)
Can't quite do it though. aplay gets -f FLOAT_LE -c 2 -r 48000 I've tried:
| sox -t raw -r 48k -c 2 -L -e float - dump-SOX-test.wav
sox FAIL formats: bad input format for `-': data encoding or sample size was not specified
| sox -t wavpcm -r 48k -c 2 -L -e float - dump-SOX-test.wav
sox FAIL formats: can't open input `-': WAVE: RIFF header not found
I'm guessing though really.
--
John.
