Stereo Tool

In the stand alone version the Tilt correction is not vissible anymore (also in v74).

About the RDS: It's very hard to test this. I again had strange texts in the display of my radio with v74. But when i used the RDS feature of the pira, it displayed no strange texts. But when i changed to another preset it was ok again on the tuner.
I didn't used AF. Maybe it depends on the time of the switch from preset to preset. If i see again strange texts i'll check it more.

This is 30 Hz HPF of Stereo Tool with phase linear on:


This is 30 Hz HPF of BBP, also phase linear:

As you see this one is much steeper.

This is 2 times 32 Hz HPF of Stereo Tool with phase linear on:


In this last picture there are 2 stereo tools chained together, so 2 times HPF.
So my request for the next beta is a checkbox to select "2x HPF".
This should give less problems with low bass on some transmitters and still be phase linear.

Thinking a bit more about it, this will proabaly increase the passband ripple.
Better would be to recalculate the filter for a higher order.

Take a look at this http://zone.ni.com/reference/en-XX/help ... r_filters/
Especially the last part:
You also can use the IFIR technique to produce wideband FIR lowpass filters and wideband FIR highpass filters. A wideband FIR lowpass filter has a cut-off frequency near the Nyquist frequency. A wideband FIR highpass filter has a cut-off frequency near zero.

I made mention of this before...months ago....when I was talking about how the filtering wasn't steep enough. I didn't have any pretty tools to show what I knew I was hearing, so I got doubted, much like the doubting about cache size.

I hate being grumpy, and I hate saying "I told you so", but I knew something needed improvement in the highpass filter, so I'll just say that I'm glad to have another voice added to that request for a HPF improvement.

Looking at those output graphs, I think you can replicate something close to the BBP output by increasing the filter frequency. The filter seems to be similarly steep (at least from the point where the level starts to drop), but it starts at a higher frequency!

I think chaining 2 highpass filters will cause a lot of artifacts. Not sure though, but it's pretty difficult to filter out low frequencies without increasing the latency and without getting too many artifacts.

The current phase linear filter is a compromise between steepness and artifacts.

BETA076:
- Veil now works on detilted audio
- Veil settings saved (STS file, not yet to VST host)

Winamp DSP plugin: http://www.stereotool.com/download/dsp_ ... 20-076.exe
Stand alone version: http://www.stereotool.com/download/ster ... 20-076.exe
VST version: http://www.stereotool.com/download/vst_ ... 20-076.dll
VST version (No SSE2): http://www.stereotool.com/download/vst_ ... 20-076.dll
Command line version: http://www.stereotool.com/download/ster ... 20-076.exe
Linux command line version: http://www.stereotool.com/download/ster ... ETA620-076 NOT AVAILABLE
Linux GUI version: http://www.stereotool.com/download/ster ... ETA620-076 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
- 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
- Save veil settings and 'not reconstruct if like original' (STS + VST)
- 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. Solved, but should use a slider to set the sensitivity.
- Fix veil and detilting display (in declipper window; separate windows for separate channels; no change in DrawWave code needed!)
- Longer history for maximum level?
- Non phase linear highpass filter: Make steeper
- Update presets? (BASS_AGC etc.) 1 day
- Remove Natural Dynamics
- Change version number 1 hour
- Release 1 hour
- 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

No you cannot. and steep is not even close.


What you can see is 1 ST with Phase-Linear HPF at 42Hz. So, not same as 2 HPF's at 32...
Here you lose much of sounds at 40Hz and even more at ~35Hz, and it looks less strong.
No, very small difference in artifacts! Only questions is Latency then.

btw, how much is latency of HPF ?

I've measured the passband ripple.
1x phase linear HPF at 30 Hz = max around 0.11 dB
2x phase linear HPF at 30 Hz = max around 0.15 dB
1x non phase linear 6th order at 30 Hz = max around 0.53 dB

So big difference in the results between phase linear and non phase linear.
A difference of 0.04 dB should not be noticeable when using 2x phase linear HPF.

Can you post a spectrum picture of what the non phase linear filter looks like?

Non-Phase Linear 30Hz.

Picture of passband ripple. This is with tone from 0 to 500 Hz.
First is with 1x phase linear
Second is with 2x phase linear (ignore the big spike it was a buffer fault)
Third is with 1x non phase linear 6th order.