Stereo Tool

No with 6.21 it was around 17-18% and i just started the old exe with current settings it have, after closing the new one. Also tried with 6.32 - it was around 23-24% . New one has around 31-32% (only sometimes it`s a bit lower to 28%) . So from 6.21 to 7.42 it looks like double the load . Settings of the different versions are identical.
Yes i always prefer the last versions because of lot of good options like MB drive that not exist in older ones. Also target level per band.
If i make some changes on last version in Advanced clipper i can make the load like in 6.32 and maximum modulation peaks around 104% , which is not that bad.


Edit: Here is the real settings

I always wonder why there is no MB options for speed like in AGC . I think that this will be very good option to have speed db/s - from 0 to 200 ? may be to 300 . And if this option is selected , others to be disabled - like speed sliders that are now with all the Non-standart and may some other sliders. Is this even possible? I think yes.

Last beta and some error for about few hours.

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Urban Junglist

STAND ALONE: http://www.stereotool.com/download/ster ... 43-006.exe

FIXED:
- Fixed one weird hangup in VLC output when passing bad parameters
- Added option to pass extra parameters to input VLC (just type them after the URL).

OLDER CHANGES:
- Carrier at +/- 19khz on the AM band. Just turning the FM off doesn't kill the stereo pilot. (fixed in 004)
- I again will say here that there is something wrong with Calibration section . Three options are identical with what they do : Test right silence , Test SSB lower and Test SSB higher. Nevermind what option you check, they causes same display behavour and same result as audio. --> NO, they are very different if you look at the actual output. Display is generated before calibration. Actual output does contain SSB or LSB audio.
- At the new Normal output settings there is a problem with "Input without processing" option . Sound is gagged/strange / I'm not at my Stereotool pc at the moment but when using a pipeline input and normal & fm output (192 kHz) , I noticed a strange effect when switching the normal output to bypass. Sounded like it could be a sampling rate thing? - Worked fine in some cases but not in others. I think it's fixed now.
- I think I also noticed a small typo - check the AM transmitter window.
- On the thorium preset it plays fine and then gets garbled and sounding like a 45rpm song playing on 33 rpm. cpu load is just 26% the rest of the presets work fine. (waiting for feedback, unable to reproduce it). --> Multipath Clipper is enabled in this preset. That explains the higher CPU load.
- VLC output still hangs on outputs that don't work (for example an invalid ShoutCast address). Probably because no data is sent through the output path, which causes the same stop hangup again. --> If Status is STOPPED, ERROR or NOTHINGSPECIAL, I don't call STOP or RELEASE. This seems to cause a small memory leak (a few kB probably) every time it happens, so it's not ideal. But better than a complete hangup. Still waiting for feedback on a question in VLC's forum.
- Add VLC output error detection & re-connect behavior (if possible without resetting the other inputs/outputs).
- VLC input: But something is strange sometimes plays like 1 or more samples delay between channels, occurs highs cancelation when switch to mono. I checked everything including exact mono sound input. -> Output was already running before connection was established. That's ok, but the start sample must be left channel -> fixed.
- VLC input with Low Quality Low Latency output hangs -> Solved by disabling LQLL output when VLC is used for input (which makes it impossible to reach low latencies anyway)
- VLC MP3 output. No room in current settings window... -> You can now add more options in the SOUT string (space separated)
4. Turn AM stuff off if FM is enabled (?) but what about asymmetrical clipping/limiting for FM? -> Extra switch in Expert mode
5. Add pre-emphasize output button for AM pre-emphasis
Fixed the mentioned and some other bugs in the previous beta. All combinations of settings seem to work now.
- Normal Output can now be partially separately processed for FM output. So you can generate, for example, a composite clipped 75 us pre-emphasized FM signal and a not or far less pre-emphasized signal for streaming, or a separate FM signal without composite clipping to feed remote transmitters, in the same Stereo Tool instance - no more need to run a separate copy for the stream! Only the clipping and limiting is done separately, so the CPU load does not go up much. Note that the lowpass filter can not be specified separately (lowpass filtering happens much earlier in the sequence and using different values would require doing almost all the processing twice).
- AM section now contains an 'AM Transmitter' Enabled switch.
- Headroom slider added to the AM section. No more need to abuse the Post Amp slider to create headroom for asymmetrical output. (Note: It's also not possible anymore, so you need to update your settings if you use AM).
- AM pre-emphasis checkbox added in the AM section, so you don't have to abuse the FM pre-emphasis anymore.
- AM pre-emphasis curve (which is slightly different from the FM curve) is now correctly implemented.
- AM signal can now go through the FM sound card output (it's now basically an FM/AM sound card), including tilt correction etc.

Bugs reported and some todo's:
- Show Lowpass filter freq in AM panel?
- C-QUAM?

More details please? Were you using VLC for input and/or output? Were you doing something special when this happened? DSP or stand alone version? etc.

Stand alone version, VLC input and normal output with pre-emph, 2 core AMD CPU, XP home, ST CPU load ~90-95%.
It's something like transmitter site PC for tests, no other software, just plays and I don't touch if don't need.
I was only 1 time.

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Urban Junglist

VLC output is just pain in the ass, have you considered libav? (not that I know what are the requirements that make usage possible, libav just seems more ready for these things, and a command-line tool is available, avconv)

Why is it a pain in the ass? Once you have an encoding string, it seems to work very quickly (and I will of course put some example encoding strings on my site in the help pages).

I checked libav and found that it has a server component called avserver: https://libav.org/avserver.html
The first line of this description is:
WARNING: avserver is unmaintained, largely broken and in need of a complete rewrite. It probably won’t work for you. Use at your own risk.

So that doesn't really seem to be a good option (for now anyway).

Have you run the same tests with older versions (7.40 and later, since 7.40 is where I added VLC input support)?

Is it possible that this happened after a dropout of VLC? Can you try what happens if you press the 'RESTART SOUND CARDS' button a bunch of times?

Latest = BETA006? I have actually added protection against VLC crashes in BETA005, are you 100% sure that this wasn't an older beta (BETA004 for example)? Please check the date of the file... Someone complained about VLC crashes after a signal (stream) drop, I added protection against crashes in VLC in BETA005 and so far this other person has been unable to reproduce it (and before I added the protection he could). This may sound like a stupid question, but it wouldn't be the first time that someone reports a bug and says to be using the latest version but is actually using an older one...

If you can reproduce it, please send me (if it's accessible for me) the VLC input string that you're using (via a PM if you don't want others to connect).

Yes, you're correct. It works fine as long as you have found the golden path. Btw have you tried to send MPX with VLC? I haven't been successful yet, made my own Python script with pyaudio to grab MPX from VAC and send over UDP to anywhere. Normal output works as is.