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Bauer stereophonic-to-binaural DSP plugin, new plugin
Segovia
post Dec 21 2006, 17:25
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I love this plugin. I've been using/testing it for about a month, and I must say I find it difficult to listen to headphones without it now. Thanks Boris. smile.gif

In my opinion, the very slight changes in frequency response are far outweighed by the benefits: a more natural soundstage and drastically reduced fatigue in long listening sessions.
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cbrigstocke
post Dec 30 2006, 19:44
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I don't know if anyone else has experienced this, but when I, using foobar2000, make MP3s with the DSP turned on, I get a ton of artifacts. It seems to happen if there are two instances of the plugin being used. I've figured out a workaround which is: 1) Don't playback music while converting with the DSP on. 2) Check "don't reset DSP..." (This prevents foobar2000 from opening up two threads since I have an AthlonX2.)

This is a great DSP and I like creating MP3s with it since I only use MP3s with my portable. Perhaps somewhere down the road someone can take a look at this. Thanks for all the work!
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w1L50n
post Jan 24 2007, 00:13
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QUOTE (cbrigstocke @ Dec 30 2006, 11:44) *
I don't know if anyone else has experienced this, but when I, using foobar2000, make MP3s with the DSP turned on, I get a ton of artifacts. It seems to happen if there are two instances of the plugin being used. I've figured out a workaround which is: 1) Don't playback music while converting with the DSP on. 2) Check "don't reset DSP..." (This prevents foobar2000 from opening up two threads since I have an AthlonX2.)

This is a great DSP and I like creating MP3s with it since I only use MP3s with my portable. Perhaps somewhere down the road someone can take a look at this. Thanks for all the work!


I found the same problem and your work around also worked for me.
But it led me to wonder about the proper way to use the DSP's in foobar. I use EAC to rip to .flac, then I use fb2k to convert to .mp3, using the Bauer plugin. So now listening to the mp3 on a portable, it's hard to hear any difference (although I tried back to back comparisons with a limited number of music types). I am assuming that in converting the .mp3, fb2k/Bauer Plugin modifies the actual music data (as opposed to something lik replaygain writing a value to tags). So, if that's right, any portable should reproduce whatever (however subtle) has been done to that data by the Bauer plugin. Are my assumptions correct? comments? anybody..anybody..
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cbrigstocke
post Jan 31 2007, 00:16
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Your assumptions are correct. The dsp modifies the audio data and then that modified data is encoded. The dsp is subtle by nature and only meant to relieve listening fatigue. As a side-note, I also apply replaygain when I make mp3s to help reduce clipping problems.
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w1L50n
post Jan 31 2007, 16:28
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QUOTE (cbrigstocke @ Jan 30 2007, 16:16) *
Your assumptions are correct. The dsp modifies the audio data and then that modified data is encoded. The dsp is subtle by nature and only meant to relieve listening fatigue. As a side-note, I also apply replaygain when I make mp3s to help reduce clipping problems.


Yes, I do that too.
I also use the 'Advanced Limiter' after the Bauer DSP. That is the extent of my .mp3 manipulation.

I convert the .flac to .mp3 inducing the Bauer then Advanced Limiter; then I run mp3Gain on them (although I have another thread going and there is some question as to that, and I haven't gotten to bottom of my confusion there yet).
Would you recommend any other DSP's? I use mp3 pretty much exclusively on a portable with a fair amount of outside noice pollution.
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Azultra
post Feb 3 2007, 15:56
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QUOTE (Madman2003 @ Dec 1 2006, 22:40) *
I made a ladspa plugin based on the bs2b library, so linux users can enjoy it as well in the future. It will most likely be included in the next tap-plugins release. (it's not my set, but i figured that would be the easiest way to get it to spread) It's currently in cvs if anyone is curious.

You've made my day! biggrin.gif
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5cylinders
post Apr 8 2007, 06:54
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great work!

Does anyone else use bs2p along with dolbyhp(foo_dsp_dolbyhp.dll) ?
I place bs2p below the dolbyhp (in DSP manager) . Is this the right order?
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Hancoque
post Apr 9 2007, 13:56
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There is no right order for these two plugins as they serve the same purpose. You should only use one of them at a time.
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5cylinders
post Apr 10 2007, 00:39
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QUOTE (Hancoque @ Apr 9 2007, 06:56) *
There is no right order for these two plugins as they serve the same purpose. You should only use one of them at a time.


Thank you !
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BaldByChoice
post May 27 2007, 20:37
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Thanks a lot for this excellent plugin! smile.gif I read earlier in this thread about a VST version of this. The link, however, doesn't work...
I do a lot of mixing through headphones ('coz I work when the family sleeps) and I would absolutely LOVE to be able to use this
in my VST host.
Any chance of getting the VST version again, please...? smile.gif

This post has been edited by BaldByChoice: May 27 2007, 21:58
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boris_mikhaylov
post Jun 4 2007, 07:15
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QUOTE (BaldByChoice @ May 28 2007, 01:37) *
Thanks a lot for this excellent plugin! smile.gif I read earlier in this thread about a VST version of this. The link, however, doesn't work...

I have not make VST plugin yet :-(
What link about you say?


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BaldByChoice
post Jun 9 2007, 13:39
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QUOTE (boris_mikhaylov @ Jun 4 2007, 06:15) *
QUOTE (BaldByChoice @ May 28 2007, 01:37) *

Thanks a lot for this excellent plugin! smile.gif I read earlier in this thread about a VST version of this. The link, however, doesn't work...

I have not make VST plugin yet :-(
What link about you say?


Oh, sorry. I must have misunderstood. I thought the link halfway through the first page contained a link to a VST version of our excellent plugin.
Would it be possible for you to make a VST version of bs2b? It would totally rule to be able to mix through my headphones when the rest of my family sleeps. smile.gif

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buktore
post Jun 9 2007, 14:45
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There is a VST Plugin that HOST Winamp Plugin.

http://www.hydrogenaudio.org/forums/index....mp;#entry497444

It's still a little bit buggy. but it's work.
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BaldByChoice
post Jun 12 2007, 09:03
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QUOTE (buktore @ Jun 9 2007, 13:45) *
There is a VST Plugin that HOST Winamp Plugin.

http://www.hydrogenaudio.org/forums/index....mp;#entry497444

It's still a little bit buggy. but it's work.


Thanks a lot! I will try this plugin today. smile.gif
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BaldByChoice
post Jun 15 2007, 11:32
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Nope, this didn't work... sad.gif
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Bad Monkey
post Aug 22 2007, 07:00
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Okay I've read this entire thread, plus looked over Boris's site, but I'm still confused about the settings for the plug-in.

Could Boris, or someone else who understands, explain - in noobish:
  • what the slider "crossfeed level" actually does (amplifies degree of processing? narrows the soundstage by using different values in the same formula? uses a different logic altogether?)
  • what the "easy" on/off does and how this affects each of the three crossfeed level settings

crying.gif

I guess what I want is a high level explanation in binaural terms.

From my own point of view I'm most interested in working out how to reproduce the most realistic sound, and it isn't possible to do that just by listening if you're not familiar with the original pre-recording sound.

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boris_mikhaylov
post Aug 23 2007, 06:06
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QUOTE (Bad Monkey @ Aug 22 2007, 12:00) *
  • what the slider "crossfeed level" actually does (amplifies degree of processing? narrows the soundstage by using different values in the same formula? uses a different logic altogether?)
  • what the "easy" on/off does and how this affects each of the three crossfeed level settings


Info from http://bs2b.sourceforge.net/ :
---
# perl bs2b-H-transform.pl 700 -5.5 -2.5 # high crossfeed level
# perl bs2b-H-transform.pl 500 -6.75 -2.25 # middle crossfeed level
# perl bs2b-H-transform.pl 360 -8 -2 # low crossfeed level
# perl bs2b-H-transform.pl 700 -8 -2 # high easy crossfeed level
# perl bs2b-H-transform.pl 500 -9 -1.8 # middle easy crossfeed level
# perl bs2b-H-transform.pl 360 -10 -1.6 # low easy crossfeed level
...
$Fc = shift; # Lowpass filter cut frequency (Hz)
$Gd = shift; # Lowpass filter gain (dB)
$Ad_h = shift; # Highboost filter gain (dB) (0dB is highs)
---
For example: *high* *easy* crossfeed level =
700 Hz Lowpass filter cut frequency,
-8 dB of Lowpass filter gain
-2 dB of Highboost filter gain where 0 dB is highboosted level
This means that feed level is 6 dB (-2 - -8) at lows and 700 Hz cutoff

QUOTE (Bad Monkey @ Aug 22 2007, 12:00) *
I guess what I want is a high level explanation in binaural terms.


From bs2b home: ---
This responses looks as a virtual stereo speakers are moved from 30 degrees azimuth (high crossfeed level) to 60 degrees azimuth (low crossfeed level).
---

And check:
http://bs2b.sourceforge.net/bs2b.html#References

http://headwize.com/projects/showfile.php?file=cmoy1_prj.htm

http://www.linkwitzlab.com/publications.ht...sfeed%20circuit


PS
I will make more flexibility in the next release by using cutoff frequency and mix (or crossfeed) level in the presets.

This post has been edited by boris_mikhaylov: Aug 23 2007, 06:16


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Bad Monkey
post Aug 23 2007, 07:09
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unsure.gif

lol thanks Boris but that was NOT noobish!

I'll try to look over those references but, although they all appear to be very interesting from a technical perspective, they do not directly help me with my questions.

I am sure I speak for many interested users of your plug-in when I say that you need to provide a concise and easy-to-understand high level explanation of what the settings mean. Not everyone has hours of time to study the issue in detail - and many not the ability in any case.

My own understanding of cross-feed in general extends to something about mixing a bit of left into right, and vice versa crying.gif ... I do know that binaural recording is very difficult on account of the variance between the shape of peoples' heads and ears, so perhaps you could recommend BS2B settings for different applications.

With respect, answering the Q by way of referencing circuit designs and dozens of technical white papers is not helpful. But thanks anyway!
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nawhead
post Aug 28 2007, 23:21
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QUOTE (Bad Monkey @ Aug 23 2007, 01:09) *
unsure.gif

lol thanks Boris but that was NOT noobish!

I'll try to look over those references but, although they all appear to be very interesting from a technical perspective, they do not directly help me with my questions.

I am sure I speak for many interested users of your plug-in when I say that you need to provide a concise and easy-to-understand high level explanation of what the settings mean. Not everyone has hours of time to study the issue in detail - and many not the ability in any case.

My own understanding of cross-feed in general extends to something about mixing a bit of left into right, and vice versa crying.gif ... I do know that binaural recording is very difficult on account of the variance between the shape of peoples' heads and ears, so perhaps you could recommend BS2B settings for different applications.

With respect, answering the Q by way of referencing circuit designs and dozens of technical white papers is not helpful. But thanks anyway!


Here's how I understand it. Someone please correct me if I'm way off here.

Crossfeed level
min : You're in a room with speakers placed almost directly left and right of you (60 degrees azimuth).
med: You're in a room with speakers at 10 and 2 'o clock.
max: You're in a room with speakers at 11 and 1 'o clock (30 degrees azimuth).

Easy OFF: Mix the left and right signals normally. Result is mild treble rolloff.

Easy ON: Highgain boost or "Easy on the mix, baby!" Not as much treble rolloff.


And to Boris, amazing plug-in! Thanks for all your hard work.

This post has been edited by nawhead: Aug 28 2007, 23:43
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Bad Monkey
post Aug 30 2007, 17:56
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Thanks nawhead.

Now, in 50 words or less biggrin.gif why would mixing the channels "normally" result in a treble rolloff? And, following, why would anyone want the "Easy" option to be anything but on?
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boris_mikhaylov
post Aug 31 2007, 15:15
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QUOTE (nawhead @ Aug 29 2007, 04:21) *
Crossfeed level
min : You're in a room with speakers placed almost directly left and right of you (60 degrees azimuth).
med: You're in a room with speakers at 10 and 2 'o clock.
max: You're in a room with speakers at 11 and 1 'o clock (30 degrees azimuth).

Not in room. There is not reverberation.

QUOTE (nawhead @ Aug 29 2007, 04:21) *
Easy OFF: Mix the left and right signals normally. Result is mild treble rolloff.
Easy ON: Highgain boost or "Easy on the mix, baby!" Not as much treble rolloff.

May be 'Easy' means much closer seating to virtual speakers. huh.gif
Longer distsnce - less highs, more mix. It's a wave phisics. wink.gif E=mc2


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boris_mikhaylov
post Aug 31 2007, 15:50
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QUOTE (Bad Monkey @ Aug 30 2007, 22:56) *
Now, in 50 words or less biggrin.gif why would mixing the channels "normally" result in a treble rolloff? And, following, why would anyone want the "Easy" option to be anything but on?

May be Chu Moy (http://headwize.com/projects/showfile.php?file=cmoy1_prj.htm) has an answers:

... decreasing the value of R1 would "widen" the soundstage and create a smoother response with the existing treble boost. ... I set R1a to 200 ohms (40% of the original value) (my note: original one is Linkwitz version).
Since all recordings are not the same, I added a "PERSPECTIVE" switch (S1) to customize the processing with an alternate R1 value. At R1b = 150 ohms, the low frequency separation between channels goes up to about 10dB and the overall output increases by about 2dB. Toggling from R1b to R1a, the soundstage appears to move further away (lower output, more narrow soundstage, slightly softened highs)."

Anyway, why asking what someone hear? Yours feelings is more interesting. wink.gif


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nawhead
post Sep 1 2007, 00:03
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QUOTE (boris_mikhaylov @ Aug 31 2007, 10:15) *
QUOTE (nawhead @ Aug 29 2007, 04:21) *

Crossfeed level
min : You're in a room with speakers placed almost directly left and right of you (60 degrees azimuth).
med: You're in a room with speakers at 10 and 2 'o clock.
max: You're in a room with speakers at 11 and 1 'o clock (30 degrees azimuth).

Not in room. There is not reverberation.


Ok, not in a room. A better description might be "virtual soundstage."
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nawhead
post Sep 1 2007, 00:30
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QUOTE (boris_mikhaylov @ Aug 31 2007, 10:50) *
QUOTE (Bad Monkey @ Aug 30 2007, 22:56) *

Now, in 50 words or less biggrin.gif why would mixing the channels "normally" result in a treble rolloff? And, following, why would anyone want the "Easy" option to be anything but on?

May be Chu Moy (http://headwize.com/projects/showfile.php?file=cmoy1_prj.htm) has an answers:

... decreasing the value of R1 would "widen" the soundstage and create a smoother response with the existing treble boost. ... I set R1a to 200 ohms (40% of the original value) (my note: original one is Linkwitz version).
Since all recordings are not the same, I added a "PERSPECTIVE" switch (S1) to customize the processing with an alternate R1 value. At R1b = 150 ohms, the low frequency separation between channels goes up to about 10dB and the overall output increases by about 2dB. Toggling from R1b to R1a, the soundstage appears to move further away (lower output, more narrow soundstage, slightly softened highs)."

Anyway, why asking what someone hear? Yours feelings is more interesting. wink.gif


Here's a more layman description of Chu Moy's PERSPECTIVE switch which can be found further down on his project page:

"With the PERSPECTIVE switch set to high crossfeed, the soundstage narrowed and the top treble softened, yet there was more depth, more dimensionality - as though it had been moved further back. Recordings that polarized the stereo presentation with instruments or vocals to the extreme left and right had an substantially improved sense of aural continuity."

So I think Boris' EASY switch is similar to Chu Moy's PERSPECTIVE switch, except logically reversed.

Easy switch OFF (bs2b) = Perspective switch HIGH (Chu Moy)
Easy switch ON (bs2b) = Perspective switch LOW (Chu Moy)

But I think with 3 Crossfeed settings, the Easy switch makes it confusing, since the overall result is the same as adding less or more crossfeed and/or moving closer or farther away from the virtual soundstage.

The bs2b plug-in has 6 distinct settings, as already documented:
CODE
3: 700 -5.5  -2.5  # high crossfeed level
2: 500 -6.75 -2.25 # middle crossfeed level
1: 360 -8    -2    # low crossfeed level
6: 700 -8    -2    # high easy crossfeed level
5: 500 -9    -1.8  # middle easy crossfeed level
4: 360 -10   -1.6  # low easy crossfeed level


If I try to interpret the settings from a purely virtual soundstage representation, "4: low easy [on] crossfeed" should be closest to the soundstage whereas "3: high [easy off] crossfeed" should be farthest away.

But in testing by ear, this is not what happens. Going from setting 6 to 1, the soundstage becomes closer, not farther away, so these 2 middle settings are counter-intuitive. And settings 5 and 2 are much too similar to my ears.

So I think it would be easier for users if you delete the Easy switch, and simplified to just 4 settings in soundstage logic order:
3 "High" (700 -5.5 -2.5)
6 "High easy" (700 -8 -2)
1 "Low" (360 -8 -2)
4 "Low easy" (360 -10 -1.6).

unsure.gif

This post has been edited by nawhead: Sep 1 2007, 07:27
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boris_mikhaylov
post Oct 9 2007, 16:14
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VST plugin released.
Thanks to Vlad Goncharov.


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