# Five Channel Soundstage



## Patrick Bateman (Sep 11, 2006)

A few years back I made a serious assault on creating the ultimate soundstage here : http://www.diymobileaudio.com/forum...ussion/60146-creating-perfect-soundstage.html

That system didn't last too long. It sounded like a big pair of headphones and it was ugly.

I've had a couple of very nice home systems, and I've noticed that most of what I listen to is basically mono. There's a TINY bit of ambiance and width, but it's mostly mono.

I've had a multichannel system in my Genesis for a few years now, and I'm fairly happy with it. So I thought I'd explore the possibility of doing a five channel in my other car.

Here's what I think are the advantages and disadvantages:

advantages:
1) Solid center image, and most of what I listen to is basically mono anyways
2) Might be more intelligible. (In a stereo system, intelligibility suffers because you're listening to two speakers.) Intelligibility is important to me because I listen to a lot of podcasts, in addition to music.
3) Potentially better dynamics, because I have a lot of room for a center channel in my car.

disadvantages:
1) Hideously ****ing complex. We're talking about something on the order of ten channels of amps, a PC, software, etc


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## Lycancatt (Apr 11, 2010)

I've always liked 5 channel stereo on the Yamaha receivers I've had, it wasn't perfect or really tunable, but it sounded good, and it did it from multiple locations, not so important from the drivers seat, but nice in a bg vehicle, or a party situation at home


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## Patrick Bateman (Sep 11, 2006)

To turn two channels into six channels, I'm using Dolby ProLogic II. I wanted to figure out what PLII is doing to the signal, so I measured it. Here's what I found:

1) When playing a sound through the left channel, the sound goes to the left channel, *but no other channels.*
2) Ditto for the right.
3) When playing a sound through both channels, some interesting things happen. First, the center plays at 100%. *But the left and the right each play at -4dB.* IE, if you're playing something in mono, you're not just getting the center channel.

The third point took a while for me to figure out. Here's what I think is going on:
First, if the center channel played at 100%, and the left and right played nothing, *the volume level would drop by 6dB.* This isn't immediately obvious, but the reason that this happens is because you're going from two speakers to one speaker. IE, if you had a mono signal playing on a pair of stereo speakers at 100dB each, you'd end up with 106dB. *But when you switch to PLII, you only have one speaker in the center.* So you have to bump up the output of the center by six dB, OR you have to feed some signal to the left and to the right. Clearly, Dolby went with the latter; when you feed Dolby PLII a mono signal, you're getting output from all three speakers.

If that last paragraph made sense, you're probably wondering why the left and the right speakers are attenuated by four dB, instead of six dB. I think this was likely a subjective judgement by Dolby. In particular, the left and the right aren't going to sum perfectly with each other, due to comb filtering. So raising the level of the left and the right to -4dB instead of -6dB is likely to combat some of the attenuation that's caused by comb filtering.

If they really wanted to go crazy with this, you'd boost the high frequencies more than the low frequencies. The reason for this is because comb filtering will have a greater impact at high frequencies than low; there's going to be frequency where all three speakers in the front are close enough that they're working together, instead of comb filtering each other.


Another reason to feed the mono signal to the center speaker AND the left and right speakers is to create some 'fake width' with mono recordings. This is something that bugged me with Opsodis, a lot of pop recordings sound too narrow because they're basically mono. So putting some of that signal into the left and the right will stretch the stage width. This is admittedly artificial. Then again, this is a bigger problem with a two channel set up than a five channel.









Here's a pic showing the behavior in Audacity. In the top track, I'm playing the left channel, and measuring both channels. We see output only on the left. In the track below it, I'm playing mono, and recording both channels. We see output on both the left and the right, but attenuated by 4dB. Not shown here is the center channel, which was showing 100% output. (0dB)

To come up with this data, I created test tracks in Audacity. Then I played them back and recorded them on the same computer. (line out to line in.)


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## Bluenote (Aug 29, 2008)

Patrick, my question is regarding PLII and music formats. Like the H800 for instance...will regular two channel stereo playback in 5.1 through PLII? I know L7 does an Upmixing from stereo to multi-channel playback; but PLII seems to require discrete 5 channel recordings and those formats are really expensive.

Btw, thanks for starting this thread...sub'd!


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## Patrick Bateman (Sep 11, 2006)

Test tracks are great, but I wanted to run some actual music through the decoder to see what it would do.

At first, it looked like the decoder was sending a SIGNIFICANT amount of sound into the center speaker. *Even more than a mono test track.* But after studying the graphs for a few minutes, it hit me: Dolby is applying a very subtle low pass filter to the left and to the right.

This is way cool. In the last post, I alluded to the idea that a filter would be useful to combat comb filtering. Looks like the folks at Dolby are one step ahead of me - it's built into the decoder.

In a nutshell, the decoder is doing the following:
1) The center is getting most of the output. Particularly at low frequency. At 100hz, the center channel is getting TEN TIMES as much output as the left speaker!! So don't skimp on the center, it's doing the heavy lifting. On a side note, this probably explains why home theater receivers give you the option of selecting "small" or "large" speakers. If you send a full range signal to your center, it's really going to get punished.
2) The left and the right are getting about 4dB less output than the center. At low frequency, the difference gets more and more pronounced; by 100hz it's 10dB.
3) The subwoofer channel is lowpassed with a second order filter. One odd thing about the sub channel is that it's output is much lower than the other channels. I'm guessing this is to preserve the signal to noise ratio of the main channels, where a low S/N would be much more audible.

Here's some measurements. To get this data, I ripped one of the typical pop songs I listen to, then I directed the OUTPUT of the soundcard into it's own inputs, and recorded the result and compared it to the original file. The song in question is "My Dark Twisted Fantasy" by Kanye West. I used a WAV as the source. I took a six second snippet of the track, where the bass was going.

















Left and right channel. Basically identical.









The center channel. Note that the shape is similar to the left and to the right, but begins to get louder and louder at low frequency. (Again, this is probably to combat comb filtering from the three channels.)









Here's the output of the sub. We only have about an octave of sound here. This is one of the reasons that low distortion subs are a big deal. We don't need our sub generating a bunch of third harmonic distortion at 150hz and dirtying up the mains.



In the spectrum analysis you can also see that it's incredibly important to have a lot of displacement in the two octaves between 80hz and 320hz. While the high frequency energy in the song is down by almost 30(!) decibels at 10khz, the output between 80 and 320 is nearly as loud as the sub output. But our center channel is reproducing that, not the sub. And the left and the right speaker aren't doing much in that range. So those two octaves are a real torture test for the center.

Of course, you could also select "small" for the center channel. That would limit how much punishment the center is subjected to.


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## Patrick Bateman (Sep 11, 2006)

Bluenote said:


> Patrick, my question is regarding PLII and music formats. Like the H800 for instance...will regular two channel stereo playback in 5.1 through PLII? I know L7 does an Upmixing from stereo to multi-channel playback; but PLII seems to require discrete 5 channel recordings and those formats are really expensive.
> 
> Btw, thanks for starting this thread...sub'd!


Yep, Prologic II was designed for music by Jim Fosgate (of Rockford Fosgate fame.) It was explicitly designed to turn two channel stereo into six channels.

_"Another one of Harman’s acquisitions was Lexicon, a highly respected name in the pro business as well as in the consumer market. Their processors were very good, but they were stuck with the aging Pro Logic. They were, however, developing a new surround technology, Logic 7, a product of the brilliant David Griesinger. Harman now had the two leading alternatives to replace Dolby Pro Logic, and Harman executives in sales and engineering became divided on which one would emerge, if in fact either would. As time went on, it started to look as if Logic 7 would become Pro Logic II, due to a higher level of support within Harman. Jim and his technology were being viewed as superfluous.

6-axis surroundAssuming that Logic 7 would win, Harman management did something that in retrospect was unwise. Jim Fosgate was terminated. Wisely, he exempted and retained the rights to some key patents he felt would be useful in the future, and he was correct. Charlie Wood offers a first-hand explanation: “Post Harman, Jim continued to work on the technology and evolve what was 6-Axis, employing the patent rights he withheld. Jim and I got Roger Dressler of Dolby Laboratories over to hear what Jim had developed. Roger returned to Dolby and convinced others there, including Ray Dolby, that Jim’s technology was superior and that if Dolby tried to re-invent it to avoid patent infringement that it would be a futile effort. Dolby then signed a generous licensing agreement with Jim and made his 6-Axis system the new Dolby Pro Logic II. That development made him a wealthy man, with over 300 million pieces of gear produced worldwide with his Pro Logic II and IIx technology incorporated within…”

I would have felt schadenfreude towards those at Harman who leveraged so hard for Logic 7, except they are good, talented people, and many are still friends of mine. Logic 7 would have made a good Pro Logic II, but for the decades and decades of tireless work without reward that Jim put in, he definitely earned it._"

History of Surround Sound Processing: The Battle for Dolby Pro Logic II | Audioholics


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## Patrick Bateman (Sep 11, 2006)

If you want to try this on your own, there's a number of ways to do it in your car and all of them kind of suck. I wish there was an easy way to do this.

1) You can stream lossless bluetooth to a BT4 receiver, and then plug the BT receiver into a ProLogic II decoder. I have a Belkin BT4 receiver sitting here ($50), and I have a PLII receiver on the way($20). Total cost is about $70. The main problem with this solution is that you have to make a cable for the PLII decoder, if you use the "Tritton" decoder that's ubiquitous on eBay. I know that using Bluetooth for audio doesn't seem "hifi" but keep in mind that it's APT lossless. So it's basically like running toslink to the trunk.

2) The method I'm using right now, because I'm a control freak, is a car PC. I'm using an Athlon 8150 cpu ($20) a mini-itx mobo ($22) and a Antec case. I have an Antec case that's about the size of a book ($60.) This is probably overkill, but I was impatient and I didn't want to wait for the Tritton decoder or make a cable for it. I'm using PowerDVD to do the PLII decoding ($55.) I tried using Corel's WinDVD but it kept crashing.
AMD is hemorrhaging money like crazy, and I'm guessing that's the reason these CPUs and mobos are so cheap.

3) It occurred to me that you could replace the Carputer with a Windows tablet. So I ordered a Windows tablet off of Groupon for $70. If you go this route, be sure to get one from HP or Dell. There are a bunch of no-name Windows tablets out there, and they look ****ing terrible. The Dell looks really nice, and it's marked down from $400 to $100 at Dell.com right now. It has plenty of horsepower to playback music. The main downside with the Dell and the HP is that you'll have to get a docking station to get all the ports you'll need. (You'll need six audio outputs.)








HP Stream. For sale at Groupon for $70. Definitely not as cool as an iPad, but the Stream should be able to use USB sound cards and any Windows software.


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## stills (Apr 13, 2008)

Hopefully by the weekend I'll have my single tweeter PLII project up and running. 

My main concern is getting enough output from my peerless 2" center.


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## Nismo (Jan 10, 2010)

Damn you Patrick! I've got a Nexus 7 with my 80PRS, and without replacing both, I can't do DPLII...and I've been mulling DPLII the last few days. In addition to that, I will be getting a used Atom-based laptop shortly.

I can't do Torque on Windows, the Windows tablets don't have GPS, and I can't (that I know of) do DPLII from Android.

Well, crap.

Eric


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## Nismo (Jan 10, 2010)

Now that I'm thinking about it, I could use the deck as a preamp, running one set of outputs to the inputs on one of these boxes (if someone can help me understand it) could allow me to get DPLII out, but this prevents any and all DSP.

It also brings into question how the processing/xovers are done on the outputs and the subwoofer. I need to be able to set these, and not being in control of this concerns me. Not even a MiniDSP 8x8 would handle this for me, as I would have 10 channels of amplification (of 11 available), and 8 channels of processing (3 way L/R, 1 ch C/SL/SR/Sub).

How are you planning to do these?

Eric


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## garysummers (Oct 25, 2010)

With regards to the center channel, the Alpine PXA-h990 has a center channel bass management scheme that allows you to direct all center channel signal below a selected frequency to the left and right channels. Alpine obviously assumed that people would not be able to install large drivers in the center channel location. The function works quite well.


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## Patrick Bateman (Sep 11, 2006)

Nismo said:


> Now that I'm thinking about it, I could use the deck as a preamp, running one set of outputs to the inputs on one of these boxes (if someone can help me understand it) could allow me to get DPLII out, but this prevents any and all DSP.
> 
> It also brings into question how the processing/xovers are done on the outputs and the subwoofer. I need to be able to set these, and not being in control of this concerns me. Not even a MiniDSP 8x8 would handle this for me, as I would have 10 channels of amplification (of 11 available), and 8 channels of processing (3 way L/R, 1 ch C/SL/SR/Sub).
> 
> ...


I was sort of hoping the center channel would be ten decibels louder or more.
If that was the case, I could really skimp on the left, right, and surround channels. Was thinking something on the order of a 2" driver.

This is not the case; the center is down by about four decibels.

If that *was* true, I was aiming to do something like a B&C 8NDL51 as a center, and then small drivers everywhere else.









Here's a pic of the 8NDL51s from my Accord. These are shallow enough, they fit UNDER the brake pedal. With that kind of depth, I could slip them into the stock center channel location. With a hundred watts the 8NDL51 will hit something like 115dB.

The setup would look something like this:

1) left, right and surrounds, each running full range, with a dedicate amp channel and passive xovers
2) Center channel running active, with an amp channel for the midrange and tweeter
3) Two subs with an amp channel for each

So eight channels of amplification. I wouldn't even need all four channels of a MiniDSP to do this.

I'm half considering an end-fire array for the center channel.


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## Nismo (Jan 10, 2010)

Patrick Bateman said:


> I was sort of hoping the center channel would be ten decibels louder or more.
> If that was the case, I could really skimp on the left, right, and surround channels. Was thinking something on the order of a 2" driver.
> 
> This is not the case; the center is down by about four decibels.
> ...


Wow,
That's quite a bit different than I expected. I have amp channels to burn, and good xovers built into my amp. I was hoping to do everything via MiniDSP if I were to go this direction (I guess Rube Goldberg must be a relative of mine!).

I don't have tweets yet, but I'm thinking I could get 3 more Faital Neo 3's, and it could work well... the other bit would be trying to get midbass to the quartet of Anarchys. I wonder if MiniDSP could strip the Center channel midbass as Gary mentioned above, and send it to L/R.

Eric


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## Patrick Bateman (Sep 11, 2006)

I need somewhere to stash all the gear for this project.
Usually I work on the front stage first. But since this one is going to require a lot of amps and a computer, I'm starting with the sub first, on the expectation that I'll cram the electronics in there.


































My initial plan was to do an Alpine SWS-10D2 in a front loaded horn. I'd built one already for my last car, a Mazda6, and it worked well. It's still serving me faithfully in my home theater. (Read about the build here : http://www.diymobileaudio.com/forum...0952-sealed-box-versus-infinite-baffle-5.html)

But over the years, I've realized that sealed boxes in cars act a bit like horns. Due to the reflection off of the front of the car, you can get an efficiency 'bump' from the reflection off of the windshield. (The sims above show the SPL and the excursion for a five cube box with dual TC Sounds fifteens, versus a convention front loaded horn with a single Alpine SWS-10D2. If you want to build the latter, it is basically the same as the 'depth charge horn' linked above.)

So I think a couple of fifteens in a plain ol' sealed box will likely outperform a single ten. You can see in the hornresp sims that a horn-loaded ten will crush any sealed twelve out there, but once you start to raise the efficiency by piling on with more cone area, the sealed boxes start to shine. Hoffman's Iron Law still rules the day, and this is a BIG sealed box - five cubic feet. 


Unfortunately, a sealed box doesn't give me anywhere to put my gear. So I think I'll do a bandpass, purely because it gives me a place to put the electronics. (Inside the vented portion of the box.)


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## Patrick Bateman (Sep 11, 2006)

OK, here's the "evolution" of the sealed box from the last post.

If it's not clear what's going on here, this is what I am doing:

1) This is a single reflex bandpass box with dual TC Sounds fifteens. Yeah this is going to be a monster  
2) The woofers are push-pull. Because, why not? Reduces harmonic distortion and it's free.
3) The port probably looks way too skinny. It's not. I'm doing a radial port. You don't see these too often. The only company I'm aware of that uses them is Polk. They call them a "power port." Basically a radial port is a conventional port, but it radiates in a ring, instead of in a straight line. (Well, it's a straight line, but a straight line that goes in all directions.) The cool thing about a radial port/power port is that you get a REDONKULOUS mouth area. For instance, my box is 38" wide. So here's how much area we get with a radial port that's 38" in diameter:
circumference = 2 * pi * radius * height
= 2 * 3.14159 * 18" * 1.5"
= 169"^2

So a "power port" with a radius of eighteen inches and a height of 1.5" winds up having a mouth area that's 169 square inches.








To put that in perspective, those monster six inch ports that the SPL dudes use are 18.84"!! So my port is NINE TIMES bigger than an SPL port. 

















Here's some pics of Polk's version. They changed it a few times. Most likely for patent reasons.

Yeah, this is some crazy port, but I've found that small ports REALLY hurt your SPL. (As the SPL guys know also.) Small ports also generate a lot of distortion.


The computer in my sub box is an Antec VESA 110


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## Patrick Bateman (Sep 11, 2006)

My Dolby processor arrived today. Cost a whopping twenty bucks.









I'd hoped to get one of the units with RCA outs, but they were all sold out. (I'm guessing it was my fault for talking about it before buying it.) But it looks like the connector is a plain ol' PS2 connector. There are six pins and a ground. I'm hoping they correspond to this:

1) left
2) center
3) right
4) left rear
5) right rear
6) subwoofer

I'll probably waste a couple of hours finding out, but I'll post the results. If it all works out, then we'll have a readily available Dolby Prologic II processor for under $40


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## Nismo (Jan 10, 2010)

Patrick Bateman said:


> My Dolby processor arrived today. Cost a whopping twenty bucks.
> 
> 
> 
> ...


Patrick,
What unit is this again, and where did you get it? I looked back and was unable to find the detail.

Thanks!


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## quietfly (Mar 23, 2011)

Nismo said:


> Patrick,
> What unit is this again, and where did you get it? I looked back and was unable to find the detail.
> 
> Thanks!


"Tritton" decoder | eBay


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## Patrick Bateman (Sep 11, 2006)

I have bad news - the Tritton decoders that are ubiquitous on eBay won't work for five channel it looks like.









I believe the problem is the stupid inline controller that leads to the headphone. The output of the decoder is six pins, and I had high hopes that those six pins would be the following:
1) left channel
2) right channel
3) center
4) sub
5) back left
6) back right

Or something like that.

Unfortunately, it appears that four of the six pins do something mysterious. (I dug around in their docs and couldn't figure out the other four pins.)

Another 'fly in the ointment' is that the Belkin documentation is without a doubt the ****tiest I've ever seen. It literally tells you NOTHING. In particular, it doesn't tell you how to set the receiver to "lossless." Then again, perhaps that is set in the transmitter? (Our phones.)

I seriously came real close to throwing in the towel on this whole project, it's been frustrating.

There ARE a couple of pieces of good news:

1) The combination of the Belkin receiver and the Tritton decoder WILL give you a way to do optical in your car without a head unit. I was *definitely* getting audio on the first two pins, and I'm fairly certain they correspond to left channel and right channel. So this combo could be bad ass for people who have noise problems in their car. My phone, the decoder, and the receiver all run on five volts, so it's incredibly easy to get this going in a car.

2) I have not given up on Prologic comompletely. I am now trying to get it to work on an HP Stream tablet. The Stream runs a full version of Windows eight, and it's a whopping $70 at Groupon. I bought a $5 USB adapter and the Stream synced up perfectly with a $40 USB sound card. So you're looking at a $120 solution that spits out five channel sound and runs all Windows applications out there. (iTunes, VLC, MPC, etc.)

I'm trying to get it to run WinDVD or PowerDVD to do the PLII decoding...


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## TheAlchemist9 (Apr 29, 2015)

I personally think the PLii upmix from my H800 sounds terrible. Alpine's Euphony mix actually sounds a lot better to me. Or plain stereo. I'm not a fan of PLii for home audio either, though... I think Logic 7 (or Neo 6) sound much better.

I'll be following this and hope that PLii or another 5-channel mix works better for you than me.


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## Nismo (Jan 10, 2010)

Has anyone figured out how to get a decoder set to 'Music' instead of 'Movie'? Also, how do you set gains on amps, if each output is different, based on incoming signal?

I found that Pioneer offers a pair of old units to do DPLII, but they're rare like hen's teeth. That unit allows digital or analog, and you can change the input, as well as control volume, if I understand properly.

Eric


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## cajunner (Apr 13, 2007)

Patrick Bateman said:


> I have bad news - the Tritton decoders that are ubiquitous on eBay won't work for five channel it looks like.
> 
> 
> 
> ...


for 50 bucks you can buy a HTIB and add another 25 for the "400 watt inverter" you'll need?

or 40 bucks gets you a real Home Theater Receiver, with good power at a Pawn shop?

then another 40 bucks for a decent power inverter?

the box I got, was the Tritton gray unit which comes with three 3.5mm jacks which I assume, are TRRS each, for the sub/center, left/right front, and left/right surround?

I don't have any literature, just the power plug and the box.
Tritton AX Pro Decoder Box Only Dolby 5 1 Surround Sound | eBay


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## Patrick Bateman (Sep 11, 2006)

That might work. Just bought the one you linked.

Where are the inputs?

Looks like it might be on the top? And if so, are they analog, digital, or both?


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## cajunner (Apr 13, 2007)

toslink and USB inputs.

maybe get an AptX bluetooth receiver that outputs optical, tape it to this thing, and there you go?

under the odd USB port it says "MIC" so it might be for gaming stuff and not a stereo input.


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## Patrick Bateman (Sep 11, 2006)

It took way, way, way, WAY too long to get this working.

But I got it working.










Here's how I did it:

1) I am using an HP Stream 7. It is $70 from Groupon and it's basically a miniature Windows laptop that fits in your hand. It's hard to appreciate how small it is until you use it; it isn't much bigger than my phone. (Your narrator may be biased as he may or may not work for HP.) https://www.groupon.com/deals/gg-hp-stream-7-32gb-7-windows-tablet
2) To get five channels of sound, I am using a cheap USB sound card. Amazon.com: Diamond External Xtreme Sound 7.1-Channel USB Audio Device: Electronics (I only paid $40 for this at Frys. I'm 90% sure that any ol' USB sound card with six channels of output will work fine, as long as there's a Windows 8 driver. You could probably find one on eBay for ten bucks.)
3) Here was the part that was REALLY tricky. The sound card wasn't fully functional until I plugged into a powered USB hub. I think what's happening here is that the USB jack on the tablet isn't supplying enough juice to the sound card. If you don't have a powered hub around, you might consider a universal USB docking station. (It MUST be powered.)

OK, that was a p.i.t.a., but it wasn't too expensive. About $120 for everything.









I know a lot of people have put PCs in their car, and I really think the Windows tablet is a better solution. In particular, I found that the monitors for car use look ****ty. Even the $200 units look clunky and washed out. The HP Stream has a 1280x800 IPS panel. You can't get a panel like that for under $100. So you're basically getting a very nice IPS panel for $70, with a free Windows license AND a computer for free! It's quite a deal.

Getting the software to work really sucked. Here's what I wound up doing:
1) download foobar2000 : Download foobar2000 and optional components
2) download the "free surround" plugin : foo_dsp_fsurround - Hydrogenaudio Forums
3) Install the plugin like this : Foobar2000:How to install a component - Hydrogenaudio Knowledgebase
4) Then configure the plugin for six channel output


I'm a little cranky right now, because it took about twelve hours of trial and error to get this to work. If you follow these instructions, you should be able to get it working in under 30 minutes.

In summary, here's a combination of free software and cheap hardware that will give you the following:
1) six channels of sound
2) a slick 7" IPS display
3) 32 gigs of storage

All for about $120

btw, those pics above aren't mine, stole them from the interweb


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## Patrick Bateman (Sep 11, 2006)

Here's some **** I tried that *didn't* work. You may have more luck with this:

1) PowerDVD offers a PLII decoder. On the Stream 7, the installer hung. I think it may be due to the lack of memory on the tablet.
2) WinDVD offers a PLII decoder. On the Stream 7, the decoder options were greyed out. I have no idea why.
3) Dolby Prologic II decoding requires a license. There are a couple of "free" programs which supposedly decode PLII. The first is called "AC3filter". The second is called "foo_dsp_pl2." The first one is available for download, but it didn't work for me. Though configured, only two channels of audio were being produced. The second is completely impossible to find online. As PLII requires a license, I'm guessing the author got a cease and desist and took it down. (The first was written by a Russian author, so perhaps he ignored Dolby's lawyers? I can only speculate.)

The reason that I bring all of this up is because the free decoder I'm using isn't a PLII decoder. But it IS written by the same author who wrote the PLII wrapper, so I'm guessing he probably knows a thing or two about PLII decoding.


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## Patrick Bateman (Sep 11, 2006)

EcoHawk said:


> I personally think the PLii upmix from my H800 sounds terrible. Alpine's Euphony mix actually sounds a lot better to me. Or plain stereo. I'm not a fan of PLii for home audio either, though... I think Logic 7 (or Neo 6) sound much better.
> 
> I'll be following this and hope that PLii or another 5-channel mix works better for you than me.


Do you have a center channel?

Check out the graphs I posted earlier in the thread, and note that there's quite a bit of bass in the center channel.

I'm actually considering using a sub for the center channel, along with the mids and the tweets. From what I can see, the center channel in a ProLogic II setup is the trickiest part; the other five channels play a role, but the center channel does most of the heavy lifting.

Another fundamental problem with a five channel setup is that width will probably suffer. I notice this in my Genesis with the Lexicon processor. It's a tradeoff between the width of stereo or Opsodis, versus the solid center of a five channel setup. Heck, if this experiment is a dud I'll probably just do Opsodis.


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## cajunner (Apr 13, 2007)

how about this, and the Tritton decoder you bought?

Adding Optical (Toslink) output to Alpine Cd Player

wouldn't that be a cool way to run surround and get optical to the trunk, in one shot?


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## Nismo (Jan 10, 2010)

If there was a simple plugin to take stereo audio into Windows, and put out DPLII, I would definitely be down for that. I have a laptop that would just need a power supply and the USB sound card!

Eric


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## Patrick Bateman (Sep 11, 2006)

Nismo said:


> If there was a simple plugin to take stereo audio into Windows, and put out DPLII, I would definitely be down for that. I have a laptop that would just need a power supply and the USB sound card!
> 
> Eric


It looks like the instructions I posted yesterday should do that. I paid about $120 but you could get it down to about $100 if you bought the soundcard off Amazon.

True, it's not DPLII, but considering the author wrote a DPLII wrapper, I think it'll probably be close. His plugin includes a lot of options that the 'real' decoder does not.

If that doesn't work, the Belkin bluetooth adapter ($50) and the older Tritton decoder should work. ($10). Not the one I posted, the one that cajunner posted. I'll know for sure next week.


----------



## TheAlchemist9 (Apr 29, 2015)

Patrick Bateman said:


> Do you have a center channel?
> 
> Check out the graphs I posted earlier in the thread, and note that there's quite a bit of bass in the center channel.
> 
> ...


Yeah, I've got Xtant 4" point-source drivers for my 5 surround channels, fourth-order high-passed at 315hz. Peerless SLS midbass and Arc Black 12 in a 2.0ft^3 @ 30hz box.

If I play with the center width and rear mix settings, I might be able to get something that I'm happy with... we'll see. If I do center bass split (center channel bass sent to L and R), I get WAY too much bass. Probably a 10+db boost over stereo. Even without center bass split, I'm getting extra bass (that I don't want) in PLii mode.


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## oabeieo (Feb 22, 2015)

Patrick Bateman said:


> A few years back I made a serious assault on creating the ultimate soundstage here : http://www.diymobileaudio.com/forum...ussion/60146-creating-perfect-soundstage.html
> 
> That system didn't last too long. It sounded like a big pair of headphones and it was ugly.
> 
> ...



Pat , I have been glued to the thread you posted . I have minihorns on my dash now , I love what you have done. Having the 6.5 midbass right next to the dash horn really makes the sound amazing IMO . The wavefronts from both are in perfect alignment and phase and travel in same direction, the impact is astonishing. Dood I tryed making a home made horn and I failed my first attempt . I did something wrong . The ES mini does so much better . I cut a Pyle wave guide and molded it around my dash and the ES mini does way better. I had to have scored something up. In parts of it I cut the flare down to about 2" long . Do you have any learning material to teach me the do's and don't about making a custom horn. . I like using these mini because pld is about 4inches simply because I can move the horn where ever I want on passenger side . Do you have any of those horns you made laying around? I would be interested in maybe getting a set from you to take a listen . I can't figure out how to use hornresp and I know I want a solid center and being the horn is on axis I don't know if the ES mini is optimized for on axis listening , there's questions I don't have answers to . Can I fill the ES mini with reticulate foam with it on axis and have it come out like ice cream on a cone like you pic shows , when I had two sets horns installed I had reticule foam strips in mouth but never filled up like that . Should I try that? I'm happy with the way it sounds . I do have some hom. The ES mini also shoots sound so intensity over to other side, do the driver side looses 8k and up and the passenger side has too much 8k and up , I can eq and change the angle of horn and tilt them in and get it to sound very good , but it would be nice if both sides had even sound ,because pld is so small to begin with .


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## Patrick Bateman (Sep 11, 2006)

EcoHawk said:


> Yeah, I've got Xtant 4" point-source drivers for my 5 surround channels, fourth-order high-passed at 315hz. Peerless SLS midbass and Arc Black 12 in a 2.0ft^3 @ 30hz box.
> 
> If I play with the center width and rear mix settings, I might be able to get something that I'm happy with... we'll see. If I do center bass split (center channel bass sent to L and R), I get WAY too much bass. Probably a 10+db boost over stereo. Even without center bass split, I'm getting extra bass (that I don't want) in PLii mode.


Do you have five midbasses, four midbasses, or two midbasses?

If you have two or four, the reason that the bass is increasing is because moving the bass from the center (where there's no midbass) to the sides (where there is.)

Earlier in the thread I did some spectrum analysis of the output of a PLII decoded signal, and an extra 10dB of bass sounds about right.


----------



## thehatedguy (May 4, 2007)

Eric's horns are meant to be off axis, no toe in or out, and not aiming them at your ears/head.

You can put foam in there, it only really has to go to the edge of the mouth of the horn. Last time I made foam inserts, I went from the reflector to the edge of the mouth.


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## Patrick Bateman (Sep 11, 2006)

oabeieo said:


> Pat , I have been glued to the thread you posted . I have minihorns on my dash now , I love what you have done. Having the 6.5 midbass right next to the dash horn really makes the sound amazing IMO . The wavefronts from both are in perfect alignment and phase and travel in same direction, the impact is astonishing. Dood I tryed making a home made horn and I failed my first attempt . I did something wrong . The ES mini does so much better . I cut a Pyle wave guide and molded it around my dash and the ES mini does way better. I had to have scored something up. In parts of it I cut the flare down to about 2" long . Do you have any learning material to teach me the do's and don't about making a custom horn. . I like using these mini because pld is about 4inches simply because I can move the horn where ever I want on passenger side . Do you have any of those horns you made laying around? I would be interested in maybe getting a set from you to take a listen . I can't figure out how to use hornresp and I know I want a solid center and being the horn is on axis I don't know if the ES mini is optimized for on axis listening , there's questions I don't have answers to . Can I fill the ES mini with reticulate foam with it on axis and have it come out like ice cream on a cone like you pic shows , when I had two sets horns installed I had reticule foam strips in mouth but never filled up like that . Should I try that? I'm happy with the way it sounds . I do have some hom. The ES mini also shoots sound so intensity over to other side, do the driver side looses 8k and up and the passenger side has too much 8k and up , I can eq and change the angle of horn and tilt them in and get it to sound very good , but it would be nice if both sides had even sound ,because pld is so small to begin with .


Eek, I totally neglected to mention that the Pyle is a screw-on.
I didn't mean for people to hack up a waveguide to make it fit! (Or maybe I did say that?)

That's fairly simple to do, but you need the right parts, and the right parts aren't obvious. Way simpler by far is to simply use a screw on compression driver. JBL 2408H-2 is a good one. There are a bunch of screw-on compression drivers from JBL for sale on eBay; it appears that screw-on is their favorite mounting method lately. Some of them are under $35, but I can only vouch for the 2408H.

As far as listening angle goes, waveguides are fairly tolerant of having you anywhere in the beam. IE, if you have a beam of 90x40, as long as you're in the beam you're fine. One "easy" way to calculate this is simple: if you can see the throat of the waveguide, you're in the beam.

What's kind of interesting is that response REALLY goes to **** outside of the beam. I think this is due to diffraction off the lip of the waveguide. But the differencie isn't subtle. Obviously, big waveguides don't suffer from this as much. In all of my underdash HLCDs that I made, if found that they were really finicky about the vertical aiming.


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## Patrick Bateman (Sep 11, 2006)

If you *do* intend to hack up one of the Pyle clones of the JBL waveguides, it's a lot simpler to use a dome. I did that with a Vifa XT25 and it worked fine. It was interesting how different a dome sounds on a waveguide compared to a compression driver, even though the response was the same. Dare I say it, the dome might just work better. I'm really dying to put my SB Acoustics tweeters on a waveguide. (Haven't had time - I'm literally typing this at an airport.)


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## oabeieo (Feb 22, 2015)

Patrick Bateman said:


> Eek, I totally neglected to mention that the Pyle is a screw-on.
> I didn't mean for people to hack up a waveguide to make it fit! (Or maybe I did say that?)
> 
> That's fairly simple to do, but you need the right parts, and the right parts aren't obvious. Way simpler by far is to simply use a screw on compression driver. JBL 2408H-2 is a good one. There are a bunch of screw-on compression drivers from JBL for sale on eBay; it appears that screw-on is their favorite mounting method lately. Some of them are under $35, but I can only vouch for the 2408H.
> ...



Hmmmmmmm, okay so that makes sense , yeah I used a screw on adaptor and was using the celestion cdx1-1425 . I hacked the piles down really far in parts and thought the dash would extend the horn , I crossed at 2.5k than tryed a bms 4550 , and went to 1.2k , now the cd10nd on esmini. I've got to try that jbl driver. , is that one with a annular diaphragm ?


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## oabeieo (Feb 22, 2015)

thehatedguy said:


> Eric's horns are meant to be off axis, no toe in or out, and not aiming them at your ears/head.
> 
> You can put foam in there, it only really has to go to the edge of the mouth of the horn. Last time I made foam inserts, I went from the reflector to the edge of the mouth.


Is there anything one could do to the mini to make it for on axis , does it need a redesign? Or is it just best suited for off axis but works on axis? I have noticed that hom changes as the horn is moved or repositioned. Seems like hom is worce off axis but on axis you can hear some sort of something down at the reflector that sounds like you can hear the decompression . It sorta sounds like the swishing sound coming from a air tip off a air compressor line centered at 3500khz . There is no perfect way , so I'm hoping the foam will filter that decompression sound out and get rid of a little hom, I can still hear hom on axis but it's not 1/2 as bad as off axis hom. Small tweaks in mounting can eliminate annoying modes especially coming from passenger side, I know there meant to be mounted straight forward and parallel but sometimes that's just not optimal because of hom. So I sacrifice imaging for tun ability. But sacrifices not to the point that imaging is done away with , I'm talking a tow in of maybe a few degrees here and there or so just little bits to get hom in check and get pattern to a usable degree.


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## oabeieo (Feb 22, 2015)

Patrick Bateman said:


> If you *do* intend to hack up one of the Pyle clones of the JBL waveguides, it's a lot simpler to use a dome. I did that with a Vifa XT25 and it worked fine. It was interesting how different a dome sounds on a waveguide compared to a compression driver, even though the response was the same. Dare I say it, the dome might just work better. I'm really dying to put my SB Acoustics tweeters on a waveguide. (Haven't had time - I'm literally typing this at an airport.)


I'm down to try just about anything if it means using a horn in a car. For this install I discovered I need the horn to play down to 1.2k . The 6.5s I got are the ones with out shorting rings and 3" coils induct at 1k also the bimming at 1100 from the 6.5. , I am stuck on these morel drivers , I would entertain trying the audax Jason mentioned in a diffrent thread . I'm holding out for Eric's UH6 6.5s . His fb page shows them and they have baskets and motors like the morels which tell me 3" coils . Looks like a morel with a paper cone and the surround from a 18sound speaker. I'm kinda excited to give them a shot . I could only imagine a highe efficiency morel/dyn type speaker would be fantastic!


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## TheAlchemist9 (Apr 29, 2015)

Patrick Bateman said:


> Do you have five midbasses, four midbasses, or two midbasses?
> 
> If you have two or four, the reason that the bass is increasing is because moving the bass from the center (where there's no midbass) to the sides (where there is.)
> 
> Earlier in the thread I did some spectrum analysis of the output of a PLII decoded signal, and an extra 10dB of bass sounds about right.



I've got 4 midbasses, but they both get signal from the front L and R channels (only 8 channels on the H800). I'll try turning down the midbass and sub bass drivers next time I run PLii... But first I have to set up my 2 12w6's


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## Patrick Bateman (Sep 11, 2006)

Patrick Bateman said:


> It took way, way, way, WAY too long to get this working.
> 
> But I got it working.
> 
> ...












I managed to get it working with the Tritton adapter that Cajunner showed me.
*This solution is probably superior, for the following reasons:*
1) this is real Dolby PLII. This isn't someone reverse engineering it.
2) I took the decoder apart, it's a real Dolby decoder
3) This solution does not require any cables between the source and the player. IE, you sit down in your car, turn on your gear, and hit "play" on your iPad or Android.

To get this to work, I used the Belkin "G3A2000" BT receiver ($50), the Tritton "AxPro" decoder ($10), a couple of 5V adapters I had lying around, and a tablet.

So it's a cheap solution.

I had some infuriating issues getting the SPDIF to work. It will probably work fine for you, but I could NOT get it to work with my Android phone. (ZMAX ZTE.) I wasted a few hours trying to figure out why; I thought that maybe the decoder wanted the audio in Dolby AC3, so I transcoded some FLAC files to AC3. That didn't fix it.

It turned out the problem was simply my phone. For some reason, it just didn't work with the Belkin BT receiver.

I knew that the PLII decoder was the real deal when I hooked it up to my Playstation 3. It played back without a hitch.

By switching from my Android phone to a Windows tablet, the whole setup worked without fail. I haven't tried it on my iPad yet.


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## Patrick Bateman (Sep 11, 2006)

I'm going to put future updates to this project here:

http://www.diymobileaudio.com/forum...udio-discussion/180497-28-weeks-later-10.html

The current thread is basically focused on HOW to do a five channel soundstage; the '28 weeks later' thread is more general, and what I learned from THIS thread will be applied to that project.]


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## TheAlchemist9 (Apr 29, 2015)

So how did it end up sounding when you got the 5.1 playing?


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## cajunner (Apr 13, 2007)

see, that's the thing I was worried about, I thought:

"headphones? they don't have to follow THX, they don't have to even make it up to Dolby standard since it's a non-standard application, they probably don't have to adhere at all to any of the benchmarks that licensing requires, because these may just be counterfeit or IP stolen, and some warehouse is shipping the chips...."


then I thought:

"well, if they are shipping these with good chips, then the actual Dolby stuff is going to be intact, the only thing to worry about is noise floor and controls, build quality" and that's when it hit me.


the headphones are about the most sensitive way to listen to an audio equipment and if they suck just a little at it, they won't sell in the market. Obviously they sell, since gamers use them for their game play and on that standard, they have to be at least kinda good.

then I thought, if the execution is at least as good as just regular radios running off modern opamps on headphone outs, and they are made to drive loads down to 32 ohms, they must be pretty decent, and if you can't hear the noise on headphones, it's a good chance that you won't hear it on high powered car audio either.


so, in all, as cheap as you can get these on the refurb/parts bin prices from ebay resellers, it's only a slight gamble to go with a true upmixed center and ambient information surround.

I know purists running Sinfoni and Mosconi A class amps would balk at tying this thing into their tune-box, but even if it's just to extract a center channel when you want to get bangin' on the in-dash movies or listen to a couple of surround encoded concert DVD's, it's probably worth the time and effort.


I mean, for what it does and if done well, it'll be just like switching in the surround on the H800 for what? 20 bucks for a pair of those cheap 3" full range on Parts Express and another 20 for the decoder, and another 20 bucks for a POS old school low power amp you probably have already?

could be a game changer, you might not want to switch it out, EVER...


and it might even beat Logic7 at it's own game if you have all kinds of outboard controls on the signal AFTER it leaves the decoder.


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## oabeieo (Feb 22, 2015)

I just bought the computer and the sound card . Woot woot . What I want to do is get the creative sound blaster live portable with eax and get that working mobile . So I got this card instead , has spdif and surround outs ,just a better card IMO and creative environmental audio which is pretty fun to tinker with . I should be able to spdif out to the Audison bit one and should sound pretty good . 

http://www.pcworld.idg.com.au/review/creative/sound_blaster_x-fi_surround_5_1/224822/


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## oabeieo (Feb 22, 2015)

Patrick Bateman said:


> I managed to get it working with the Tritton adapter that Cajunner showed me.
> *This solution is probably superior, for the following reasons:*
> 1) this is real Dolby PLII. This isn't someone reverse engineering it.
> 2) I took the decoder apart, it's a real Dolby decoder
> ...


Patrick can you help me? 
I got the computer you posted , I got a 6 channel sound card as you said to do . 
Besides a usb hub how do I connect the tablets micro usb to the hub or the sound card because sound card has mini usb jack to full size usb cable? How do I do that part is there a special cable I have to get? 


hosting image


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how to do a screen shot


image sharing sites


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## Patrick Bateman (Sep 11, 2006)

the table has a female microusb port, so you need a male microusb on one side, to a female usb on the other.

I'm using something called a "usb on the go" adapter.

I found that the sound card wouldn't work properly unless I had a powered USB hub. Basically the tablet isn't providing enough amps on the USB port to power the sound card.

I think Cajunner's solution, of using the Tritton decoder, works better. I still have my tablet but I don't intend to use it in my car.

If you opt to go with the tablet, the cables are easy to find. Office Depot or Best Buy will have it.


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## oabeieo (Feb 22, 2015)

Patrick Bateman said:


> the table has a female microusb port, so you need a male microusb on one side, to a female usb on the other.
> 
> I'm using something called a "usb on the go" adapter.
> 
> ...


Oh thank you! Ok I never seen a cable like that. I will get one. I really hope I can get this sound card to work. It has some sound field stuff in the software that allows you to manage effects very well , you can add reverb , two tap, chorus , and it has eq plug ins that is fully parametric and you can select the center freq down to a decimal point i.e. Could have a 3683.63hz eq center with a variable q to be a sharp notch or a wide multi oactave q. .... You can go into effects like reverb and program how much reverb , room size, can detune the room and have like a 2hz swing to the reverb with sub frequency detuned . Ajust how close early and late reflections are and there volume .... It goes on and on its a sound editors dream. I would be stoked to get it to work. I would put spotify on the tablet go interest the sound card and go optical out to a helix or my Audison bit one. I'm kinda excited to play with it and get some rear speakers


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## fullergoku (Jun 21, 2009)

Patrick Bateman said:


> It took way, way, way, WAY too long to get this working.
> 
> But I got it working.
> 
> ...


I'm curious about since you are using Foobar2000 could you use this tablet to run the APL software for tuning then out optically into a Alpine H800 and maybe also do the tuning of the alpine on this tablet?


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## cajunner (Apr 13, 2007)

fullergoku said:


> I'm curious about since you are using Foobar2000 could you use this tablet to run the APL software for tuning then out optically into a Alpine H800 and maybe also do the tuning of the alpine on this tablet?


that would be awesome, and if you could do a sendspace of the APL software, on the down low...


haha..

uh, sorry got carried away there.

If the tablet performs the functions of the APL-1, then you wouldn't need the hardware at all?

and if it's going into a DSP that is configured by the tablet, it doesn't have to be connected, or the software doesn't have to run concurrently as the APL software, right?


so is the APL software going to run inside the Foobar shell, or VST plug-in, or is it going to have to be engaged for the music to run through....

it gets complicated for me, can somebody "do up" a tablet with the software and set it for a three way with my settings, then send it to me to see if I likey or not?

that would be best, I think...


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## redit (Jan 14, 2012)

Patrick Bateman said:


> HP Stream. For sale at Groupon for $70. Definitely not as cool as an iPad, but the Stream should be able to use USB sound cards and any Windows software.


Sorry for going OT, but could that HP Stream be used to configure/tune a miniDSP?


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## Patrick Bateman (Sep 11, 2006)

redit said:


> Sorry for going OT, but could that HP Stream be used to configure/tune a miniDSP?


Yep. You'll need a USB OTG adapter, about $10 at Fry's or Amazon.

The HP Stream is basically a Windows PC that happens to be a tablet for about seventy bucks. Dell sells one for about $100 that's an inch bigger with twice the ram.


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## redit (Jan 14, 2012)

Patrick Bateman said:


> Yep. You'll need a USB OTG adapter, about $10 at Fry's or Amazon.
> 
> The HP Stream is basically a Windows PC that happens to be a tablet for about seventy bucks. Dell sells one for about $100 that's an inch bigger with twice the ram.


Right on, thanks dude.


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## Nismo (Jan 10, 2010)

Resurrecting this...ftp://ftp.saitek.com/pub/support/trittonmanuals/90302N_TRI_AXPro_UG.pdf

This shows the 'mic' port as being a USB connection to a game system. I don't know if it will work with Windows (or better yet, Android). The issue I'm having here, is that I'm going Nexus 7 > Hifimediy DAC > JVC media deck (for now) > Tritton > MiniDSP > ADC/DAC in Kicker amps. I'm seriously looking at changing from analog to digital 8x! The issue I'm now seeing is that my deck will probably be intercepting the LF/RF signal AFTER the Tritton for BT calling (which is honestly the only purpose), which seems like a good idea...but it doesn't allow me to mute the audio to the center, LR/RR, or sub. What I'm considering is using a switch to kill power during phone calls to the Tritton, but I don't know if that will cause popping.

This just seems like a complicated PITA to get Pro Logic II. I know if I ran a Pioneer AXM-P8000/DEQ-P8000 combo or an Alpine PXA-H800, I wouldn't likely have these issues.

Anyone want to chime in with a good suggestion on what to do?

Eric


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## Patrick Bateman (Sep 11, 2006)

That setup makes my head hurt. 

But I can tell you that the device that cajunner recommended worked for me.

I had to buy two different devices to get it to work. IIRC, the device that worked for me had analog outputs, not digital.

In the end, I wound up going back to a stereo setup. I'm still doing five channel in my Genesis with Lexicon Logic7.


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## Alextaastrup (Apr 12, 2014)

cajunner said:


> that would be awesome, and if you could do a sendspace of the APL software, on the down low...
> 
> 
> haha..
> ...


Yes, APL has a VST plugin which is working perfectly in Foobar2000 - but only for two channels. In this case you do not need the hardware. This plugin raplaces the box completely. And this solution is 3 times cheaper than APL1 processor. 

If with the APL1 box you are limited to 16 presets, the freedom of selection is simply unlimited in the case of the APL plugin.

You might be interested in making XO from your Windows tablet using dephonica software.
https://dephonica.com/version3/


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## Nismo (Jan 10, 2010)

Patrick Bateman said:


> That setup makes my head hurt.
> 
> But I can tell you that the device that cajunner recommended worked for me.
> 
> ...


I agree completely! I think I may end up doing a separate speaker for the Bluetooth...eliminating a couple AD/DA conversions. I can now run my DAC in digital to digital mode, and I'll be down to Nexus -> Tritton -> MiniDSP -> amps.

I wish I knew a way to cut out the minidsp analog out and amp analog in...but I know that would be significantly more difficult than I can imagine.

If I don't like the Tritton, I can always cut it out and keep my deck in the signal chain instead.

Eric


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## Patrick Bateman (Sep 11, 2006)

I bought one of these surround sound encoders, detailed earlier in this thread, but I was thinking about how I could recreate the effect without a hardware encoder.

Here's my thinking:

The surround sound encoder I'm using is designed for videogame headphones. Maybe it has great parts inside, *but why would it?* It seems like a bit of a waste to run the signal through a potentially dodgy encoder. And reviews on other PLII encoders seem to indicate that many of them have issues.

When it comes to surround sound encoding, there's a lot of moving parts, and a control freak like me is keen to have control over those parts.

So...

It seemed to me that an ideal way of doing this would be to do the surround encoding myself. Basically turn surround sound into five channel in software, on a PC, then record the files to AC3.


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## garysummers (Oct 25, 2010)

Patrick Bateman said:


> I bought one of these surround sound encoders, detailed earlier in this thread, but I was thinking about how I could recreate the effect without a hardware encoder.
> 
> Here's my thinking:
> 
> ...



The method your describe is how I generate my up-mixed material to play in my car. I use the "Perfect Surround"( google it and try the demo) up-mixing plug-in on my PC. I load the stereo music into "Reaper" , a sound editing program. Then the VST plug-in is applied and 5.1 24 bit-48Khz .wav files are generated. Those are then DTS encoded and a 5.1 DVD-video is burned. Audio only of course. Just sharing my method!


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## Patrick Bateman (Sep 11, 2006)

garysummers said:


> The method your describe is how I generate my up-mixed material to play in my car. I use the "Perfect Surround"( google it and try the demo) up-mixing plug-in on my PC. I load the stereo music into "Reaper" , a sound editing program. Then the VST plug-in is applied and 5.1 24 bit-48Khz .wav files are generated. Those are then DTS encoded and a 5.1 DVD-video is burned. Audio only of course. Just sharing my method!


Your Mercedes was the thing that got me thinking about surround sound again.

I have surround sound in my Genesis, using the Lexicon stereo that's an option. And I do enjoy it, and 90% of the time I listen in five channel, not two.

But the thing that's really compelling about surround, at least for me, is the ability to add some 'ambience' to crummy recordings.

IE, if the music I listened to was really well recorded, surround sound might just make things sound worse. But I listen to a lot of music that's recorded TERRIBLY. My favorite music genre is EDM, and my 2nd favorite is old school punk rock. Both are basically mono.


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## Patrick Bateman (Sep 11, 2006)

The easiest way to turn stereo into five channel is to use a hardware processor. There's a handful of them, I believe the MS-8 can do it, and there's an old Fosgate processor that can do it. Jim Fosgate, of Rockford-Fosgate, invented Dolby Prologic II.

The second easiest way is to use software. Gary posted a link on this page, and there's also encoders from Corel and Minnetonka.

But if you're incredibly cheap... You DIY it.

First step is to grab Sox from here:

SoX - Sound eXchange | HomePage


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## Patrick Bateman (Sep 11, 2006)

There's a thread at videohelp that shows how to convert your stereo files to multichannel here: 

How to Convert 2 channel stereo wave into DD5.1 AC3 448 / AC 640

The next few posts are mostly based on that.


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## garysummers (Oct 25, 2010)

Patrick Bateman said:


> Your Mercedes was the thing that got me thinking about surround sound again.
> 
> I have surround sound in my Genesis, using the Lexicon stereo that's an option. And I do enjoy it, and 90% of the time I listen in five channel, not two.
> 
> ...


The up-mixing plug-in that I use, same as Andy is using in his new processor, does not ADD ambience but extracts the L-R, ambient information and moves it to the surround field. This process is much more successful than the common "Rear Fill" application. Even if you are able to derive an L-R signal and send it to the rears, you have not removed this L-R signal from the left and right. It has just been duplicated and sent to the rear channels. This I believe will not yield the surround effect that you're talking about and that we've experienced in true surround decoding.
The reason I use the "Perfect Surround" plug-in is that it is the most faithful in re-creating the front LCR stage as well as deriving a satisfying rear surround field.


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## Orion525iT (Mar 6, 2011)

The Penteo software isn't that expensive in my opinion. They even had a sale on the premium Penteo 7 ($349), but sadly it ended on Nov 30th.

The real holy grail to me would be to find something to do this real time with any stereo input so you could easily switch between streaming and recorded material. It would most likely take some serious processing power I assume, but I don't see it as an impossibility. As it is, you have to encode everything beforehand which is a bit of a hindrance.

I fully sold my soul to single seat and am committed to trying a Opsodis like arrangement. But if I wasn't going that direction, I would spring the $199 for the 5.1 Penteo. Heck, even the $500 for the 7.1 pro suite doesn't seem like too much in the grand scheme of things.


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## garysummers (Oct 25, 2010)

Orion525iT said:


> The Penteo software isn't that expensive in my opinion. They even had a sale on the premium Penteo 7 ($349), but sadly it ended on Nov 30th.
> 
> The real holy grail to me would be to find something to do this real time with any stereo input so you could easily switch between streaming and recorded material. It would most likely take some serious processing power I assume, but I don't see it as an impossibility. As it is, you have to encode everything beforehand which is a bit of a hindrance.
> 
> I fully sold my soul to single seat and am committed to trying a Opsodis like arrangement. But if I wasn't going that direction, I would spring the $199 for the 5.1 Penteo. Heck, even the $500 for the 7.1 pro suite doesn't seem like too much in the grand scheme of things.


The Penteo Surround folks are currently working on what you speak of, realtime in-vehicle processing of stereo material into 5.1 surround. Andy is the first to receive the chip version of the process. I know they are talking to other car audio processor manufacturers about implementing Penteo into their processors.
They are big names you know! Patience! ?


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## Jscoyne2 (Oct 29, 2014)

I have a feeling andy's dsp is gonna be game changing.


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## Orion525iT (Mar 6, 2011)

garysummers said:


> The Penteo Surround folks are currently working on what you speak of, realtime in-vehicle processing of stereo material into 5.1 surround. Andy is the first to receive the chip version of the process. I know they are talking to other car audio processor manufacturers about implementing Penteo into their processors.
> They are big names you know! Patience! ?


Ahh, I didn't realize Andy had a chip on hand. I was under the impression he was utilizing preprocessed source material. I still think there will be sonic compromises to any approach like this; the space is simple too small optimize equally for two seat or more. But I am willing to eat crow if proven wrong.


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## Focused4door (Aug 15, 2015)

cajunner said:


> for 50 bucks you can buy a HTIB and add another 25 for the "400 watt inverter" you'll need?
> 
> or 40 bucks gets you a real Home Theater Receiver, with good power at a Pawn shop?
> 
> ...


Cheap inverters are often pretty noisy. Even true sine wave inverters aren't always as clean as you would expect.

I think I would skip the inverter and pull the receiver step down transformer and put in a good DC-DC converter. This falls into the don't try this at home category, since it likely involves voltages high enough to be fatal.

May open a different can of worms than dealing with an inverter, but that is my thoughts.

After 4k receivers got popular the old stuff sure did get cheap though.



EcoHawk said:


> I personally think the PLii upmix from my H800 sounds terrible. Alpine's Euphony mix actually sounds a lot better to me. Or plain stereo. I'm not a fan of PLii for home audio either, though... I think Logic 7 (or Neo 6) sound much better.
> 
> I'll be following this and hope that PLii or another 5-channel mix works better for you than me.


I will second that, Logic 7 always sounds better to me.


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## oabeieo (Feb 22, 2015)

Jscoyne2 said:


> I have a feeling andy's dsp is gonna be game changing.


We'll see...gosh dang I sure hope so. 
Having a palethra of processors daisychained is kinda clunky.


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## Patrick Bateman (Sep 11, 2006)

Orion525iT said:


> The Penteo software isn't that expensive in my opinion. They even had a sale on the premium Penteo 7 ($349), but sadly it ended on Nov 30th.
> 
> The real holy grail to me would be to find something to do this real time with any stereo input so you could easily switch between streaming and recorded material. It would most likely take some serious processing power I assume, but I don't see it as an impossibility. As it is, you have to encode everything beforehand which is a bit of a hindrance.
> 
> I fully sold my soul to single seat and am committed to trying a Opsodis like arrangement. But if I wasn't going that direction, I would spring the $199 for the 5.1 Penteo. Heck, even the $500 for the 7.1 pro suite doesn't seem like too much in the grand scheme of things.


My stock system is from lexicon and it already offers this. (Harman owns the patents on Logic7, a competitor to DTS NEO6 and Dolby Prologic II)

I find that 80% of my stuff sounds better in five channel.

Maybe this processing hasn't spread to most cars because Microsoft, Bose and Google dominate the market for in-car entertainment, while Dolby and Harman have all the patents. This might change with Samsung's acquisition of Harman.


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## Patrick Bateman (Sep 11, 2006)

Patrick Bateman said:


> The easiest way to turn stereo into five channel is to use a hardware processor. There's a handful of them, I believe the MS-8 can do it, and there's an old Fosgate processor that can do it. Jim Fosgate, of Rockford-Fosgate, invented Dolby Prologic II.
> 
> The second easiest way is to use software. Gary posted a link on this page, and there's also encoders from Corel and Minnetonka.
> 
> ...




After you install Sox, you'll want to modify your path as detailed here: https://www.java.com/en/download/help/path.xml

I'm doing the work in Windows, so this will vary if you're using the linux or OSX versions of SOX


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## Patrick Bateman (Sep 11, 2006)

I might copy the Penteo setup instead of the Prologic II setup. Trying to figure out if it can be done with SOX.

From the Penteo maual:

_"In Penteo Music mode, some centre is removed from the centre speaker and mixed into the front pair at a low level, resulting in a wider centre, yet remaining solid and stable in the centre speaker itself. All other features (bass management, phantom fronts) remain the same."_

And some deeper detail:

_From the outset, a number of criteria had to be met.

* The audio quality would be as sonically artefact-free as technologically possible at every stage, even when soloing individual output channels.

* The latency would be the lowest possible while still accurately performing the DSP processing.

* The upconverted 5.1 stereo would downmix transparently back to the original stereo.

* All surround field placement would be determined by the original stereo mixer’s panning choices.

* The stereo image would simply bend in shape from stereo’s traditional flat, left-to-right movie screen shape into a horseshoe shape, with extreme left mapped to the left (rear) extreme, centre mapped to the precise centre, and extreme right mapped to the right (rear) extreme, using the same relative left-to-right panorama that the original stereo mixer had defined.

* A firm, stable centre channel, with no side-to-side wobbling.

* By default, bass should be redirected from the centre to the front left and front right speakers, as those often are home listeners’ only full-range speakers.

* The centre width should be variable from hard-and-discrete to full-front-width.

* The rear channels should share no sounds in common, as they define the extreme ends of the panorama; nothing in stereo would ever be analogous to rear-centre.

* There should be discrete separation between at least the rear left and centre as well as the right rear and centre.

* There will be no phase alteration or phase-matrix processing: all channels must remain 100% in-phase.

* There will be no added delays, reverb, or bandwidth filters (other than the above-stated bass management).

* An LFE channel will be created by simply duplicating any sounds below 100Hz.

Penteo achieves these criteria by matching the sonic patterns created by each sound wave in the stereo mix and then, with DSP, separating out the sounds based on their left-to-right stereo placement. It then distributes the mix accordingly, and the result is a direct translation of a flat stereo mix into 5-point surround.

In the traditional remixing process, any sound source can be manipulated to appear in a specific point in the left to right range by changing the relative volume levels in the left versus right speaker channels. Instead of looking at stereo as two parts, left and right, Penteo treats a stereo source as one whole range: a left to right flat panorama, similar to a flat movie screen stretched between the two speakers. Using DSP, Penteo is able to subdivide the left-to-right spectrum as setup by the original stereo mixer and then it slices the panorama into those groups of singers, actors, or instruments."_


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## Patrick Bateman (Sep 11, 2006)

Here's a quick 'guesstimate' on the Penteo music mode, based on what's in their manual. 

LEFT CHANNEL = AS IS
RIGHT CHANNEL = AS IS
CENTER CHANNEL = (0.7071 LEFT CHANNEL + 0.7071 RIGHT CHANNEL) BANDPASS 70 20000 
LFE CHANNEL = (0.7071 LEFT CHANNEL + 0.7071 RIGHT CHANNEL) LOWPASS FILTER 120
REAR LEFT CHANNEL = left channel - (left channel + right channel)
REAR RIGHT CHANNEL = right channel - (left channel + right channel)

From their user manual:

_" In Penteo Music Mode, material panned to the center appears loudest in the center speaker, but some also appears in the front left/front right pair. This mode is optimized for music, keeping the center channel vocalist loudest in the center, but not so discrete as to offend listeners who are accustomed to a broad center vocalist coming from the full width of the stereo field. The front pair also receives bass diversion, if selected. See Bass Redirect Toggle, p26. The rear pair receives the extracted pure sides only."_

And a quote from the Penteo dude, from avsforum:

"But again I digress. With Penteo, I'm attempting to upmix the original masters to the absolute best quality that I can, simply extracting the extreme pan positions and rechanneling them to the surround locations. It's not a new Mona Lisa with the original paint formulations; it's the old girl herself. In fact, the front two channels are the original 2-channel mix, super-mastered to the utmost possible quality. With Penteo, I mathematically find the center, pure left and pure right, and channel them to the surround speakers. That's all. The original mix is untouched."









LEFT CHANNEL = AS IS
RIGHT CHANNEL = AS IS
CENTER CHANNEL = (0.7071 LEFT CHANNEL + 0.7071 RIGHT CHANNEL) BANDPASS 70 20000 
LFE CHANNEL = (0.7071 LEFT CHANNEL + 0.7071 RIGHT CHANNEL) LOWPASS FILTER 120
REAR LEFT CHANNEL = 0.8718 LT CH WITH -90 DEG PHASE SHIFT + 0.4898 RT CH WITH +90 DEG PHASE SHIFT
REAR RIGHT CHANNEL = 0.4898 LT CH WITH -90 DEG PHASE SHIFT + 0.8718 RT CH WITH +90 DEG PHASE SHIFT

For comparison's sake, here's the Dolby Prologic II matrix. Note that there will likely be quite a bit of comb filtering between the left, center, and right channel in PLII. This is because the center has music from the left channel, but so does the left channel. (To improve separation, you might consider removing or reducing the signal that's in common.)


Wheeler's statements on avsforum contradict the Penteo manual. According to his statements on avsforum, the left channel and the right channel are intact. According to his manual, " material panned to the center appears loudest in the center speaker, but some also appears in the front left/front right pair." I personally would prefer the second option; I anticipate that it would create a stronger center channel. This is because the second option would subtract some or all of the sound that's common between the left and the center channel.

For instance, in the PLII setup, you have 100% of the sound from the left channel going to both the left channel and the center channel. I think you could improve separation by eliminating some of that from the left channel. (To accentuate the center.) You would do this like this:
center = left + right
left = (left - center)


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## garysummers (Oct 25, 2010)

In the Penteo Composer 5 VST plug-in that I use, the user is able to vary the amount of center and surround information extracted from the stereo signal.
From my experience the left and right channels are not left intact. The process does not just duplicate information from left and right into center and the rears but removes a user determined amount and redirects it to the proper location.


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## Patrick Bateman (Sep 11, 2006)

Yeah I've been tinkering with a way to do this with open source, and I think I have it sorted out:

First, we create a mono signal

Next, we create a 'hard left' and a 'hard right'. These two signals are the left channel and the right channel *but with the center removed.*

If we were doing three channel, we could stop at this point and call it a day. At this point we've turned stereo into three channels.

To get five channels, we create two new channels that are a blend of the ORIGINAL left and the 'hard left' channel. And then repeat the process with the original right and the 'hard right' channel.

We can adjust the ratio in those two channels to increase or decrease the separation from the centre channel.

As noted in the docs, this turns flat two channel into something that's shaped like a horseshoe.

And as noted in the Penteo docs, we're not changing the phase or adding delay. Due to this, it's possible to turn the five channel back into two channels.

Hopefully I'll have the script knocked out tonight. SOX is not user friendly.


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## garysummers (Oct 25, 2010)

Patrick Bateman said:


> Yeah I've been tinkering with a way to do this with open source, and I think I have it sorted out:
> 
> First, we create a mono signal
> 
> ...


Penteo 5.1 composer is only $199 and it works very well!
Just a thought! ?


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## Patrick Bateman (Sep 11, 2006)

This script needs to be cleaned up. Seems to be working.

Here's how to run it:

1) Install Sox as mentioned in the previous post
2) Using notepad, create a script using the code listed below. I named my script "prologic2.bat"
3) put the script in your path, or in the same directory as your music files
4) and then run it like this:

prologic2.bat myfile.wav

Sox can deal with a number of file formats, so your music doesn't have to be in WAV. It could also be in mp3 or flac. I've only tried it with stereo files, so I'm not sure how it would behave with a mono or six channel file.

Here's the script:

echo resample to 48kHz and lower volume to avoid clipping
sox -S -c 2 %file% -r 48k stereoInput.wav gain -h

echo create mono channel
sox -S stereoInput.wav -c 1 mono.wav remix 1,2

echo create left channel
sox -S stereoInput.wav -c 1 left.wav remix 1

echo create right channel
sox -S stereoInput.wav -c 1 right.wav remix 2

echo create hard-left channel
sox -M left.wav mono.wav left-plus-mono.wav
sox -S left-plus-mono.wav -c 1 hard-left.wav remix 1v0.5,2v-0.5

echo create hard-right channel
sox -M right.wav mono.wav right-plus-mono.wav
sox -S right-plus-mono.wav -c 1 hard-right.wav remix 1v0.5,2v-0.5

echo create new-left channel
sox -S left-plus-mono.wav -c 1 new-left.wav remix 1v0.5,2v-0.25

echo create new-right channel
sox -S right-plus-mono.wav -c 1 new-right.wav remix 1v0.5,2v-0.25

echo create LFE channel
sox -S stereoInput.wav -c 1 lfe.wav lowpass 120 remix -

echo merge channels
sox -S -V -M new-left.wav new-right.wav hard-left.wav hard-right.wav mono.wav lfe.wav multichannel.wav

echo normalize + make sure it's 16bit
sox -S -V -G multichannel.wav -b 16 normalize.wav 

echo convert to AAC
qaac normalize.wav


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## Patrick Bateman (Sep 11, 2006)

I know the last few posts were confusing and dry, but I pondered the Penteo and Dolby ProLogic II schemes, and I have some good news:

*We can get REALLY close to what they're doing, using nothing but MiniDSP.*

This simplifies things a great deal:

1) it means you don't need a Windows tablet for a source
2) it means you don't have to chase down a dodgy processor off of EBay to do ProLogicII
3) it means you don't need to install a home theater receiver in the trunk of your car
4) it means you don't have to process your entire music library using a $500 VST plugin
5) it means you can turn two channel into five channels in real time, using something as simple as an iPod or a head unit as a source









The 'trick' is to look and see what the ProLogic II matrix is doing:

1) the left channel is untouched
2) the right channel is untouched
3) the center channel is a sum of the left and the right

We can do ALL of that in MiniDSP. No problemo.

The only tricky part are the surrounds. Dolby/Fosgate use a mix of the left signal, mixed with an inverted copy of the right, with a 90 degree phase shift. The Posts from Penteo imply that they're doing something like "L-R", but with no phase shift. (This is why Penteo can turn their matrix back into stereo, with no phase anomalies.)

I'm thinking we can do a simple "L-R" signal for the surrounds. "L-R" is just what it sounds like, it's what you get if you nuke the common sounds out of a combined signal. It blows a big hole in the center of the stage. *Increasing the volume of the "L-R' component makes the stage wider.* This trick is wildly abused by musicians already.

So...

I'm thinking about a simplified surround setup, as illustrated in the pic above. Five discrete channels, produced by MiniDSP, in real time. No tablets, laptops, car PCs or processors.

I think the 'key' to getting this right is understanding how stereo works. Basically the entire system will need to be designed specifically for five channel, in particular the center speaker will need to be a beast, because we want it to handle some abuse, and also be directional enough to minimize interference with the left and the right speaker.

For anyone taking notes, here's the Dolby ProLogic II Matrix. This isn't speculation, this is right out of their documentation:


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## felix509 (Dec 17, 2006)

I will just leave this here for consideration. Seems to get some good reviews..

Surround Master

Manual


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## Patrick Bateman (Sep 11, 2006)

Thank you!


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## Patrick Bateman (Sep 11, 2006)

It's amazing how quickly projects come together when I use off-the shelf parts. My old standby, the Pyle PH612S, works like a charm as a center channel. Be sure to lop off the sides, to make it a better match for the windshield. It took less time to put this together than it took to type this post. TWELVE BUCKS


















Of course it's a Synergy Horn


For anyone playing along at home, here's the parts list:

Celestion 4" closed back midrange, $32 at Loudsepakers Plus (same driver as Synergy Horn)
JBL 2408H-1 compression driver. This one used to cost about $80, but it's up to to $150 for some reason. Any ol' ring radiator will work, but this one is particularly well-suited because it's a screw on. If you're up to the task of making an adapter, you could use one of the other BMS drivers. Do NOT use an adapter, they screw up the wavefront


Although this variation is cosmetically the best, I'm going a different route because I want a narrower coverage angle, to reduce the interaction between the left and right channel and the center channel. Stay tuned.


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## Orion525iT (Mar 6, 2011)

Patrick Bateman said:


> I know the last few posts were confusing and dry, but I pondered the Penteo and Dolby ProLogic II schemes, and I have some good news:
> 
> *We can get REALLY close to what they're doing, using nothing but MiniDSP.*
> 
> ...


Might be a bit OT, but is there anyway with the minidsp that you can send rear channel L-R surround to drivers that also receive a normal signal? It's a bit weird to do this, but in my car I am using some 8" Type Rs to cover 30-150hz. They are stuffed in the rear quarters and fed a stereo signal. They can play higher than 150hz, but it can't be the main signal because it will start to pull the stage back. So can the subs be utilized further with a combined signal of normal stereo and some version of fill or surround?


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## Patrick Bateman (Sep 11, 2006)

Orion525iT said:


> Might be a bit OT, but is there anyway with the minidsp that you can send rear channel L-R surround to drivers that also receive a normal signal? It's a bit weird to do this, but in my car I am using some 8" Type Rs to cover 30-150hz. They are stuffed in the rear quarters and fed a stereo signal. They can play higher than 150hz, but it can't be the main signal because it will start to pull the stage back. So can the subs be utilized further with a combined signal of normal stereo and some version of fill or surround?


I can't figure out any obvious way to send L-R to any speaker using a MiniDSP where there are speakers *also* getting left or right signals.

IE, you can use the surround plugin to send L-R, but if you do that, you *can't* send L or R.

Due to that, I have to use *two* MiniDSPs.

I had a couple of ideas of how to work around this:

1) I think if you pre-encode the channels into mono and mid-side stereo, you might be able to turn them *back* into left and right using MiniDSP. If true, then you could produce all four channels with *one* MiniDSP. (left, right, center, and surround)
2) The other idea I had was to *physically* produce L-R in the car itself. To do that, you simply have two drivers in your surround speaker. One driver gets the left channel, the other driver gets the right channel, you invert the second driver, and that gives you "L-R"



As far as doing a *ratio* of L-R, you could do that three ways:

1) using SOX. This would require you to pre-encode everything you play in your car
2) using the miniDSP surround plugin. The catch is that you can't mix, so you'd have to *physically* mix the two, by literally putting the two speakers near each other and varying the ouput of each.


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## Focused4door (Aug 15, 2015)

Patrick Bateman said:


> It's amazing how quickly projects come together when I use off-the shelf parts. My old standby, the Pyle PH612S, works like a charm as a center channel. Be sure to lop off the sides, to make it a better match for the windshield. It took less time to put this together than it took to type this post. TWELVE BUCKS
> 
> 
> 
> ...


Shame that a car manufacturer doesn't build something like that into the dash. Probably would add minimal cost to the dash. Plastic dash + plastic waveguide instead of center channel grill.


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## Patrick Bateman (Sep 11, 2006)

I decided to try a different horn instead of the Pyle PH612. This is the Pyle PH12S. It's coverage angle is about half as wide as the Pyle PH612. This has a few advantages for a center channel:

1) Because the angle is narrower, there will be less interference between the center channel and the left and right speaker.
2) Because the angle is narrower, the on-axis output is higher
3) Because the on-axis output is higher, it makes it easier to crossover from the midrange to the tweeter. Basically the output is higher, so that gives us some flexibility with the crossover point.
4) Because it's deeper, there's room to put the midrange on the TOP instead of the sides. This leaves room to mount woofers on the waveguide, which will allow the center channel to be truly full-range. This will be a three-way Synergy Horn, not a two-way.

On the downside, the horn is bigger, because the depth is larger.









One 'trick' with the mounting plates for the midrange is that I drilled some holes into the horn body. The idea here is that the epoxy will fill in those holes and they'll act like 'anchors' for the epoxy. I learned this from one of my bike forums.


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## Patrick Bateman (Sep 11, 2006)

The "obvious" way to wire this thing up is using three amps and two miniDSPs. It would look something like this:

amp 1 / mini dsp 1 : left channel, right channel
amp 2 / mini dsp 2 : center channel surround channels
amp 3 : subwoofer

There's a thread that describes how to do this with one amp : http://www.diymobileaudio.com/forum...112658-old-school-passive-surround-sound.html

I had high hopes for that, but no dice. I'm using Class D amps and it appears that they'll blow up with this wiring scheme.

This might be a little wacky, but *I can't see any reason why you couldn't do the L-R mix right at the loudspeaker.* 










Polk did this in the SDA, why not do it in our surrounds?

So in this scenario, the setup looks like this:


amp 1 / mini dsp 1 : left channel, right channel, center channel, surrounds
amp 2 : subwoofer

Besides saving $200 in DSP, amplification, and software, it also reduces the number of wires a LOT. That was becoming a real issue in this project. My midbasses are under my seat, and due to that, I don't have a lot of room to hide electronics. Basically my electronics have to fit under my dash, or else I have to run RCAs and speaker wires all the way to the trunk. Which I don't want to do.

To add some detail to how this will be wired:

1) I will use a MiniDSP with the 2x4 advanced plugin. (https://www.minidsp.com/products/plugins/2x4-advanced-detail)
While it might not be immediately obvious why you want the "2x4 Advanced" instead of the "2x4", the reason is that the 2x4 advanced can give you a mono channel. I believe that was intended for a sub, but there's nothing stopping you from using it for a center channel.
2) Two of the four channels of MiniDSP will be used for the center. One for the tweeter, one for the midrange
3) One of the four channels of MiniDSP will be used for the left speaker, one for the right

Where things get interesting is that I'll have two additional speakers wired along with the left speaker, and two along with the right. So I'll create the "L-R" channel physically; I'll literally have a left channel in the back with a right channel right on top of it, and the right channel will be wired out of phase. This setup will create an "L-R" signal, but acoustically, not electronically. The downside to this setup is that it's a waste of power, and it requires four surround speakers instead of two. But speakers are cheap; the surround satellites that I intend to use cost under $10 each. So it's still cheaper than buying one more amp, one more MiniDSP, and the software. More importantly, it simplifies everything.


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## nstaln (Feb 11, 2009)

I am in the midst of installing and setting up my first 5.1 in a car.

My plan is to use the factory Bose 6-CDHU into an Alpine pxa-h800/RUX combo.

I will have 8x100rms and 2x500rms on tap for power via a pair of JL HD900/5's.

Speakers will be AudioFrog GB10 tweets and GB25 mid's through GB2510c crossovers. Tweets in A-pillar mid's in upper door. 

Midbass will be a pair or AF GS693's in the lower front door, band-passed for midbass duty only. 

Rear fill will be a pair of AF GS62's in lower rear doors and the center channel will be a single AF GS42 firing up and slightly forward from the dash. 

Suwoofer duty will be handled by a single TC Sounds TC1000 15 4ohm dvc with each sub channel going to each coil. Sealed.

Vehicle is a 2010 Nissan Rogue SL.

Any advice you can give of set-up/tuning etc? Any pitfalls to look out for? I'll be tuning with REW and a minidsp mic (I forget the model#). I have never tuned using REW so this will all be kind of new.


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## Patrick Bateman (Sep 11, 2006)

nstaln said:


> I am in the midst of installing and setting up my first 5.1 in a car.
> 
> My plan is to use the factory Bose 6-CDHU into an Alpine pxa-h800/RUX combo.
> 
> ...


I believe Andy and Gary are using relatively similar speakers for their rear surround as the front speakers. They may even be identical.

For Gary's setup, this makes sense; he literally works on movies for a living (check out his IMDB page, it's epic). So Gary can easily demo mixes in the car that are genuine, five channel movie soundtracks.


My application is quite a bit different. I'm not using ANY sources that are legitimately five channel. All of my sources are stereo, turned into five channel.

Due to that difference, the demands on my surround channels are quite mild. I did an analysis of the content of those channels using Audacity, and there'v very very little content in the surround channels.

Here's why that is:
_In the Dolby ProLogic II scheme, nearly 100% of the mono signal has been removed from the surrounds._ In the MiniDSP scheme detailed above, that rises to 100% removal of the mono signal. I believe Penteo does the same.

Due to the fact that the mono signal has been eliminated from the surrounds, the surrounds can be quite puny.

If possible, I'd put the surrounds somewhere with an unobstructed path to your ears. For instance, I'm putting mine at shoulder level. The reason for this is that these wavelengths are fairly short, and early reflections might cause issue. YMMV

The flip side of this is that the center channel needs to be pretty beefy. In particular, *the tweeter of the center channel quickly becomes the weak link in the entire chain.* This is because the tweeter frequently distorts before anything else. (This is why I went with a compression driver.)



TLDR: If you're using "real" five channel material for a source, you might consider using relatively large speakers for your surrounds. If you're faking it with Dolby Prologic II or the like, you can downsize the surrounds in a big way, because the mono signal is non-existent or virtually non-existent in the surrounds.


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## garysummers (Oct 25, 2010)

Just for the record, I do not play movie soundtracks in my car.
I play the 5.1 discreet music recorded for certain motion pictures, that I have access to.
I don't and never will watch movies in my car.


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## Rrrrolla (Nov 13, 2008)

I know I used to be able to get L-R from a bridgeable amp just by wiring from L+ to R+. Ive done it with amps in the past and ran that signal to the rears. I'm quite certain you could use the same amp and get the center channel by wiring it in a traditional bridged config, L+ to R-. I believe it sounded best with the rears run out of phase with each other. I'm pretty sure that any bridgeable amp can sum or subtract right at the speaker outputs. I dont recall the amp ever getting hot from this type of wiring... I tried this because I used to have an old surround sound home amp and this was exactly how they created the surround outputs. It was litterally a L+ to R+ connection on a stereo amp.

With a c-dsp, you could use the high level inputs. That way you could just invert wire inputs (just flip the polarity from inputs 1 and 2) on 3 and 4 input and then sum in the routing matrix.


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## Patrick Bateman (Sep 11, 2006)

garysummers said:


> Just for the record, I do not play movie soundtracks in my car.
> I play the 5.1 discreet music recorded for certain motion pictures, that I have access to.
> I don't and never will watch movies in my car.


Agreed, I just wanted to point out that the requirements for the surround speakers really depends on whether the source has "real" information in the surrounds, or if it's just derived from a stereo source.

When I was messing around with SOX, I discovered something neat, which is that the AAC file format has support for something like 50(!) discrete channels. Although this is neat, none of the musicians I listen to are using more than two channels. In fact, the music I listen to is basically mono.


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## Patrick Bateman (Sep 11, 2006)

Rrrrolla said:


> I know I used to be able to get L-R from a bridgeable amp just by wiring from L+ to R+. Ive done it with amps in the past and ran that signal to the rears. I'm quite certain you could use the same amp and get the center channel by wiring it in a traditional bridged config, L+ to R-. I believe it sounded best with the rears run out of phase with each other. I'm pretty sure that any bridgeable amp can sum or subtract right at the speaker outputs. I dont recall the amp ever getting hot from this type of wiring... I tried this because I used to have an old surround sound home amp and this was exactly how they created the surround outputs. It was litterally a L+ to R+ connection on a stereo amp.
> 
> With a c-dsp, you could use the high level inputs. That way you could just invert wire inputs (just flip the polarity from inputs 1 and 2) on 3 and 4 input and then sum in the routing matrix.


I believe that the amp channels in a Class T amp are bridged already. To make things more confusing, my Sure amps are advertised as "T amps", which would imply Tripath. But Tripath has been kaput for years, and my amps have a Texas Instruments chipset iirc.

In the thread here http://www.diymobileaudio.com/forum...112658-old-school-passive-surround-sound.html someone mentioned that many A/B amps will produce a bridge signal if you wire them as a passive matrix.


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## nstaln (Feb 11, 2009)

This is the 5.1 info from Alpine:

Surround Sound: The Alpine PXA-H800 System Integration Audio Processor supports up to 5.1-channel surround sound from multi-channel and 2-channel sources.

Dolby Digital/DTS: Enjoy true 5.1-channel surround sound just like your home cinema system. You can hear multi-channel sound from movie and music sources in the Dolby Digital and DTS Digital Surround formats. With Dolby Digital, the dynamic range is compressed so that powerful sound can be achieved at regular volume levels. This compression can be cancelled to achieve an energetic sound with even great power, like the sound in a movie theater. This can be set to Standard or Maximum.

Dolby Pro Logic II: The PXA-H800 also provides Dolby Pro Logic II decoding, so you can even enjoy stereo recordings in surround sound. You can choose between a Music or Movie mode. Music mode allows you to adjust the center width which provides the optimum vocal localization by adjusting the center channel position in between the center speaker and the L/R speakers.

Euphony: Euphony makes it possible to enjoy 5.1-channel sound from 2-channel sources. It creates natural surround sound from CD or iPod with wider soundscapes and harmonics. It especially enhances vocal and instrumental quality, for a superior sound field and no listener fatigue.

Rear Mix: This function mixes the front channel audio signal with the audio signal output from the rear speakers, improving the sound in the vehicle's rear seat. This is used in a system where there is no subwoofer and rear speakers can product lower frequency sounds than the front speakers. The Rear Mix can be set to -6, -3, 0, +3, or +6 dB.

Center Bass Split: Turning on this function sends the center channel low frequencies equally to the front left and right speakers. this enhances the overall sound when using a small center speaker. This is used when the center speaker has a small diameter and cannot produce low frequency sounds. You can set the Center Bass Split function to 200, 225, 250, 280, 315, 350, 400, 450, or 500Hz.


These look like some interesting choices…I can't seem to find much info on the 'Euphony' up-mixing.

The bass-split feature looks helpful as well.


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## Focused4door (Aug 15, 2015)

felix509 said:


> I will just leave this here for consideration. Seems to get some good reviews..
> 
> Surround Master
> 
> Manual


That looks interesting, and seems to get decent reviews but appears you would need to add a DSP with 5 inputs that could run simultaneously to be able to time align for a car. 



garysummers said:


> Just for the record, I do not play movie soundtracks in my car.
> I play the 5.1 discreet music recorded for certain motion pictures, that I have access to.
> *I don't and never will watch movies in my car.*


You should at least catch a drive in theater sometime, especially in CA where the weather is nice enough to do it year round.


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## truckerfte (Jul 30, 2010)

garysummers said:


> Just for the record, I do not play movie soundtracks in my car.
> I play the 5.1 discreet music recorded for certain motion pictures, that I have access to.
> I don't and never will watch movies in my car.


Lol, just curious, after spending all day working on movies at the office, can you enjoy going to see a flick on a Fri night?


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## Patrick Bateman (Sep 11, 2006)

Here's the frequency response and distortion of the Synergy Horn posted on the previous page. *This is the response of the midrange driver alone.*

This performance isn't all that great. I believe the issue is that the ports are a little bit too small. Enlarging them should lower the distortion and flatten the response.

This measurement is with no filtering, except for a 250hz high pass to protect from over excursion.








Here's the response of the compression driver on the Synergy Horn. This is pretty darn good. Distortion is basically nil and the response is great, except for a dip in the very top octave, which probably won't be audible.

This measurement has a first order low pass at 20,000hz. That's to offset the low pass effect of a constant directivity horn, and also to protect the tweeter from over excursion.

Make no mistake about it, this tweeter and horn combination is tough to beat.

*With the midranges and the compression driver working in Synergy, it should be trivial to achieve six octaves of response.* (300hz-20kzh)
To put that in perspective, the onobtanium horns in Richard Clark's Grand National covered five octaves.

But why stop there?! I got space to burn, and this is going to be a THREE way Synergy Horn. Seven octaves of output from a point source, with efficiency in excess of 100dB.


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## garysummers (Oct 25, 2010)

truckerfte said:


> Lol, just curious, after spending all day working on movies at the office, can you enjoy going to see a flick on a Fri night?


When you spend 50 to 80 hours of a week on average, in the dark, staring at a movie screen, when Friday comes, your looking for something else to do for fun.


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## Patrick Bateman (Sep 11, 2006)

I haven't been doing a lot with my 3D printer lately, because my prints have been failing about 90% of the time. (This is something I didn't know about 3D printers; the more you use them, the worse they perform.) But I tried making some satellites out of wood and PVC for this five channel project, *and they just looked awful.*

So I recycled these 3D printed satellites that I made for a previous project. *It turned out really well.*

There are a few 'new' tricks here:

1) A few months back I discovered that 3D printed loudspeakers leak like a sieve, and due to that, you MUST make them airtight. I used to use liqud nails for subfloors, a water based adhesive. This time around, I used mortite. The advantage of mortite is that it adds more mass, and it never dries. Basically it behaves like sound deadening material.
2) I've tried using mortite on the *outside* of 3D printed enclosures. Don't do that, it doesn't work. This is because paint won't stick. So this time around, I wrapped the enclosure in fiberglass. This gives you something similar to a constrained-layer-damping enclosure. (To do it properly, you'd add another layer of fiberglass to the mortite that's on the inside of the enclosure.
3) While it wasn't intentional, the fact that the enclosure has no sharp edges means that the fiberglass came out really smooth, which saves me a lot of time sanding and finishing the enclosure. It still needs to be painted though.

Put those two things together, and you have a good looking enclosure that's paintable, airtight, and deadened.


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## Patrick Bateman (Sep 11, 2006)

The center channel for my system is largely a copy of Bill Waslo's "Small Syns" Synergy Horn. I heard it at Bill's place, and it sounds exceptional.









Bill's speaker is a three way:
1) it uses a Tymphany compression driver for the highs
2) it uses a Celestion TF0410MR for the mids, the same midrange used in the Danley Synergy horns
3) it uses a pair of Faital woofers for the bass. They're 6.5" iirc.

More details can be found in the thread over at diyaudio.

My center channel speaker is set up like this:
1) it uses a JBL 2408H-1 tweeter
2) it uses a Celestion TF0410MR, as above
3) I'm trying to figure out what I can fit for the bass

OK, here's the problem I'm facing right now:

My center channel speaker is starting to fill up the dash in a hurry. Here's a pic of just the horn, you can see how it could cover up the whole center of the dash quickly:










Over at diyaudio, xrk971 documents a way of making a horn using a full range driver at the apex, instead of a compression driver. The advantage of going this route is that it simplifies things, it makes it less expensive, it makes it smaller, it makes it lighter. Here's a pic:










I am wondering if I should go a similar route. My center channel speaker is working nicely, and I am making good progress. *But it is getting big.*

XRK 971's speaker uses an SB Acoustics SB65 as both a midrange and a tweeter, and a pair of Dayton eights for the midbass.









Here's the measured response of the Celestion TF0410MR on my Pyle horn









Here's the *predicted* response of a Celestion TF0410MR on two different size horns. The first horn is 90 degrees x 90 degrees. (grey curve) The second horn is 45 degrees x 45 degrees (black curve). You'll notice that reducing the beamwidth of the horn raises the ouput level quite a bit, about six decibels. That's equivalent to quadrupling the input power(!) You'll also notice that the *predicted* response rolls off below 1000hz, while the *measured* response plays about an octave higher. This is fairly normal for Hornresp; it tends to exaggerate the high frequency rolloff of a driver.

From the sims you can see that a TF0410 has an efficiency approaching 100dB on a 90 degree horn, and if you reduce the coverage angle to 45 degrees it goes all the way up to 105dB. *This is an insane level of output, and way more than I need.* This is one of the weird things about Synergy Horns, it's possible to generate and absurd level of output in the midrange. The limiting factor in a Synergy Horn is nearly always the tweeter. The midrange is just loafing. (And that's why I got away with 2" midranges in 2009.)









Here's a comparison of the Celestion TF0410MR versus the SB Acoustics SB65 *on the same horn.* This shows the following:
1) horn loading the SB65 raises it's efficiency over 10dB
2) Although the TF0410MR is quite a bit louder, the SB65 is pretty darn loud

The key to all of this is the horn loading, basically all of the energy is 'funneled' into a narrow angle. The narrower the angle, the higher the output, for the most part.

So...

I'm now wondering if I could replace my three way with a TWO way. The same bandwidth, just fewer drivers and lower efficiency. (It will still be very efficient.)

Here's a pic of XRK971's horn:


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## thehatedguy (May 4, 2007)

Why would you need a 3 way center?


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## Patrick Bateman (Sep 11, 2006)

thehatedguy said:


> Why would you need a 3 way center?


The center channel is a Synergy Horn

Here's the response of the midrange on the horn:










Due to the rolloff at 300hz, I need a midbass to fill in the hole between the subs and the midrange

While it's true that I have a midbass array already, I don't have any midbasses for the center channel, only the left and the right


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## Patrick Bateman (Sep 11, 2006)

I was looking at my dash, and thinking that I could still squeeze a three way center channel up there if I put the midrange UNDER the horn, instead of on top of it, and then put the midbass at the top.









To squeeze three drivers into the center, I'll need a waveguide that's wider and shorter. The Pyle horn that I was using before measured 10" x 10". The one pictured above measures 13.5" x 6.5".









When finished, it should look like a miniature version of the Danley SH-64, the speaker at the top of this stack.









If anyone out there wants to build one of their own, it's really simple. Get a piece of plywood, and draw two vertical lines spaced 318mm apart. (12.5") The left line is 15mm (.6") tall. The right line is 165mm (6.5") tall. Now connect those two lines and you'll get a piece of plywood that looks like the one pictured above. You'll need two of these pieces, they form the sides of the waveguide.









Now for the top of the waveguide. Get a piece of plywood, and draw two vertical lines spaced 285mm apart. (11.2") The left line is 15mm (.6") tall. The right line is 335mm (13.2") tall. Now connect those two lines and you'll get a piece of plywood that looks like the one pictured above. You'll need two of these pieces, they form the tops of the waveguide.

Ideally you'd make all your cuts at a 45 degree angle, so the parts fit together tightly and all the angles come out right. But tbh, if you do it with a hand saw it won't be the end of the world, because it's the throat that really matters and we'll make a perfectly circular throat using a 1" hole saw.


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## seafish (Aug 1, 2012)

I keep thinking that you could slip that large a*s horn right into the dash where the HU is if you could relocate the HU into a center console or somewhere else?


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## thehatedguy (May 4, 2007)

If your upmixer has some signal steering, then anything below the XO point for the center will be kicked out to the fronts where you'll have more room for midbasses. The center is going to have a lot of MB information... probably too much for only the center to do.


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## Patrick Bateman (Sep 11, 2006)

thehatedguy said:


> If your upmixer has some signal steering, then anything below the XO point for the center will be kicked out to the fronts where you'll have more room for midbasses. The center is going to have a lot of MB information... probably too much for only the center to do.


I'm doing the upmixing via MiniDSP, so I have total control over the upmix.

Details on the previous page.

In a nutshell, I wouldn't "steer" so much bass to the center channel in a large room, but in my car, I have serious issues trying to get bass out of the left and the right channel. And they're already quite close to the center. So there will be a fair amount of midbass going to the center channel.


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## Patrick Bateman (Sep 11, 2006)

Bad news, the plywood waveguide didn't work out. I built it this morning, and it actually looked ****tier than the Pyle waveguide it replaced.

So I guess I'll stick with 3D printed waveguides, or waveguides from real companies. *My plywood waveguides look like ****.*





































Since I couldn't make room for a three way waveguide on the dash, I'll stick the midbass in the 'stock location for a center channel, and drop my Synergy Horn on top.


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## Izay123 (Jun 9, 2009)

Patrick Bateman said:


> Bad news, the plywood waveguide didn't work out. I built it this morning, and it actually looked ****tier than the Pyle waveguide it replaced.
> 
> So I guess I'll stick with 3D printed waveguides, or waveguides from real companies. *My plywood waveguides look like ****.*
> 
> ...




Have you thought of trying out a tweeter/midrange combo that ALREADY come with engineered waveguides? I love how my Boston SPZs sounded in the BMW low/forward in the doors --& the one I'm using as a center for my current car sounds great--even with factory amplification. I'm also running the passives --& I think i will continue to do so-Just biamped-because they're 24db slopes with The Q designed specifically for this (coaxial) alignment. If you want serious output, either of the sets below will GET DOWN with about 500watts RMS to each mid/150 per tweet. If you want more midbass, I wonder if you could PORT/build a TH for one in the dash...


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## grinkeeper (Jun 26, 2015)

first page has a talk about the radial port. I have a set of polk speakers with the radial port design and they worked really well. I thought it was always a great idea but had no idea how to design such a thing.


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## Patrick Bateman (Sep 11, 2006)

Well that sucked. I invested about twenty hours over the last ten days on this project, and when I hooked it up, there were a couple of huge issues:

1) I could hear a 'cracking' sound coming from one of the midranges. I think I may have cracked one of the woofer frames when I tightened the screws. Those tiny little midranges are really susceptible to this.
2) The soundstage was really forward. Although the tweeter is only six inches closer to the driver than when it's in the corners, it sounded jarringly 'in your face.'


BUMMER


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## grinkeeper (Jun 26, 2015)

I’m sure your research and efforts won’t let this die


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## Bnlcmbcar (Aug 23, 2016)

I have a center speaker idea and wouldn't mind going half on a set with someone since only 1 woofer, 1 Tweeter, and 1 crossover is needed.

So far I'm thinking of running a Morel Virtus Nano 602 passive off of 1 channel from Helix DSP Pro (70hz to 20khz)

It is 6-3/4" component speaker system with a woofer mounting depth of only 11/16" and a 1-1/8" Tweeter. (Bing has posted positive comments on their surprising ability for midbass given their shallow depth.)

I figure their super shallow depth would allow for an easier install in the center dash area than a smaller speaker with greater mounting depth. My hope is that they would also provide better midbass than smaller 3" or 4" drivers.

*Also if anyone wants more info or help on setting up rear surround channels this thread might help:

http://www.diymobileaudio.com/forum/technical-advanced-car-audio-discussion/307730-how-rear-fill-helix-dsp.html


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## rton20s (Feb 14, 2011)

Bnlcmbcar said:


> I have a center speaker idea and wouldn't mind going half on a set with someone since only 1 woofer, 1 Tweeter, and 1 crossover is needed.
> 
> So far I'm thinking of running a Morel Virtus Nano 602 passive off of 1 channel from Helix DSP Pro (70hz to 20khz)
> 
> ...


If you have enough channels on the DSP and with your amps to power them separately, you can purchase the shallow drivers individually. I'm not sure of a readily available source for the tweeter though.

https://www.madisoundspeakerstore.com/approx-6-7-woofers-morel/morel-powerslim-6-ultra-shallow-woofer-4-ohm/


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## Patrick Bateman (Sep 11, 2006)

I've been consistently stunned by how much output you can get from an AuraSound Whisper, so just for curiosity's sake, here's how EIGHT whispers compare to ONE of the Morels:

*name:* Morel PowerSlim 6 vs EIGHT AuraSound Whispers
*depth:* 0.56" vs 1.2"
*SD:* 141 vs (8 x 13.2)
*xmax:* 2.5mm vs 3mm
*displacement:* 35cc vs 32cc
*power handling (rms):* 80 vs 120
*price:* $166 vs $80



















It's such a strange driver. Basically the displacement is the same as the Morel, power handling is 50% higher, cost is half. As a bonus, you could arrange those eight drivers however you want.

Not saying the Morel is bad, they make great stuff, just saying the Whisper is strangely capable.


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## rton20s (Feb 14, 2011)

Patrick, why do 8 NS2 Whispers when you can do 10 Tectonic Elements TEBM35C10-4s for the same price?


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## Patrick Bateman (Sep 11, 2006)

rton20s said:


> Patrick, why do 8 NS2 Whispers when you can do 10 Tectonic Elements TEBM35C10-4s for the same price?


The 2" driver is going to lose because it's cone area is the same as the Whisper but it's xmax is lower.

But the 3.5" driver looks like a winner. If you use seven of them you get higher power handling than the Morel, higher displacement, and it's cheaper.

The Aurasound NS3 is an obvious option here, but it really doesn't work too well for ultra-shallow applications because the driver is so darn deep for it's size.

This is definitely some interesting food for thought. I had 8NDL51s under the seat of my Mazda, but they were always about 1/2" too deep to fit all the way under the seat. But it's really hard to find anything under 3" deep with a lot of output. I don't care if the driver is twelve inches wide, I just want it to be really really shallow. Most of the shallow mount tens and twelves are over 3" deep, so the obvious solution to get a lot of displacement with an ultra low height is to use an array. To get all of the sound to come from the sides of the car, for stage width, I use bandpass boxes.

*name:* Morel PowerSlim 6 vs SEVEN Tectonic Elements TEBM65C20F-8 3-1/2" BMR Full-Range Speaker 
*depth:* 0.56" vs 2"
*SD:* 141 vs (7 x 37.2)
*xmax:* 2.5mm vs 5mm
*displacement:* 35cc vs 130cc
*power handling (rms):* 80 vs 210
*price:* $166 vs $93


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## Bnlcmbcar (Aug 23, 2016)

rton20s said:


> If you have enough channels on the DSP and with your amps to power them separately, you can purchase the shallow drivers individually. I'm not sure of a readily available source for the tweeter though.
> 
> https://www.madisoundspeakerstore.com/approx-6-7-woofers-morel/morel-powerslim-6-ultra-shallow-woofer-4-ohm/


Only 1 channel left for center:

Active 3-Way Front - 6 channels

Passive 2-Way Rear fill - 2 channels

Subwoofer - 1 channel

I have a 2016 Civic Touring that has a 10 channel 5.1 DTS Neural head unit. I use 2 setups via my Helix DSP Pro.

1st bypasses head unit by connecting my iPad Mini to the digital coax input of DSP. This is for personal stereo SQ listening. Center channel is not activated and the wrest of the system is tuned for driver seat.

2nd is for when I have passengers. To do this I currently have 8 of 10 stock amp channel outputs connected to Helix DSP Pro speaker level inputs (missing 2 are the rear Tweeter signals since Helix only has 8 inputs). This maintains the stock DTS Neural for center channel steering and rear surround steering (plus goodies like Apple CarPlay and navigation).

So that's why I'm looking for a 1 channel passive solution for my center


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## Patrick Bateman (Sep 11, 2006)

@barasingha asked me how to do a Dolby Prologic II Matrix using MiniDSP. Here's the best that I could come up with, on short notice.









This is the Dolby PL II Matrix. The front channels are easy:

1) The left channel is untouched. It's exactly the same as you have in a 'regular' stereo setup.
2) The right channel is untouched. It's exactly the same as you have in a 'regular' stereo setup.
3) The center channel is mono, _with a caveat._ The left signal and the right signal are combined to make a mono center channel, *but the input is attenuated to 0.7071% of the original.* I don't think that Jim Fosgate used the number "0.7071" arbitrarily. If you draw a right triangle, the hypotenuse is 1.4142 as large as the other two sides. So methinks Jim Fosgate set the level of the center channel using the good ol' Pythagorean Theorem. (Understand that when you take two signals that are 0.7071% of their original volume, they will combine to yield an output that's 1.4142% as loud.)









Here is the default routing for the sound in a MiniDSP using the 2x4 Advanced plugin. This will give you two left outputs, and two right outputs.









If you push a few buttons, you get this. Now we have the following:
1) one left output
2) one right output
3) two mono outputs

Boom! You're done. That's all you need to do a three channel ProLogic II setup with MiniDSP. (OK, admittedly you have to attenuate the output of the center by 29.3%)

If you really want to kick it up a notch, add in DSP delay! You can use DSP delay to line up the wavefronts, so that you have a center channel that's locked to the center of your dash, along with a right channel that's nice and wide.


*You could stop at this point and you'd have a really nice three channel setup.*

But I'm bonkers, so let's take it a little further...


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## Patrick Bateman (Sep 11, 2006)

Doing the rear channels is way trickier.

The most elegant way to do it is with MiniDSP. The back left channel is .8717% of the left channel, mixed with 0.4898% of the right. The left signal is inverted before it's mixed with the right.

All of this can be done with MiniDSP. I do not see any way to invert the left channel in MiniDSP, so you would have to invert it before you feed it into your MiniDSP.

This inversion creates a bevy of problems:

1) You now need a second MiniDSP, that's $80
2) If you do the inversion using a cable, I believe that has the potential to create issues with your DSPs and your amplifiers. *Someone with a better understanding of electronics would need to chime in here.* I know from personal experience that splitting and combining RCA cables passively can lead to a bevy of issues. Again, I'm no EE, but I understand that electronics are expecting to see a certain input and output impedance, and splitting and inverting things passively is just begging for trouble.

If it were me, there are three simple ways to do the rear channels:

1) I think the simplest and most elegant way to get an inverted signal is to simply use two loudspeakers at each of the rear surround locations. 









Companies have been doing this forever. In Home Theater, you're typically doing the processing at the receiver. But if the head unit in your car can't do PL II, there's nothing stopping you from doing the processing for the rears AT THE SPEAKER. In this case, one speaker gets .8717% of the signal, the other speaker gets .4898% of the signal, and the former is inverted. The net effect is that the two drivers are out of phase at low frequencies. This is important, because it keeps the sound from 'wandering' to the back of the car. In the car, the speaker wouldn't look like this; you'd just create a plate and put it where the stock location for the rear speakers are. There would be two drivers on the plate, and the processing for the rears would be done with resistors and an inversion on one of the speakers. 

The second way of doing it is to wire the rear speakers like this:

https://cdn.instructables.com/FSE/V48J/FU0HZ8CM/FSEV48JFU0HZ8CM.MEDIUM.jpg

I don't recommend doing this. Many amplifiers will blow up if you bridge them. Don't do this wiring unless you know your amplifier well.


The third way of doing it is with "mid side processing." (https://theproaudiofiles.com/mid-side-processing/)

Mid-side processing makes my head hurt, but it's basically a way of creating a stereo signal using the following two signals:

1) mono (L+R)
2) L-R

Read the article for more info. In a nutshell, it is possible to create the following signals from "mono" and "L-R":
1) Left channel
2) Right channel
3) mono channel

In other words, you can create five channels from two channels, if those two channels are "mono" and "L-R".


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## Jscoyne2 (Oct 29, 2014)

Patrick Bateman said:


> @barasingha asked me how to do a Dolby Prologic II Matrix using MiniDSP. Here's the best that I could come up with, on short notice.
> 
> 
> 
> ...


Instead of using 2 channels for mono, one could simply use a single channel as mono and a y splitter at the amp. yea?

I also dont understand the statement " the left is inverted when its mixed with the right" when your table says there is a 90 degree phase shift. I thought invert was 180 degrees?


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## Patrick Bateman (Sep 11, 2006)

Jscoyne2 said:


> Instead of using 2 channels for mono, one could simply use a single channel as mono and a y splitter at the amp. yea?


Someone with an EE would have to chime in here.

As I understand it, *any kind of passive splitting or summing of the RCA cables is begging for trouble.*

For instance, simply taking one mono output from the MiniDSP and splitting it at the input of your amplifier is going to lower the input level to your amplifier. This could raise your noise floor, which will really suck for your center channel. You will also need to get out a microphone to set the SPL levels to compensate for the voltage drop. Obviously, if you know what you're doing, no worries. But splitting introduces a bunch of variables that can screw things up.

I found this out the hard way when i first tried to do a three channel set up in the front of my car. The obvious solution was to make an RCA cable that provides a mono channel, a left channel, and a right channel. Everything I've read indicates that this can blowup your amp and your source. An EE would know better than me though.

Until someone can explain how to do this properly, I'm going to continue using MiniDSP to do the splitting and summing. 

I believe you'll be alright if your head unit has front and rear outputs that are electrically isolated. The 'trick' though, is that you're never 100% sure that they're isolated. (I would imagine that any quality head unit would be.)

This article explains it better than I can:

_"Have you ever used a stereo-to-mono “Y-cable” or TRS (Tip/Ring/Sleeve)-to-TS(Tip/Sleeve) cable for combining two audio outputs together, or for summing the Left and Right channels from a single stereo output to mono? For example, maybe you wanted to connect the stereo outputs of your computer, CD player, iPod, iPad, Android tablet or phone into a single 1/4″ input channel on an audio mixer. Or maybe you needed to sum a stereo signal to mono for connecting to a single subwoofer. Or maybe you were mixing a song in your home recording studio and needed to check your mixes in mono on a single “grot box”, like the Auratone or one of its clones (Avantone Mix Cube, Behringer Behritone C5A, C50A, etc.). If you’re anything like me, you’ve probably done this, but what you may not know is this: a Y-cable or stereo-to-mono cable used to SPLIT a signal into two outputs is being used properly. A Y-cable used to MIX or COMBINE two signals into one input is being abused, and may even damage your equipment!

Here’s the rule: outputs are low impedance, and must only be connected to high impedance inputs. You should never tie two audio outputs directly together! If you do, each output tries to back-feed into the other, and drive the very low impedance of the other output, forcing both outputs into current-limit. At best, this can cause signal loss, audible distortion (popping and clicking sounds), and weird phasing effects. At worst, over time it can actually cause damage to your equipment.

I learned this the hard way when gigging with my band. We were playing at a large outdoor picnic and using our own PA system. We normally run some backing tracks (mostly synth pads) off of one of our band member’s iPhone. In this case, we were using an Android tablet instead. We connected it up to the system using a HOSA CMP-105 cable. I had used this same cable in the past with a couple of other devices without any trouble. But when we started the tracks this time, we were getting constant pops and clicks, the sound was distorted, and everything had a weird phasey sound to it (think phaser pedal for an electric guitar). We fooled with it for a few minutes, but never could find the problem, so we ultimately had to abandon using those backing tracks for the gig.

I later did some research and discovered that what I needed (for the reasons listed above) was a stereo-to-mono summing cable with resistors to prevent the outputs from back-feeding into each other. I began the search, but lo and behold–I could find no such cable readily-available on the market. It’s a widely-known problem, but for some reason no one has ever addressed it with a ready-made cable. There are cables on the market that *claim* to do it, including the Hosa CMP-103, CMP-105, and CMP-110. But none of them do it *properly*. These cables simply short the two outputs (tip/ring or left/right) together, and do not utilize any resistors. This works ok on some devices, but doesn’t work at all for others. And given enough time, even the outputs on devices where it *appears* to work could eventually be burned out. When I couldn’t find a ready-made cable to do this, I decided to build my own."_

The Stereo-to-Mono Summing Cable That No One Makes | Late Reflections – the Silent Sky Studios Blog


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## Jscoyne2 (Oct 29, 2014)

Patrick Bateman said:


> Someone with an EE would have to chime in here.
> 
> As I understand it, *any kind of passive splitting or summing of the RCA cables is begging for trouble.*
> 
> ...


Why Not Wye? Your answer is here

Im referring to sending a mono signal thru one rca and then splitting it with one of these. Its used commonly in bridging amps.


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## Patrick Bateman (Sep 11, 2006)

Yes, that will lower the voltage seen by the input of the amplifiers. This could introduce noise into the system.

It's particularly problematic with MiniDSP, because MiniDSP has 2V outputs IIRC, and is particularly prone to noise.

Back in the day, Richard Clark used a bunch of DIY devices to raise the voltage level on the outputs of the head unit, for this very reason. (He had four amps, which was quite a lot for that era.)


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## Barasingha (Sep 26, 2017)

Thanks for all the Dolby insights.

One question though: How can the center channel be 1.4x louder than the left and right, when theaters are calibrated for equal sound level from all channels?

Does the 0.7071 attenuation of each channel do something besides making a mono signal? 

Barasingha


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## Patrick Bateman (Sep 11, 2006)

Barasingha said:


> Thanks for all the Dolby insights.
> 
> One question though: How can the center channel be 1.4x louder than the left and right, when theaters are calibrated for equal sound level from all channels?
> 
> ...


Theaters don't use Dolby PL II. Dolby PL II is designed to 'upmix' from stereo to three channel.

The reason why Jim Fosgate opted to make the center 41.4% louder is open to debate, but it appears to be based on the Pythagorean theorem. Most stereo receivers give you the option to make the center louder or quieter so you might tweak the output level to taste.

Note than an increase of 41.4% is just *barely* audible. And increase of 100% yields an increase of 3dB. 

I'd argue that most people wouldn't be able to perceive a difference of 1dB. In the car, we're typically listening at levels of 90-100dB.


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## 156546 (Feb 10, 2017)

Patrick Bateman said:


> @barasingha asked me how to do a Dolby Prologic II Matrix using MiniDSP. Here's the best that I could come up with, on short notice.
> 
> 
> 
> ...


Well, almost. What isn't indicated completely in the wikipedia documentation of any of these matrix processors is the steering angle calculator and channel level control adjustments that happen on the fly thousands of times per second according to the computed angle. 

THIS DOES NOT WORK.


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## rton20s (Feb 14, 2011)

GotFrogs said:


> Well, almost. What isn't indicated completely in the wikipedia documentation of any of these matrix processors is the steering angle calculator and channel level control adjustments that happen on the fly thousands of times per second according to the computed angle.
> 
> THIS DOES NOT WORK.


So, the bottom line is...

If you want to run a center channel, pick up an Alpine H800 (or discontinued Alpine processor), a discontinued JBL MS8, or break into Audiofrog while Andy is at home on his fancy new toilet seat and swipe a prototype Multiseat upmixing processor?


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## brumledb (Feb 2, 2015)

Or find an old school Rockford RFQ5000. 


Sent from my iPhone using Tapatalk


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## 156546 (Feb 10, 2017)

Patrick Bateman said:


> Yes, that will lower the voltage seen by the input of the amplifiers. This could introduce noise into the system.
> 
> It's particularly problematic with MiniDSP, because MiniDSP has 2V outputs IIRC, and is particularly prone to noise.
> 
> Back in the day, Richard Clark used a bunch of DIY devices to raise the voltage level on the outputs of the head unit, for this very reason. (He had four amps, which was quite a lot for that era.)


Voltage is constant in a parallel circuit. Adding a splitter doesn't reduce the level until there are so many amplifier input stages in parallel that the output of the preceding piece of equipment cannot supply enough current to drive the load. That's why the preamp input impedance is usually onthe order of 1k -10k--so very little current is required.

Additionally, .707 is a 3dB decrease from 1. We use the change in voltage formula for that: 20log (v1/v2) So, in the dolby matrix above, both the right and the left channel are attenuated by 3dB at the input to the sum and the result of the sum is +3dB for the mono center signal.


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## 156546 (Feb 10, 2017)

Barasingha said:


> Thanks for all the Dolby insights.
> 
> One question though: How can the center channel be 1.4x louder than the left and right, when theaters are calibrated for equal sound level from all channels?
> 
> ...


Theaters don't use upmixing. They use discrete surround. Big difference. Discrete surround is 6 individual streams that are decoded in the processor. That's what Dolby Digital is. The person mixing the movie can place sounds discretely in the channels or can mix sounds across several channels to create movement of the sound or to place the sound in between speakers.

The stereo track in a movie (the one that would have been on a VHS tape) doesn't include the possibility for 5 individual streams. So, movies mixed for playback over an upmixer include the surround channel out of phase. The PL2 upmixer "decodes" that out of phase signal as rear. 

When music is mixed for 2-channel, the guy doing the mix isn't often thinking about playback over an upmixer. Consequently, what happens in the rear speakers is sort of catch as catch can. This can produce unwanted steering of sounds. 

For PL2, L7 and other matrix upmixers, there are mixes that break the code--a person speaking in the center and a steady state surround signal breaks the code. 

Here's why (as I mentioned in a previous post). PL2 and L7 also include a signal steering matrix. This is the secret sauce and isn't usually documented in simple explanations of the matrix. Look for the patents and you'll see it. It wasn't included in the analog versions of upmixing because it couldn't be. DSP made this possible. 

The algorithm analyzes the level and the phase in the right and left channels. It then computes a steering angle for the sound a few thousand times per second. When a signal is computed to be mostly mono, the algorithm computes a steering angle that's front center. The rear speakers are attenuated. the right and left speakers are attenuated. If a signal is computed to steer front right, then the rear speakers are attenuated. The left speaker doesn't play, and the center speaker is attenuated. The opposite is true for signals that steer left.

In some versions of these algorithms, the left (or right) rear channel polarity is reversed for signals steered to the front. 

This was developed to mitigate the stage narrowing effect of the summed center signal. 

And this is why these things can't resolve the signal I mentioned above. When a person is speaking in the front center, the ENTIRE signal is steered center. That kills the steady state surround signal to the rear.

In a movie, that signal might be someone speaking during a rainstorm. If one were mixing the movie for playback over PL2, then one would reduce the level of the rain shower that was mixed in out of phase to mitigate this problem. It's somewhat convenient to do so because the speech would be the focus anyway. 

In discrete surround, this wouldn't be a problem.


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## 156546 (Feb 10, 2017)

Here:

https://encrypted.google.com/patents/EP1387601A2?cl=en


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## Patrick Bateman (Sep 11, 2006)

GotFrogs said:


> Voltage is constant in a parallel circuit. Adding a splitter doesn't reduce the level until there are so many amplifier input stages in parallel that the output of the preceding piece of equipment cannot supply enough current to drive the load. That's why the preamp input impedance is usually onthe order of 1k -10k--so very little current is required.
> 
> Additionally, .707 is a 3dB decrease from 1. We use the change in voltage formula for that: 20log (v1/v2) So, in the dolby matrix above, both the right and the left channel are attenuated by 3dB at the input to the sum and the result of the sum is +3dB for the mono center signal.


I knew an EE would answer this. Thanks Andy!


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## Patrick Bateman (Sep 11, 2006)

GotFrogs said:


> Well, almost. What isn't indicated completely in the wikipedia documentation of any of these matrix processors is the steering angle calculator and channel level control adjustments that happen on the fly thousands of times per second according to the computed angle.
> 
> THIS DOES NOT WORK.


Well that's a ****ing bummer 

I studied this yesterday and came to a couple conclusions:

1) Older documentation seems to indicate that there was no steering the the front three channels of Dolby Prologic II. Everything that I found indicates that there IS steering in the rear channels. I'm not expert on this (Andy is), but this seems to make sense.

Here's an example:

In the song "Mercy" by Kanye West, nearly the entire song is out-of-phase. This creates a big wide soundstage, the effect was definitely intentional. If you ran that song through a phase inverter, you're going to wind up with a mono signal (because the original track is out-of-phase, by design.) So if you played the song through an upmixer that inverts phase, you'll wind up with a front stage that's out-of-phase, and a mono signal in the rear channels. (Because the upmixer is flipping the polarity.)

Clearly, this is not A Good Thing. This illustrates the need for some type of steering.

I've never noticed any artifacts like this in my Genesis, which uses the "Logic7" processing from Lexicon/Harman. But the center channel really dominates in that system, the other four channels are barely audible.

2) I reviewed some of the newer documentation from Dolby, and it talks a lot about Dolby Prologic II steering, but very little discussion of how it's done. The documentation seems to imply that the spec has evolved over the years.


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## Patrick Bateman (Sep 11, 2006)

Here's some discussion of ProLogic II steering from it's inventor, Jim Fosgate. This is from over 20 years ago, off of Usenet:

_""For years I have dreamed of designing the perfect Matrix surround
processor (final solution to this technology). I remember thinking when
I started, that I could conquer it in a few months. Boy was I wrong,
been working close to 25 years now. It turned out to be one of the
biggest design challenges in the audio business. No doubt about it,
this technology has humbled me many times. I think it represents one of
the biggest design challenges in audio."

"Like other designers I started with the feedforward concept that Mr.
Schieber patented in the late 60s. In fact I worked with Peter for a
number of years. There were many problems to solve in those early
circuits to get close to the ideal decoder. To obtain good separation
the steering logic must respond quickly to changes in dominant signal
direction. But it must also be very precise and smooth. Fast and smooth
are both opposites. To sound smooth the logic must be running slower.
Seemed like one was trying to defy the laws of physics. You couldn’t
seem to get both. Designers were faced with serious tradeoffs. The
logic had to be slow to sound smooth and anomaly free, but then there
was not enough speed to get the separation required to make it sound
discreet. Quite a juggling act for sure."

"As time went on additional circuits (band aids) were added to fix the
problem, variable speed ETC. The idea was to let the logic run faster
or slower depending on the demands of the program. The real challenge
was to control these circuits, not possible to get perfect control.
Different types of audio material have different dynamics, classical is
slower, rock faster, and electronic and choral works must be very
smooth. This is why several modes were required to accommodate the
different dynamics. The logic could be tailored to each type of music.
Problem was this requiring the user to know which mode to use. Much
better if one mode could play everything and sound good."

"Another problem was that test signals did not correlate well with real
audio signals. I found out on the earlier circuits I could go for best
performance on test tones or real audio, but not both. This problem had
to be fully understood to find the limitations."

"Here were the design goals for the new PL II technology:

1-Symultanious discreet-like separation close to an AC3 and DTS digital
system

2-Logic system must play everything without requiring the user to
switch modes

3-Large open spacious sound field where the speaker locations seem to
disappear

4-Pin Point imaging

5-Play normal stereo material

6-Play surround encoded material

7-No detectable circuit action or anomalies

8-Simple less expensive circuit

9-More accurate feedback steering logic concept that could be patented

10-Audofile sound qualities

"As you can imagine these features have never been possible in a matrix
decoder before. I determined it could only happen with a breakthrough
concept capable of improving the circuit accuracy and dynamic
performance by a factor of at least 10 of any previous technology. I
figured if feedback could reduce the distortion in an amplifier it
could be used around the steering logic to reduce the circuit errors
(distortions)."

"My 6 axis design has been in the marketplace for years now. I guess it
has become sort of a standard as far as matrix decoders go so it will
be useful to use it as a comparison to the new PL II. I was very happy
with that design and felt it was clearly superior to other technologies
at the time. One test I use is to A-B the sound of the decoder against
regular stereo. Ideally the front soundstage should be the same when
listening from the sweet spot. I usually perform this test with the
backs turned off."
"This test revealed that the front soundstage was not quite “right on”
with some material. It is interesting to note that almost any decoder
design can sound excellent on one piece of material and not so good on
others. The goal of course is to make it work well on everything. We
where choosing demo material carefully for 6 axis. I knew the
soundstage could be deeper, wider, more discreetly separated, and with
less audible decoding action, and better imaging. In short it needed to
be better all the way around, big bummer when I realized this. I kept
working on 6 axis well after it went into production and had it
sounding better, but not as good as I wanted. I finally came to the
conclusion that this old feed forward concept was as good as it would
ever get no matter how many circuit band aids I added. That is why I
started over with this new concept."
"How does PL II stack up against 6 axis? Well the difference isn’t
subtle at all! In the first few seconds you will be aware that this is
a completely different animal. It is wall to wall surround. (That is
what one of my audiophile friends called it the first time he heard it)
Through the years my family and friends have heard all my design
changes. Like many designers I would make improvements that seemed
pretty big to me, but my listening audience could not hear it. Not so
this time, everyone who has heard it has been at a loss for words. Even
the grandchildren can hear the improvement. (They can hear much better
than us older folks anyway) "

"The sound stage is very deep, wide, and imaging is tight and focused.
There is much better simultaneous separation, and ambiance recovery.
Can’t hear the circuit working at all. Real smoooooooth. Stereo decodes
so well one listener remarked that it sounded better than most of the
discreet digital mixes he had heard. One of the tests performed was to
take the 5 channel outputs from a Dolby AC3 decoder, encode that into
two channels, and feed that into the decoder. Next we compared the
output of the encoder/decoder to the original 5 channels using the A-B
switch. Mr. Roger Dressler from Dolby Labs was here in my sound room
when we did this. I will never forget what happened. We started playing
the first cut on the James Taylor Live At The Beacon Theater DVD. We
both sat down and listened for awhile than threw the switch which was
on a long cable from the switcher. Nothing happened, we switched it
back and forth several times and decided it was not working. We got up
started pulling cables and found to our amazement that the switch was
working, we just couldn’t tell position A from B. Extended listening
did reveal that they were some minor differences but it was very hard
to tell which was which. We did this same test with Movies. They were
also excellent. Some differences, but the essence of the performance is
there. Didn’t think a Matrix Decoder could ever do that."

"How does PL II compare to PL I? Big difference, on movie or surround
encoded material PL II is much closer to AC3 than PL I. Bigger, more
open, wider soundstage, simultaneously discreet, better focus, lots
more fun. On stereo there is bigger difference, PL I is not very happy
with most stereo recordings. PL I has center build up and loss of width
on most recordings when the center channel is on, PL II does not. This
is not a design fault with PL I; it just was not designed to play non-
encoded stereo material."

"How does PL II compare to Circle Surround? I do not believe in
negativity so I will try to keep things on the positive. The circle
unit has its place. It is low in cost and can sound very good on some
material. I think it compares to some of my designs before 6-axis. I
can hear problems on some recordings and it works pretty well on
others. If you like it enjoy it. Very big difference when compared to
PL II."

"How does PL II compare to Lexicon? Keep in mind this technology is a
feedforward concept like 6 axis, I think it was close to 6 axis but I
prefer the 6 axis myself, bet that’s no surprise to you. The lexicon
seems to loose logic steering sometimes on stereo material. Which
results in partial collapse of the soundstage. Dosent do this on
everything, seems to work better on encoded material. We did AB
comparison on several stereo test cuts and thought the difference was
greater than when comparing PL II to 6 axis. Here again if you like it
enjoy it. I think they have all ways been my number one competitors."

"How close have I come to my design goals? Pretty close, in fact I
think it sounds better than I originally though possible. PL II will
loose a contest for pure separation compared to a discreet AC3 or DTS
system, but not by as much as you might think. On the other side of the
coin the soundfield is sometimes larger with the PL II, depends on the
mix. It is lots of fun for sure, and that’s the whole idea. PL II is
not meant to compete with the AC3s and DTSs of the world but rather to
compliment them. It allows one to enjoy excellent surround from any two-
channel source, even recordings made as far back as 1957. It’s really
fun to hear some of the old classics in surround. Did you know that
stereo was a compromise from day one? The early work done by Western
Electric and others indicated that at least three channels were
required to reproduce a correct stereo soundstage. Unfortunately two
channels were all that they could get on a stereo LP, so that’s what we
got. Now one can hear stereo the way it should be heard, with three
front channels. Some of the early audio pioneers played with wide-stage
stereo with added center channel. All these concepts had problems
because when a center channel is summed from the left and right it
narrows the soundstage and interferes with the imaging. It takes a
state of the art high separation decoder to do it right."
------------------
Jim Fosgate_


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## Patrick Bateman (Sep 11, 2006)

That last post might be confusing, because Jim Fosgate compares "Pro Logic II" to "Circle Surround."

If I'm not mistaken, Jim Fosgate made those comments before Dolby licensed his technology. All of this gets super confusing, Jim is 72 years old and I've only seen one article that documents the history of these things.

But as I understand it, Harman acquired one of Jim's companies in the 80s, and Jim became a Harman employee. At some point he was terminated, but he was able to take his work with him, which he later licensed to Dolby. Harman went with Logic7. That's what's in my car. Andy used to work for JBL, which Harman owns. (Does MS-8 use Logic7?) Harman is now owned by Samsung. Jim Fosgate was also a consultant at Rockford Fosgate during the 80s, which is one of the reasons that RF has offered various upmixers over the past thirty years. (Rockford Corporation - Company Profile, Information, Business Description, History, Background Information on Rockford Corporation)

Read more here: History of Surround Sound Processing: The Battle for Dolby Pro Logic II | Audioholics

While all of this stuff might sound academic, I think it's worth serious study. I'd be hard pressed to tell the difference in sound between two amplifiers, but surround upmixers have an unmistakable signature. The Penteo stuff sounds pretty darn good and I'm excite to see when AudioFrog releases it. You can try it yourself by purchasing the software and doing the upmix on your PC and encoding the file in six or eight channels. (AAC supports multichannel, and you can play AAC on any ol' Windows tablet, even an $80 one.)

Aren't patents fun? Half of the **** I know about audio is from reverse engineering patents.


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## Patrick Bateman (Sep 11, 2006)

As Erik noted here(1), and as Andy Wehmeyer noted on the previous page, *the solution that I proposed with MiniDSP is fairly useless without steering.*

There seems to be a zillion ways to do this, and all of them have various issues:

1) There are processors you can buy on Amazon or eBay. It's really difficult to ascertain if they're legitimately doing Dolby Prologic II. (IE, are they using legitimate licensed silicon, or did they just slap a fake DPL II logo on their box, with their own processing?)

2) You can get legitimate Dolby PL II processing in various HT receivers. But a lot of us are doing our own DSP and processing, so we don't WANT to use a home theater receiver. That's the situation I'm in. *I want to use MiniDSP and a HTPC source.* That means a HT receiver does me no good.

3) PowerDVD and various other software players can do the decoding. But then you have to spend $80 and you're tied to their software. This is a p.i.t.a. if you're doing a HTPC or you want to do Dolby PL II in a car. (I don't want to deal with the PowerDVD interface on my Windows tablet, the screen is too small.)


Here's a method which I think is (fairly) elegant:

Step 1: download foobar 2000 (foobar2000)
Step 2: download the "freesurround" plugin (foo_dsp_fsurround)
Step 3: download ffmpeg (Builds - Zeranoe FFmpeg)
Step 4: load your song or songs into Foobar 2000, and configure the input and output plugins accordingly. I am using "wav" for the output format, and I am using the "FreeSurround" DSP processing.
Step 5: Once you have Foobar2000 convert your track or tracks, you will have a six channel wav file for each song.
Step 6: If you have a HTPC, you might just play those WAVs. I have a tablet with limited space, so I convert the six channel WAV to a six channel AAC file. AAC supports something like a hundred channels, so you can go crazy with the surround options. Here's the command line:

ffmpeg -i [wav file] -c:a aac -b:a [bit rate] [aac file]

for instance:

ffmpeg -i Mercy.wav -c:a aac -b:a 384k mercy.aac

My six channel WAV was 176MB, and encoding in AAC at 384K reduced the size to 16MB.

I used "Mercy" by Kanye West for my testing. A great deal of the song is out-of-phase, and the vocals float all over the stage. So it's a bit of a torture test for an upmixer.









Here's what the stereo track looks like in Audacity. Your basic pop track with maximum loudness.









Here's the track after processing by FreeSurround. You six distinct channels. Most impressively, I listened to the isolated surround channels and the steering did a (fairly) excellent job of keeping vocals out of the surrounds. Andy or Erik would have to chime in on how the steering works, but listening to the surrounds by themselves reminded me a bit of a noise gate. The steering seems to 'detect' when there's something in the surrounds that belongs in the mains, and it seems to attenuate that dynamically. If you look at the content in the surround, and you see those big spikes, that was when vocals wound up back there, and the steering quickly attenuated it. (As noted earlier in this thread, the song "Mercy" is largely out of phase, so an upmixer WITHOUT steering would send the vocals to the back channels. This is because an unsteered upmixer does a phase reversal. So the out-of-phase lyrics on the original song would be inverted into mono and send to the surrounds. Again, a bit of a 'torture test' for an upmixer.

Upmixing really seems to be an art, so if any of you have messed around with Penteo surround or the like, chime in with your opinions.

(1) http://www.diyaudio.com/forums/digital-line-level/308266-diy-prologic-ii-1


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## Patrick Bateman (Sep 11, 2006)

I did another test. In this test, I created a stereo file with pink noise. The pink noise is slooowly faded from one channel to the other. You can see this on the clip on the left. _See how the sound starts on one side of the stage then fades to the other?[/b]

Then I converted it to five channel with FreeSurround. It seems to be doing it's job. In the FreeSurround clip, you can see the following:

1) The sound starts on the left
2) Then it moves to the center channel
3) Then it moves to the right
4) There's nothing sent to the surround
5) There's some LFE

Seems to be doing it's job._


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## Mike-G (Dec 25, 2008)

Patrick Bateman said:


> In the song "Mercy" by Kanye West,
> Clearly, this is not A Good Thing.


:laugh:Sorry, couldn't help myself


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## 156546 (Feb 10, 2017)

These matrix processors were designed to decode a movie soundtrack that was "encoded" into 2-channel signal. 

The example of the Kanye West track is instructive. Mixing methods designed for 2-channel to increase stage width are NOT intended to be decoded. In PL2 or L7, the out of phase components would be steered rear. 

So, it's useful to categorize upmixers into two groups: those designed for movies and adapted to music (PL2, L7, DTS Neural) and those designed for music playback (Circle Surround, Audiofrog Multiseat, Harman QLS...who knows if there are others).

Penteo was designed as an upmixer for studios. Its primary use is in converting a 2-channel signal into multichannel SO THE RESULT CAN BE INCLUDED IN THE DISCRETE SURROUND MIX. The primary consideration for an upmixer designed for that duty is that it should be perfectly down-mixable back to the original stereo. Matrix processors CANNOT do this. The version in Audiofrog Multiseat (and what will also be Perfect Surround Automotive) has a lot of changes required for great performance in AUTOMOBILES based on available speaker locations and sizes, consumer expectations and for simple tuning.


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## cyanogen (Oct 24, 2017)

Due to the current lack of 7.1 processors designed for the car, I'm considering picking up an Outlaw Audio 975 and either attempting a 12V conversion or running it on an inverter. Besides having the ability to decode various multichannel formats, it also supports a bunch of upmix algorithms including Dolby IIx and IIz, and DTS Neo:6. Has a smallish footprint, it's highly configurable, and the price is right too. 

Anyone tried one of these? What could possibly go wrong?!


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## Bnlcmbcar (Aug 23, 2016)

I’ve been considering running the Outlaw unit as well. One could possibly fit it in a passenger glove compartment and power it with a small pure sine wave inverter like the Samlex PST 120.

120W 12VDC to 120VAC Pure Sine Wave Power Inverter | PST-120-12

I could very much be wrong but I believe one of Andy’s prototype DSP’s was implementing a version of the DTS Neural up-mixer. My 2016 Civic stock headunit has DTS Neural processing to 10 active channels that I have been experimenting with channeling and EQing them through a Helix DSP Pro. I can achieve a decent multiseat listening experience but I am limited in the tuning capabilities since the headunit signal already has crossovers and time delays set for the channels.

The Outlaw 975 is appealing to get better ‘steered’ channels to tune but as stated earlier all those processing formats were meant for movies and not music. I’m curious to know the differences though between the different processing that happens between DTS Neural and DTS Neo if any?

I’m thinking/hoping that Outlaw 975 processed channels ran through a multi channel DSP for fine tuning could yield some decent results for a multi seat listening experience.

I also remember reading on one of the threads about a user using the Surround Master unit from Involve Audio... wonder if it worked out?


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## cyanogen (Oct 24, 2017)

Bnlcmbcar said:


> I’ve been considering running the Outlaw unit as well. One could possibly fit it in a passenger glove compartment and power it with a small pure sine wave inverter like the Samlex PST 120.
> 
> The Outlaw 975 is appealing to get better ‘steered’ channels to tune but as stated earlier all those processing formats were meant for movies and not music. I’m curious to know the differences though between the different processing that happens between DTS Neural and DTS Neo if any?
> 
> ...


Nice thing about the Outlaw is that it supports a bunch of formats, and the corresponding music modes as well. My plan is to feed a Mosconi 8to12 with it.


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## Bnlcmbcar (Aug 23, 2016)

Many times I find myself willing to sacrifice stereo width for a compromised listening experience that my passenger can enjoy as well.

I thinks that’s the only way to ‘currently’ achieve a more audiophile grade surround sound processing for multi seat listening (outside of a CarPC or the aged dedicated 12volt solutions of course). It will take a bunch of fine tuning on the DSP end but certainly possible in my opinion.

At least until the Audiofrog unit comes to fruition.


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## cyanogen (Oct 24, 2017)

I pulled the trigger and bought a used Outlaw off someone on eBay. The system that's being put into my car right now is basically designed for 7.1, with center, side coaxes, and rear hatch surrounds. I'm running full active 3-way fronts, so if I feed the DSP with the Outlaw it will use all inputs and outputs perfectly. I'm guessing that not a whole lot of TA is needed if surround processing is running.. Or should everything be time aligned to the center between the driver and passenger with surround processing bypassed first? This will get interesting.


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## SPLEclipse (Aug 17, 2012)

I guess that depends on what kind of tuning you can do with the Outlaw. You might want to use a combination of crossover/TA settings on both units.


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## Patrick Bateman (Sep 11, 2006)

cyanogen said:


> I pulled the trigger and bought a used Outlaw off someone on eBay. The system that's being put into my car right now is basically designed for 7.1, with center, side coaxes, and rear hatch surrounds. I'm running full active 3-way fronts, so if I feed the DSP with the Outlaw it will use all inputs and outputs perfectly. I'm guessing that not a whole lot of TA is needed if surround processing is running.. Or should everything be time aligned to the center between the driver and passenger with surround processing bypassed first? This will get interesting.


I would argue that DSP delay is even more important with a five channel setup.

You want to get your mic out and insure that every wavefront is reaching your ears at the same time. I've heard some systems where that's been done and it's pretty amazing.

Alternatively, if you're doing a two seat system, you'd want a balance between the left and right seats.


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## cyanogen (Oct 24, 2017)

Patrick Bateman said:


> I would argue that DSP delay is even more important with a five channel setup.
> 
> You want to get your mic out and insure that every wavefront is reaching your ears at the same time. I've heard some systems where that's been done and it's pretty amazing.
> 
> Alternatively, if you're doing a two seat system, you'd want a balance between the left and right seats.


Doing full the 7 channels actually. It's a Tesla Model X with 3 row seating, so we are putting good coaxials in the rear doors (side channels) and a pair of widebanders in the far back hatch area (rear surrounds). My hope is to get a good sounding stage in as much of the car as possible. The front 3-way components definitely need some offset relative to driver locations. Once I get the Outlaw, I'll run a bunch of tests to sort out what it's actually doing delay wise in IIx / Neo6 mode.


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## 156546 (Feb 10, 2017)

Bnlcmbcar said:


> Many times I find myself willing to sacrifice stereo width for a compromised listening experience that my passenger can enjoy as well.
> 
> I thinks that’s the only way to ‘currently’ achieve a more audiophile grade surround sound processing for multi seat listening (outside of a CarPC or the aged dedicated 12volt solutions of course). It will take a bunch of fine tuning on the DSP end but certainly possible in my opinion.
> 
> At least until the Audiofrog unit comes to fruition.


It isn't necessary to sacrifice width to get a center image in both seats.


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## Bnlcmbcar (Aug 23, 2016)

GotFrogs said:


> It isn't necessary to sacrifice width to get a center image in both seats.


Oh I believe you on that! I just haven’t been able to do so myself yet.

“Please understand that PLII doesn't affect the stereo soundstage, other than to display the in-phase part of the program over the three front channels—the out-of-phase or randomly phased signals are sent to the rear. When switching between stereo and PLII, you'll see that the stereo soundstage stays intact but has greater depth and width. Sometimes you're not even aware of the extra speakers, until you turn them off and the soundstage collapses back to stereo.
-Fosgate (2004 Stereophile Interview)

I want to obtain an upmixed tune like that and quality tuning requires dedicated effort, time, and dillegence. It’s safe to say there is not too many people who possess your level of skill and knowledge on this subject matter. I’m still on my long tuning journey. Mining the gold nuggets you and several others lay on this site. I already saw the general process for tuning you posted on another thread. I’m taking notes:smart:

Any chance your able to offer any insight or gist (protect those patents!) on how the surround upmix processing differs between DTS NEO and DTS Neural? Or the DTS/Dolby upmixers vs what being implemented in Audiofrog multi seat?


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## 156546 (Feb 10, 2017)

There are a bunch of differences. First, in the matrix upmixers, the center is a summed L and R and the steering angle computer turns the levels of the channels up and down according to the calculated vector to maintain stage width. Second, the rears are a L-R signal. 

In our Upmixer, the sound field in the stereo recording is separated into mono (same in L and R), Intermediate information (similar but not exactly the same in left and right--the sounds between the center and the left or the right) and differential information (only right and only left). The mono info is sent to the center. The intermediate and differential info is spread over the left and right, the sides and the rears--kind of like a horseshoe if you were to turn all of the dials to 11. 

This provides great stage width, a stable center and no artifacts in the sides and rears.


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## KillerBox (Jan 7, 2011)

I am subscribing but, I can't add much because you all are over my head. 

I will say this my MS-8 - 7.1 Logic 7 (that set up as close as possible to Andy's Mercedes) produces the most realistic live concert sound that I have ever experienced in a vehicle. 

For the first time in my life, I find myself searching for live recordings instead of studio recordings. I got goosebumps the other night listening to U2 Live. I like U2 but, not that much!

So if something else comes along better, I am 100% a buyer!


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## Bnlcmbcar (Aug 23, 2016)

GotFrogs said:


> There are a bunch of differences. First, in the matrix upmixers, the center is a summed L and R and the steering angle computer turns the levels of the channels up and down according to the calculated vector to maintain stage width. Second, the rears are a L-R signal.
> 
> In our Upmixer, the sound field in the stereo recording is separated into mono (same in L and R), Intermediate information (similar but not exactly the same in left and right--the sounds between the center and the left or the right) and differential information (only right and only left). The mono info is sent to the center. The intermediate and differential info is spread over the left and right, the sides and the rears--kind of like a horseshoe if you were to turn all of the dials to 11.
> 
> This provides great stage width, a stable center and no artifacts in the sides and rears.


Wow that sounds (pun intended) awesome!

If I am understanding it correct, it will still be a 5.1 or 5.2 form factor but your route is going more along the lines of widening the front stage and ‘wrapping’ it around the listener from left to right vs the other upmixers trying to place the listener in between a front and back stage?

Isolating that intermediate information must have been a painstaking process whether you achieved it in the analog domain or with digital wizardry. I’m excited. 

Icing on the cake is the elimination of those pesky artifacts.


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## 156546 (Feb 10, 2017)

Most of the matrix upmixers were originally designed to extract an encoded rear surround signal from a movie on a VHS tape. So, yes, they are designed to provide the ability of the movie remixer to place sounds in the back.


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## Patrick Bateman (Sep 11, 2006)

Hey Andy,

Isn't it too cost prohibitive to have your own chip manufactured? And if so, is this something that could be done in software and sold to miniDSP?

I'm picturing this:

1) you write the software in audiomulch
2) you sell it to miniDSP
3) you collect a royalty on the sales

Just an idea, because it seems like a DSP might be 'generic' enough to run your upmixer, and that simplifies the process of getting this to market


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## 156546 (Feb 10, 2017)

Patrick Bateman said:


> Hey Andy,
> 
> Isn't it too cost prohibitive to have your own chip manufactured? And if so, is this something that could be done in software and sold to miniDSP?
> 
> ...


The code is available for license.


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## Kevmoso (Jun 4, 2013)

GotFrogs said:


> The code is available for license.


I would like to purchase a license to run this 'upmixer' on the minidsp c-dsp 8x12 that I will buy at the same time.

Do hashtags work here to get manufacturers attention? 
#MINIDSP #MINIDSP #MINIDSP #MINIDSP #MINIDSP


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## Patrick Bateman (Sep 11, 2006)

Kevmoso said:


> I would like to purchase a license to run this 'upmixer' on the minidsp c-dsp 8x12 that I will buy at the same time.
> 
> Do hashtags work here to get manufacturers attention?
> #MINIDSP #MINIDSP #MINIDSP #MINIDSP #MINIDSP


You can contact Penteo here : Contact Us

If I'm not mistaken, you'd have to work with the folks from Penteo and MiniDSP to make this happen.

This wouldn't be the first time they made a plugin for us; years ago MiniDSP created a plugin for this forum. (Heck, it may even be this thread. I think Werewolf was involved?)

I tried to learn AudioMulch once and it was over my head. For me, it's easier to use a Windows tablet and the existing upmixers that are already available on Windows. (Here's some food for thought: http://www.diymobileaudio.com/forum/783898-post286.html )

Having said that, I've heard Penteo in Gary Summer's car and it's probably the best upmixer I've heard. (Gary made the mix on a computer and recorded it to multichannel DVD iirc.) Plus, he does mixing for a living so that sure helps


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## cyanogen (Oct 24, 2017)

Got my Outlaw but won't be able to try it out for a few weeks. On another note, holy schnikes does DTS Neural:X sound good on my new home setup!


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## Patrick Bateman (Sep 11, 2006)

It doesn't surprise me that Andy is so passionate about multichannel, when it's done properly it's a real problem solver. I can see how people love two-channel if they have the opportunity to sit equidistant between a set of speakers, but as soon as you have to listen off-axis, multichannel starts making a lot of sense.


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## Nismo (Jan 10, 2010)

Patrick,
You got me started down this wormhole...I was almost able to buy an RFQ-5000, but it fell through. I ended up with a Pioneer DEQ-P8000 and it's dash controller. I haven't installed this yet, but bought it specifically because of DPL II. I can't hang with the big boys in the $$ department, so I'm kinda stuck with what I can pick up used.

Eric


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## ultimatemj (Jan 15, 2009)

BUMP

Any progress made on this topic???

I recently acquired an E92 BMW that has the worst speaker placements for 2 seat stereo of any vehicle I've owned (PLD for LvR vocals is huge). That said, it is set up with a center, rear, and side surround....just needs a better multiseat signal.

The processing in the Enhanced Premium (S752) creates a high and tight center image, but the stage width is super narrow 

These BMWs need a proper multiseat DSP...

I *NEED* a proper multiseat DSP!


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## Zoom_M5 (Oct 28, 2017)

I got tired of waiting and recently scored a RFQ-5000. Today the UMI-1 arrived today (thanks Andy!) so I can test it. Looks like I'm still gonna be playing the waiting game on a pair of Arc PS8-50's or the Helix V-Twelve...


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## hokis78 (Jan 11, 2020)

I’m going to list most of the practical ways to amplify a home-theater system and rank them with ultimate sonic performance as the goal. Convenience, ease of use and décor will count for nothing in these rankings so they may move around considerably when you apply your own personal needs to the mix. Just bear in mind that this ranking considers only ultimate sound quality and nothing else. Cost is a big consideration in building a home theater, but cost is not a factor in the ratings for this table. Since the front three channels carry the most information, they are given more weight. The home-theater subwoofer is not part of this table, but it is an important element of the home-theater sound package.


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## hokis78 (Jan 11, 2020)

Sometimes specific products perform so well that they break out of the mold. The Belles 150A Hot Rod amp ($1499 stereo, $1999 three-channel) is an example…you’d be hard pressed to find five mono amps for anywhere close to $3500 that sound as good as a 3+2 Belles Hot Rod setup. Because of all of these issues, don’t consider this chart a bible for amplification selection in your home theater. There will always be exceptions to the rules laid out here. You might want to use the chart as a sounding board for potential choices you might need to make when designing or refitting your own home theater. 

You can check the latest best instant pot for the kitchen. Having superb quality. Check it’s reviews. Best instant pot


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## KillerBox (Jan 7, 2011)

Who is talking about a home theater?


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## Patrick Bateman (Sep 11, 2006)

Someone over on diyaudio sent me an email about upmixing.

At first, I was going to tell them the following:

Basically, upmixing is really challenging. I have an upmixer in my car, and I generally leave it on. But I'm not thrilled with how it sounds. Basically, it tends to narrow the soundstage, and you can get a convincing phantom center with carefully placed component speakers.

I think you can achieve a five channel soundstage using software, and it will exceed the quality of what my Lexicon Logic 7 is doing. For instance, Gary Summers has played me some tracks that he's upmixed on his computer, and they sound light years better than what my Logic 7 does.

A lot of the magic is in the design of the upmixer. Upmixing is a complex business. I am too lazy to dig into the patents right now, but as I understand it, some of the earliest upmixers were just adding a mono center channel, and the left and right channels were still stereo. So, as you can imagine, this will tend to 'pull' the image towards the center of the car. At the same time, it can be pleasant, because you can fit a really big speaker in the center of your dash.

If I recall correctly, the upmixer that Andy Wehmeyer was working on a few years ago uses the same software that Gary Summers used to do his five channel mixes. Those two know way, WAY more about upmixing than I do. Here's an article about the upmixing used in the movie "13 Hours", which Gary worked on.









Oscars: The '13 Hours' Post Sound Team Brings Authentic Intensity to Michael Bay’s Modern War Thriller - Sound & Picture


Michael Bay is best known for directing the Transformers film franchise. So when you see his name as director on the modern-day war film 13 Hours you expect




www.soundandpicture.com





Okay, if you read everything up to this point, you would think "upmixing is hard, I should stick to stereo..."

*But hold on a minute...*

While researching an answer for the person who emailed me, I found something new and interesting. It's an open source project that uses artificial intelligence to do upmixing. Basically they trained an AI model to look for what the human voice sounds like. I messed around with it, and it does a fairly amazing job of taking a stereo track, and breaking it into TWO stereo tracks. One stereo track for the singer, and another stereo track for everything else (minus the singing.)

You guys might have fun tinkering with this, to see what you can come up with.

Here's a few ideas that I had:

1) You could take the stereo track that's acapella, downmix it to mono, and use it for a center channel. Then the other stereo channel, that has everything BUT the voices, would be routed to the left and the right speaker.

2) You could do the same as option one, but you could mix in a fraction of the original stereo signal, the one that HASN'T been split, into the left and the right.

The reason why you would want to experiment with the mix, is because the AI software is removing _everything_ but the vocals from the acapella output. IE, with a conventional upmixer, if there were drums in the original song, and the drums are present in the left and the right channels of the stereo mix, the drums would also be in the center channel. The AI software doesn't work like that, the AI software basically breaks the track down into individual instruments.

The AI software also has the option of creating FIVE stereo sources. I didn't mess with that too much, because it didn't know what to do with the tracks that I had. It might work fine on music that _only_ had a singer and a guitar and a piano. But that's not the type of music I listen to. So I stuck with the option of splitting the stereo source into an Acapella output, and an output that has everything but the Acapella output.

There's a few different ways to run the AI software. The simplest by far is to use an online portal: Ezstems: Get audio stems online with Spleeter in seconds.

I installed the software into a Linux VM running on my Windows desktop. The reason that I went that route, is because I think the only practical way to do this is to process your entire music library. Basically you would take your stereo library of music, convert it into three or five channels, and then you would put that (processed) library onto a USB drive, or possibly use some type of HTPC in your car or a tablet. A Windows tablet would be especially good, because there are many inexpensive USB sound cards with 5-7 channels of output.

If you want to install the software on Linux, here's how: deezer/spleeter


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