Hey hey,
I just released wavetral, a tool to convert between single cycle waveforms and
harmonic spectra in different formats. It supports audiofiles, plaintext
representations (one value per line), Waldorf Microwave II/XT wave dumps. It
can furthermore read Csound ftables as stored by ftsave and ftsavek and it can
write a Yoshimi script, which can be run loccally (runlocal) in the wave
context of the AddSynth voices and PadSynth.
Converting between waveforms means resampling a waveform and converting
between harmonics allows to decrease harmonics or fill higher harmonics with
0s to meet certain requirements.
You can clone the git repo here:
git clone https://github.com/jeanette-c/wavetral.git
Stranglely, when I tried I wasn't on the master branch, so I had to do
cd wavetral
git checkout master
git pull
Maybe it's just my git setup.
Best wishes,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
For whatever reason,
I feel like I've been wanting you all my life <3
(Britney Spears)
Hey hey,
what is the easiest way to find libfftw3 with cmake? I've seen a few
find_package scripts, but they don't appear to work. The source compiles, but
gets linking problems.
I've seen something in Yoshimi using pkg_check_modules but I don't completely
understand the following mechanism nor am I completely sure which module to
include to get that function.
Could someone please help?
Best wishes,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
For whatever reason,
I feel like I've been wanting you all my life <3
(Britney Spears)
Hi all,
After a long period of testing, having no negative feedback I decided to
issue the SoundTracker 1.0.2 release. Comparing to the last pre-release,
1.0.2-pre2, this is only the bugfixing release with few small
improvements only. But some critical bugs are closed included that
causing crash sometimes after manipulations with instruments. Here are
main features of the 1.0.2 release comparing to 1.0.1:
* Almost all user actions are logged and can be undone/redone;
* Sample editor has a new compact and comprehensive design;
* PulseAudio output driver is created.
You can find more new features in NEWS file.
ST-1.0.2 can be downloaded here:
https://sourceforge.net/projects/soundtracker/files/latest/download
Any feedback is welcome in SoundTracker mailing list:
soundtracker-discuss(a)lists.sourceforge.net
For those who didn't hear about the SoundTracker before: this program is
a score music editor and sequencer with tablature-like notation (such
kind of editors called as "trackers" and originating from the
demo-scene) with some facilities for editing samples and instruments.
Regards,
Yury.
Old dog here; trying to learn a new trick.
I've found QProcess in Qt5 very useful and Qt Creator helps by
immediately complaining about my mistakes. I found a line in
QJackCtl like '#include <jack/jack.h>' and was surprised there
were no complaints when I pasted it in.
'locate jack.h' found these two, amongst many others, in a
familiar folder and it all begins to make sense:
/usr/include/jack/jack.h
/usr/include/jack/weakjack.h
I've used 'jack_transport' via QProcess successfully, but
I feel like there's a giant leap in my understanding ahead
and a simple basic example, or tutorial, would help.
--
Long time ago I 'typed in' a whole 1kB on a hexadecimal keypad
and got to play space invaders on my Tangerine Microtan 65 :)
Hey hey,
I have a problem. I'm trying to understand the Waldorf Microwave 2/xt's
waveform data dump. The SysEx manual seems clear and direct enough, but I
can't get anywhere.
In short: these synthesizers can load 8bit single cycle waveforms. Only a half
cycle is sent, the second half is created by "mirroring"/negating the first
half:
Wave[64+n] = -Wave[63-n]
To store 8bit samples in the SysEx format, which can only use 7bit, the
samples were split into nibbles. Here's what the SysEx manual says about
conversion to a signed char:
"... Not(e) that samples are not two's complement format, to get a signed
byte, the most significant bit must be flipped:
signed char s = Wave[n]^0x80; ..."
The whole wave dump format is defined as: "f0 3e 0e Dev_id 12 location_1
location_2 "128 bytes of wave data" chksum f7"
Wave Data:
index range description
0 0-0f sample1, most significant nibble
1 0-0f sample 1, least significant nibble
...
Knowing all that, I tried a few things, but still couldn't get any sensible
result. Here's a wave dump of wave 1, which I assume is the sine wave
Complete SysEx dump:
https://www.dropbox.com/s/v50wq3x55bh52l8/full_wav_dump.syx
Only the wave data (128 bytes):
https://www.dropbox.com/s/q5rcm3cvtfp8cdu/pure_wave_data.dat
Could someone please help me on how to convert one sample from this data and
vice versa, please?
Best wishes,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
And I love the way with just one whisper
You tell me everything <3
(Britney Spears)
Hey hey,
I've used libsamplerate through the src_simple interface to resample a
audiofile of a single cycle wave. The output does not look right though:
In the input values go from positive to negative floats. In the output values
go from small positive floats to big positive floats. The process itself is
error free and both used and generated frames are exactly to my expectations.
Please, does anyone have an idea why this could be? Or if this is an error at
all?
Best wishes,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
If only I could trade the fancy cars
For a chance today,
it's incomparable <3
(Britney Spears)
I need a command line sound file player which I can somehow control,
while playing, to go immediately to a new position in the same file
and keep playing without missing a beat. And a way to pause/continue.
I'm currently using sox (from a Qt QProcess) and I can stop it with
a 'kill' QProcess, but I seem to find that 'kill one start another'
leaves some overlap (but it could be me).
Thanks for any thoughts. If it could also output feedback on the
current play position, it would save me tracking the position in Qt
with a timer, which works (and looks quite good on a QLCDNumber) but...
--
Thanks again, John.