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.
Hey hey,
I'm trying to convert a single cycle wave to a harmonic spectrum. I know that
in theory this is a job for standard FFT. I wondered if there isn't a
"simpler" way given a few assumptions about the input.
The input waveform is exactly one single cycle
The waveform has only harmonic overtones.
That means that only integral multiples of the base frequency need to be
considered and the specific Nyquist frequency is determined by the number of
samples (I think).
I treid searching, but it appears that I don't have the right keyword to hand,
getting only very general results bordering on the off-topic. :)
Can anyone help? A good keyword, name of an algorithm or a general name for
this specific task would be helpful enough, I suppose.
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
She's so lucky,
she's a star
But she cry, cry, cries in her lonely heart... <3
(Britney Spears)
Hello all,
zita-njbridge 0.4.8 is now available at the usual place.
Changes:
* added --ipv4 and --ipv6 options
* removed some unused code
* some minor fixes
Ciao,
--
FA