Stereo Tool

I don't really understand why, but the asymmetrically clipped audio is ok now. Output is good. I still need to add a limiter to really ensure that the signal doesn't surpass -75%, but for now it all looks fine (haven't seen overshoots of more than 3% yet).

I would say this version is good enough for testing... (don't put it on a big transmitter yet, until I've added the 75% limiter).

Will build it & post it tomorrow.


Image is attached which shows Left and Right audio (asymmetrical, bottom clipped around -75%, as you can see there are some bigger overshoots so I really need to add a limiter); L+R audio (100% vs 125% modulation; 100% is 50% in this screenshot), L-R audio (maximum is 100% modulation, 50% in this image). Except for the spikes at the bottom in the top 2 images I think this is perfect.

Not if you can help this simple circuit ...

_________________
by GAP
"Less is More" (Bob Katz)

I'll give you some background on the report I filed earlier today. It started when I thought I could hear a difference in the audio listening to a stream with VLC => HiFi Cables => Stereo Tool and using the built in VLC in Stereo Tool. So I opened VLC on my desktop, got the stream playing and tried to do an A/B by switching the inputs in Stereo Tool. It was fine switching from HiFi Cables to internal VLC, but switching back broke Stereo Tool. I wasn't able to restart the audio card. I had to kill it in the task manager and restart, then I was back in business.

I'll do some more testing tomorrow. I'm not calling this a bug, but an interesting finding.

I enjoy Stereo Tool. Keep up the good work!

Just tried it again and this time it did a relaunch. I have restart on crashes checked.

STAND ALONE: http://www.stereotool.com/download/ster ... 43-011.exe
WINAMP/DSP: http://www.stereotool.com/download/dsp_ ... 43-011.exe

CHANGED:
- Added C-QUAM clipper; HARD LIMIT still needs to be adjusted to remove some peaks that are left in L and R (to the negative side). Should be usable for small scale tests now.
- Added Event FM License (lower price, works 3 days at a time)
- Added auto-start with Windows option
- Added tests for exception handling

TO DO:
- C-QUAM: Adjust HARD LIMIT.
- Fix crash reported by bhelbush (it's in the AddNextBuffer function when called in the AudioRecorder thread)
- Show Lowpass filter freq in AM panel?

OLDER CHANGES:
Improved limiter behavior. The limiter (in all compressors) now works much tighter so it causes less pumping and at the same time it causes less distortion.
- Support ASIO as well in the watchdog
- Improve restart behavior (make sure that the old instance is really closed).
- Memory usage of stand alone version reduced.
- Merged in older changes in a separate branch. should not have any noticeable effects.
- Lower CPU load when using streaming with low pre-emphasis values. "Hole protection" must be set to LQ mode if it's disabled for non-preemphasized audio.
- Make SCA frequencies configurable for HAM radio
- Real watchdog behavior: Regularly check for sound card input & output
- Added restart-on-crash behavior (enable 'Watchdog' in the Configuration panel).
- 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).
- 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.

This watchdog behavior works as follows: If you enable the check, it will act if no audio passes through an enabled audio input or output for more than 1 minute. If so, it will trigger a 'RESTART SOUND CARDS' button click. It will attempt this 4 times, if NO DATA passes through the sound card AT ALL during all these attempts, if the 'Restart' watchdog setting is enabled Stereo Tool will close and a new instance will be started.

Did you have 'restart on crashes' enabled the first time when this happened, and it didn't restart?

You write you weren't able to restart the audio card - was Stereo Tool hanging completely at that point or did that happen after you pressed the RESTART button? Since you say you had to kill it in Task Manager.


Also I'm a bit confused. The input buffer in your screenshot is shown in blue, which means that normal audio (not from VLC) should have been running. At least it should mean that the sound card was opened succesfully (I think - will check the code). But based on what you write it was dead completely?


And you can reproduce it right? Then I can try adding more info to the exception so I can see *where* it crashes (I have just looked through the code and I don't see any places where I missed handling VLC call exceptions locally - but you still see one so it seems to be coming from somewhere else; I think I need to add function info to the crash info)...

Hans,

One thing I've noticed with less-than brick-wall aggressive processing, is that the AGC will noticeably pull the drive to the multiband section (its job, yes?). I imagine this was the purpose of the EQ section in your processor, but it is still noticeable to me.

A popular leveler uses a 4-band, frequency dependent AGC, for placement ahead of a multiband section.

Would something like this be a worthwhile option to the wideband AGC that Stereo Tool uses?

Thanks for considering.

-Jack

The AGC has up to 3 bands (bass, mids, highs) or do you mean something else?

Unless I've missed how to set it up, the 3 "bands" in the AGC appear to allow different time constants on the same wideband section.

I'm referencing the functionality of the "Ariane" leveler.

It seems to be nearly impossible or at least difficult to correctly log the actual exception in C++ code (unless I'm going to use OS and compiler specific features, which would mean that it won't work on Linux or Mac etc.).

For now, I'm just going to make sure it logs a more precise location of the crash so I can figure out where it happened (function names instead of thread names). The amount of code under 'AudioRecorder' is pretty small so it should be possible to drill down to the cause in a few iterations...

Question on the AM operation : When running normal AM, will the Left and Right inputs be combined into mono at the sound card output? And a question not exclusive to AM, is it possible to have the automation system and ST on 1 computer and use only the onboard sound card? That is : Automation>Sound card in>ST>sound card out or will I need two sound cards? It seems that Breakaway had a "pipeline" that showed up on my automation (Station Playlist) that may or may not have made this possible.