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PostPosted: Mon Mar 20, 2017 1:04 pm 
Site Admin
User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 8891
Windows 32 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-031.exe
Winamp DSP: http://www.stereotool.com/download/dsp_ ... 25-031.exe
VST: http://www.stereotool.com/download/vst_ ... 25-031.dll

Windows 64 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-031.exe
VST: ----://www.stereotool.com/download/vst_stereo_ ... 25-031.dll (NOT BUILT)

CHANGES:
- Fixed slider behavior for values < 0 (clicking left/right, adjusting with cursor keys or mouse wheel, minimum value was not reachable)
- Fixed 'Flat tops' mode when 'Adjust slopes' is disabled.(left as-is for when this is enabled for compatibility reasons)
- Fixed weird jumps in band behavior when 'Adjust slopes' is disabled (left as-is for when this is enabled for compatibility reasons)
- Change version number

OLD CHANGES:
- Try new limiter mechanism - to test it disable 'Legacy mode' in the lmiter settings
- Linux plugin: Close and resize don't work.
- Linux plugin: Window close should call a callback in the Generic plugin.
- Change slope calculation in MB??? Wait for feedback from Wes and Matt --> Yes, change it
- Fix for compiler bug (crash in 026)
- SST: Add widget with text for version number, preset name etc.
- Font stuff: Głośniki - display as Glosniki (file in C:\temp\hd_v7.sts)
- Linux version no GUI in Generic plugin - Rivendell. - Probably fixed (more tests needed)
- Put Wasapi cards on top, KS below that. Or, better, order them alphabetically!
- Fix mono Bypass output and drops.
- Fixed crash when switching to/from Bypass mode (threading issue)
- Make 'Ignore high frequencies' enabled by default on ARM.
- Add WASAPI mode that uses exclusive sound card control to set sample rate.
- Add real PEQ, with low/high shelves
- Put PEQ in SST interface
- Fixed bug in Bypass code that might have caused all kinds of weird things.
- Natural Dynamics: Changed the dB value calculation for "Reduce for jump above" (display only); changed default setting to 6 dB (instead of 13). This reduces the effect of ND if the input signal is more dynamic. With some more tweaks I can probably make it sound better.
- Raspberry Pi version: Stokkemask causes silence.
- Hard Limit: Check report of overshoots - Solved, at least for any logical situations
- Fixed daylight saving time bug introduced in 8.23.
- PFFFT resampler!!! YEAH! 20% CPU load reduction on the Pi :)
- Generic plugin in debug mode shows PEQ too often assert, why?
- In the AM configuration When we use Asymmetry, the output markers show the red color all the time even if with fm transmitter it continues to generate pilot tone at the AM output. As you can see in the attached figure. Thank you for your attention .... -> Fixed pilot issue, restored Headroom slider.
- It appears when RDS RT Text is sent to StereoTool and it contains a “/” that ST separates it into two messages. Is this how you intended it to work? PS Text doesn’t appear to be affected. -> Fixed (but PS should also have been affected, odd).
- The DSP-Plugin still works correct. But somehow some trial features are activated when i am using the commandlinetool. What you have to know is this, that i am also totaly blind so i cant use the interface for configuring. I allways thought that the commandlineversion uses the DSP_Stereo_Tool.ini created by the winamp-plugin to set the different options. I think something changed cause i am getting the message that the program cant find or cant read the dsp_stereo_tool settings in the user-folder. So the stereotool_cmd.exe uses some settings wich are not completely available in the free Version. So i am getting this two beeps every minute. -> Oops, it was accidentally using the .ini file without .ini at the end... so not the Winamp settings file (which is the default).
- Linux and Raspberry versions: Now linking PortAudio statically.
- Make PortAudio work properly on Linux - it works but with weird warnings and it takes a long time to start. -> Fixed by removing sample rate check on startup. --> Appears to be a PortAudio ALSA backend issue!!! For now, keep the Jack version as well.
- Allow Linux version no GUI to run on Raspberry Pi without X11. (Hm, Windows version probably too - already basically there for service). - TODO: Test if it really works
- Fixed: Raspberry Pi version: AGC, Multiband 1 & 2 and some other filters can cause dropsouts
- - Fixed: Raspberry Pi version: Clipper can cause silence
- Fixed: All versions: Using Composite Clipper with Strictness 1 or 2 + Hard Limit can cause weird drops in the audio. Fixed, Hard Limit works different from before now! (and sounds better! Hint: Use this same trick for all limiters!!!!!!)
- PortAudio linked statically on Mac
- GiPHouse build warnings
- PortAudio weird jumps in buffer filling and glitches
- Put all PortAudio calls in 1 thread - was needed for WASAPI
- Made PortAudio work on Mac - weird hack needed (?)
- Synchronize PortAudio input and output starts
- Make sure that in case of no input audio, the error is reported for the input. Added the watchdog tick to be always called when the callback is called, even if no audio is being played.
- Make buffer filling 50% on start - Wait for first input to return. Only then allow outputs. Also needed for VLC (so the delay can be big!) In fact, the best way might be to connect it to when the first callback of the input sound card happens... enable the other callbacks INSIDE that input callback. But if I do that I need special solutions for VLC and ASIO. And that won't take the calculation time into account, and it might actually be a good thing to take that into account to make the buffer at least not too empty.
- Updated PortAudio to latest version
- Added FM/HD synchronziation slider. You will hear glitches when adjusting it, but no complete dropouts as with changing the buffer size.
- Changed PortAudio callback block size to let PortAudio decide the best size. This might slightly reduce latency and also help against hiccups.
- Added display of supported sample rates (and MME) to sound card selection.
- SST help
- Delossifier Spectral hole filler CPU load is much higher in 8.23 than in 8.00 - no obvious change in the code. - Wasn't true, slightly optimized it anyway.
- Removed old WINMM audio card code, replaced by PortAudio.

TODO:
- Change default Limiter settings in MB
- Add and update presets
- Fix Declipper display when Threads > 1 -> Buffers were too small and data was getting overwritten before it was displayed.
- Check new preset "soft" levels
- WASAPI input glitches (reported by 1 person)
- Fix Declipper display in plugin mode when Threads > 1. --> Seems kinda ok now, no idea why
- Remove some presets
- Add Wes' latest preset - maybe remove older?
- Implement EXTRA auto-tilt correction

- Test if FM lossy link protection improves reception!!! -> Yes! (But it does also reduce audio quality quite a lot).
- Possibly also lower latency (based on what I read it should be possible to get close to ASIO latency - thread priority must be set too in that case, both with a flag and a value).
- Check if BASS is usable for streaming...
- Limiters: Make same change as in Hard Limit (from l + l * .01 to l + .01).
- Compare Stokkemask vs Multipath Clipper
- Maybe support ISO 226 curves for AGC and other side chains instead of ITU1770? See http://www.sengpielaudio.com/Acoustics226-2003.pdf ; curves look similar to the "old" curves but less to the new ones.
- Send out RDS texts
- Fix 31N
- Show low input level warning for inputs below - say - around -9 dB or so. For extended periods. Reason; I hear on a local station here that they probably have a too low input level and they didn;'t know that that was a problemn, but the sound on one of the built-in presets wasn't as consistent as it should be, and some tracks sound muffled probably due to too much noise gating.


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PostPosted: Mon Mar 20, 2017 6:10 pm 

Joined: Sun Feb 03, 2013 2:39 pm
Posts: 237
Thanks, everything fixed except this:

Without flat tops the slopes are still not following the settings. They are now to steep for distances above an octave.

Use only 2 far away bands e.g. 100 Hz and 10 kHz, no flat tops, automatic steepness correction off. The slope will be far off the given one. Also try moving the bands frequency interactively. You'll see the curves are not only shifting to one side but also slightly changing form. If the bands are closer together, e.g. 500 Hz and 2000 Hz, the slope will be (near?) correct.Image


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PostPosted: Mon Mar 20, 2017 6:32 pm 
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User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 8891
Ok. Now this one makes sense (and isn't easy to fix). If you set 'Flat frequency response' to 100%, you'll see that the outer lines are nearly perfect (except that at a large distance the crossover can move below -6 dB). The problem is that to be neutral when there's no compressor action, the sum of all the bands must be 1 (or if you prefer, 0 dB). And the slopes are compensated for that. In this extreme case, the compensation is also extreme... and that's why the slopes are too steep - they are calculated correctly (as the 'FFR' shows) but then adjusted to be flat.

Since this is an extreme case, and not easy to solve, I'll leave it for now.


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PostPosted: Tue Mar 21, 2017 12:24 pm 
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Joined: Thu Oct 25, 2012 3:22 pm
Posts: 216
Location: Poland
I think, band filters should be changed to adjust crossing point instead actual solution.

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PostPosted: Tue Mar 21, 2017 2:51 pm 

Joined: Sun Feb 03, 2013 2:39 pm
Posts: 237
@hans
OK, though it's a bit funny that the slopes were not steep enough with automatic correction and are now too steep (in such extreme situations).

Btw what's the 'natural' distance for two bands (without flat tops). With automatic steepness correction it seems to be one octave, i.e. (band1, band2) = (fc/sqrt(2), sqrt(2)*fc) Hz which will reach the given slope + 6 dB at (fc/2, 2*fc) Hz without further manual correction. However without automatic steepness correction I just found out that setting band1 = band2 = fc will exactly reach the given slope.

So can we just turn off automatic steepness correction and set two bands (without flat tops) to the same value and get a 'perfect' crossover there with everything following the settings? Or are there any drawbacks?


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PostPosted: Wed Mar 22, 2017 6:12 am 
Site Admin
User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 8891
Windows 32 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-033.exe
Winamp DSP: http://www.stereotool.com/download/dsp_ ... 25-033.exe
VST: http://www.stereotool.com/download/vst_ ... 25-033.dll

Windows 64 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-033.exe
VST: http://www.stereotool.com/download/vst_ ... 25-033.dll

CHANGES:
- Fix Declipper display when Threads > 1 -> Buffers were too small and data was getting overwritten before it was displayed.
- Change default Limiter settings in MB
- Add and update presets
- Check new preset "soft" levels

OLD CHANGES:
- Fixed slider behavior for values < 0 (clicking left/right, adjusting with cursor keys or mouse wheel, minimum value was not reachable)
- Fixed 'Flat tops' mode when 'Adjust slopes' is disabled.(left as-is for when this is enabled for compatibility reasons)
- Fixed weird jumps in band behavior when 'Adjust slopes' is disabled (left as-is for when this is enabled for compatibility reasons)
- Change version number
- Try new limiter mechanism - to test it disable 'Legacy mode' in the lmiter settings
- Linux plugin: Close and resize don't work.
- Linux plugin: Window close should call a callback in the Generic plugin.
- Change slope calculation in MB??? Wait for feedback from Wes and Matt --> Yes, change it
- Fix for compiler bug (crash in 026)
- SST: Add widget with text for version number, preset name etc.
- Font stuff: Głośniki - display as Glosniki (file in C:\temp\hd_v7.sts)
- Linux version no GUI in Generic plugin - Rivendell. - Probably fixed (more tests needed)
- Put Wasapi cards on top, KS below that. Or, better, order them alphabetically!
- Fix mono Bypass output and drops.
- Fixed crash when switching to/from Bypass mode (threading issue)
- Make 'Ignore high frequencies' enabled by default on ARM.
- Add WASAPI mode that uses exclusive sound card control to set sample rate.
- Add real PEQ, with low/high shelves
- Put PEQ in SST interface
- Fixed bug in Bypass code that might have caused all kinds of weird things.
- Natural Dynamics: Changed the dB value calculation for "Reduce for jump above" (display only); changed default setting to 6 dB (instead of 13). This reduces the effect of ND if the input signal is more dynamic. With some more tweaks I can probably make it sound better.
- Raspberry Pi version: Stokkemask causes silence.
- Hard Limit: Check report of overshoots - Solved, at least for any logical situations
- Fixed daylight saving time bug introduced in 8.23.
- PFFFT resampler!!! YEAH! 20% CPU load reduction on the Pi :)
- Generic plugin in debug mode shows PEQ too often assert, why?
- In the AM configuration When we use Asymmetry, the output markers show the red color all the time even if with fm transmitter it continues to generate pilot tone at the AM output. As you can see in the attached figure. Thank you for your attention .... -> Fixed pilot issue, restored Headroom slider.
- It appears when RDS RT Text is sent to StereoTool and it contains a “/” that ST separates it into two messages. Is this how you intended it to work? PS Text doesn’t appear to be affected. -> Fixed (but PS should also have been affected, odd).
- The DSP-Plugin still works correct. But somehow some trial features are activated when i am using the commandlinetool. What you have to know is this, that i am also totaly blind so i cant use the interface for configuring. I allways thought that the commandlineversion uses the DSP_Stereo_Tool.ini created by the winamp-plugin to set the different options. I think something changed cause i am getting the message that the program cant find or cant read the dsp_stereo_tool settings in the user-folder. So the stereotool_cmd.exe uses some settings wich are not completely available in the free Version. So i am getting this two beeps every minute. -> Oops, it was accidentally using the .ini file without .ini at the end... so not the Winamp settings file (which is the default).
- Linux and Raspberry versions: Now linking PortAudio statically.
- Make PortAudio work properly on Linux - it works but with weird warnings and it takes a long time to start. -> Fixed by removing sample rate check on startup. --> Appears to be a PortAudio ALSA backend issue!!! For now, keep the Jack version as well.
- Allow Linux version no GUI to run on Raspberry Pi without X11. (Hm, Windows version probably too - already basically there for service). - TODO: Test if it really works
- Fixed: Raspberry Pi version: AGC, Multiband 1 & 2 and some other filters can cause dropsouts
- - Fixed: Raspberry Pi version: Clipper can cause silence
- Fixed: All versions: Using Composite Clipper with Strictness 1 or 2 + Hard Limit can cause weird drops in the audio. Fixed, Hard Limit works different from before now! (and sounds better! Hint: Use this same trick for all limiters!!!!!!)
- PortAudio linked statically on Mac
- GiPHouse build warnings
- PortAudio weird jumps in buffer filling and glitches
- Put all PortAudio calls in 1 thread - was needed for WASAPI
- Made PortAudio work on Mac - weird hack needed (?)
- Synchronize PortAudio input and output starts
- Make sure that in case of no input audio, the error is reported for the input. Added the watchdog tick to be always called when the callback is called, even if no audio is being played.
- Make buffer filling 50% on start - Wait for first input to return. Only then allow outputs. Also needed for VLC (so the delay can be big!) In fact, the best way might be to connect it to when the first callback of the input sound card happens... enable the other callbacks INSIDE that input callback. But if I do that I need special solutions for VLC and ASIO. And that won't take the calculation time into account, and it might actually be a good thing to take that into account to make the buffer at least not too empty.
- Updated PortAudio to latest version
- Added FM/HD synchronziation slider. You will hear glitches when adjusting it, but no complete dropouts as with changing the buffer size.
- Changed PortAudio callback block size to let PortAudio decide the best size. This might slightly reduce latency and also help against hiccups.
- Added display of supported sample rates (and MME) to sound card selection.
- SST help
- Delossifier Spectral hole filler CPU load is much higher in 8.23 than in 8.00 - no obvious change in the code. - Wasn't true, slightly optimized it anyway.
- Removed old WINMM audio card code, replaced by PortAudio.

TODO:
- Fix Declipper display in plugin mode when Threads > 1. --> Seems kinda ok now, no idea why
- WASAPI input glitches (reported by "Primary User", waiting for feedback)
- Remove some presets
- Compare sound against 8.23 (ok)
- Add Wes' latest preset - maybe remove older? Also put in SST.
- Try making dynamic PNR -> Done, test if it works, add warning that it's HEAVY.
- SST fix noncompliant presets
- SST quick setup screen: Loudness - Just a copy of the clipper drive slider / Aggressiveness - Slider that controls the attack and release speed multipliers of both multibands. Slider that increases or decreases the density of both multibands (without affecting the volume) / Bass - A slider for Bass In Your Face, A slider for MB band mix of band 1/2 (taking band frrequencies into account), or maybe an equalizer after the MB's. A slider for MB band density for bands 1/2 (taking band frrequencies into account), without affecting the volume. / Highs - Highest MB bands extra drive and density.
- MicroMPX resampler! Eng...
- Implement EXTRA auto-tilt correction
- Re-order presets
- Release Declipper plugin with BaOne service mode fix

- Test if FM lossy link protection improves reception!!! -> Yes! (But it does also reduce audio quality quite a lot).
- Possibly also lower latency (based on what I read it should be possible to get close to ASIO latency - thread priority must be set too in that case, both with a flag and a value).
- Check if BASS is usable for streaming...
- Limiters: Make same change as in Hard Limit (from l + l * .01 to l + .01).
- Compare Stokkemask vs Multipath Clipper
- Maybe support ISO 226 curves for AGC and other side chains instead of ITU1770? See http://www.sengpielaudio.com/Acoustics226-2003.pdf ; curves look similar to the "old" curves but less to the new ones.
- Send out RDS texts
- Fix 31N
- Show low input level warning for inputs below - say - around -9 dB or so. For extended periods. Reason; I hear on a local station here that they probably have a too low input level and they didn;'t know that that was a problemn, but the sound on one of the built-in presets wasn't as consistent as it should be, and some tracks sound muffled probably due to too much noise gating.


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PostPosted: Wed Mar 22, 2017 8:38 am 

Joined: Sun Feb 03, 2013 2:39 pm
Posts: 237
hvz wrote: - Change default Limiter settings in MB
Just for safety: Yes you did, but this is for the legacy limiter because the new one is off.

You also changed the legacy 'Flat frequency response' value and the RMS block sizes. Just mentioning it for you accidently did so because you normally don't touch these old settings.


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PostPosted: Wed Mar 22, 2017 3:19 pm 
User avatar

Joined: Thu Oct 25, 2012 3:22 pm
Posts: 216
Location: Poland
I've asked before, if you would be able to add ^2 to digital compressors in multiband ii will be very nice.

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PostPosted: Wed Mar 22, 2017 11:32 pm 
Site Admin
User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 8891
Windows 32 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-034.exe
Winamp DSP: http://www.stereotool.com/download/dsp_ ... 25-034.exe
VST: http://www.stereotool.com/download/vst_ ... 25-034.dll

Windows 64 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-034.exe
VST: http://www.stereotool.com/download/vst_ ... 25-034.dll

CHANGES:
- Fix Declipper display in plugin mode when Threads > 1. --> Seems kinda ok now, no idea why
- WASAPI input glitches (reported by "Primary User", waiting for feedback)
- Added "Continuous learning" mode to Dehummer - EXTREMELY HIGH CPU LOAD!!! (But if it works I can optimize it later)
- Remove some presets
- Compare sound against 8.23 (ok)
- Add Wes' latest preset - maybe remove older? Also put in SST.

OLD CHANGES:
- Fix Declipper display when Threads > 1 -> Buffers were too small and data was getting overwritten before it was displayed.
- Change default Limiter settings in MB
- Add and update presets
- Check new preset "soft" levels
- Fixed slider behavior for values < 0 (clicking left/right, adjusting with cursor keys or mouse wheel, minimum value was not reachable)
- Fixed 'Flat tops' mode when 'Adjust slopes' is disabled.(left as-is for when this is enabled for compatibility reasons)
- Fixed weird jumps in band behavior when 'Adjust slopes' is disabled (left as-is for when this is enabled for compatibility reasons)
- Change version number
- Try new limiter mechanism - to test it disable 'Legacy mode' in the lmiter settings
- Linux plugin: Close and resize don't work.
- Linux plugin: Window close should call a callback in the Generic plugin.
- Change slope calculation in MB??? Wait for feedback from Wes and Matt --> Yes, change it
- Fix for compiler bug (crash in 026)
- SST: Add widget with text for version number, preset name etc.
- Font stuff: Głośniki - display as Glosniki (file in C:\temp\hd_v7.sts)
- Linux version no GUI in Generic plugin - Rivendell. - Probably fixed (more tests needed)
- Put Wasapi cards on top, KS below that. Or, better, order them alphabetically!
- Fix mono Bypass output and drops.
- Fixed crash when switching to/from Bypass mode (threading issue)
- Make 'Ignore high frequencies' enabled by default on ARM.
- Add WASAPI mode that uses exclusive sound card control to set sample rate.
- Add real PEQ, with low/high shelves
- Put PEQ in SST interface
- Fixed bug in Bypass code that might have caused all kinds of weird things.
- Natural Dynamics: Changed the dB value calculation for "Reduce for jump above" (display only); changed default setting to 6 dB (instead of 13). This reduces the effect of ND if the input signal is more dynamic. With some more tweaks I can probably make it sound better.
- Raspberry Pi version: Stokkemask causes silence.
- Hard Limit: Check report of overshoots - Solved, at least for any logical situations
- Fixed daylight saving time bug introduced in 8.23.
- PFFFT resampler!!! YEAH! 20% CPU load reduction on the Pi :)
- Generic plugin in debug mode shows PEQ too often assert, why?
- In the AM configuration When we use Asymmetry, the output markers show the red color all the time even if with fm transmitter it continues to generate pilot tone at the AM output. As you can see in the attached figure. Thank you for your attention .... -> Fixed pilot issue, restored Headroom slider.
- It appears when RDS RT Text is sent to StereoTool and it contains a “/” that ST separates it into two messages. Is this how you intended it to work? PS Text doesn’t appear to be affected. -> Fixed (but PS should also have been affected, odd).
- The DSP-Plugin still works correct. But somehow some trial features are activated when i am using the commandlinetool. What you have to know is this, that i am also totaly blind so i cant use the interface for configuring. I allways thought that the commandlineversion uses the DSP_Stereo_Tool.ini created by the winamp-plugin to set the different options. I think something changed cause i am getting the message that the program cant find or cant read the dsp_stereo_tool settings in the user-folder. So the stereotool_cmd.exe uses some settings wich are not completely available in the free Version. So i am getting this two beeps every minute. -> Oops, it was accidentally using the .ini file without .ini at the end... so not the Winamp settings file (which is the default).
- Linux and Raspberry versions: Now linking PortAudio statically.
- Make PortAudio work properly on Linux - it works but with weird warnings and it takes a long time to start. -> Fixed by removing sample rate check on startup. --> Appears to be a PortAudio ALSA backend issue!!! For now, keep the Jack version as well.
- Allow Linux version no GUI to run on Raspberry Pi without X11. (Hm, Windows version probably too - already basically there for service). - TODO: Test if it really works
- Fixed: Raspberry Pi version: AGC, Multiband 1 & 2 and some other filters can cause dropsouts
- - Fixed: Raspberry Pi version: Clipper can cause silence
- Fixed: All versions: Using Composite Clipper with Strictness 1 or 2 + Hard Limit can cause weird drops in the audio. Fixed, Hard Limit works different from before now! (and sounds better! Hint: Use this same trick for all limiters!!!!!!)
- PortAudio linked statically on Mac
- GiPHouse build warnings
- PortAudio weird jumps in buffer filling and glitches
- Put all PortAudio calls in 1 thread - was needed for WASAPI
- Made PortAudio work on Mac - weird hack needed (?)
- Synchronize PortAudio input and output starts
- Make sure that in case of no input audio, the error is reported for the input. Added the watchdog tick to be always called when the callback is called, even if no audio is being played.
- Make buffer filling 50% on start - Wait for first input to return. Only then allow outputs. Also needed for VLC (so the delay can be big!) In fact, the best way might be to connect it to when the first callback of the input sound card happens... enable the other callbacks INSIDE that input callback. But if I do that I need special solutions for VLC and ASIO. And that won't take the calculation time into account, and it might actually be a good thing to take that into account to make the buffer at least not too empty.
- Updated PortAudio to latest version
- Added FM/HD synchronziation slider. You will hear glitches when adjusting it, but no complete dropouts as with changing the buffer size.
- Changed PortAudio callback block size to let PortAudio decide the best size. This might slightly reduce latency and also help against hiccups.
- Added display of supported sample rates (and MME) to sound card selection.
- SST help
- Delossifier Spectral hole filler CPU load is much higher in 8.23 than in 8.00 - no obvious change in the code. - Wasn't true, slightly optimized it anyway.
- Removed old WINMM audio card code, replaced by PortAudio.

TODO:
- SST: Check SSB mode default settings -> Seems ok.
- Try making dynamic PNR -> Done, test if it works, add warning that it's HEAVY.
- SST quick setup screen: Loudness - Just a copy of the clipper drive slider / Aggressiveness - Slider that controls the attack and release speed multipliers of both multibands. Slider that increases or decreases the density of both multibands (without affecting the volume) / Bass - A slider for Bass In Your Face, A slider for MB band mix of band 1/2 (taking band frrequencies into account), or maybe an equalizer after the MB's. A slider for MB band density for bands 1/2 (taking band frrequencies into account), without affecting the volume. / Highs - Highest MB bands extra drive and density.
- SST fix noncompliant presets
- MicroMPX resampler! Eng...-> Adjusted values, logging added. Retest. -> Changed maximum speed to half the maximum adjustment speed... crude, but works.
- Low latency mode sounds BAD with vinyl, highpass filter phase linear? -> Yup was accidentally using phase linear HPF, oops!
- SST: Preset jitter in debug mode.
- SST quick setup screen: Add Bass Harmonics (Phat Bass)?
- SST quick setup screen: Replace Loudness by "Extra loudness" slider so it always starts at 0 dB if you didn't change the preset.
- Implement EXTRA auto-tilt correction
- Re-order presets
- Check if command line version can output MPX
- Release Declipper plugin with BaOne service mode fix

- SST: Figure out how to handle ND
- 9sg: Add full MONO mode.
- 9sg: Windowed pilot start/stop for AutoPilot.
- Test if FM lossy link protection improves reception!!! -> Yes! (But it does also reduce audio quality quite a lot).
- Possibly also lower latency (based on what I read it should be possible to get close to ASIO latency - thread priority must be set too in that case, both with a flag and a value).
- Check if BASS is usable for streaming...
- Limiters: Make same change as in Hard Limit (from l + l * .01 to l + .01).
- Compare Stokkemask vs Multipath Clipper
- Maybe support ISO 226 curves for AGC and other side chains instead of ITU1770? See http://www.sengpielaudio.com/Acoustics226-2003.pdf ; curves look similar to the "old" curves but less to the new ones.
- Send out RDS texts
- Fix 31N
- Show low input level warning for inputs below - say - around -9 dB or so. For extended periods. Reason; I hear on a local station here that they probably have a too low input level and they didn;'t know that that was a problemn, but the sound on one of the built-in presets wasn't as consistent as it should be, and some tracks sound muffled probably due to too much noise gating.


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PostPosted: Fri Mar 24, 2017 2:24 pm 
User avatar

Joined: Wed Nov 19, 2008 7:44 pm
Posts: 1083
Location: Bulgaria
I just try the soundcard behavour in win7 SP1 .
Results: Output
Wasapi --> not work (cannot open stream.... )
Kernel --> not work (no error)
MME --> Works

Input:
Same
Same
Same

Overall result : only MME works.
Audio is onboard Realtek ALC888 - almost latest drivers


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