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PostPosted: Thu Mar 16, 2017 1:29 am 

Joined: Sun Jan 04, 2009 2:54 pm
Posts: 169
Location: Finland
Quote:
I`m on XP- may be the driver is not the same
mine is 5.10.0.7513
Well that's your problem then, WASAPI is available from Windows Vista onwards.


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PostPosted: Thu Mar 16, 2017 2:48 am 
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User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 11185
Windows 32 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-026.exe
Winamp DSP: http://www.stereotool.com/download/dsp_ ... 25-026.exe
VST: http://www.stereotool.com/download/vst_ ... 25-026.dll

Windows 64 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-026.exe
VST: http://www.stereotool.com/download/vst_ ... 25-026.dll

CHANGES:
- 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!

OLD CHANGES:
- 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:
- GiP recordings
- Add widget with text for version number, preset name etc.
- Linux plugin: Close and resize don't work.
- Add and update presets
- Change slope calculation in MB??? Wait for feedback from Wes and Matt --> Yes, change it
- Change version number
- 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: Thu Mar 16, 2017 3:06 am 

Joined: Wed Nov 20, 2013 6:42 pm
Posts: 54
32 bit standalone crashes every 8 secs or so, even in bypass mode. Last beta worked great. Running Windows 10 home 64bit.

Emailing screen shot.


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PostPosted: Thu Mar 16, 2017 8:28 am 
User avatar

Joined: Thu Oct 25, 2012 3:22 pm
Posts: 237
Location: Poland
Bad news, Now I don't have sound from VLC or something else is wrong, I hear and see also somo tone and crash:

Image

_________________
Urban Junglist


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PostPosted: Thu Mar 16, 2017 11:38 am 
Site Admin
User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 11185
Ok, good news: It reproduces here too. Looks like a memory alignment issue (which would be odd). Will try to fix it asap.


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PostPosted: Thu Mar 16, 2017 11:39 am 
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Joined: Wed Nov 19, 2008 7:44 pm
Posts: 1164
Location: Bulgaria
beta26 on startup


Attachments:
error333.JPG
error333.JPG [ 16.8 KiB | Viewed 4375 times ]
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PostPosted: Thu Mar 16, 2017 11:42 am 
Site Admin
User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 11185
Ok, good news: It reproduces here too. Looks like a memory alignment issue (which would be odd). Will try to fix it asap. Only release mode is affected (that's why I didn't notice it before).


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PostPosted: Thu Mar 16, 2017 1:21 pm 
Site Admin
User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 11185
Wow. This is the strangest compiler bug that I have ever seen.

So I use a string - "Stream (via VLC)" - at multiple places. (And I just added a new one to correctly sort the sound cards based on their names and still keep this one at the bottom).

So in one place I used:
strcpy(some_variable, "Stream (via VLC)");

In another I now added:
strstr(some_other_variable, "Stream (via VLC)");

For the strcpy, the compiler sees that it can perform the full strcpy with just 2 SSE instructions (it's a 16 character/byte string). So, it loads the string into an SSE2 register and then stores it. This however requires that the string is aligned on 16 bytes, and until now it always was.

But now I created that strstr line, which created another copy of this string which didn't have to be 16 byte aligned, and it wasn't.

So far so good. But in a final optimization step, the compiler checks if strings occur multiple times, and if so, it throws all of them away but one. And the one it kept wasn't 16-byte aligned. Which caused the code to crash. Ugh :(


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PostPosted: Fri Mar 17, 2017 2:34 am 
Site Admin
User avatar

Joined: Mon Mar 17, 2008 1:40 am
Posts: 11185
Windows 32 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-027.exe
Winamp DSP: http://www.stereotool.com/download/dsp_ ... 25-027.exe
VST: http://www.stereotool.com/download/vst_ ... 25-027.dll

Windows 64 bit:
Windows stand alone: http://www.stereotool.com/download/ster ... 25-027.exe
VST: http://www.stereotool.com/download/vst_ ... 25-027.dll

CHANGES:
- Fix for compiler bug (crash in 026)
- SST: Add widget with text for version number, preset name etc.

OLD CHANGES:
- 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:
- Linux plugin: Close and resize don't work.
- Change slope calculation in MB??? Wait for feedback from Wes and Matt --> Yes, change it
- Change version number
- Linux version: Window close should call a callback in the Generic plugin.
- Add and update presets
- Add Wes' latest preset
- 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: Fri Mar 17, 2017 8:05 am 

Joined: Sun Feb 03, 2013 2:39 pm
Posts: 333
Quote:
- Change slope calculation in MB??? Wait for feedback from Wes and Matt --> Yes, change it
What's that about?


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