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A better version of The Blue Rain Coat


STC

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I have previously requested audiophiles to give feedback on my new attempt to attempt to play the existing stereo files with something similar to what Polk is currently doing. However, since no one in this forum provided any feedback, I dropped the idea . The feedback are important as I require many good systems to evaluate the sound for it to work with different speakers and position. Not all audiophiles follow the equilateral triangle rule so the only way I could get this right is by getting inputs from other users.

 

Anyway, I just posted it to the Youtube without making reference to it in any forum. Finally after two months it somehow caught Brian of SlossAudio's attention. You can check out his profile at  https://www.slossaudio.com/work.html . Here is what he commented : -

 

"I'm using Focal cms 65s in a 60deg setup and wow, it doesn't even hardly sound like the same song. Interestingly, I found listening to the (Ambiophonic?) version uncomfortable at first because it's SO wide. However, after multiple listens, the separation of instruments does seem much better. It feels like everything in the mix has way more space to breathe and it makes the other version seem cluttered. I'm curious about your process, also curious how this sort of a thing translated between different systems that are set up differently. Have you ever mixed something on a crosstalk-free system and seen how it translates back to a regular stereo system? Also, i'm not really sure what's going on when I stand off axis and which one I like better, i'm definitely getting the majority of the effect in the sweet spot."

 

I hope you guys could try out for yourself. 

 

https://youtu.be/UptuwWoQ1O8

 
 
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7 hours ago, JoeWhip said:

I played it on my system this morning. The fidelity on the initial tracks was irritating. Your processed tracks did sound better and decluttered the center of the soundfield and spread out the vocals. On the Warnes track, instruments were pushed out laterally as well as front to back. There was more ambiance too. The vocal hung right in the middle. 


Thank you very much for the feedback. I was hoping to hear comments related to depth and stage and you have have provided that information which is very important. 
 

I would appreciate if you could describe the angle of the speakers and equipment used. You PM me directly if you want to keep the information private. I will also PM you a full 24/96 version of two or three processed tracks in a week or two. 
 

Once again thank you!

 

Warm regards,

ST

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1 hour ago, Ralf11 said:

you can get some hot HF ones like Sandy has too


I am trying keep this thread unmoderated. Grow up and settle your score like a grown adult. WTF Sandy’s headphone got to do with this thread???????  Or WTF headphones got to do with OP??????  This is a effort for people with normal hearing with well formed pinnae who can and have the luxury to have a decent loudspeakers playback at home. Which part of it is difficult to understand????  
 

@sandyk maybe you are looking for this kind of posts to put forward your side of the story for Chris’s attention. 

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32 minutes ago, tmtomh said:

 

It is a very interesting effect, and it appears to be the converse (if not exactly the opposite) of an ambience filter: it sounds like it adds out-of-phase information from each channel to the opposite channel, but to reduce crosstalk rather than to add reverb or the feeling of a larger hall/room.

 

It's an interesting effect, and because it doesn't seem to add reverb per se, it doesn't sound nearly as artificial or processed as conventional ambience processing.

 

It is not adding anything. And the original signal was unaltered in anyway except in this case it was upsampled for recording purpose. You are supposed to hear the sound as close to what you hear with headphones but a direct comparison cannot be made unless your headphones have your pinna correction filter. 

 

 

32 minutes ago, tmtomh said:

 

That said, I do tend to agree with semente - the effect is rather extreme and a bit artificial-sounding, and I would think a subtler/more modest application of this DSP would be better for longer-term listening, to more different kinds of material. I have plenty of music in my collection - and this is regular rock, not chamber music, acoustic jazz, or whiz-bang-soundstage audiophile recordings - where the L-R soundstage already is quite wide. This effect would make such music sound bizarre and totally unpleasant. 

 

But it is an interesting effect that I believe can benefit some musical sources. I would like to learn more about it - is anything like this available as an off-the-shelf DSP plugin?

 

This Is something I developed for my own Ambiophonics setup and realized it wasn’t too sensitive to exact speakers position and allowed greater flexibility . So I thought I could developed a different one for conventional stereo setup like what Polk latest speakers are doing. That the simplified version of XTC was the one you are hearing in the video. I am surprised with the comments of the stage width but I think I know why. They were referenced to a recording where the exact position of the instrument is predetermined and recorded accurately. However, I only tested them with the cheap Edifier and Sonic gear speakers. So it can be the speakers Vertical dispersion that is affecting the soundstage. I can make the stage smaller. What really mattered here was to have the separation between the instruments. I think I have crossed the first hurdle and now I need to figure out the excessive stage. 
 

How big is the stage you think?  90 or 120 degrees?  If you get it bigger looks like the theory of ITD for localization may need to be rewritten. 

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16 minutes ago, tmtomh said:

 

Thanks! I am confused, though: how can the DSP cancel/reduce L-R crosstalk without adding (or subtracting via the addition of out-of-phase info) something to the original signal?


This is going to be hard to explain. :)  I will do my best....

 

Unplug your right speaker and play your favorite track at a very low volume in a quiet room. The level must be very low so that when you plug your right ear with your finger you should not be able to hear the left speaker‘s sound. You will notice change in timbre ( and also a shift in the position but that’s not relevant to the question you ask). 
 

This is the original of the left channel and received by your left ear. Now you put your headphones by muting the right channel and you are hearing the same as what your left ear heard with the loudspeaker setup with the right speaker muted. NOTE: There will be difference in the tonality anyway due to pinna function where you may find the sound of headphones a little less bright compared to the sound of the loudspeakers. The pinna modify the frequencies from 2 to 5KHz where else when the same sound played through the headphones it hits directly to the ear canal without pinna shaping effect. The final tonal balance can vary depending on the type of the headphone and how much bigger it is covering the ears. Earbuds completely eliminate any role of the pinna function so again all these will sound different from one another even if the loudspeaker and the headphones got identical frequency response measurements. 
 

The only way to find out is to take the measurement inside the ear canal just before the ear drum and apply the correction filter to the headphones. But once you do that you will sort of externalize the headphone sound and you can feel the sound to be coming from the front. ( This is what Ambiophonics about but then it also involves moving the speakers very close AND it is not related to your question but it is the basis for the DSP development. This is just the gist to explain why there is no DSP involved in the original sound so do not read too much into it because this itself is very complicated subject. 
 

Now going back to your question, my DSP ( actually there is no program yet and this is done by using DAW  with just delays and phase), presents the original signal untouched from the speaker to the ear. The ‘DSP’ is only concerned with the unwanted sound reaching the opposite ear which is what happens when you put your headphones as the opposite ear cannot hear the sound meant for the respective ear/channel.  
 

Can you follow this far? :)  

 

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6 minutes ago, tmtomh said:

But when you write "this is done by using DAW with just delays and phase," that is the alteration of the original signal, yes? And it includes altering the phase and timing of some of the signal, yes?


The original signal is not touched. All the cancellation is done by the copy of the the original signal. if you take the measurement of the first sound emerges each speakers of the processed and unprocessed file, they are identical. What is altered is the copy of the original signal delayed and inverted and send to the other channel. 
 

A better way to understand and to do is, Have one pair of speaker that is producing the original recording and another pair for the processed file. It is important that the original signal is produced untouched to avoid any coloration otherwise this method is no different from any other DSP attempt. 

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5 minutes ago, STC said:


The original signal is not touched. All the cancellation is done by the copy of the the original signal. if you take the measurement of the first sound emerges each speakers of the processed and unprocessed file, they are identical. What is altered is the copy of the original signal delayed and inverted and send to the other channel. 
 

A better way to understand and to do is, Have one pair of speaker that is producing the original recording and another pair for the processed file. It is important that the original signal is produced untouched to avoid any coloration otherwise this method is no different from any other DSP attempt. 


The processed file I refer to the above is the processed file with system of having two pair of front speakers. In my example since the whole thing is utilizing one pair of front speakers. Those processed files will be a different type but the original sound is reproduced in bothe instances untouched. 

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24 minutes ago, tmtomh said:

 

Again, many thanks for your patience and detail.

 

I do not mean to sound disagreeable or cranky, but "the copy of the original signal delayed and inverted and sent to the other channel" is exactly what I hypothesized several comments ago when I said it sounds like a phase-inverted portion of each channel is added to the other channel. You said that was not the case - but it turns out it is indeed the case.


Yes you are right. Rereading your post  and my response, I have inadvertently misled you. Crosstalk cancellation works with phase inversion and I took it for granted that it will already be known when XTC is referred to. 

 

I was overly eager to emphasize that RACE was based by not altering the original signal And reproducing them in the original form. It only deals with the copies. Sometimes it is hard to convey what I wanted to say because I am seeing this from a different POV and not understanding others POV on this subject is different and from different angle. 
 

Sorry for the confusion. 

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3 minutes ago, sandyk said:

 

I did not post that link to attack your video. Even compared with the original LPCM version the improved soundstage and instrument separation should be quite obvious. I do agree with others though that the effect is currently a little extreme.


I wasn’t attacking you. I thought you may wanted a better resolution which is fair considering I am in a high end forum. I think I am just more preoccupied with my thoughts that I respond without really trying to understand what the other person trying to say. Anyway, try listening to the hires file and tell me what you think?  Thanx. 

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  • 2 months later...

The stereo at 60 degrees progress here. In the previous version there were comments about the stage being too wide. Frankly, this is not easy to address because I really have no idea how big was the original stage.  I am really on my own recording to determine the correct stage and so far the feedback is encouraging. 
 

Here are two samples and which one you perceive better depth and stage? State whether you used headphones or loudspeakers. The volume level was not matched so you need to adjust from your side. 
 

the lossless files:-

 

https://1drv.ms/u/s!Avexw_l7DM5sgqt9Jry6P8b5N9jPXw

 

 

https://1drv.ms/u/s!Avexw_l7DM5sgqt88Tx48P1asifoMg

 

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15 minutes ago, John Dyson said:

I use normally use headphones and used heaphones for the comparison -- the B recording is more natural sounding, it appears that the A recording was matrixed somehow.   The B recording IS FeralA, and sounds REALLY GOOD when decoded.   A snippet of B, but decoded is here. (At low levels llater in the recording, if not encoded would jerk badly.)   The response balance might not be perfect -- there is a small amount of room for adjustments for frequency response balance and still be within spec.

 

B.mp3 2.1 MB · 12 downloads


Thanks for taking your time to listen but did you download the old file?  Your sample is not the files in my post. BTW, my samples were from Chesky and I don’t think DolbyA was involved here. 

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7 minutes ago, John Dyson said:

He might have started with a FeralA master.   All I know is what I hear...  If there isn't any jerking in my result (because of the low levels), then *some kind* of compression is being undone by the attempted decoding.   I cannot really 100% claim that it is feralA, but I can definitely note the compression.   I'll review the old file also -- I was just astounded when I heard the recording and heard the compression (my hearing is 10000% attuned to compression FOR SOME REASON :-).)   I don't know if other people can hear compression -- you know about the perception thing, a huge part of it is the 'brain' processing.  Mine is trained to hear those things.  I promise that I will look at the other recording poste haste.

 

BTW -- the YouTube copy was also FeralA (the 'tell' on that is the boosted highs at the low levels. I did do the decode on that also.)

 

*  I REALLY DO NOT WANT TO PESTER PEOPLE ON THE *FeralA* SUBJECT, but stereo imaging is very significantly modified by dynamic range compression, and I wanted to make the *compression* known, the FeralA is secondary.

 

John

 

 


Are you referring to post #33?  

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47 minutes ago, John Dyson said:

Yes...   I also successfully decoded the file 'Allegro*.wav'. (Not 100% if 100% correct, but the decoding is reasonably ok.)

 

I didn't try to decode A.wav because it twisted my perception on my headphones.  (Headphones are the only thing that I trust for my work, so I am not critical enough & my listening is technically incompetent on speakers.)

(As always, sorry for the mp3 on Allegro , but it communicates what I am talking about..)

 

Also, I included a better decoded B.wav...  It is sometimes tricky, especially since I just developed new, simpler and more accurate formulas....

 

John

 

Allegro Energico e Passionato.wav-decoded.mp3 1.91 MB · 0 downloads

B.wav-decoded.mp3 2.44 MB · 0 downloads

 

 

But this  is my version of specially crosstalk cancelled for typical stereo setup at 60 or so degrees.

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55 minutes ago, JoeWhip said:

Sorry but I lost track of this thread as I have been away a good bit since the original post. My speakers are Vandersteen 3A Signatures with no toe in at all. In a dedicated room out into the room about 4 feet and about 6 feet apart.

 

No worries. Hope you met the penguins :) .

 

Try the recording in post #33. 

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7 hours ago, John Dyson said:

I am SOOO sorry for diverting the topic.  Your evaluation/comparison project is good.  FORGET anything about the word 'FeralA', as I was more intending that 'dynamic range compression' being a variable often not considered.  FeralA happens to be one of the manifestations of dynamic range compression.

 

JOhn

 

 

You are contribution is important. Anyway I only one feedback so I have to make use of it. :) 

 

I usually let music students and non audiophiles to do the blind tests live. Audiophiles live in their own world with their own rules which they themselves cannot be sure of. 

 

In your case, with only one valuable feedback, I am unable to make use of it because your decoded version was  A. You also mentioned that you decoded the original file and the decoded version sounded much better than the original. However, your sample was a decoded file mine. So the version you thought better was my version which you decoded. Anyway, now it is too late as the opinion will be clouded due to this confusion. Thanks for the contribution.

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