Stereo Tool

I figured you'd notice as well...

Disable the Advanced Extreme Highs Protection (uncheck)
Move the "Strictness (CPU)" slider to 0.
Make sure you have "Take some shortcuts (CPU)" checked.
Also, unless you have sound card issues, make sure to uncheck "Use oversampling to avoid inter-sample spikes..." in the upper right corner.

When you do all that, there's still a modest increase in usage (20-30% increase, meaning 10-20 CPU percent) coming from the implementation of the "Strict" checkbox for the bass clipping right beside the "Always clip bass below" slider... You might could also disable that and be back to where you were, and depending on your settings, you may not notice a difference in the audio quality. Maybe after a while Hans can find any ineffecient code that might exist in that section.

Ah yes I forgot to uncheck oversampling. But everything else I did. It all works fine that way.

There are _some_ tracks where loud highs cause the level of other sounds to be dropped.
Example is Bojcha's "mina04voice" track: Play that with everything off, only FM output (pre-emphasis 50) and Loudness (x4.00), and the 'S' at around 35 seconds sounds really crappy - the lows and mids are briefly dropped by 7 (!) dB. Now I want to fix that (also someone - don't remember who - complained about it).


I have now redesigned the filter, it does almost nothing in most cases, but in some bad cases it increases the low/mids by upto 3 dB. It can probably still be improved further (to better preserve transients), but it's getting close. Building now.

20-30%?! I'm measuring 8%, and that's with only the clipper turned on (so bascically the clipper CPU usage went up by 8%, so for Stereo Tool in total it should be around 2%). That was measured with oversampling though, so without it the difference will be bigger.

The biggest issue is that I need to perform 12 FFT transforms, and I don't see a way around it (well I could reduce it to 10, maybe even 8, at the cost of getting bigger overshoots. And I do that when you lower the strictness slider.)

BETA117:
- Redesigned Advanced Extreme Highs Protection. Should have FAR less impact on the audio now. (One more change to better preserve transients might be helpful, although it will not make a very big difference.) Highs clipping is at +6 dB (200%) most of the time, and never drops below 0 dB (100%) anymore.

Winamp DSP plugin: http://www.stereotool.com/download/dsp_ ... 20-117.exe
Stand alone version: http://www.stereotool.com/download/ster ... 20-117.exe
VST version: http://www.stereotool.com/download/vst_ ... 20-117.dll
VST version (No SSE2): http://www.stereotool.com/download/vst_ ... 20-117.dll
Command line version: http://www.stereotool.com/download/ster ... 20-117.exe
Linux command line version: http://www.stereotool.com/download/ster ... ETA620-117 NOT AVAILABLE
Linux GUI version: http://www.stereotool.com/download/ster ... ETA620-117 NOT AVAILABLE

TODO:
- Fix loading changed multiband frequencies
- Add buffer and filter for SCA output (SCA1 ok, SCA2 ok)
- Finish AGC improvement - make mono value configurable (replace checkbox by slider) 1 hour --> NO, not needed - anything else needed? -> NO
- Fix Punch
- Check what to do with new filters (such as bass AGC) - keep them, remove them, change them? --> KEEP
- Save new BASS_AGC setting in VST mode
- Loudness: Annoying cracking sound in bass. Slightly present in 5.00, worse in 6.00, maybe even worse in 6.10. Only when bass is too loud. Much worse than in Final Limiter (at same input level!) - so this clearly indicates a bug. Most likely cause: The filter that was added to remove bass artifacts....... - No, it's the louder bass. But it can be fixed by changing some settings. Default settings updated, and behavior for 'not Very strict' improved. Also Deep bass boost and Very deep bass protection are enabled for latency 512 now.
- Fix crash at program close
- Fix VST plugin version (does not run)
- Dynamically drop 'Allow louder highs, even if it causes vibrations' to 0 when bass filter suspects noticeable voice vibrations.
- Reduce Loudness CPU load
- Check and remove static variables
- Finish new de-essing filter (check what to do with the settings, remove at least some!)
- Convert Multiband input to MONO, then use arrays [2][4096] --> should give speedup. - FAILED
- Natural Dynamics: Fix or remove transient boost
- Natural Dynamics: Add expected + strength slider per band
- Finish declipping filter (clipping level detection + level reduction in dB). 1. Figure out why removal of unwanted frequencies causes flat lines at high quality setting with small overlap. This causes distortion, with this fixed repairs are MUCH better. 2. Fix MP3 correction, automatically scale down when this deteriorates the sound. --> TOP part fails!
- Declipping filter: Fix low latency behavior
- Always oversample clipping (configurable)
- Declipping filter: Change detection at lower input levels.
- "Test Right Channel" in "FM Transmitter Calibration" does not work since v6.10 (standalone) (bojcha)
- Declipper window close function - check! - Seems ok
- Scopes black background?
- Add AGC start level
- Declipping filter: Add comparting of sample history to make sure loud bursts are still detected properly (now, with 16 blocks of history, 65 samples are removed even when ignoring just 0.1%). I should also check the current block (probably with a margin of a factor 2).
- Declipping filter: Use reliability of tilt detection to determine movement speed; slowly move to default tilt (configurable!)
- Declipping filter: Far too many samples are marked as 'maybe/probably' clipped. Histogram not used or not cleared?
- Declipping filter: Optimize tilt detection for performance. SSE2 for maximum. And keep separate smaller histograms to determine the maximum (should perform much better).
- Declipping filter: Cleaned histogram must be mirrorred. After cleanup, mirror center is place where SIGMA above == SIGMA below! Only use on the lower end. Then run cleanup again! The result should contain approximately the same amount of data above and below. If not, that indicates incorrect detection.
- Declipping filter: Add a long term histogram, and use maximum of short term and long term start clipping level detection.
- Declipping filter: Check if 'No declipping when tilt cannot be determined' can safely be turned on.
- Declipping filter: Add setting and saving of new sliders (history size, percentage of highest samples to drop, tilt detection range start, end and precision)
- Declipping filter: Check why higher Quality settings causes cracking sounds. Answer: Rounding error, and too low filtering values to remove it.
- Move pre-emphasis to end of processing
- AGC: Separate 'Remove remaining peaks' settings for band 1 and 2
- Channels L/R swap in stand alone version when changing filtering/quality (eldoradofm)
- Declipping filter: Optimize peak matching for performance: Move determination to extra preprocessing step DONE; try to change if statements to min/max or something IMPOSSIBLE.
- Fix changing value of 'final limit' in Multiband when changing Loudness settings.
- Save bandpass filter order
- Save all new settings, also through VST interface
- Declipping filter: Change default values for Maximum distortion and Step size (see Bojcha's post)
- Try to improve reconstruction match (should remove artifacts for low bass + soft highs) ■ FAILED
- Declipper: Fix loud bass with soft higher frequencies issue. Do something with loudest frequency vs. peak level - if it cannot fit (Madonna, Metallica) there must be clipping, otherwise there might not be. On top of this I should somehow try to preserve deformed bass sounds (if they are not hard clipped). Maybe this will work: Find loudest frequency, if it's very low. Than assume that it's clipped at the maximum restoration level (?); generate a new tone (using lowpass filter) of the same frequency and clip that at some margin below the maximum peak level. Then filter the input signal based on the FFT of the clipped signal. Finally analyze this new signal; if it is indeed clipped the filtered output will resemble a clipped waveform (visible in sample histogram; no tilt issues!) instead of a sine wave. Use the output of this as input to determine how strict the clipping detection for the input waveform should be (eg. based on normal or sinusoid signals). This is all way too difficult and it just won't work. Here's the real solution: Detect if a loud bass sound is present, if so make sure that symmetrical clipping detection is used, possibly with automatic 'maybe clipped' level adjustment. This should only slightly affect Metallica (which still sounds very reasonable with symmetrical enabled), but has a big effect on DBN - Jack is Back (which is far less affected with symmetrical enabled; auto likely/maybe adjustment based on symmetrical should help even more). Using veil solution now
- Move veil processing to detilted signal
- Switch to symmetrical detection if very loud very low frequencies are present NO, not needed anymore because artifacts are fixed using another mechanism
- Fix veil and detilting display (in declipper window; separate windows for separate channels; no change in DrawWave code needed!)
- Fix declipper display cleanup
- If the reconstructed samples are close to the original samples, and there are a lot of 'clipped' samples OR there are very loud low frequencies present, don't declip. Also add texture based filtering, and use a slider to set the sensitivity. Also increase the effect of bass and reduce overall level.
- Save veil settings and 'not reconstruct if like original' (STS + VST)
- Fix clipper (idea: Clip, filter, filter diff against clip diff, apply).
- Can I combine 2 FFT windows into 1? That would give half of the improvement of the smooth window, without CPU load increase.
- Old clipper seems to sound better. (New with less steps also sounds good). Use new clipper AFTER old clipper, and use already clipped and cleaned signal as 'clean_input'. Or try a bunch of short clips, than a long one to clean up, then some short (but not very short) ones to clean up pre/post ringing.
- Declipper EXTREMELY slow when feeding DC input.
- Big window output is smaller than normal one - no it's not.
- Bass AGC RMS display does not work (always displays 0.00)
- Longer history for maximum level?
- In FM calibration 38 kHz is NOT working. EDIT: And more bugs: 23 kHz is not at 23 kHz 28 kHz is not at 28 kHz 33 kHz is not at 33 kHz 43 kHz is not at 43 kHz 48 kHz is not at 48 kHz 53 kHz is not at 53 kHz. - not true, tested and all seems to work fine. There is another issue though: This filter still uses the very old windowing method and sounds crappy (artifacts). To be looked into for the next Stereo Tool version.
- Remove Natural Dynamics - Partially done, does the crash come from here? No, for some reason ANY call on my system to InternetOpenUrl in a Winamp plugin causes crashes. But this happens even with version 3.03, so it's not something new - it's only something new on my system...
- When "Enable FM transmitter calibration" is OFF and you click "Calibrate FM transmitter ..." and then close calibration screen again, then "Enable FM transmitter calibration" is turned ON automatically.
- Use DLL name in INI file name (requested by Jazler)
- Add registration to command line versions
- AGC: Added 'Force level below' slider for improved peak control. If the 'down speed' causes the volume to be much higher than the calculated 'target' volume, with this slider you can force the volume to go down anyway if the difference gets too big.
- Registration key must be written to DEFAULT ini file. And for stand alone version, ini file StereoTool.ini should automatically be converted to stereo_tool.ini to match existing ini files.
- FM Calibration beep tones are still not corrected.
- Stand alone: Check box to FM Calibration on/off missing in the FM transmitter calibration window.
- Declipper artifacts ( viewtopic.php?f=14&t=3065&p=12527#p12527 ): Adjust defaults or increase bass sensitivity. - No, replaced MORE switch by slider.
- Add extra bass clipping step to clipper.
- Fix Loudness display
- Smarter lows volume drop against highs protection: First clip using reduced highs (highs protection filter), then do it again but do NOT lower the lows below the level found in the 1st step. - NO! No longer needed.
- VST: Bass AGC up/down speeds setting problem.
- Optimize clipper code: Reflection, smaller overlap (stronger reflection possible?); optimize RMS search filter; optimize voice vibration removal
- HPF bad since BETA080, especially with declipper enabled
- Highs vs bass distortion protection: Fix window edges.
- Add highs distortion sensitivity slider(s)?
- AGC hangups
- DC caused by clipper
- Fix pre-ringing artifacts
- Checkbox to combine bands 1+2 of Multiband
- Phase linear highpass filter if non-phase-linear is selected to avoid DC offset.
- Non phase linear highpass filter: Make steeper
- Strict bass clipping
- Extra HPF step either just before Singleband or - if that's not used - after the bass protections for Loudness. Maybe also for phase linear HPF! AVAILABLE, WAITING FOR FEEDBACK
- Finish blind interface WAITING FOR FEEDBACK TODO: TEST: Only display this window when needed.
- RDS issue reported here: viewtopic.php?f=15&t=3703&p=11524#p11524 partially solved
- Remove or simplify 'Advanced highs protection', look at other protection mechanisms.
- Robot effect when changing to lower latency (?)
- Save new clipper, bandpass, multiband options (STS+VST)
- SSE version crashes (reported in email by Francis)
- Update presets? (BASS_AGC etc.) 1 day
- Change version number 1 hour
- Release 1 hour
- Add composite clipper
- Add lowpass filter for stereo signal (will cause a lot of extra latency!) - it might be possible to avoid this latency using a Hilbert transform
- Add smarter clipping detection. Maybe something much simpler suffices: Current clipping detection with threshold + flat line detection
- Declipping filter: Automatically override the 3 clipping level sliders if the clipping level is detected very clearly (clear thin spike in sample value histogram). - NONSENSE, this is already done by the histogram function. But it can be made a BIT better - I think - by automatically LOWERING the 'always clipped' slider if a lot of data is present at the highest few bins (but care is needed for DBN - Jack is Back like tracks)
- Declipping filter: Create a new histogram after declipping, clean up and subtract from original.
- Change stand alone version to 32 bits.
- Optimization: Reduce CPU usage when input is silence.
- Natural Dynamics: Attempt to reduce number of bands by 2 (memory + cpu DIV 2), using bands with sloppy slopes (no interaction between bands -> memory usage DIV 2, out-of-cache reads gone, hopefull final filter gone too -> CPU DIV 2 again).
- Natural Dynamics: Voices, especially in chorus, still sound weird.
- Natural Dynamics: Smooth out different bands more to reduce low-bitrate-MP3-like sounds.
- Natural Dynamics: Optimize situation with no interaction between bands for performance.
- Natural Dynamics: Save settings in VST version
- AGC: Sudden loud kicks (32:40) - much faster drop! IDEA: IGNORE FAST 2ND STAGE DROP IF THE EFFECT IS LOW.
- AGC: Try frequency separator implementation (2 or 3 different volume measurements, take lowest). Should protect loud voices much better.
- FM Calibration causes artifacts and latency.

- Highpass before Multiband etc? Not for now.

When I stated the percentage, I stated it relative to the previous beta. My 20-30% is based on:

38 - 50 -----> 44 - 65

20% of 38 is 7.6. 36 + 8 = 44
30% of 38 is 11.4. 36 + 11 = 47

20% of 50 is 10. 50 + 10 = 60
30% of 50 is 15. 50 + 15 = 65

So, my "10-20 CPU percent" was a bit off, it's actually 8-15% as viewed in Task Manager in raw increase. The 20-30 was the delta.

Note: The following are Task Manager figures, not deltas...

Strictness slider at 0, CPU shortcuts checked, no oversampling, strict checkbox checked...

I can get back into the upper 30s - low 40s now with Advanced Extreme Highs Protection disabled and the GUI in the tray. If I bring up the GUI there's an increase to the lower 50s. If I enable AEHP, there's another increase to the upper 50s - mid 60s. AEHP on with GUI in the tray gives low-mid 50s.

So, what I'm experiencing is a GUI up vs. tray issue...although the AEHP does indeed still cause additional load, which I'm unsure of the quality value of using it right now...

Beta117 seems to be okay. With completely new settings from scratch i´m happy with it, CPU load is low, ~10% now. I think you could take it for 6.20 Release.

@Brian: I understood that

But my measurement is done similarly: I just let Stereo Tool run at max speed for a minute, and then I check how much data it has processed.

Then, with only the clipper (Loudness) enabled, switching Strict on and off makes a difference of 8%, not 20-30%...

About AEHP: It has very little effect most of the time. But there are some cases where the sound gets better. Turn it on if you can - or if you play music or use settings that require it (75us FM pre-emphasis might be one of those), off otherwise. I'll also check if I can optimize it further, it might be possible to make the filter more accurate (less reduction when it's not needed) while at the same time reducing the CPU load a lot.

We're taking different measurements. You're testing that process in isolation. My measurment is with your product as a whole, with all of my normal options enabled.

The testing method you're using is that it only tests that isolated process. That, in industry terms, would be a "unit test" or maybe even as far as "integration test", but it is not a "system test". A system test would test the entire StereoTool product, soup to nuts, as it were. You would have various test cases / test scenarios, in which you test for, among other things, performance. With testing like what you're doing, sure you can get the increase in load on that specific process, but you don't get a picture of the cummulative effect. IMO, you have a tipping point somewhere that is unoptimized that when tested by itself doesn't look so bad, but you don't see these issues until people report them due to not testing for the cummulative impact when it joins up with another process and causes issues. Also, IMO, your 2MB+ cache processor masks some of these issues.

Ok, I've figured out now why the Advanced Extreme Highs Protection filter works fine for certain sounds ('S' sounds mainly) and not at all for others (brief sinusoids, such as xylophones). What impacts other sounds is NOT the peak level, but the RMS level. And I need to work on shorter pieces of audio - at least for detecting the dynamic clipping level.