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PostPosted: Thu Feb 06, 2014 9:18 am 
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Quote:
Quote:
At the moment I'm testing on other PC win7 64 and FM out arrtifacts are smaller but SSB mode don't work.
Few hours without crash..
Can you *please* answer these 3 questions:
1. Does Stereo Tool crash, or the whole Windows OS?
2. Was SSB enabled when it happened?
3. How long did it run (approx) before it crashed? A second, an hour, a day?
1. Maybe it's not problem in ST because I testes again few hours and nothing happens.
2. Probably yes
3. 3rd times about dozen minutes.

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PostPosted: Thu Feb 06, 2014 1:46 pm 
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Quote:
Edit: Turning off declipper and oversampling in Bojcha's newest preset has the CPU load at about 70%, which is approximately where the classic MB + Advanced Clipper goes. The preset only uses 7 bands though, but perhaps that is enough. I think I had tried with 9 bands + Advanced Clipper, and that didn't go so w..w..w..w...we...we...we..well...l...l...l. :lol:
The MB has been optimized a lot, CPU load is at least 30% lower than before. So that probably helps as well :)


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PostPosted: Thu Feb 06, 2014 2:27 pm 
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But was it Stereo Tool that crashed (then it MUST be a Stereo Tool issue) or the whole Windows (then it could be anything)?

SSB is fixed now.
Edit: Rest of the behavior also seems to be back where it was... Only 'weird' sample rates (input sample rates below 32 kHz) don't support composite clipping or limiting at this moment. IN FACT, THEY CAUSE CRASHES!!!

Hm, since I don't see ANY use for working with input sample rates below 30 kHz, I'm disabling the FM features (stereo, RDS) for such sample rates.


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PostPosted: Thu Feb 06, 2014 3:44 pm 

Joined: Mon Jan 21, 2013 3:01 pm
Posts: 126
Quote:
Edit: Rest of the behavior also seems to be back where it was... Only 'weird' sample rates (input sample rates below 32 kHz) don't support composite clipping or limiting at this moment. IN FACT, THEY CAUSE CRASHES!!!
32 kHz samplerate? Who will use that? Is there a possibility to give a warning that's not possible to use that low sample rates? At least 44,1 kHz could be the minimum for normal stereo processing without FM?

Running beta 024 at transmitter site, supersound and without any problems :)


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PostPosted: Thu Feb 06, 2014 4:26 pm 
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Quote:
Quote:
Edit: Rest of the behavior also seems to be back where it was... Only 'weird' sample rates (input sample rates below 32 kHz) don't support composite clipping or limiting at this moment. IN FACT, THEY CAUSE CRASHES!!!
32 kHz samplerate? Who will use that? Is there a possibility to give a warning that's not possible to use that low sample rates? At least 44,1 kHz could be the minimum for normal stereo processing without FM?

Running beta 024 at transmitter site, supersound and without any problems :)
Lol I was just editing the post above. If you use such low sample rates, all the FM features (except pre-emphasis) are now switched off. Not the best possible solution but it makes my life a lot easier (I can now always assume that I *never* need to upsample more than 4 times. Before I just didn't know).


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PostPosted: Thu Feb 06, 2014 5:30 pm 
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I agree that sample rate below 44.1 Khz must disable FM features. Why such a low quality will be processed ? May be it`s not the best solution but it`s the right solution in my opinion.


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PostPosted: Thu Feb 06, 2014 11:46 pm 
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Posts: 88
I try the latest builds on my little stream on a regular basis. The latest build sounds great - ekeing out new freshness with the recent changes to the AGC.

I'm using the WinAmp plug-in with RadioBoss.

My only concern - I've been unable to copy/paste my registration key, so I have to re-type it every time I want to use a new build.

Would be nice if I could copy paste the reg key...

Thanks,
Dave

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PostPosted: Fri Feb 07, 2014 1:59 am 
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Quote:
I try the latest builds on my little stream on a regular basis. The latest build sounds great - ekeing out new freshness with the recent changes to the AGC.

I'm using the WinAmp plug-in with RadioBoss.

My only concern - I've been unable to copy/paste my registration key, so I have to re-type it every time I want to use a new build.

Would be nice if I could copy paste the reg key...

Thanks,
Dave
Control+V doesn't work to paste?
When you update, you shouldn't have to type in the registration key - it should still be there!


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PostPosted: Fri Feb 07, 2014 3:00 am 
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Windows stand alone: http://www.stereotool.com/download/ster ... 41-027.exe
Winamp DSP: http://www.stereotool.com/download/dsp_ ... 41-027.exe
VST32: http://www.stereotool.com/download/vst_ ... 41-027.dll

Behavior should be similar to that of BETA023 (including the FM output behavior)

Changes:
- Bugs in beta's after BETA023 should be fixed now. The extra thread is still gone though.
- FM output for input sample rates below 30 kHz is no longer possible. This makes things a lot easier (I know that I will never need more than 4 times upsampling so I don't have to change buffer sizes depending on the input sample rate).

Older changes:
- Get rid of chain2() thread. This should also allow reducing the ASIO latency by 1 step (usually 1.5 ms). Hm.... Or not? I'm confused :( Well at least the thread is gone now.
- New ASIO behavior: Push samples, read them back directly from buffer, skip whole Chain2 stuff.
- Old Hard Limit for composite limiting was slightly tighter input level was very high. And had no overshoots; the new one does.
- Lot of things from separate thread moved to main code. This includes a lot of changes that affect different sample rates and might be buggy. -> Was indeed buggy
- Memory usage reduced by 60 MB -> No, was only in my VMware environment. Removed this change. Hm now other people are confirming it. Will add this change again.
- Removed some multi-threading, replaced it by doing everything in the same thread. I would have expected a small deterioration in performance, but on my pc it's actually running faster!
- Memory usage should have been a lot lower but appears to be nearly unaffected. I don't understand why... Oddly, if I turn some compiler optimizations off, it uses about 60 MB less!

- Hard Limit for composite clipper caused very soft clicks every block!!! Also in older versions...
- Composite Limiter was running in a separate thread, and taking 2 ms extra latency. The new version does not do that anymore, and returns a cleaner spectrum, but it requires a bit more horse power from the PC because it doesn't run on a separate CPU anymore.
- Added HQ mode Not available, for testing only
- Improve Multiband3 and Singleband2 limiting and (to a lesser extent) compression for low latency settings. LQ output should sound similar to normal output! Fixing this will also improve audio at lower latency settings. Compressor is probably more or less ok, limiter is pretty horrible, also at lower latencies!
- Fixed Phase Rotation frequency effects at low latencies (need to compensate for loss at certain freqs in low latency modes)
- Fixed AZIMUTH behavior at lower latencies
- AGC behaves slightly differently for lower latencies - Kinda OK. With shorter block size the drop for short spikes is bigger, which leads to a slightly lower overall output level. But I cannot easily fix that. Other differences are fixed now.
- Clipper (probably only ABDP) does not work well for latency 128. Yup -> if I lower the top bass freq from 400 to 200 Hz it's MUCH better. Fixed.
- Something removes low bass in low latency modes. -> EQ and other things. -> Improved. Difference is still large though.
Rewrote LQ Low Latency monitoring to use the normal processing code. Works reasonable, sound resembles that of the normal latency EXCEPT for the bass limiters and to a lesser extent the compressors in the multiband section. Memory usage for plugin version is reduced by more than 20 MB. Stand alone version might use slightly more than before.
- Fixed FOX TV Carbon Coder R128 normalization issue Waiting for feedback.
- Moved a lot of threads into a single thread. Might improve hiccups that some people have reported.
- Added Power Highs (it's in the same window as Power Bass).
- Moved Power Bass and Power Highs to before the wideband AGC to improve volume level consistency.
- Sudden fast rise of bass or highs is limited, new slider 'Release boost' added. I'm not really sure yet if this is ok; if there's a loud high or low sound it can push the band down a lot, and it comes down slower than before. If needed I can add something to allow it to come back faster after a short spike. Waiting for feedback first though.
- Sidechain checkbox removed (without that doesn't exist anymore).

Attempt #2: Redesigned Simple Clipper. Reduced CPU load.
- Reduced the memory usage
- Fixed most of the Stereo Image artifacts!!! "Deprecated" is removed from the sliders that were marked with it. See (*) for a cool new possibility!
- Removed some more unnecessary steps (AZIMUTH 2x, Stereo Boost 2x).
10 remaining.
Fixed 'Post filter for DC offset' problem.
52. Check CPU load. Start with checking if there's anything left that uses the 'unnecessary steps'. Sevdah Web preset: Data still gets converted 58 times... I think I need to do this one first, it should have some effect on the CPU load. 28 removed - next convert the 2 IIR filters so they can be optimized and the merge/split around it can be removed. I'm not measuring any effect from this though (but it makes the code simpler which is also good)
53. Noise Gate/Stereo Boost: Pre-calculate 1-cos() and sqrt() values.
55. Check MemoryPool behavior for cache improvements -> No effect measured, and might make behavior less constant.
56. Check if we can go in opposite direction for each next step to improve cache.
57. Check if lazy reverse FFT is an option. -> No, difficult and gain does not even seem to be measurable.
58. Created a separate class that performs the processing chain. Currently the same code is repeated twice (once for normal processing, once for low latency processing) - which means that a lot of code is duplicated and it's difficult to add extra chains. Most, not all, of that code is now moved elsewhere.

TO BE DONE:
- Declipper display is broken.
- Fix ASIO buffer size calculation. Probably need an offset to allow easy upgrades from older versions.
- Add some code to New/Delete in _DEBUG mode to test for never used memory (large blocks filled with 0's).
- Spread over cores is not constant, which causes differences in performance. I *think* it might be the chain2() code that causes this. Actually it might be a good idea to get rid of that completely...
- Composite Limiter effect no longer visible in GUI. (Is that bad?)
- Reduce stand alone version memory usage (unused low latency thread items can be removed.)
- Add watchdog for stand-alone version. Both built-in and separate.
- For ASIO HQ mode, add redundancy protection.
- Make Low Latency monitoring latency configurable (between 128 and 512, current value is 256).
- Check low latency buffer filling issues (?)
- OPTIMIZE SOUND CARD HANDLING CODE (USES EXTREMELY AMOUNT OF CPU POWER).


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PostPosted: Fri Feb 07, 2014 9:53 am 
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Links are up now!


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