Correcting for soundstage?

Discussion in 'Sound Science' started by oblongdarts, Nov 15, 2017.
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  1. Whitigir
    You can only know where the positioning supposed to be when you are the engineers who recorded and edited the track
     
  2. jgazal
    Glad that you clarified.

    Are you sure oblongdarts and the company he mentions had in their minds the same definition of spatial distortion you've just mentioned when they wrote the following assertion?

     
    Last edited: Nov 16, 2017
  3. 71 dB
    I don't know their definition, never heard of them. So, not sure at all.
     
  4. RRod
    What do you mean by 'electronic crossfeed'? Convolving the PRIR for a given (mono) channel should already yield a stereo signal including the crossfeed.
     
  5. jgazal
    I was imagining the following feature in the Realiser processor:

     
  6. RRod
    Ah I see, so it's optional crosstalk cancellation for playback scenarios (e.g. binaural recordings) that benefit from it?
     
  7. jgazal
    It is very hard to be precise with terminology. I wouldn’t say it is crosstalk cancellation since “eletronic crossfeed” or if you prefer “digital domain crossfeed” need to be accounted for if your goal is to emulate with headphones the acoustic crosstalk one would have when playing more than than one channel in more than one loudspeakers. In other words, acoustic crosstalk is not inherently to headphones playback, so crossfeed is electronically/digitally added.

    Binaural recordings benefit from perfect crosstalk cancellation. Stereo recordings with natural ITD and ILD also.

    But there are stereo recordings with unnatural ITD and ILD in which tracks are purely assigned to one channel. When listening to such recordings you may benefit not from avoiding acoustic crosstalk or not adding digital crossfeed, but from adding less dB of digital crossfeed than dB of crosstalk one would have with more than one loudspeakers in a room.

    Does it make sense?

    Anyway:

     
    Last edited: Nov 24, 2017
  8. Erik Garci
  9. Erik Garci
    That would be much easier than manually muting the microphones during measurements, and just about any PRIR could be used.

    Allowing fractional values would be even better, such as 0.5 (-6 dB) or 0.1 (-20 dB).
     
    Last edited: Nov 24, 2017
    jgazal likes this.
  10. jgazal
    I don’t know if my question really address the issue, but let’s wait for they answer:

     
    Last edited: Nov 26, 2017
  11. ironmine
    It's easy to achieve this effect (with speakers, not headphones) with the help of an mid-side equalizer (I tried it with FabFilter Pro-Q 2, it's one of the best, most transparent VST equalizers):

    Create two points in Pro-Q:
    1) Mode - "Mid", type - "Low Shelf", frequency - 400 Hz, Q - 0.3, Gain - -4 dB.
    2) Mode - "Side", type - "Low Shelf", frequency - 400 Hz, Q - 0.3, Gain - +4 dB.

    By doing so, we exaggerate the differences between the left and right channels in low frequencies.
    You will perceive this effect as the widening of stereo image and the clearing of the middle.

    If your speakers are situated too narrowly (not enough distance between them), this trick can help you to achieve a better stereo image and a wider soundstage.
     
  12. sander99
    [Edit: better skip the rest of this post go to my next post, as I discovered a serious flaw, I leave this for who is interested in my line of thought]

    If or as long Smyth doesn't implement your requested cross-talk control, maybe this is helpful for your experimentation plans:
    I have an idea for creating a PRIR (or preset combining speakers from 2 special PRIRs) with wich you could "regulate" the amount of crosstalk. This PRIR or preset would use 2 x N channels of the Realiser for N speakers, each channel must be input twice (using the analog multi channel inputs). By changing the input levels you can regulate the amount of crosstalk.

    For simplicity I will first descibe the idea for one normal pair of stereo speakers.

    Suppose input channel 1 is left input channel, input channel 2 is right input channel, then a normal PRIR measurement results in a PRIR that makes the Realiser do the following calculation:

    LeftOut = TransferFunctionSpeaker1ToLeftEar(InputChannel1)
    + TransferFunctionSpeaker2ToLeftEar(InputChannel2)

    RightOut = TransferFunctionSpeaker1ToRightEar(InputChannel1)
    + TransferFunctionSpeaker2ToRightEar(InputChannel2)


    When you make a cross-talk free PRIR you effectively take out half (in this case one of the 2) of the summation parts out of each channel, you get:

    LeftOut = TransferFunctionSpeaker1ToLeftEar(InputChannel1)

    RightOut = TransferFunctionSpeaker2ToRightEar(InputChannel2)


    With a similar method (plug out the opposite mic everytime) you could create a PRIR that takes out the other half (in this case the other of the 2) of the summation parts, this would be a cross-talk-ONLY PRIR:

    LeftOut = TransferFunctionSpeaker2ToLeftEar(InputChannel2)

    RightOut = TransferFunctionSpeaker1ToRightEar(InputChannel1)


    Now make a preset in which you select:
    speaker 1 of the cross-talk free PRIR to be used for input channel 1
    speaker 2 of the cross-talk free PRIR to be used for input channel 2
    speaker 1 of the cross-talk only PRIR to be used for input channel 3
    speaker 2 of the cross-talk only PRIR to be used for input channel 4
    (Alternatively you can make one 4 channel PRIR in one go using 2 real speakers, each fed with the summation signal of 2 output channels of the Realiser; left with output channels 1 and 3, right with output channels 2 and 4.)

    Now input the left source channel to input channels 1 and 3, and the right source channel to input channels 2 and 4.
    By attenuating the signals in to inputs 3 and 4 you can reduce the amount of cross-talk to whatever fraction of the real-world cross-talk you like
    (or you could even increase the cross-talk if you wanted).
    I hope the Realiser allows independent attenuation of individual virtual speakers internally, if not you would have to do this attenuation externally before the inputs 3 and 4.

    If you generalise this idea you can use 14 channels (if you have an A16) to do something similar for a 7 channel speaker set-up. I guess you then would not reduce cross-talk for the center speaker, and for the other speakers reduce the cross-talk with an amount that is depending on which position that speaker would be on.

    (You could maybe even do something similar for a 16 speaker set-up, using both user A and B parts of the A16 (now possible because we use only 16 input channels, and sent each input channel to 2 different speakers!), put all cross-talk free virtual speakers under user A, all cross-talk-only virtual speakers under user B, add the headphone-output signals together (use a mixer), now the only question is: can you attenuate individual virtual speakers independently from other speakers, for one user? If not you can only reduce cross-talk equally for all channels by attenuating the user B output - although you could skip for example the center channel by using a normal not cross-talk free measurement for it under user A, and a dummy muted measurement for user B or something like that - I don't know exactly what possibilities there are with the A16.)

    You could also increase the ITD by somehow delaying the cross-talk only virtual speakers, or decrease the ITD by somehow delaying the cross-talk free speakers. Let's hope the A16 has the possibility to delay individual virtual speakers, and with enough precision. Otherwise it is only possible with the "double-input" scenario for 8 speakers maximum, and the delay must be done exteranlly for the proper input signals.
     
    Last edited: Dec 11, 2017
  13. sander99
    On second thought (ahum), there is one big problem with my proposal:
    The correct relative timing between the cross-talk free part and the cross-talk only part of each speaker will be lost if they are measured seperatly.
    Sorry, I will have to think again...

    But first I have to get some sleep.
     
  14. sander99
    But I think I have a solution already, but only if indeed the A16 allows independent attenuation of individual virtual speakers - and independent for each user - internally (with respect to controlling the amount of cross-talk) and for influencing ITD the A16 should allow independent delay of individual virtual speakers independent for each user. And the bonus of my new idea: you don't have to do a special measurement with muting a certain mic at a certain moment, it can be done with any PRIR, up to 16 channels!
    How? Like this:

    Load the same PRIR for both user A and B.
    Use the left channel of user A headphone output for the left channel of your headphone.
    Use the right channel of user B headphone output for the right channel of your headphone.
    (This can be done by making a special cable, but probably simpler and better - safer to avoid strange left right asymmetric output load for the amplifier? - is to simply use the analog outputs and an external amplifier: connect a stereo RCA/cinch cable, one side: left into user A out left, right into user B out right, other side normally to your stereo headphone amp or other amp with headphone output.)

    Now if for example you want to reduce the cross-talk of the left front speaker, you just have to attenuate the left front speaker for user B (that supplies the right ear).
    If you want to reduce the cross-talk of the right front speaker, you just have to attenuate the right front speaker for user A (that supplies the left ear).
    If you want to increase ITD for the front left speaker, you delay the front left speaker for user B (that supplies the right ear).
    etc. etc.

    [Edit: all this provided the user A and B parts work perfectly in sync]
     
    Last edited: Dec 11, 2017
  15. Erik Garci
    Great idea. However, you would give up head-tracking unless you wear two head-tops.
    The A8 lets you adjust levels in increments of 1 dB. Per speaker and per user (but not mute it per user). To minimize crosstalk, you would set one speaker to +12 dB and the other speaker to -12 dB.
    The A8 lets you adjust delays in increments of 1 ms which is too coarse for ITD purposes. Per speaker and per user.
    I have not tested sync on the A8.
     
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