Stereo Crosstalk Elimination (reduction) Par Excellence!

[STC]

It's not the volume. I think it's my implenentation of the XTC. It needs more work to iron out the details. It's not intensity. It's an excess spread on some recordings and a sort of boxy tonality that I pick up on with music I'm more familiar with. Interestingly it passed other tests just fine, like ambient wind or surf sounds. Those are complex, broad spectrum and tend to reveal hollow or boxy tones pretty well.

I have made of videos of XTC and stereo of classical music before but if you give me a track and about 1 minute duration where you perceive what your perceived, I can make a recording with DPA in ear microphones and we all can learn something from there. I no longer trust my own ears as after 100s of setting with XTC I could be biased .

 

[STC]

BTW guys, this is OT but here is the frequency response of different XTC. I wish I could also get and actual audio samples. Wish you too could shares yours. I would add them to the blog.

Frequency response of XTC – uBACCH and Ambiophonics

A fellow Ambio user shared his experience of using uBACCH. Here are the waveform chart of the XTC compared with the original files. UBACCH – top is XTC and bottom the original file Free Ambio…

ambiophonics.wordpress.com

 

[LIΟN]

Thanks for following my website. It was started as place of reference if I am really making a progress or is it all in my head .

Basically, he is talking about the recursive nature of cancellation. Ralph’s think the ideal cancellation is 60 db recursive attenuation. For an example, your left speakers sound will reach the right ears about 70 μs delayed for speakers placed at 20 degrees. This can be calculated. The right speakers should send the signal coincide with arrival of left speakers sound waves so that both will cancel out at the right ear. However, since the right speakers cancellation sound will also be heard by the left ear, another signal is send by the right speakers so that the right ear and this will be delayed by 140 μs. And it continues and hence the term recursive was used.

if you put the speakers at 60 degrees then the average delay will be 220 μs and the first cancellation signal of the left speaker will start at 440 μs.

Hope this paper helps.

http://www.filmaker.com/papers/RGRM-RACE_rev.pdf

Thank you! I've already read the Race PDF as well but I think there was a bit of confusion in understanding. Thank you for correcting that.
I think one more thing that I was worried about implementing this filter has been solved.!

 

[LIΟN]

BTW guys, this is OT but here is the frequency response of different XTC. I wish I could also get and actual audio samples. Wish you too could shares yours. I would add them to the blog.

Thanks for share. If it's recorded in stereo for each channel (L,R), I think we can explore it a little bit more.
I couldn't check the capture below. I checked it now!
I only have experience with uvacch and now I'm trying to make a filter like ubacch style XTC myself. The cable hasn't arrived yet, so I can't messure it. Also I'm going to combine speakers and headphones(Using Impulcifer) at the same time, so I'm trying to approach it by changing my mind a little bit from the concept of regular xtc.

 

[Tim Link]

I did some work tonight with REW, calculating the predicted combined response at each ear based on the differences in timing of each signal from the 3 speaker array with about 1 foot spacing and listing about 7 feet back to a side panned signal. I think this clearly shows why it works so well. Below 1000 Hz it's mostly phase that gives us directional info. At 1500Hz it's a mix. By 2000 it's mostly level difference. You can see that the array starts to create a level difference around 1000 Hz that gets stronger for a while until it looses power, and eventually at higher frequencies actually goes backwards. This has prompted me to roll off the extreme highs on the side channels. I thought I was hearing that! My attempts at measuring were showing that reversal too.

edit update: rolling off the highs might not be the best approach to the problem up above 10k. Better is to turn off the matrix above 10k and just run regular stereo in the left and right speakers of the array. It sounds natural without sounding dull, and while the stereo separation is pretty weak, at least it's not reversed. I don't think our stereo perception is great up there anyway.

In any case, I think this calculated response difference of between each ear for a side panned signal provides a solid explanation why the 3 speaker array works to create a wide stereo soundstage. This doesn't show the phase difference that's happening below 1000Hz, but it is happening.

 

[Tim Link]

On DIYaudio forum someone reported trying this three speaker array, complaining that it's main problem was that it didn't work at all for people off axis. That's a good point, and it inspired me to try something I've been thinking about - adding side ambient channels, just a pair of speakers playing left and right, way out wide, delayed and attenuated to give that kicker effect. Tonight I got a couple of the Sony SSCS-5s hooked up to try it out. I delayed them about 20ms compared to the center array, and put them about 6 dB down from the center. The effect when seated in the middle is very nice. It's a richer, bolder sound and I definitely prefer it. So that's a big win. I crossed the Sonys over at 600Hz so it's just mid and treble ambience. Down lower didn't sound as good.

So it sounds great in the sweet spot, but what does it do off axis? Well, it dramatically improves the off axis listening experience. There's a real soundstage now, and the center stayes locked in place. So this is a very good thing to do. I've been contemplating that the center array doesn't hit the pinnae the right way for side panned sounds, so there needs to be something really out there. A pair of delayed and attenuated speakers seems to do a great job of making this a better sounding and more family friendly system.

 

[tjcinnamon]

On DIYaudio forum someone reported trying this three speaker array, complaining that it's main problem was that it didn't work at all for people off axis. That's a good point, and it inspired me to try something I've been thinking about - adding side ambient channels, just a pair of speakers playing left and right, way out wide, delayed and attenuated to give that kicker effect. Tonight I got a couple of the Sony SSCS-5s hooked up to try it out. I delayed them about 20ms compared to the center array, and put them about 6 dB down from the center. The effect when seated in the middle is very nice. It's a richer, bolder sound and I definitely prefer it. So that's a big win. I crossed the Sonys over at 600Hz so it's just mid and treble ambience. Down lower didn't sound as good.

So it sounds great in the sweet spot, but what does it do off axis? Well, it dramatically improves the off axis listening experience. There's a real soundstage now, and the center stayes locked in place. So this is a very good thing to do. I've been contemplating that the center array doesn't hit the pinnae the right way for side panned sounds, so there needs to be something really out there. A pair of delayed and attenuated speakers seems to do a great job of making this a better sounding and more family friendly system.

Is this something you could draw in Microsoft Paint with delays, crossovers, and volume? I’m struggling to follow the evolutions.

You are doing a foot apart with the DIY horns? How are they wired and is it a 2 channel amp doing all of the work? Is the matrix still the same?

 

[Tim Link]

Is this something you could draw in Microsoft Paint with delays, crossovers, and volume? I’m struggling to follow the evolutions.

You are doing a foot apart with the DIY horns? How are they wired and is it a 2 channel amp doing all of the work? Is the matrix still the same?

Yes, still 1 foot apart with the DIY horns. I'm using two Denon receivers in tandem now since I ran out of channels. The DIY horn array is using active crossovers. There is a separate amp for each tweeter, but the side midwoofers are sharing an amp channel with the speakers wired in parallel, out of phase with each other. No reason not to do that unless your amp can't handle the lowered impedance.
All channel mixing, EQ, crossovers, and time delay is being done on my Mac Mini. I'm using HDMI out to a 7 channel Denon Receiver to power everything except the two ambient channels. A USB to Toslink converter is sending signal out to the 2nd Denon receiver to run the two delayed stereo channels. The time delay and attenuation on the side channels will likely be room dependent, adjusted by ear to preference.

I just measured with my laser pointer that I'm currently sitting with ears about 84 inches back from the center speaker in the array, and about 118 inches from the side stereo channel tweeters, which are sitting on the bass cabinets in the corner of the room.

 

[tjcinnamon]

Yes, still 1 foot apart with the DIY horns. I'm using two Denon receivers in tandem now since I ran out of channels. The DIY horn array is using active crossovers. There is a separate amp for each tweeter, but the side midwoofers are sharing an amp channel with the speakers wired in parallel, out of phase with each other. No reason not to do that unless your amp can't handle the lowered impedance.
All channel mixing, EQ, crossovers, and time delay is being done on my Mac Mini. I'm using HDMI out to a 7 channel Denon Receiver to power everything except the two ambient channels. A USB to Toslink converter is sending signal out to the 2nd Denon receiver to run the two delayed stereo channels. The time delay and attenuation on the side channels will likely be room dependent, adjusted by ear to preference.

I just measured with my laser pointer that I'm currently sitting with ears about 84 inches back from the center speaker in the array, and about 118 inches from the side stereo channel tweeters, which are sitting on the bass cabinets in the corner of the room.
View attachment 324327

Thank you!!!

For the L+R, when digitally mixing LR does it increase the volume?

For the L-R does the inverse right signal heavily attenuate the in phase left unless it’s hard panned?

 

[Tim Link]

Thank you!!!

For the L+R, when digitally mixing LR does it increase the volume?

For the L-R does the inverse right signal heavily attenuate the in phase left unless it’s hard panned?

For L+R, yes, mixing two identical signals yields a +6 dB. (I think that's right.) I have meters running after the mix to make sure I'm not digitally clipping. I'm using -3.2 dB on each channel before mixing it just to make sure that it doesn't clip. That seems to be enough for most everything because I'm not seeing or hearing anything on the meters.

For L-R and R-L, yes anything that's in phase and equal in volume is completely attenuated, which means anything panned center doesn't come out of those side speakers at all. The amount of attenuation depends on how equal the left and right signals are. If they're in phase but slightly different volumes you'll get partial attenuation. One interesting situation is when the left and right channels in the recording are identical but out of phase with each other. In that case the center channel will be silent and the side channels will end up playing just left in the left speaker and right in the right speaker.

I just did a test with a minus zero dB slow sweep, and it appears that my channel mixing software is automatically attenuating to prevent clipping. That's a little worriesome as it might be using some compression to do that. I turned off all the pre-attenuation and still no clipping.

Answering your question I just learned something - I'm not actually seeing the mixed result because the signal going into the output device block is not mixed yet. I'm sending both left and right signals to the same channel on the amp so I can't actually see the result of the mix. So my meters should be reading a max of -6. So I do need -6 dB attenuation to ensure there's no compression or clipping.

 

[tjcinnamon]

For L+R, yes, mixing two identical signals yields a +6 dB. (I think that's right.) I have meters running after the mix to make sure I'm not digitally clipping. I'm using -3.2 dB on each channel before mixing it just to make sure that it doesn't clip. That seems to be enough for most everything because I'm not seeing or hearing anything on the meters.

For L-R and R-L, yes anything that's in phase and equal in volume is completely attenuated, which means anything panned center doesn't come out of those side speakers at all. The amount of attenuation depends on how equal the left and right signals are. If they're in phase but slightly different volumes you'll get partial attenuation. One interesting situation is when the left and right channels in the recording are identical but out of phase with each other. In that case the center channel will be silent and the side channels will end up playing just left in the left speaker and right in the right speaker.

I just did a test with a minus zero dB slow sweep, and it appears that my channel mixing software is automatically attenuating to prevent clipping. That's a little worriesome as it might be using some compression to do that. I turned off all the pre-attenuation and still no clipping.

Answering your question I just learned something - I'm not actually seeing the mixed result because the signal going into the output device block is not mixed yet. I'm sending both left and right signals to the same channel on the amp so I can't actually see the result of the mix. So my meters should be reading a max of -6. So I do need -6 dB attenuation to ensure there's no compression or clipping.

That’s great and thank you for doing the research!!

I’m trying to figure out how to do it passively using a MiniDSP or trying to invert and mix signals. I could pull it off with 3 minidsp’s but for LCR, I’d only use a Flex, so now we’re at $1500.

I want it to work outside of a bespoke music setup but for all content like HDMI (movies, games, etc…)

I wish MiniDSP would have a 4 analog input 8 analog output Flex option

 

[Tim Link]

That’s great and thank you for doing the research!!

I’m trying to figure out how to do it passively using a MiniDSP or trying to invert and mix signals. I could pull it off with 3 minidsp’s but for LCR, I’d only use a Flex, so now we’re at $1500.

I want it to work outside of a bespoke music setup but for all content like HDMI (movies, games, etc…)

I wish MiniDSP would have a 4 analog input 8 analog output Flex option

A miniDSP should have a mixer on-board, allowing you to change L and R to L+R and L-R. All you need are those too channels coming out of the miniDSP to setup the thee speaker array. I find using a computer to be the easiest way to do it. With HDMI output into a reciever I get up to 8 channels if the reciever has that many amps, and that provides coordinated volume control too. For a higher quality implementation I've been considering getting a good USB 8 channel DAC/preamp and some good amps. But that adds up quick. I got a used Denon reciever for $150 bucks.

 

[tjcinnamon]

A miniDSP should have a mixer on-board, allowing you to change L and R to L+R and L-R. All you need are those too channels coming out of the miniDSP to setup the thee speaker array. I find using a computer to be the easiest way to do it. With HDMI output into a reciever I get up to 8 channels if the reciever has that many amps, and that provides coordinated volume control too. For a higher quality implementation I've been considering getting a good USB 8 channel DAC/preamp and some good amps. But that adds up quick. I got a used Denon reciever for $150 bucks.

The bummer for the minidsp is that the mixing matrix happens before the phase inversion. Meaning I can get L+R quite easily but the L-R or R-L requires more units at least 2.

I’m going to see if there are analog mixers I could use. Once I get L+R together.

https://docs.minidsp.com/product-manuals/flex/images/signal-flow/flex/flex-signal-flow.svg

 

[Tim Link]

The bummer for the minidsp is that the mixing matrix happens before the phase inversion. Meaning I can get L+R quite easily but the L-R or R-L requires more units at least 2.

I’m going to see if there are analog mixers I could use. Once I get L+R together.

https://docs.minidsp.com/product-manuals/flex/images/signal-flow/flex/flex-signal-flow.svg

It makes sense that the miniDSP folks wouldn't be expecting people to be mixing out out of phase signals.

 

[tjcinnamon]

It makes sense that the miniDSP folks wouldn't be expecting people to be mixing out out of phase signals.

I think if I can find an audiophile quality mixer, I could just plug it into that and do my mixing there. Then I could get away with one miniDSP.

 

[Tim Link]

I think if I can find an audiophile quality mixer, I could just plug it into that and do my mixing there. Then I could get away with one miniDSP.

I've been thinking about making something like that. I don't think it would be too complicated to make an analog mixer with very high fidelity using op amps. But I've zero experience with making such things.

 

[Tim Link]

I think if I can find an audiophile quality mixer, I could just plug it into that and do my mixing there. Then I could get away with one miniDSP.

I've been looking in to this and it's a bigger problem than I thought. A channel mixing device with sum and differencing for a hifi system isn't something readily available. Maybe this is something a little Raspberry pi kit could do?

 

[tjcinnamon]

I've been looking in to this and it's a bigger problem than I thought. A channel mixing device with sum and differencing for a hifi system isn't something readily available. Maybe this is something a little Raspberry pi kit could do?

Haha! I was literally just reading about freeDSP and HifiBerry. I just moved on to Camilla DSP. I’ll look and see if there are boards. For home theater application (I think) it requires analog inputs and outputs.

 

[tjcinnamon]

I've been looking in to this and it's a bigger problem than I thought. A channel mixing device with sum and differencing for a hifi system isn't something readily available. Maybe this is something a little Raspberry pi kit could do?

This could be an All-In-One unit https://motu.com/products/avb/624

The Motu Ultralite is very well reviewed on ASR. Difference is that this has the DSP built in. It’s all in one box. Otherwise, with an external DSP an audio interface (like a non DSP interface) could work.

 

[tjcinnamon]

This could be an All-In-One unit https://motu.com/products/avb/624

The Motu Ultralite is very well reviewed on ASR. Difference is that this has the DSP built in. It’s all in one box. Otherwise, with an external DSP an audio interface (like a non DSP interface) could work.

I just checked the 624 and it can mix channels and has some methods to invert the phase on some inputs. Alternatively, as a worst case, a modified XLR coupler would invert that phase. and could be done with Y cables on the input.

That 624 could be an all in one. 4 inputs (L, R, -L, -R), 8 outputs with effects and matrixing.