Stereo Tool

To many flat tops small and big through phase rotation filter makes audible artifacts on loud voice.
With external phase rotator plugin before ST is ok.

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Urban Junglist

Latest Linux CLI version works great on my "crappy" Debian system

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I just use ST with liquidsoap to enhance a streaming radio station I like...

Would be good to move the Phase Rotator ...

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by GAP
"Less is More" (Bob Katz)

BETA724-037 (RELEASE CANDIDATE)
Windows stand alone: http://www.stereotool.com/download/ster ... 24-037.exe
Winamp DSP: http://www.stereotool.com/download/dsp_ ... 24-037.exe
Windows VST: http://www.stereotool.com/download/vst_ ... 24-037.dll
Windows command line: http://www.stereotool.com/download/ster ... 24-037.exe
LINUX stand alone: http://www.stereotool.com/download/ster ... ETA724-037
LINUX command line: http://www.stereotool.com/download/ster ... ETA724-037

DONE:
- Linux: cranil7 crash fixed
- Hangup reported by Bojcha -> Input Close() was doing things in the wrong order!
- Full audio restart if there's a Normal or FM output buffer underrun in the stand alone version!
- Wunjo: Input gain sometimes set to -99 dB on preset load. - For now added safety override that doesn't allow it to get below 0 dB.
- Check BS412 compliance of new FM presets
- Change version number
- Linux: Remove GLIBC_2.15 dependency (Debian is still on 2.13), now only dependent on 2.11).
- Windows: Plugin Latency correction does not work - fixed.
- Windows: DSP/VST FM output broken (too low sample rate)
- Add presets
- Restart behavior still not ok: Input buffer too full doesn't cause a restart -> black bar in input now causes a restart. Unless it decreases rapidly.
- Linux: Final check with valgrind on test system
- Linux: Jack buffers 'leak' on startup. -> Immediate restart added after startup.
- Linux: Keep sound card settings if Jack not available or output not available.
- Linux: Buffer filling now always same after restart. Not sure yet why, on Windows it's fine. There is some code that I disabled in Linux because it caused issues - I'm starting to think I do still need it. --> Now fixed and working perfectly (and better than ever before) on both Windows and Linux!
- Windows: Bojcha reports that restarting after buffer empty doesn't work properly. Might be fixed.
- Low latency audio quality improved (turned off phase rotation, which had a very bad effect on the quality).
- Linux: Apparently outputs are mixed (see description from Radiofreak).
- Linux: Sound card buffer display incorrect
- Low latency is not de-emphasized when Normal output is disabled
- Linux crash with specific settings - also affected Windows
- Windows: 24 bit sound card - Already 24/32 bit when using ASIO, and 24/32 bits is not supported in WINMM mode!
- Windows: Channel swapping in Windows reported by Bojcha -> Threading error, fixed.
- ^C/^V/other keys don't work in SAM...
- All: Low latency sound sometimes bad (VERY bad) - also in older versions. - was a resampling error (44.1 in, 176.4 out)
- Crackling and static in low latency output
- Loading and saving .sts file (and hence also starting and closing of Stereo Tool) is a lot faster.
- Entering registration key changed, and there's an overview now of which features are registered. See Configuration -> License
- AM stations: Added asymmetric clipping
- Slightly improved the mouse hold (slow slider movements) vs. non-hold detection.
- Rewrote all OS specific code for GUI and sound cards.
- Errors are now displayed in sound card bars when they are red.
- Bug fix: With Multicore disabled, Normal output doesn't work in the stand alone version.
- Low latency monitoring in ASIO wasn't really low latency - fixed

TODO:
- Release

- Wunjo: Input gain sometimes set to -99 dB on preset load. (Safety override already added)
- Wunjo: VLC new track Jack connection lost -> auto-reconnect needed.
- Bojcha: If possible to add on list fix for MB compressors meters (and might possible audio) interact wrongly with Burst compressors AND limiters.
- Linux: RDS time zone detection (radiofreak)

Hi Hans. Is the non-Asio 24 bit support a windows limitation?

If an AoIP driver is used for the input sound device (Axia, Logitek etc - typically 24 bit 48 kHz systems with a nominal operating level of -20dbfs), then for FM broadcast we use a different output device (Julia, Marian etc). The catch 22 currently seems to be that if we select the ASIO mode to allow the 24 bit resolution required to properly support those input levels, we then lose the ability to use the ST "synchronise with different output sound card" function which does not support ASIO.

Any thoughts on this? Thanks.

You're using the AoIP for input with ASIO and non-ASIO output to the sound card? I guess I should add support for synchronizing that as well then!

In our application I don't think (?) it likely matters if the input and/or output is ASIO or not - Axia AoIP and Julia drivers support both. As long as I can select a mode with 24 bit input resolution and also have output synchronisation active, that would be good.

Sometimes Display is out of synchronization and i must reload the settings to get it in touch with the music . I don`t know why is that .And after some CPU load or some operations i see that display is again out of synchro. Talk for VST plugin . is this reproducing ? I guess that if buffer fills to the max after load settings then display is not in synchro. When is to the half filled then is in synchro. But every time the buffer is filling differentially , sometimes full , sometimes half . Hans ?


I other words : Synchronize with Fm output is not working ?! At least not every time?!
In plug-in panel checkbox

The URLs to the last Linux-Betas are dead...

Sorry, they are working now!