BBE Sonic Maximizer Software vs. Hardware
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- KVRian
- 1050 posts since 4 Nov, 2003 from Washington DC
I was thinking about getting a hardware BBE Sonic Max unit but I already have the software version. Do you guys think it's redundant to buy the hardware since the software is said to be pretty authentic? Or is the hardware miles ahead of the plug-in version?
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- KVRAF
- 1655 posts since 3 Mar, 2009 from Colorado Springs
It's basically a variable state filter with a treble exciter, nothing going on there that can't be more than adequately duplicated by software.
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- KVRian
- Topic Starter
- 1050 posts since 4 Nov, 2003 from Washington DC
Cool mate! Thanks for the feedback. It'd be interesting to hear some audio examples of the hardware compared to the software. Can anyone kindly furnish these? Should be pretty educational!
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- KVRAF
- 5703 posts since 8 Dec, 2004 from The Twin Cities
If you don't mind submitting an email for an awesome free plugin,CLAS is, as it's authors put it:
I love this thing....similar to BBE treatment, but without being "in your face!". Psychoacoustic compression is similar to many classic compression curves, producing the most boost at low sound levels, and progressively tapering off to no-boost at extreme loud levels. This is more pleasing to listen to than typical linear compression, and accounts for the appeal of many classic analog compressors. The effect, in CLAS, is much like dynamic Loudness Contouring.
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- KVRAF
- 14739 posts since 19 Oct, 2003 from Berlin, Germany
But it's unfortunately windows only. Unless they changed that lately.
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- KVRist
- 293 posts since 2 Aug, 2002 from Odense - Denmark
The BBE plug sounds nothing like the hardware version. I've worked with both. Hardware version sounds really sweet. Plugin version sounds really hard.
And in the end, the love you take is equal to the love you make.
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- KVRian
- Topic Starter
- 1050 posts since 4 Nov, 2003 from Washington DC
Aha! This is exactly what I was waiting for!Kim Olesen wrote:The BBE plug sounds nothing like the hardware version. I've worked with both. Hardware version sounds really sweet. Plugin version sounds really hard.
Would you mind putting up some audio examples? I would really appreciate it! Cheers!-
- KVRist
- 293 posts since 2 Aug, 2002 from Odense - Denmark
Unfortunately i can't. The HW unit i worked with is located in a studio far away from where i live.dj ray wrote:Aha! This is exactly what I was waiting for!Kim Olesen wrote:The BBE plug sounds nothing like the hardware version. I've worked with both. Hardware version sounds really sweet. Plugin version sounds really hard.
And in the end, the love you take is equal to the love you make.
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- KVRAF
- 1729 posts since 26 Feb, 2008
I'll have to agree, I've used both both (quite a few different hardware models) and there is something definitely lacking from the non-hardware stuff.Kim Olesen wrote:The BBE plug sounds nothing like the hardware version. I've worked with both. Hardware version sounds really sweet. Plugin version sounds really hard.
Snare drums samples: the new and improved "dither algo"
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- KVRAF
- 1655 posts since 3 Mar, 2009 from Colorado Springs
I have a hardware unit that I could run something through with a DI out, anyone with the software want to participate?
The thing is, it's totally doable, but I've never used the BBE software. The actual BBE circuit is pretty darn clever but not at all magical or fancy, it's just that the IC where the stuff takes place is proprietary and not available to the general public. But using a bunch of well researched info, patents, design docs, etc. and some smart reverse engineering, a gentleman on another forum (freestompboxes.org) actually makes daughter boards which use discrete components in a workalike circuit to either replace fried ICs in out-of-warranty units, or for use in new circuits. So I'm not just making this up, it really is a fairly ordinary thing compared to the extremely complex analog modeling that is being done today even by freeware makers. If BBE's official plugin is doing it wrong, I have to wonder why, as it's not something that is somehow mystical or out of reach for simulation, you know?
The thing is, it's totally doable, but I've never used the BBE software. The actual BBE circuit is pretty darn clever but not at all magical or fancy, it's just that the IC where the stuff takes place is proprietary and not available to the general public. But using a bunch of well researched info, patents, design docs, etc. and some smart reverse engineering, a gentleman on another forum (freestompboxes.org) actually makes daughter boards which use discrete components in a workalike circuit to either replace fried ICs in out-of-warranty units, or for use in new circuits. So I'm not just making this up, it really is a fairly ordinary thing compared to the extremely complex analog modeling that is being done today even by freeware makers. If BBE's official plugin is doing it wrong, I have to wonder why, as it's not something that is somehow mystical or out of reach for simulation, you know?
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- KVRAF
- 1729 posts since 26 Feb, 2008
Well, actually - I thought it was common knowledge that really what was going on was simple lf and hf phase rotation (similar to stardust vst) followed by an enhancer. Nothing crazy.Agreed wrote:I have a hardware unit that I could run something through with a DI out, anyone with the software want to participate?
The thing is, it's totally doable, but I've never used the BBE software. The actual BBE circuit is pretty darn clever but not at all magical or fancy, it's just that the IC where the stuff takes place is proprietary and not available to the general public.
Snare drums samples: the new and improved "dither algo"
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- KVRAF
- 1655 posts since 3 Mar, 2009 from Colorado Springs
Nothing about the BBE "process" seems to be common knowledge, haha. They take good advantage of obscurity, I don't know if people would be willing to pay several hundred dollars if they knew everything going on in the box. But then again maybe they would, it's simple but very effective for certain things. Here's all the relevant info on the topic that I've gathered:
I've been working a DIY BBE implementation for September's FX-X. After reading all the info I could find, including BBE chip datasheets, I came to a pretty good idea of what it was about and was able to build a nice sounding circuit. I still need to get the gear to record some samples, thus I haven't posted this yet.
As an avant premiere, the things going on can be grouped in three main categories:
1) A larger delay to the lower frequencies than the mids and than the highs. In particular, we are talking about a 1.5 msec delay for frequencies below 100 Hz or so, which drops gradually as the frequency increases. Can be implemented by cascading two properly tuned first order all-pass networks.
2) A shelving-type boost equalizer for the highs and the lows, centered around 750 Hz or so. It is a second-order type shelving response thus, when highs and/or lows are boosted then impact on mid frequencies is minimal. IMHO this is key to the good sound of the device.
3) A dynamic expansion of the highs based on the input level of the audio material. This is the tricky part to implement.
For instance, 1) and 2) can be easily implemented around a state-variable filter (that one which has low-pass, band-pass and high-pass output simultaneously), and then properly weighting and adding each band. This sounds really good by itself, better than any EQ curve I've tried before. Also, in spite of the propaganda that says the BBE process can be applied to any kind of music, speakers and gear, I played with the Q and center frequency of the original filter and found a combination which I liked even better for my guitar/combo setup. This is what I'll post as a simplified DIY version, which is finally built around 4 opamps and some R's and C's.
I also developed (in the sim) a dynamic expansor to implement the 3rd aspect, but then this requires 10 opamps and 2 JFETs, thus it is not what I'd call DIY friendly, and the simpler version makes such an improvement to my tone that I don't feel the need for that now.
As a disclaimer, I must add that the implementation I described probably matches the early version(s) of the BBE process, however more elaborated processing could be going on in newer versions. Nevertheless the improvement of the simple circuit is amazing considering it is just a cleverly designed eq.
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Quote from: Mark Hammer on October 18, 2006, 11:05:37 AM
This thread - http://www.diystompboxes.com/smfforum/i ... ic=48921.0 - discusses some DIY possibilities towards the end, and also names some of the various commerially available BBE chips that could be used to make a DIY pedal/unit. Just note that what the "process" does is different than a mere EQ or "exciter" circuit. Those can certainly render more punch to your tone, but they do not attempt to align the fundamentals and harmonics the way the BBE process does.
Just note that what the "process" does is different than a mere EQ or "exciter" circuit. Those can certainly render more punch to your tone, but they do not attempt to align the fundamentals and harmonics the way the BBE process does.
Mark, based on the datasheets on the process implementations I studied, and in particular the NJM2150 (New Japan Radio) and BA3888S (Rohm) datasheets, I can say that the BBE process is not that special or particularly more complex than other audio signal processing we are used to, which are Equalizing and Expanding. My personal opinion is that most of the fancy pseudotechnical descriptions behind BBE aim for marketing and mojo so as to raise the technical aspects of process itself to a sublime level.
I RECOGNIZE and AGREE with the good sonic qualities of the process. That's out of question here, however I'd like to demistify the obscurantism around the technical aspects of this subject, as every time someone talks about the process I tend to perceive a halo of awe and supernatural respect on the subject.
The "Process" is just a mix of high-frequency eq + dynamic expansion of the already existing high-frequency content, i.e. it doesn't generate new harmonics like an exciter would do. "Dynamic Presence" would be a perfectly good name, and perhaps indeed more representative of what's really going on with the highs. Of course the term "Process" sounds more technical and add some extra mojo.
In particular, the NJM2150 implementation is just an Eq, as it doesn't have the high frequency dynamic expansion part. Well, either that, or the dynamic expansion part was totally removed from the datasheet, which I find questionable as the actual circuit values and the detailed frequency and phase (which together allow reconstructing the group delay) response curves are presented in great detail.
The BA3888S has essentially an identical Eq implementation, with the addition of the dynamic highs expansion and a noise gate to cut down hiss when no signal is present.
Both datasheets state clearly that they won't sell the IC's unless you have a signed agreement with BBE Sound Inc.
As for the Bass, it is a simple matter of eq'ing the lows via a 2nd order shelving frequency response, which happens to be already available as the main filter is based on a state variable topology which already has low-pass, band-pass and high-pass outputs. The "Lo Contour" name for the bass is again marketing in my opinion. It could be perfectly called "Bass", "Bottom", "Deep Bass" or "Bass Boost" and do a better justice to what's going on with the low frequencies.
Regarding the "time alignment" between the lows, mids and highs, I can state that the 1.5 msec delay mentioned in the related literature is a consequence of the state variable filter itself. When the "Lo Contour" and "Process" are set to the minimum, and leaving apart the dynamic expansion of the highs, one ends up with a 2nd order all-pass filter that has a group delay of exactly 1.5 msec below 100 Hz or so. Adjusting the "Process" knob and the dynamic expansion on the highs do not affect the group delay on the lows, thus this particular "time delay" maintains. When the "Lo Contour" is increased the 1.5 msec time delay varies a little, but remains essentially within +/- 10% of its original value or so, thus the delay relationship between the different frequencies maintains. In other words, the frequency-selective delay can be thought of rather as a consequence of the Eq implementation.
Ah, but someone might argue that the big merit of the Eq lies in the inherent time alignment it introduces, beyond the high and low frequency equalization, and that this is essential to the good sound. Well, that's what I first thought, however when I decided to try the delay alone, i.e. having the filter set just as an all-pass, A-B testing revealed the difference in the perceived attack and brightness when playing chords was really small. There were differences, but very subtle. I tried different delay times and Q's for the pure 2nd order all-pass network in an attempt to find an optimum, however I would rather describe the options as different, but none of them significantly better than the others. Perhaps this time alignment is better noticed when reproducing the material through triaxial speakers with passive crossovers, as each band will be delayed differently and there is more to correct. I don't know for sure, and my interest here goes to the guitar domain where most of the times you don't have crossovers.
In summary, the simpler Eq only implementation (without highs expansion) made a BIG improvement in my guitar & amp sound, however when I tweaked the frequency and Q of the eq filter the sound I could get became out of this world! Really.
In particular, for the neck humbucker pickup and the amp with all knobs at noon, my favorite setting was 12 dB boost for the highs, 6 dB boost for the lows, central filter frequency reduced to 70% its original value, and filter Q increased a bit. Now the 6th and 5th strings produce fat and deep bass without detracting from the clarity of the highs. Doing chord work produces very nice cutting sounds with the strums. Notes across all strings sound balanced. I know that I am sounding like a marketing person, but I am not trying to sell anything here.
I recognize that the dynamic expansion part might add an extra dimension to the sound, but currently my sound improved so much that I don't feel like I need more for now, especially with the customized tuning of the eq/filter.
Quote from: oldrocker on October 18, 2006, 12:13:16 AM
I know why people ask if there's a DIY for a Sonic Max. Because it sounds like an easy thing for a DIY guru to make. But I guess there is more to it than meets the eye. With effects like phasers, flangers, chorus and wah's all cloned or redesigned by all knowing effects and electronic gurus, it seems hard to believe a simple tone enhancement effect would be so long in coming. I guess if it was that easy to re-create BBE wouldn't be doing as good as they are selling their units.
I think the lack of commercial clones is due to the fact that the process is patented, and neither the name nor the implementation might be used by a manufacturer without being running into trouble.
The lack of DIY clones can be justified considering that the IC's that implement the BBE process are not sold to the general public, thus you cannot readily clone an existing device.
The option left is to reverse engineer the process itself in order to make a custom designed implementation, while dealing with the different sources of obscure pseudotechnical information that lead to believe it is something so special that it just can't be made up.
I think the above reasons justify why there hasn't been an attempt of a DIY version... until now.
I insist that my intent here has been only to demistify the technical beliefs around this, without detracting from the sonic improvement of the process which is real. In fact, I admire how Mr. Berry was able to market and protect his invention. Just think how it is spread massively in audio and TV consumer equipment!
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Quote from: Doug_H on October 18, 2006, 03:34:05 PM
FWIW, my bass player tried it and didn't like it. He plays a Jazz bass derivative and I don't think there's enough hi-freq content in it (or maybe his amp) for the BBE to make it interesting. So the "process" control didn't do much and the bass boost just farted out the lows on his amp. It might be more interesting in a crisper sounding bass rig that's bi-amped, using an Alembic or Rick or something.
Yes, this kind of circuit doesn't do much when highs are not there in the first place. Probably an Aphex Aural Exciter or a similar device would do better for him.
On the other hand, the BBE process is clearly targeted for the hi-fi audio market, i.e. 20 Hz to 20 kHz, thus it is fine tuned for music rather than to a particular musical instrument (I am not aware that the pedal versions are tuned differently for instruments).
For guitar playing, I found a great improvement when reducing the filter's frequency to 70% of its value--well, in fact to 66% or 1.5x less of its original value. I have the feeling that for bass dropping the frequency of the filter to 50% or so might be a much better match for the instrument in terms of the Process knob. Regarding the Lo Contour knob, maybe dropping the frequency might do good as well as it might reinforce the bass closer to the point where the speaker's response is falling, who knows.
One thing I forgot to mention based on my experiments is that the state-variable filter implementation allows you to CUT bass as well. Indeed, I fully cut the bass and with the highs raised I got a very VOX-like sound which was quite appealing also.
I've been working a DIY BBE implementation for September's FX-X. After reading all the info I could find, including BBE chip datasheets, I came to a pretty good idea of what it was about and was able to build a nice sounding circuit. I still need to get the gear to record some samples, thus I haven't posted this yet.
As an avant premiere, the things going on can be grouped in three main categories:
1) A larger delay to the lower frequencies than the mids and than the highs. In particular, we are talking about a 1.5 msec delay for frequencies below 100 Hz or so, which drops gradually as the frequency increases. Can be implemented by cascading two properly tuned first order all-pass networks.
2) A shelving-type boost equalizer for the highs and the lows, centered around 750 Hz or so. It is a second-order type shelving response thus, when highs and/or lows are boosted then impact on mid frequencies is minimal. IMHO this is key to the good sound of the device.
3) A dynamic expansion of the highs based on the input level of the audio material. This is the tricky part to implement.
For instance, 1) and 2) can be easily implemented around a state-variable filter (that one which has low-pass, band-pass and high-pass output simultaneously), and then properly weighting and adding each band. This sounds really good by itself, better than any EQ curve I've tried before. Also, in spite of the propaganda that says the BBE process can be applied to any kind of music, speakers and gear, I played with the Q and center frequency of the original filter and found a combination which I liked even better for my guitar/combo setup. This is what I'll post as a simplified DIY version, which is finally built around 4 opamps and some R's and C's.
I also developed (in the sim) a dynamic expansor to implement the 3rd aspect, but then this requires 10 opamps and 2 JFETs, thus it is not what I'd call DIY friendly, and the simpler version makes such an improvement to my tone that I don't feel the need for that now.
As a disclaimer, I must add that the implementation I described probably matches the early version(s) of the BBE process, however more elaborated processing could be going on in newer versions. Nevertheless the improvement of the simple circuit is amazing considering it is just a cleverly designed eq.
-------------------
Quote from: Mark Hammer on October 18, 2006, 11:05:37 AM
This thread - http://www.diystompboxes.com/smfforum/i ... ic=48921.0 - discusses some DIY possibilities towards the end, and also names some of the various commerially available BBE chips that could be used to make a DIY pedal/unit. Just note that what the "process" does is different than a mere EQ or "exciter" circuit. Those can certainly render more punch to your tone, but they do not attempt to align the fundamentals and harmonics the way the BBE process does.
Just note that what the "process" does is different than a mere EQ or "exciter" circuit. Those can certainly render more punch to your tone, but they do not attempt to align the fundamentals and harmonics the way the BBE process does.
Mark, based on the datasheets on the process implementations I studied, and in particular the NJM2150 (New Japan Radio) and BA3888S (Rohm) datasheets, I can say that the BBE process is not that special or particularly more complex than other audio signal processing we are used to, which are Equalizing and Expanding. My personal opinion is that most of the fancy pseudotechnical descriptions behind BBE aim for marketing and mojo so as to raise the technical aspects of process itself to a sublime level.
I RECOGNIZE and AGREE with the good sonic qualities of the process. That's out of question here, however I'd like to demistify the obscurantism around the technical aspects of this subject, as every time someone talks about the process I tend to perceive a halo of awe and supernatural respect on the subject.
The "Process" is just a mix of high-frequency eq + dynamic expansion of the already existing high-frequency content, i.e. it doesn't generate new harmonics like an exciter would do. "Dynamic Presence" would be a perfectly good name, and perhaps indeed more representative of what's really going on with the highs. Of course the term "Process" sounds more technical and add some extra mojo.
In particular, the NJM2150 implementation is just an Eq, as it doesn't have the high frequency dynamic expansion part. Well, either that, or the dynamic expansion part was totally removed from the datasheet, which I find questionable as the actual circuit values and the detailed frequency and phase (which together allow reconstructing the group delay) response curves are presented in great detail.
The BA3888S has essentially an identical Eq implementation, with the addition of the dynamic highs expansion and a noise gate to cut down hiss when no signal is present.
Both datasheets state clearly that they won't sell the IC's unless you have a signed agreement with BBE Sound Inc.
As for the Bass, it is a simple matter of eq'ing the lows via a 2nd order shelving frequency response, which happens to be already available as the main filter is based on a state variable topology which already has low-pass, band-pass and high-pass outputs. The "Lo Contour" name for the bass is again marketing in my opinion. It could be perfectly called "Bass", "Bottom", "Deep Bass" or "Bass Boost" and do a better justice to what's going on with the low frequencies.
Regarding the "time alignment" between the lows, mids and highs, I can state that the 1.5 msec delay mentioned in the related literature is a consequence of the state variable filter itself. When the "Lo Contour" and "Process" are set to the minimum, and leaving apart the dynamic expansion of the highs, one ends up with a 2nd order all-pass filter that has a group delay of exactly 1.5 msec below 100 Hz or so. Adjusting the "Process" knob and the dynamic expansion on the highs do not affect the group delay on the lows, thus this particular "time delay" maintains. When the "Lo Contour" is increased the 1.5 msec time delay varies a little, but remains essentially within +/- 10% of its original value or so, thus the delay relationship between the different frequencies maintains. In other words, the frequency-selective delay can be thought of rather as a consequence of the Eq implementation.
Ah, but someone might argue that the big merit of the Eq lies in the inherent time alignment it introduces, beyond the high and low frequency equalization, and that this is essential to the good sound. Well, that's what I first thought, however when I decided to try the delay alone, i.e. having the filter set just as an all-pass, A-B testing revealed the difference in the perceived attack and brightness when playing chords was really small. There were differences, but very subtle. I tried different delay times and Q's for the pure 2nd order all-pass network in an attempt to find an optimum, however I would rather describe the options as different, but none of them significantly better than the others. Perhaps this time alignment is better noticed when reproducing the material through triaxial speakers with passive crossovers, as each band will be delayed differently and there is more to correct. I don't know for sure, and my interest here goes to the guitar domain where most of the times you don't have crossovers.
In summary, the simpler Eq only implementation (without highs expansion) made a BIG improvement in my guitar & amp sound, however when I tweaked the frequency and Q of the eq filter the sound I could get became out of this world! Really.
In particular, for the neck humbucker pickup and the amp with all knobs at noon, my favorite setting was 12 dB boost for the highs, 6 dB boost for the lows, central filter frequency reduced to 70% its original value, and filter Q increased a bit. Now the 6th and 5th strings produce fat and deep bass without detracting from the clarity of the highs. Doing chord work produces very nice cutting sounds with the strums. Notes across all strings sound balanced. I know that I am sounding like a marketing person, but I am not trying to sell anything here.
I recognize that the dynamic expansion part might add an extra dimension to the sound, but currently my sound improved so much that I don't feel like I need more for now, especially with the customized tuning of the eq/filter.
Quote from: oldrocker on October 18, 2006, 12:13:16 AM
I know why people ask if there's a DIY for a Sonic Max. Because it sounds like an easy thing for a DIY guru to make. But I guess there is more to it than meets the eye. With effects like phasers, flangers, chorus and wah's all cloned or redesigned by all knowing effects and electronic gurus, it seems hard to believe a simple tone enhancement effect would be so long in coming. I guess if it was that easy to re-create BBE wouldn't be doing as good as they are selling their units.
I think the lack of commercial clones is due to the fact that the process is patented, and neither the name nor the implementation might be used by a manufacturer without being running into trouble.
The lack of DIY clones can be justified considering that the IC's that implement the BBE process are not sold to the general public, thus you cannot readily clone an existing device.
The option left is to reverse engineer the process itself in order to make a custom designed implementation, while dealing with the different sources of obscure pseudotechnical information that lead to believe it is something so special that it just can't be made up.
I think the above reasons justify why there hasn't been an attempt of a DIY version... until now.
I insist that my intent here has been only to demistify the technical beliefs around this, without detracting from the sonic improvement of the process which is real. In fact, I admire how Mr. Berry was able to market and protect his invention. Just think how it is spread massively in audio and TV consumer equipment!
------------------------
Quote from: Doug_H on October 18, 2006, 03:34:05 PM
FWIW, my bass player tried it and didn't like it. He plays a Jazz bass derivative and I don't think there's enough hi-freq content in it (or maybe his amp) for the BBE to make it interesting. So the "process" control didn't do much and the bass boost just farted out the lows on his amp. It might be more interesting in a crisper sounding bass rig that's bi-amped, using an Alembic or Rick or something.
Yes, this kind of circuit doesn't do much when highs are not there in the first place. Probably an Aphex Aural Exciter or a similar device would do better for him.
On the other hand, the BBE process is clearly targeted for the hi-fi audio market, i.e. 20 Hz to 20 kHz, thus it is fine tuned for music rather than to a particular musical instrument (I am not aware that the pedal versions are tuned differently for instruments).
For guitar playing, I found a great improvement when reducing the filter's frequency to 70% of its value--well, in fact to 66% or 1.5x less of its original value. I have the feeling that for bass dropping the frequency of the filter to 50% or so might be a much better match for the instrument in terms of the Process knob. Regarding the Lo Contour knob, maybe dropping the frequency might do good as well as it might reinforce the bass closer to the point where the speaker's response is falling, who knows.
One thing I forgot to mention based on my experiments is that the state-variable filter implementation allows you to CUT bass as well. Indeed, I fully cut the bass and with the highs raised I got a very VOX-like sound which was quite appealing also.
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- KVRian
- Topic Starter
- 1050 posts since 4 Nov, 2003 from Washington DC
Drums and vocals would be perfect DJ Z! Thanks so much!
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- KVRist
- 293 posts since 2 Aug, 2002 from Odense - Denmark
Better do it on a finished mix that hasn't yet been mastered. Even better yet no electronic music, but a real band.DJ Z wrote:I'll upload some stuff guys...just tell me what would be a good example for you?
drums?
vocals?
And in the end, the love you take is equal to the love you make.