# APL 1012 - 10ch FIR-based DSP - Review



## Hanatsu (Nov 9, 2010)

_Few years back I reviewed the APL software and APL1 hardware unit. I already concluded that the APL had a great algorithm to calculate the system sound power response. The APL1 unit is a two channel equalizer which connects in series before the DSP unit in a typical car audio system. You can read the full review of it here (it's a bit related to this, the measurement part is discussed in detail there...): 

APL1 Review_

*With that said, here's the review of the 1012 DSP:*

Before I installed the 1012, I used a MiniDSP C-DSP and an APL1 for input EQ. It worked fine but it's always a bit tedious to setup two different processors. For years now I've tried to create a perfect coherent stage in my car but to pull some of things off, like correcting subwoofer delays - you require a lot of processing power with FIR-based filtering. Getting the acoustic crossovers right is also a challenge, often there's cancellations in the stopband of the HP/LP filter and this does affect the stage more or less depending on setup. To attain a "perfect" acoustic crossover, you need two set of speakers (midrange/tweeters for example) that both operate in their optimum sound power range and got a flat response in each respective area before equalization. For a 3" driver the sound power response is optimal to slightly above 3kHz (the driver got a frequency response that's omnidirectional into all axis). 

You can attain an optimal power response in the crossover stopband in two ways basically, one is to lower the crossover frequency and the other is to use steep slopes (tall order filters). The drawback of lowering the crossover frequency is that you need a tweeter (in this example) that goes lower without audible distortion. The drawback of using tall order filters with conventional filters is that you get massive phase distortion (group delay). If you got a setup like mine with tweeters that doesn't like to go very low using tall order filtering is the preferable option. Now to the point... The APL 1012 gives us tall order filtering with the option to completely eliminate the delay caused by the crossover. With this we can perform pre-equalization of every speaker in the system and then place the crossovers on top of the equalized response to attain a near perfect controlled rolloff into every axis. This is very important since the majority of the sound we hear in an automotive environment comes from reflections that cannot be equalized individually (of course). 

That's the theory and I must say that it really does work in practice as well. Even though I had put a lot of time into setting up my old system, this processor did wonders when it came to staging. Most likely due to more powerful equalization and optimized crossovers. Within half an hour I managed to get a better sounding setup than the one I had before with little effort and I haven't even made the post-corrections yet. The idea is that you can setup EQ and crossovers/delays then run the measurements once more to get a more accurate response. A quick TDA measurement showed that multiple small issues had disappeared, primarily in the midbass/midrange region. After the first listening session I noticed that the stage both jumped upwards, got some improved depth and that midbass/sub integration was as good as I've ever had in my car. With a balanced subwoofer volume the upfront bass is amazing now, stage stays stable with most content. 

_Having said that - A processor, no matter how good cannot be a substitute for a good install. (Just felt obliged to clarify that fact). _





Anyways, the APL1012 in comparison to the APL1 is complete DSP with EQ, crossovers and time alignment which eliminates the need for an external DSP as you required with the APL1 unit. The version I have features ten output channels, two analogue inputs and a SPDIF input but there's a version available that also got TOSlink. With 10 channels you could either run a 5-way setup or a common 4-way with rear channels for ambiance etc. The unit got more processing power than most DSPs on the market with cascadable FIR-filters for each channel with 4096 taps available at the moment (it's the firmware that's the actual limitation in the current version). Crossovers are 4th or 8th order and both linear phase and minimum phase filters with group delay compensation are available. You can use the built in equalization function to generate target curves (global EQ) or EQ individual channels. You can also use up to 45ms of delay for each channel which is really good for rear channel experimentation or subwoofer GD compensation. Since the DSP is FIR-based you can also experiment with linear phase filtering. For instance, you could basically make a zero delay vented subwoofer output, I would wait until there are more taps available on the subwoofer channels for increased accuracy though.

The process is pretty simple. If you are familiar with the APL1 unit, you know how the filters are generated in APL Workshop or TDA EQ. For the APL1 unit, the filters are stored in left and right channels to be used as input EQ for each side respectably. In APL1012 unit, each of the filters are instead used per channel (per speaker) and the crossovers are applied on top of these filters. The big advantage of this is that you actually EQ each channel individually before you cascade the crossover on top the filter, this way the stopband will be perfectly equalized with no phase nulls in the actual acoustic crossover region. You can either invert the IR or use the built-in group delay compensation function to cancel out the substantial delay of the tall order filters. 

(To clarify, it runs the full APL equalization on EACH speaker) 

While working in the C5 software you don't need to upload filters manually as you got the option to directly control the units from the software just like any other DSP software. The software is pretty straightforward to use once you get the hang of it. The software caters both to the less experienced and the advanced user with the cascadable filter option. 

The actual hardware are high quality as usual with respectable brand components internally. You can connect volume, sub gain, balance pots and an input selection switch from the two external mini XLR ports which are an improvement over the previous units where you had to open up the chassis. Volume and subwoofer pots are smooth and work really good in a 'linear' fashion from 0-100% output, I believe you get around 6m (~18ft) length of cable. The EQ mode switch work just the APL1 with the exception that modes 1 through 5 are reserved for each set of output filters, still you got 16 modes in total so there's plenty available if you wanna swap between different EQ curves or filters. It also got balanced RCAs which eliminates ground loops if you ever should run into that problem.

All in all I highly recommend this unit, used right it offers (in my opinion) the most processing power available among any of the readily available DSPs. It's a sizable unit compared to many other DSPs but I like the simple and clean design of the unit but it shouldn't be too much of an issue to find a spot for it. I don't think anyone would be disappointed with this unit and it will probably take a while before you feel the need to upgrade to anything else...

A manual is included this time around, it describes what the various functions in the software does so don't worry too much about setting it up 

For price, please contact Raimonds Skuruls. As far as I understand the price is different as units are built with somewhat different features and some accessories like the external controls are optional.

Here's some pictures of the packaging (I let that speak for itself) and the included C5 software:





Here's how the C5 software looks:

(Main window where you upload all your filters, manage delays, muting etc.)

It accepts *.fir / *.wave files (Workshop, TDA EQ format or an IR as wave). Details about workshop is in the APL1 review << 



Crossover section, you choose your crossover points and the lowpass/highpass are placed at the same place. No need to mess around with different crossover settings/slopes in this type of setup. Here's a standard 4-way setup with minimum phase crossovers (you can also choose linear crossovers in the dropdown menu).



Every time you upload a filter you get to see the frequency response, phase, group delay of the actual measurement. You can also view all curves of the different speakers in a separate graph window.



Here's an example how you can use the volume, sub and EQ mode switch. I simply set my headunit on a fairly high output (below clipping of course) and use the controls directly connected to the DSP.



Even though it's in the manual, I will write it here as well. When using the C5 software, do not, I repeat not use space or any special symbols in any of the filenames, project folders etc. It kept me busy for a while when setting up my system, safe to say... I will do a separate little video how to measure and how to use the software in the how-to section later!

At the time of writing this, the unit isn't included on the website. Here's a link to the official APL website: http://aplaudio.com/conc2/

Raimonds is also active on this forum, so you could PM him directly.


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## GreatLaBroski (Jan 20, 2018)

Wow that's nice. Is that 4096 taps available per channel?!


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## Elgrosso (Jun 15, 2013)

Many thanks Han’! You've just made it harder to resist 
I can see 2/3 frequencies where it would be useful in my setup


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## seafish (Aug 1, 2012)

Sounds REALLY nice!!!

Can you provide the dimensions and at least ballpark the price??

Do you know if the APL C5 software works on Windows Vista 32 bit (c 2006)??

Do the 3 knob controllers come with the unit or are they something you built??

TIA


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## bbfoto (Aug 28, 2005)

Looks great. Thanks for the excellent write-up! :thumbsup:

THIS is what I was waiting for and hoping the APL1 would evolve into.  Nice work Raimonds! :thumbsup:

I would also like to know the Length x Width x Height dimensions if you have a moment to measure. 

Also, what is the pixel resolution of the software window? Is it scalable?


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## Hanatsu (Nov 9, 2010)

GreatLaBroski said:


> Wow that's nice. Is that 4096 taps available per channel?!



Yes. Hardware is capable of more, as far as I understand 8192 taps will be available at some point as it is a programming ’issue’ atm.


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## Hanatsu (Nov 9, 2010)

bbfoto said:


> Looks great. Thanks for the excellent write-up! :thumbsup:
> 
> 
> 
> ...




I think it’s 297x210 can’t remember height right now. It’s almost fullscreen on my laptop with 1680x1050 res, haven’t tried to scale it. Will have to check that


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## Hanatsu (Nov 9, 2010)

seafish said:


> Sounds REALLY nice!!!
> 
> 
> 
> ...



Works on Win7. If a Mathlab 2013 8.1a runtime version is available it should work, there is a 32 and 64bit version of the software. Dimensions are 297x210 maybe 60-70 high. I’ll check later, no idea what it costs, sorry.

The three potentiometers came with the unit and there’s plastic knobs included but I bought other ones instead. I built the holder myself though.


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## Jscoyne2 (Oct 29, 2014)

You can safely assume it'll be in the 1-2k price line. Guaranteed. Though atleast the software comes with it this time. That's nice. 

UI is still garbage, with the price he puts on these. You would assume they would fix that without the excuse of "its for professionals"


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## GreatLaBroski (Jan 20, 2018)

Hanatsu said:


> GreatLaBroski said:
> 
> 
> > Wow that's nice. Is that 4096 taps available per channel?!
> ...


So 40,960 total taps available for all channels, with 81,920 total taps available after the firmware is sorted out? At 44khz sample rate that’s about 1-2 sec of total delay. That’s absolutely monsterous. But I’m sure the price matches the performance.


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## thehatedguy (May 4, 2007)

It's more than 1k...guaranteed.



Jscoyne2 said:


> You can safely assume it'll be in the 1-2k price line. Guaranteed. Though atleast the software comes with it this time. That's nice.
> 
> UI is still garbage, with the price he puts on these. You would assume they would fix that without the excuse of "its for professionals"


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## Holmz (Jul 12, 2017)

Jscoyne2 said:


> You can safely assume it'll be in the 1-2k price line. Guaranteed. Though at least the software comes with it this time. That's nice.
> 
> UI is still garbage, with the price he puts on these. You would assume they would fix that without the excuse of "its for professionals"


There are not a lot of option for FIR based DSPs...
Whether FIR is needed or not is a different argument. 

Obviously I went for it.
I understand FIR filters, and group delay, so it seemed to be the best option, and I had been leaning heavily towards a minDSP DDRC-88A.


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## GreatLaBroski (Jan 20, 2018)

Holmz said:


> Jscoyne2 said:
> 
> 
> > You can safely assume it'll be in the 1-2k price line. Guaranteed. Though at least the software comes with it this time. That's nice.
> ...


Yeah I myself have 3x MiniDSP 2x4 HDs. I’m planning on doing some opamp / cap mods which should be good fun.

You’re right that there’s not a lot of FIR filter options at this point in time. But, there’s another product with FIR filters which will enter the market in Q3-Q4 which is currently unannounced.

I’m thinking the DSP market is going to heat up soon.


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## thehatedguy (May 4, 2007)

This is more or less a Lake style processor for the car...so even at 2k, it's half of what a Lake would cost you and runs on 12 volt.


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## GreatLaBroski (Jan 20, 2018)

Holmz said:


> Jscoyne2 said:
> 
> 
> > You can safely assume it'll be in the 1-2k price line. Guaranteed. Though at least the software comes with it this time. That's nice.
> ...


Yeah I myself have 3x MiniDSP 2x4 HDs. I’m planning on doing some opamp / cap mods which should be good fun.

You’re right that there’s not a lot of FIR filter options at this point in time. But, there’s another product with FIR filters which will enter the market in Q3-Q4 which is currently unannounced.

I’m thinking the DSP market is going to heat up soon.


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## GreatLaBroski (Jan 20, 2018)

thehatedguy said:


> This is more or less a Lake style processor for the car...so even at 2k, it's half of what a Lake would cost you and runs on 12 volt.


Not sure it does nearly what an LM44 would do, but I see the price / value comparison you’re making. It’d be nice to see a toslink input.


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## captainobvious (Mar 11, 2006)

Thanks for the review! Looking forward to your instructional writeup


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## thehatedguy (May 4, 2007)

You can get it with digital input- AES/3, S/PDIF, and toslink are options. And can be Dante ready on request too.

There is a high rez version that can do 24/96 and/or 24/192. Those versions have 16384 taps on the two lowest bands.


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## Hanatsu (Nov 9, 2010)

GreatLaBroski said:


> So 40,960 total taps available for all channels, with 81,920 total taps available after the firmware is sorted out? At 44khz sample rate that’s about 1-2 sec of total delay. That’s absolutely monsterous. But I’m sure the price matches the performance.



No, that’s not exactly how it works. Sample rate is 48kHz to begin with (I think this can be 96kHz but it’s also software limited atm) and all five ”channels” are different units. They work independant from each other. Can’t give you the numbers right now but the delay isn’t noticable for me at least with the current setup.


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## GreatLaBroski (Jan 20, 2018)

Hanatsu said:


> GreatLaBroski said:
> 
> 
> > So 40,960 total taps available for all channels, with 81,920 total taps available after the firmware is sorted out? At 44khz sample rate that’s about 1-2 sec of total delay. That’s absolutely monsterous. But I’m sure the price matches the performance.
> ...


Oh no I wasn’t saying that the unit would have an output delay of 1-2 seconds, but rather that it has the delay capability to do enough taps to equal 1-2 seconds.


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## Jscoyne2 (Oct 29, 2014)

So what your saying is I should wait till December before purchasing a nicer dsp..

Sent from my XT1710-02 using Tapatalk


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## LumbermanSVO (Nov 11, 2009)

Jscoyne2 said:


> You can safely assume it'll be in the 1-2k price line. Guaranteed. Though atleast the software comes with it this time. That's nice.
> 
> UI is still garbage, with the price he puts on these. You would assume they would fix that without the excuse of "its for professionals"


I work in the "pro" video world, and the UI for this hardware isn't bad at all compared to what I regularly come across. Workshop isn't as pretty or intuitive as REW, but I get FAR better results, in significantly less time, with Workshop than I do with REW. That's a trade off I'm willing to make.


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## Weightless (May 5, 2005)

Designing a pretty GUI takes time and costs money. I'm in a similar line of work and I agree. I've seen far worse interfaces on much more expensive systems. 

I'll take functionality any day over spending more money on something that really isn't needed for the task. 

At least it's not command line driven. 

Sent from my SM-N920P using Tapatalk


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## oliverlim (Dec 5, 2016)

Hanatsu said:


> I think it’s 297x210 can’t remember height right now. It’s almost fullscreen on my laptop with 1680x1050 res, haven’t tried to scale it. Will have to check that


Its 270 x 205 x 35mm. Thats my unit..


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## oliverlim (Dec 5, 2016)

thehatedguy said:


> You can get it with digital input- AES/3, S/PDIF, and toslink are options. And can be Dante ready on request too.
> 
> There is a high rez version that can do 24/96 and/or 24/192. Those versions have 16384 taps on the two lowest bands.



Thats right. U tell Raimonds what you want and he will customise it for you. Mine has Coax and toslink inputs.

I do wish that it can function as a desktop dac though.


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## Hanatsu (Nov 9, 2010)

oliverlim said:


> Its 270 x 205 x 35mm. Thats my unit..




Yeah I mixed it up. These are the correct numbers


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## Spartak802 (Jul 18, 2016)

oliverlim said:


> Its 270 x 205 x 35mm. Thats my unit..


I have mounted an extra plate behind the rear seats (sedan) for my present apl1 and amplifyer on one side, and the hand-made crossovers plus amp for the subwoofer on the other side. Looks great. There is even a place to replace the existing apl1 unit with this new monster one. Highly recommended.


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## Holmz (Jul 12, 2017)

GreatLaBroski said:


> Yeah I myself have 3x MiniDSP 2x4 HDs. I’m planning on doing some opamp / cap mods which should be good fun.
> 
> You’re right that there’s not a lot of FIR filter options at this point in time. But, there’s another product with FIR filters which will enter the market in Q3-Q4 which is currently unannounced.
> 
> I’m thinking the DSP market is going to heat up soon.


I looked at three miniDSP, but there are no FIR filters there.

There was a French unit and I think some other system. I am not sure how I find a unit which is not announced , but I suspect that the APL will be good enough for me 

I am not sure that the DSP market will heat up. Why pay more for a unit that is only a little better?


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## GreatLaBroski (Jan 20, 2018)

Holmz said:


> I looked at three miniDSP, but there are no FIR filters there.
> 
> There was a French unit and I think some other system. I am not sure how I find a unit which is not announced , but I suspect that the APL will be good enough for me
> 
> I am not sure that the DSP market will heat up. Why pay more for a unit that is only a little better?


MiniDSP 2x4HD, the MiniShac, the OpenDRC series, and the C-DSP 8x12 (eventually, needs firmware update for it) all support FIR.


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## thehatedguy (May 4, 2007)

The C-DSP?


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## GreatLaBroski (Jan 20, 2018)

thehatedguy said:


> The C-DSP?


Yes, their C-DSP 8x12 has FIR compatibility however the dev team has yet to release it in a firmware update. It says it in the spec sheet: “IIR/FIR (depending on plugin)”.


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## Holmz (Jul 12, 2017)

GreatLaBroski said:


> Yes, their C-DSP 8x12 has FIR compatibility however the dev team has yet to release it in a firmware update. It says it in the spec sheet: “IIR/FIR (depending on plugin)”.


^That^ is more like "existing in theory", than actually existing in the natural world..

I had been leaning the miniDSP way, but the support team and forum did not seem helpful enough for what I wanted, so I was in the fence... I am not sure how many people need to say FIR is a waste of my time, before we all agree that it seems that I am just too set in my ways.

I was about ready to dive in, and the choice was 88A-BM or the APL.

I suspect that they both end up at the same place in flatness/EQ - just the APL was "about ready to ship", and the miniDSP was "just needing a plugin".


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## GreatLaBroski (Jan 20, 2018)

Holmz said:


> ^That^ is more like "existing in theory", than actually existing in the natural world..
> 
> I had been leaning the miniDSP way, but the support team and forum did not seem helpful enough for what I wanted, so I was in the fence... I am not sure how many people need to say FIR is a waste of my time, before we all agree that it seems that I am just too set in my ways.
> 
> ...


Not exactly, the C-DSP 8x12 uses the ADSP21489 chip, which is the same as the MiniDSP 2x4 HD. The 2x4 HD does FIR today. I don’t think it’s a stretch to expect the 8x12 to do it. I do, however, have reservations about how whether or not there will be enough taps to effectively equalize / phase correct the sub-channel via FIR.

Either way you seem kind of defensive about it which isn’t necessary. The APL seems like an excellent DSP, just out of my price range. I didn’t go with the 8x12 myself because of the fact that the MiniDSP support / dev team is unwilling to communicate product roadmaps and I don’t feel like waiting on them. I got a good deal on 3x 2x4 HDs and I’m doing Toslink into all 3. I’m a DIY’er so I’ll be upgrading opamps, upgrading caps, building my own isolated dc/dc power supply and really tuning it for my setup. That’s beyond what most can do and I think an all-in-one solution like the APL is a better option for most.


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## Holmz (Jul 12, 2017)

GreatLaBroski said:


> Not exactly, the C-DSP 8x12 uses the ADSP21489 chip, which is the same as the MiniDSP 2x4 HD. The 2x4 HD does FIR today. I don’t think it’s a stretch to expect the 8x12 to do it. I do, however, have reservations about how whether or not there will be enough taps to effectively equalize / phase correct the sub-channel via FIR.
> 
> Either way you seem kind of defensive about it which isn’t necessary. The APL seems like an excellent DSP, just out of my price range. I didn’t go with the 8x12 myself because of the fact that the MiniDSP support / dev team is unwilling to communicate product roadmaps and I don’t feel like waiting on them. I got a good deal on 3x 2x4 HDs and I’m doing Toslink into all 3. I’m a DIY’er so I’ll be upgrading opamps, upgrading caps, building my own isolated dc/dc power supply and really tuning it for my setup. That’s beyond what most can do and I think an all-in-one solution like the APL is a better option for most.


I am not feeling as defensive as normal 
I'm just trying to outline my thought process, which may help to understand whether my decision was rational. It may not be, but knowing the process allows for a better discussion of whether the solution was supported by the process, or whether the thought process was flawed. To me the thought process is likely more important, as we can arrive at some agreement.

Ideally one would have the sub channel resampled (Filter-tune-decimate) down to a lower rate... So something like 48k or 96k resampled down to something like 2k samples/second for the sub-channel... which supports a Nyquist of 1k. Then 1k taps is plenty, and 129 or 257 taps would be good too... unless one actually wants the sub to get in the hundreds of Hz range, which one may want in a home setting or some front sub arrangement?? (It would probably be rare, but maybe a horn set up with subs going to 300 Hz would justify it??)

I basically looked at multiple (3) 2x4s and thought, "self, just get a single unit". Which was the APL or the 88A.

I don't think there is bad choice here, but I did decide to trade $ for better perceived support and time savings, and less complexity than multiple MiniDSPs.

Whether that makes sense or not we will see when the unit gets here, and how well the whole thing works in a system.

So I welcome your thoughts... (whether I seem defensive or not.)


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## Elgrosso (Jun 15, 2013)

GreatLaBroski, I’m really curious about the mods you planned, but I’m sure you’ll let us know.


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## GreatLaBroski (Jan 20, 2018)

Holmz said:


> I am not feeling as defensive as normal
> I'm just trying to outline my thought process, which may help to understand whether my decision was rational. It may not be, but knowing the process allows for a better discussion of whether the solution was supported by the process, or whether the thought process was flawed. To me the thought process is likely more important, as we can arrive at some agreement.


Totally reasonable, I'm sorry for reading too much into your last post and accusing you of being defensive.



Holmz said:


> Ideally one would have the sub channel resampled (Filter-tune-decimate) down to a lower rate... So something like 48k or 96k resampled down to something like 2k samples/second for the sub-channel... which supports a Nyquist of 1k. Then 1k taps is plenty, and 129 or 257 taps would be good too... unless one actually wants the sub to get in the hundreds of Hz range, which one may want in a home setting or some front sub arrangement?? (It would probably be rare, but maybe a horn set up with subs going to 300 Hz would justify it??)
> 
> I basically looked at multiple (3) 2x4s and thought, "self, just get a single unit". Which was the APL or the 88A.
> 
> ...


I think you made the right choice. If I was strictly looking to use a DSP, I'd avoid running multiple smaller DSP units. I'm just looking for a project. Since I was able to get the MiniDSPs for a little bit over $100 each, it actually enabled me to feel okay with modding them (and potentially blowing up a board if things go horribly wrong).

Also in my case I'm going to be using a stock-headunit-to-toslink adapter which will retain volume control in the headunit and output a digital signal. That digital signal goes to a splitter which splits it into 3 connections which goes to the MiniDSPs. If I was using RCA's out from my headunit or from a high/low level converter then that whole setup wouldn't make much sense as I'd be passing the signal through too many different devices and hurting signal integrity. Digital won't have that same problem.

The last component is the selecting of which preset for the DSP. I'll use an arduino and 3d print a set of 4 switches to help facilitate the communication of which preset is being used and send it to the 3 DSPs.

Between the MiniDSPs, the cost of the upgrade opamps and parts, and the arduino, I'm still below $500. And that gets me 3x the processing power of the 8x12 C-DSP, plus I expect higher sound quality after the mods. This comes with the price of a heck of a lot more hassle and engineering. Plus if I wasn't doing volume over toslink, I'd need to also make a volume controller. But that stuff is for fun, for me at least. But I probably wouldn't have gone this route if MiniDSP had equipped their 8x12 with FIR out of the box. There's also another DSP coming to market later this year, if they can release it on time, which should be really really promising.



Elgrosso said:


> GreatLaBroski, I’m really curious about the mods you planned, but I’m sure you’ll let us know.


I'll do a build thread for it. I'll definitely be A/B testing them since I have 3, but I'm trying to figure out a way to non-subjectively measure the results. 

That shouldn't be a big deal, but I know it's been a major point of contention with this forum, relating to amplifiers. Some believe that amplifiers all sound the same. Yet it's pretty well established that opamps sound different. For those who disagree, go on hifi forums and try saying that to them. They do and I've heard the differences. It's kind of what timbre is to speakers, except for solid state chips. Timbre isn't able to be expressed with waterfall charts and frequency graphs. Since different amps use different opamps and circuit architectures shouldn't it be that it's possible that different amps sound different? Anyways, I shouldn't even start that discussion here seeing how prior threads have gone. :laugh:

As a quick rundown of what I'm doing, I'm planning on: 

Replacing the JRC2068 opamps with AD8599 opamps
Replacing the output DC filtering caps with Silmic II audio 100uf 6.3v caps (expecting it to improve bass attack and depth)
Upgrading the two clock oscillators with higher quality crystals (might skip this since I should have nice power)
Upgrading the linear voltage regulator to it's bigger brother with 2x output capabilities
Adding Panasonic ECWFE 1uf film capacitors to the new opamps as bypass caps.
I may also upgrade the SMT 1000uf power supply cap with something bigger since it seems to get warm, but I'll need to find a replacement that fits. 
I'm going to make a DC/DC isolated power supply for the 3 MiniDSPs using a Cincon EC4SBW-24S12 20 integrated DC/DC converter module to provide clean isolated power.
All in all, I'm expecting these mods to make a pretty big improvement in sound quality, not that it's bad now.

I have all class G/H amps too, and I'm considering recapping and upgrading the opamps on them too, even though 2 of them are NIB. They are using JRC 5532's which really aren't that bad though.

Anyways, sorry for getting off track OP. I'd love to hear some more subjective impressions of the APL DSP. It's a beauty. Same goes for you Holmz, when you get yours I'd love to see you do a thread on it and get some in-depth feedback on it.


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## Hanatsu (Nov 9, 2010)

Holmz said:


> I am not sure how many people need to say FIR is a waste of my time, before we all agree that it seems that I am just too set in my ways.


I'm very impressed by FIR filtering to say the least. The things you can do with it far outshines any traditional IIR-type filters. I would never go back to a traditional DSP-setup nowadays.


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## Hanatsu (Nov 9, 2010)

GreatLaBroski said:


> Not exactly, the C-DSP 8x12 uses the ADSP21489 chip, which is the same as the MiniDSP 2x4 HD. The 2x4 HD does FIR today. I don’t think it’s a stretch to expect the 8x12 to do it. I do, however, have reservations about how whether or not there will be enough taps to effectively equalize / phase correct the sub-channel via FIR.
> 
> Either way you seem kind of defensive about it which isn’t necessary. The APL seems like an excellent DSP, just out of my price range. I didn’t go with the 8x12 myself because of the fact that the MiniDSP support / dev team is unwilling to communicate product roadmaps and I don’t feel like waiting on them. I got a good deal on 3x 2x4 HDs and I’m doing Toslink into all 3. I’m a DIY’er so I’ll be upgrading opamps, upgrading caps, building my own isolated dc/dc power supply and really tuning it for my setup. That’s beyond what most can do and I think an all-in-one solution like the APL is a better option for most.


The 2x4 HD got very limited FIR capabilities. At least for lower frequencies with 2048taps in total. It's like 11Hz resolution at 48kHz (unless it allows for lower sampling that is). It's usable with midrange/tweeters but pretty unusable with anythings lower than that. It's in the modal range you want the added resolution that high amount of taps give you. I remember I looked at this and passed on this DSP for that reason.


----------



## GreatLaBroski (Jan 20, 2018)

Hanatsu said:


> The 2x4 HD got very limited FIR capabilities. At least for lower frequencies with 2048taps in total. It's like 11Hz resolution at 48kHz. It's usable with midrange/tweeters but pretty unusable with anythings lower than that. It's in the modal range you want the added resolution that high amount of taps give you. I remember I looked at this and passed on this DSP for that reason.


It can do about 4096 on a single channel if you reassign taps from unused channels. I’m splitting my DSPs into (Front Left Midbass/Mid/High) / (Front Right Midbass/Mid/High) / (Subwoofer / Rear Fill L/R).

I can move rear fill to the other DSPs if I really need all 4096, but that should be enough.


----------



## Holmz (Jul 12, 2017)

GreatLaBroski said:


> It can do about 4096 on a single channel if you reassign taps from unused channels. I’m splitting my DSPs into (Front Left Midbass/Mid/High) / (Front Right Midbass/Mid/High) / (Subwoofer / Rear Fill L/R).
> 
> I can move rear fill to the other DSPs if I really need all 4096, but that should be enough.


I know a bit about DSP... so I know that you would be better off sampling the sub channel at the lowest rate, or resampling it down to a low rate. You can goggle multirate processing or resampling if that helps to get a handle on it...

Then at a 2k sample rate you get FFT bins can be 1-Hz apart and you have reduced the processing load as the bins are few and the sample rate is low. And 2k is about 10x what you really need with an 80-100 Hz sub bandwidth.

Or you use a $hit load of bins at a high sample rate... Both yeild exactly the same output, but I am not sure that resampling is feasible in these DSPs.


----------



## Holmz (Jul 12, 2017)

Hanatsu said:


> I'm very impressed by FIR filtering to say the least. The things you can do with it far outshines any traditional IIR-type filters. I would never go back to a traditional DSP-setup nowadays.


Brother,

Symmetric FIR filters are nice (and sort of causal in the mathematic sense), but the concept of asymmetric filters and group delay decolouring means that one can pretty much correct everything except for perhaps the impulse response of the speaker. One can get the output of the speaker to be much closer to the input signal.

So I figured why start with an IIR DSP(?)... when I can enter in with a FIR based DSP(?). It seemed to be more towards being on the best (leading) edge of technology. One can do this on a computer, but Raimonds is doing it on an imbedded ASIC chip or something like that. I support that R&D and hence I got his product. Hopefully it has enough of a following to stay supported and in-production. It seems from the specs to be pretty special. Over time the approach should be common place, as it is technologically optimal.


----------



## Elgrosso (Jun 15, 2013)

GreatLaBroski said:


> Totally reasonable, I'm sorry for reading too much into your last post and accusing you of being defensive.
> 
> 
> 
> ...


Very cool, I don’t know much about all this but I’m curious, as my 2x8 with better components definitely sounds better than previous ones.
There’s also the new mini shd for soon I hope, not much infos yet. But I don’t want to derail too much


----------



## Spartak802 (Jul 18, 2016)

Holmz said:


> Over time the approach should be common place, as it is technologically optimal.


Not sure. It is a hard task to predict future development of the market on the technological basis. VHS was not the best in its time, but became a winner anyway.


----------



## captainobvious (Mar 11, 2006)

I like how these can be ordered a la carte with some different options.


Is there any way the *manual* can be *posted *here so that I can take a look at it? I'm an APL1 owner now and of course very interested in this unit. I could potentially ditch my current DSP and use this as my single solution.


Also, regarding the output delay...I know Hanatsu you had mentioned it does 45ms delay. The software allows you to input up to 100ms. Is the limit actually 45ms or 100ms? That's a nice improvement over most available dsp's.


And also, what is the output voltage on the RCA outs?




Thanks!


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## Mic10is (Aug 20, 2007)

captainobvious said:


> I like how these can be ordered a la carte with some different options.
> 
> 
> Is there any way the *manual* can be *posted *here so that I can take a look at it? I'm an APL1 owner now and of course very interested in this unit. I could potentially ditch my current DSP and use this as my single solution.
> ...



more importantly whats the input voltage


----------



## B5I8 (Feb 7, 2009)

Mic10is said:


> more importantly whats the input voltage


APL1012 has "-10dBV" and "+4dBu" input and output levels which is about 2.5 and 8 Volts.
The output voltage must be devided by two in unbalanced mode.

That is from an email I got from Raimonds.


----------



## SQ Audi (Dec 21, 2010)

I too have been in contact with Raimonds. I look forward to getting my processor from here. Possibly...This or a DBX solution.


----------



## subterFUSE (Sep 21, 2009)

SQ Audi said:


> I too have been in contact with Raimonds. I look forward to getting my processor from here. Possibly...This or a DBX solution.


dBx processors are junk. Very light on features and most of them have 6 outputs or less.


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## Hanatsu (Nov 9, 2010)

Mic10is said:


> more importantly whats the input voltage


Specifications (different versions mentioned here, some specs are common with all versions though)

Inputs

2 x RCA balanced, “-10dBV” “+4dBu” levels, 7 kOhm impedance
1 x Coax S/PDIF (up to 96 kHz)
1 x Optical TOSLINK (up to 96 kHz)
1 x USB HD audio (up to 192 kHz)
1 x Remote In
1x USB configuration in

Outputs
10 x RCA, 4V output voltage (half of “+4dBu”), 8V in balanced mode
1 x Remote Out

Operating bandwidth
24 kHz for 48 kHz Sample Rate version unit
48 kHz for 96 kHz Sample Rate version unit

DSP resolution
Unlimited FPGA processing

DSP type
FPGA + Audio signal processor

Signal converters
Cirrus Logic CS4385 CS4272

Signal-to-noise ratio digital input
117 dBA

Signal-to-noise ratio analog input
114 dBA

Total harmonic distortion analog input
0.0003% at 1 kHz at -15dBFS
0.002% at 1 kHz at -0.1dBFS

Crossover / Equalizer
Unlimited options offered by APL Workshop and APL C5 programs

Configuration
APL C5 program

Memory
11 preset memory recalled by 16 position remote switch
Auto restart of DSP
watch dog served restart in case of DSP fault

Time alignment
0 – 85 ms 
0 – 28 m
Step 0.02 ms(48 kHz SR) 0.01 ms (96 kHz SR)

Protection
“Musical” output level limiters at all outputs allowing using loudspeakers at they full power with 10 times peak levels untouched


Crosstalk
>90 dB

Operating voltage:
7.1V – 15.7V

Current drawn , power on
500 mA for 48 kHz SR version, 12V power supply
550 mA for 96 kHz SR version, 12V power supply

Current drawn , power off
1.5 mA

Max remote output current 50 mA

Remote IN voltage 7.5 ÷ 15 VDC (1mA)

Fuse 
2A PPTC Resettable Fuse


Dimensions (H x W x D) 
36 x 270 x 200 mm
1.42 x 10.63 x 7.87 “

Weight
1.618 kg


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## LumbermanSVO (Nov 11, 2009)

Hanatsu said:


> 1 x USB HD audio (up to 192 kHz)


Whaaaaaaaat?!

Does this mean I can plug my iPhone directly into it?


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## Hanatsu (Nov 9, 2010)

LumbermanSVO said:


> Whaaaaaaaat?!
> 
> Does this mean I can plug my iPhone directly into it?


Not my one at least. You'll have to ask Raimonds about the different versions...


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## LumbermanSVO (Nov 11, 2009)

Email sent!

This would simplify things greatly! No more silly HDMI audio strippers, BT streamers, or other nonsense between the phone and processor.


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## crackinhedz (May 5, 2013)

Hanatsu said:


> I will do a separate little video how to measure and how to use the software in the how-to section later!


Curious if you have made a tutorial on this, think Im gonna swap my Mosconi 6to8v8 + APL VST for one of these units. I learn visually, so your vid would be very welcomed!


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## Holmz (Jul 12, 2017)

crackinhedz said:


> Curious if you have made a tutorial on this, think Im gonna swap my Mosconi 6to8v8 + APL VST for one of these units. I learn visually, so your vid would be very welcomed!


me too


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## sq2k1 (Oct 31, 2015)

Just seen this and now it has the old curiosity wanting to know more... So I am definitely keeping an eye on this.


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## crackinhedz (May 5, 2013)

Hanatsu said:


> The three potentiometers came with the unit and there’s plastic knobs included but I bought other ones instead.


Could you elaborate more regarding these knobs, specifically what you ended up using instead of the plastic knobs that came provided? Did you get them from Raimonds or somewhere else?

Just purchased the 48khz model, hope to have in hand in a few weeks!


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## thehatedguy (May 4, 2007)

Had a little miscommunication, mine is under way.


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## Hanatsu (Nov 9, 2010)

crackinhedz said:


> Curious if you have made a tutorial on this, think Im gonna swap my Mosconi 6to8v8 + APL VST for one of these units. I learn visually, so your vid would be very welcomed!




Next week. I’m super busy right now. Finishing a large job this weekend then I’ll start working on this


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## Hanatsu (Nov 9, 2010)

crackinhedz said:


> Could you elaborate more regarding these knobs, specifically what you ended up using instead of the plastic knobs that came provided? Did you get them from Raimonds or somewhere else?
> 
> 
> 
> Just purchased the 48khz model, hope to have in hand in a few weeks!



I bought mine from tme.eu.

The ones that are supplied with the unit are plastic and quite large (22-23mm in diameter). I got myself aluminum ones with smaller outer diameter.


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## crackinhedz (May 5, 2013)

Would you still use TDA for time alignment, or does C5 incorporate some type of TA measurement built in?

edit: nevermind, I saw Raimonds answer in the APL1 thread. You still use Workshop and TDA as usual, then C5 for management in the unit.


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## crackinhedz (May 5, 2013)

Hanatsu said:


> I bought mine from tme.eu


I visited this website but there were literally about 500 "knobs".  :blush:

Many looked like they were used in other configurations, so didn't know how to narrow it down to something that works with the APL. Is there a specific 'type' or 'size' I should look for? Thanks for the guidance!


----------



## Hanatsu (Nov 9, 2010)

crackinhedz said:


> I visited this website but there were literally about 500 "knobs".  :blush:
> 
> 
> 
> Many looked like they were used in other configurations, so didn't know how to narrow it down to something that works with the APL. Is there a specific 'type' or 'size' I should look for? Thanks for the guidance!




722 to be exact. Mine are OKW A1418260. At least the smaller two. 

You want 6mm shaft diameter, then choose whatever style, material, size from the filter menu.


----------



## crackinhedz (May 5, 2013)

thanks! ?


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## captainobvious (Mar 11, 2006)

I'm still looking for a manual for using the software for this unit. Is this not posted somewhere yet?


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## LumbermanSVO (Nov 11, 2009)

Not to knock Raimonds, but he seems to operate in the same way a lot of the Pro-A/V world operates: No manuals

Last week I finally went to a class to learn a pice of software I've already been using for three years. It just seems like the way things work in the Pro-A/V world. 

I *should* have some time to work something up next week, if I do I'll run it by Raimonds for accuracy before posting it.


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## Hanatsu (Nov 9, 2010)

There is a manual ready.

I was gonna make a basic workflow to go along with it before posting, should be up soon. Just waiting for Raimonds to reply.


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## Holmz (Jul 12, 2017)

hdrugs said:


> May I ask what are the drawbacks to a fir filter
> Added noise, group delay, solution artifact ect... if any..
> ...


One drawback might be that some understanding of DSP theory would be needed to really make the most of it.

On the flip side it means that one may not need to understand convolution, or the maths behind the filters, but certainly understanding the fundamentals of what is happening are an advantage.

If one was a EE using MATLAB, then it would be ideal... and if one was theory adverse, then it may not be an advantage unless they had an installer with was familiar with FIR based DSP theory.

But I may be somewhat biased...


----------



## Hanatsu (Nov 9, 2010)

Sorry, been really busy for a few days. I'm working on the promised workflow guide. I'll have it done by the end of the week.


----------



## LumbermanSVO (Nov 11, 2009)

Hanatsu said:


> Sorry, been really busy for a few days. I'm working on the promised workflow guide. I'll have it done by the end of the week.


I'm very curious how close my workflow got to yours, considering that I just poked at stuff in C5 to see what it does.


----------



## an.vamv (Apr 24, 2017)

Hello there. 

Here are some photos of my APL1012 .

I bought the 96Khz version with coax, optical and USB HD audio in (up to 192kHz/24bit). Total cost of the unit as a bundle together with TDA, Workshop, and C5 was 1800Euro delivered.

I have also attached a photo of my volume , sub volume and preset selector installation.


This is quite an amazing and unique unit ! Once one understands the philosophy of it ( a procedure which actually took me some time as i was totally unfamiliar with the whole APL idea ) the goal of achieving great sound in the car becomes a matter of 2-3 hours.
The solidity of the soundstage and the front bass integration that i managed to get is exemplary. 

The basic outline of the procedure is the following :

1. We use Workshop to take measurements for each driver (way) even without APL1012 connected. We connect the measurement soundcard's output to respective amplifier's input. Keep measurement level (volume) low (1-2 Volts on driver) to not damage high frequency drivers.
So for example for a 4way system we get 8 initial individual measurements Tweeter x 2 (L+R), Midrange x 2 (L+R) , Midbass x 2 (L+R), Sub x 2 (L+R).
Do not use space or any symbols in any project's name or filters. 

2. We import these initial .fir files that we got from Workshop to respective way/channel in C5 as EQ1. We activate L+R for each way in EQ1.

In C5 Unit 1 must be subwoofer, Unit 2 Midbass, Unit 3 Midrange and Unit4 Tweeter. 

3. We go to the "Curves" tab to have a look at the curves. 

4. On the main tab we activate crossovers for L/R in every channel that 
we are going to use by setting all checkboxes for crossovers on .We create our crossovers in "Crossover" tab. Unit must be connected to PC and ready to receive filters. We either use minimum phase Linkwitz Riley or linear phase max selectivity FIR filters. Raimonds suggests to use minimum phase L/R filters in the car. We select either 4th or 8th order . We set the desired acoustic frequencies of the crossover filters for each way. If use minimum phase filters we enable group delay correction .

5. Run the TDA program. Check levels and time delays basically between ways. TDA program can also be used to measure time delays between L/R. Adjust gains and time delays in C5 accordingly. 

6. After crossovers and time delays are set Run Workshop again for left channel all ways together and then right channel all ways together.

7. These two .fir filters we got from workshop one for the left side and one for the right side we import them in C5 in "EQ2" as master. 

8. Last thing to do is go to the PEQ tab to work on the parametric equalizer.We either use separate settings for each way/channel by selecting respective unit's number or we select "M" to apply a master PEQ curve for all ways. 

The usual settings Q,Gain and frequency can be adjusted together with the type of filter and order. Following types are available: 
BP: Bandpass ("normal" EQ filter) , Lsh: Low Shelf , Hsh: High Shelf
LshX: Low Shelf /w. adjustable 'order' , HshX: High Shelf /w. adjustable 'order', Lp X: Lowpass /w. adjustable 'order' , Hp X: Highpass /w. adjustable 'order'

I chose to load Raimonds' MP1 curve as input curve and apply it as MasterEQ by slightly modifying it. To activate MasterEQ one must go to the main page and check all MasterPEQ boxes for every way and channel.

One can load his own curves to use for PEQ.

9. Run TDA again to confirm results. Modify where necessary.

Well basically this is the procedure. The most important thing is to take careful and consistent sound power frequency response measurements with Workshop as per Raimonds suggested "painting" method and analyze these measurements to detect and try to correct any problems with the setup or installation . TDA program is extremely helpful and necessary in the process.

One thing i can not stress enough : the genius designer of this system Raimonds Skuruls is an extremely helpful and patient person even to a novice like myself. He has patiently answered dozens of emails i have sent him to help me get the grasp of the programs and their operation.
He even helped me by connecting with Teamviewer to my computer the first time after i had APL1012 installed to check my measurements and help me kick start the whole project. Raimonds' service is top notch !

To sum up, APL1012 is a unique product. By using sophisticated FIR filters which perform amplitude frequency response and phase frequency response correction and with the use of tools that offer valuable information about the system's performance, it offers the possibility to achieve amazing sound very quickly .

Most unreservedly and highly recommended !

Cheers 

Andreas


----------



## Spartak802 (Jul 18, 2016)

*Raimonds Skuruls is an extremely helpful and patient person* - fully agree with this statement. Technical support from the Acoustic Power Lab is certainly a winner compared to the compatitors. Respect, Raimonds!


----------



## Hanatsu (Nov 9, 2010)

Sorry for the wait. A basic workflow is already included in the pdf file.

https://drive.google.com/open?id=1sMad-ujMQbrxfMOzGlNTeZpGExsYpMcx

Edit: Dropbox don't work for some reason...


----------



## LumbermanSVO (Nov 11, 2009)

I you email it to me, I'll throw it up on my server for you. My email address is myusername at gmail


----------



## Hanatsu (Nov 9, 2010)

LumbermanSVO said:


> I you email it to me, I'll throw it up on my server for you. My email address is myusername at gmail


Link should be fixed now, uploaded to google drive instead.


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## LumbermanSVO (Nov 11, 2009)

It works, thanks!


----------



## Raimonds (Jun 13, 2014)

Hello Friends,

The first revision of the Manual is now ready for your evaluation!
Thanks Hanatsu for help and efforts!

Your comments and suggestions are highly appreciated!

Because each user has different experience and has different actual questions about use and workflow of the set of APL1012, Workshop, TDA and C5.

I am thinking about creation of some dedicated question-answer place.
I would like to know your suggestions where and how to create it to have it clear and easy usable as possible.

BR,
Raimonds


----------



## Durgesh (Sep 18, 2014)

Hanatsu said:


> Sorry for the wait. A basic workflow is already included in the pdf file.
> 
> https://drive.google.com/open?id=1sMad-ujMQbrxfMOzGlNTeZpGExsYpMcx
> 
> Edit: Dropbox don't work for some reason...


Thanks! This is very helpful.

Sent from my SM-G950F using Tapatalk


----------



## Hanatsu (Nov 9, 2010)

Raimonds said:


> Hello Friends,
> 
> The first revision of the Manual is now ready for your evaluation!
> Thanks Hanatsu for help and efforts!
> ...


Yeah, probably a good idea. The "how-to" subtopic section might be the correct place. There are a lot of things the apl1012 can do together with TDA, workshop and c5 and I think us users can share a lot of good info once more start to use it.


----------



## an.vamv (Apr 24, 2017)

Raimonds said:


> Hello Friends,
> 
> The first revision of the Manual is now ready for your evaluation!
> Thanks Hanatsu for help and efforts!
> ...



Raimonds, I second Hanatsu's opinion about the creation of a sub-section with Q&As soonest possible. There are several questions that users might have about APL philosophy and workflow as well as potential problems with software installation,compatibility under different OS which must be shared and discussed to everyone's benefit.


----------



## crackinhedz (May 5, 2013)

Is there anywhere to download the actual C5 program? Was hoping to get an early start loading and getting familiar with, or does C5 come with the APL unit via install disc/usb flash?

Is there any advantage to using the included volume/sub-level knobs vs. using head unit volume, amplifier gain knobs etc?


----------



## LumbermanSVO (Nov 11, 2009)

After you purchase a unit, Raimonds will email you with links to the software, and some basic instructions.


----------



## crackinhedz (May 5, 2013)

LumbermanSVO said:


> After you purchase a unit, Raimonds will email you with links to the software, and some basic instructions.


Thanks, I emailed him, must've been busy as I didn't get a download link yet.


Also for anyone that may know, can you speak about the "group delay" option? New concept for me coming from a mosconi 6to8v8. I have ported subwoofers in an suv, is this the same group delay that can produce from the port, or am I misunderstanding the concept, or why the need to use this?


----------



## Holmz (Jul 12, 2017)

crackinhedz said:


> Thanks, I emailed him, must've been busy as I didn't get a download link yet.
> 
> 
> Also for anyone that may know, can you speak about the "group delay" option? New concept for me coming from a mosconi 6to8v8. I have ported subwoofers in an suv, is this the same group delay that can produce from the port, or am I misunderstanding the concept, or why the need to use this?


The other way would be to learn and use MATLAB and there is a gnu freeware version of that. (Which I forget the name of).


----------



## Hanatsu (Nov 9, 2010)

crackinhedz said:


> Thanks, I emailed him, must've been busy as I didn't get a download link yet.
> 
> 
> Also for anyone that may know, can you speak about the "group delay" option? New concept for me coming from a mosconi 6to8v8. I have ported subwoofers in an suv, is this the same group delay that can produce from the port, or am I misunderstanding the concept, or why the need to use this?


Unless for got FIR filtering you cannot control the phase response in any accurate manner. IIR got allpass filtering but it's hardly usable when it comes to group delay corrections. I don't think the Mosconi DSP support FIR but idk.

Edit: Sorry, misread the question.

- Yes, you can compensate for a vented enclosure's group delay making a vented box with processing far superior to a sealed enclosure.


----------



## almatias (Nov 16, 2012)

Where do buy this processor?


----------



## subterFUSE (Sep 21, 2009)

almatias said:


> Where do buy this processor?




Email Raimonds


Sent from my iPhone using Tapatalk


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## crackinhedz (May 5, 2013)

Hanatsu said:


> Unless for got FIR filtering you cannot control the phase response in any accurate manner. IIR got allpass filtering but it's hardly usable when it comes to group delay corrections. I don't think the Mosconi DSP support FIR but idk.
> 
> Edit: Sorry, misread the question.
> 
> - Yes, you can compensate for a vented enclosure's group delay making a vented box with processing far superior to a sealed enclosure.


thanks, I was just trying to understand how/why someone would use this option. Is it necessary?


----------



## Holmz (Jul 12, 2017)

crackinhedz said:


> thanks, I was just trying to understand how/why someone would use this option. Is it necessary?


It basically takes the input signal and does the reverse (or inverse) of the subwoofers group delay, and then the speaker output then looks closer to the input signal.
Some say that the ears are way less sensitive to the group delay at subwoofer frees, and others that it matters.
Generally an objective measurement (hard analysis) from comparing the speaker's output getting closer to the input signal (truth) is considered better, and one trys to avoid soft subjective "analysis".

In your particular case, as you already have the Mosconi, then maybe one of the miniDSPs that do FIR could just do the subwoofer? 
Whether that is easy, or less expensive in time or $, could be a factor. Starting from scratch, with money not factored in, then it seems easier and more straight forward.


----------



## Hanatsu (Nov 9, 2010)

crackinhedz said:


> thanks, I was just trying to understand how/why someone would use this option. Is it necessary?


Here's some information. I believe GD at lower frequencies are audible. I had some paper on some listening test but I can't find it now.

Low frequency group delay equalization of vented boxes using digital...


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## SQ Audi (Dec 21, 2010)

Raimonds said:


> Hello Friends,
> 
> The first revision of the Manual is now ready for your evaluation!
> Thanks Hanatsu for help and efforts!
> ...


Raimonds, 

Did you ever get my email reply to you? Very curious about the process.


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## crackinhedz (May 5, 2013)

Holmz said:


> It basically takes the input signal and does the reverse (or inverse) of the subwoofers group delay, and then the speaker output then looks closer to the input signal.
> Some say that the ears are way less sensitive to the group delay at subwoofer frees, and others that it matters.
> Generally an objective measurement (hard analysis) from comparing the speaker's output getting closer to the input signal (truth) is considered better, and one trys to avoid soft subjective "analysis".
> 
> ...


Thanks for a detailed explanation! 

I currently have the 6to8 and APL vst, should receive my 1012 soon!


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## Durgesh (Sep 18, 2014)

Hanatsu have you tried rear ambiance with the 5th channel. Can you pls explain how to do that.


Sent from my SM-G950F using Tapatalk


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## Holmz (Jul 12, 2017)

Hanatsu et. al.,

I see there is a USB input point for digital files...
I often am out of range of any wireless links for streaming services.
I have a CD player and radio, but if I wanted to use a SSD or some other deal to shove music in, then what would be recommended?

It would need to be something that does not rely on the head unit for searching and selecting the files.

Secondly... The head-unit does have a digital in, so I could go in there and then come out of the head unit's RCAs...
So if there is some player that works that way, then that would work.

The old iPad is a 64GB, and is being relegated to a map server for navigation, and does not seem able to hold much volume of music... But I am using it for podcasts and audio-books.

The home computer is OSX, so I would need a media server that can work with that or be loaded via Parallels running an older Windows.

I am more concerned with volume of music and SQ, rather than cost.


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## Hanatsu (Nov 9, 2010)

Durgesh said:


> Hanatsu have you tried rear ambiance with the 5th channel. Can you pls explain how to do that.
> 
> 
> Sent from my SM-G950F using Tapatalk



Haven’t tried it yet but I assume you go 4-way so there’s no crossover on the 5th way then set delays to a highish number (20-25ms or even more, experiment with it). You need to bandpass the response in the parametric EQ tab, try 300-4000Hz. Measure the speakers fullrange with workshop as usual to attain the filter before that. Perhaps invert one of the speakers so they are out of phase, could use TDA to perfect delays, then 180deg phase inv on one channel.


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## Hanatsu (Nov 9, 2010)

Holmz said:


> Hanatsu et. al.,
> 
> 
> 
> ...



As of now, I have absolutely no idea what the USB audio-in is. Raimonds will have to explain how that works or if it’s an optional feature.


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## Durgesh (Sep 18, 2014)

Hanatsu said:


> Haven’t tried it yet but I assume you go 4-way so there’s no crossover on the 5th way then set delays to a highish number (20-25ms or even more, experiment with it). You need to bandpass the response in the parametric EQ tab, try 300-4000Hz. Measure the speakers fullrange with workshop as usual to attain the filter before that. Perhaps invert one of the speakers so they are out of phase, could use TDA to perfect delays, then 180deg phase inv on one channel.


Thanks. I'll try once I install the unit.

Sent from my SM-G950F using Tapatalk


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## subterFUSE (Sep 21, 2009)

For rear fill, you need to mix the left and right inputs together into a mono signal, with one of the inputs polarity inverted.

So, Left + and Right - mixed to the left rear speaker.

Then, Right + and Left - going to the right rear speaker.

Doing this will phase cancel any of the center stage information from the audio signal, leaving only the far left and right stage sounds.

Then you will need a ton of delay. More than 30 ms is a good start.


I don’t know if the APL 1012 is capable of doing a proper L-R, R-L mixed mono output?

Worst case, you can do this with a little miniDSP box for $200.


Sent from my iPhone using Tapatalk


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## Hanatsu (Nov 9, 2010)

Can always physically connect speakers in 70’s fashion ”quadrophonic mode”. I’ve done it without mono mixing before, I swapped channels and inverted one. Can’t think of a good way of doing it on this DSP.

I’m going with the ”signal difference” thing. Psuedo rear ambience or whatever it’s called

Edit; (As spl said above, tapatalk keeps bugging can’t see all posts...)


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## Holmz (Jul 12, 2017)

A Father's Day gift box came in on Friday...
1) Sivert Hoyem CD "Lioness"
2) fan belts
3) socket wrench
4) APL 1012


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## an.vamv (Apr 24, 2017)

Holmz said:


> Hanatsu et. al.,
> 
> I see there is a USB input point for digital files...
> I often am out of range of any wireless links for streaming services.
> ...


Holmz,
I use APL’s USB audio input for sending digital audio up to 192Khz/24bit over USB from my Raspberry Pi3 .
The RPi3 is placed in the trunk powered by a 12-5V DC-DC regulator . I have attached an SSD and a 128GB flash disc to it . I have installed Volumio which is a free Linux based operating system tuned exclusively for music playback. The RPi3 becomes a headless digital audio player which is controlled over Wi-Fi from my android mobile phone or my android Dashboard Multimedia System.
I would suggest for better quality to send direct digital audio to APL instead of going into your head unit’s digital in and then from its RCA output to APL, as like this you have one DA-AD conversion less . In case your APL does not have USB input there are SPDIF output piggyback boards in the market for Raspberry which you can use to send direct coax digital to APL.


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## subterFUSE (Sep 21, 2009)

an.vamv said:


> Holmz,
> I use APL’s USB audio input for sending digital audio up to 192Khz/24bit over USB from my Raspberry Pi3 .
> The RPi3 is placed in the trunk powered by a 12-5V DC-DC regulator . I have attached an SSD and a 128GB flash disc to it . I have installed Volumio which is a free Linux based operating system tuned exclusively for music playback. The RPi3 becomes a headless digital audio player which is controlled over Wi-Fi from my android mobile phone or my android Dashboard Multimedia System.
> I would suggest for better quality to send direct digital audio to APL instead of going into your head unit’s digital in and then from its RCA output to APL, as like this you have one DA-AD conversion less . In case your APL does not have USB input there are SPDIF output piggyback boards in the market for Raspberry which you can use to send direct coax digital to APL.



For car, do you recommend the Rasberry Pi over the Volumio Mini86?

https://volumio.org/product/volumio-mini86/


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## Hanatsu (Nov 9, 2010)

I'd go with an AppleTV (if you got an Apple phone or iPod) to feed the APL via wifi. Probably my next project.

A 7" touchscreen combined with a Raspberry Pi or something would be pretty cool as well. Digital audio is great for car audio in several ways...


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## bbfoto (Aug 28, 2005)

Good info above, thanks.

I haven't tried these myself, but from my research, if you will be using an add-on DAC/SPDIF output board with the Raspberry Pi3, I've read from guys who are using them in very Hi-End home systems that the available DAC/SPDIF output boards are "okay" but not great compared to their other stand-alone units (which of course are many times more expensive, so theirs that). 

But if you're going straight into the APL USB input, it's a non-issue.


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## subterFUSE (Sep 21, 2009)

Hanatsu said:


> I'd go with an AppleTV (if you got an Apple phone or iPod) to feed the APL via wifi. Probably my next project.
> 
> A 7" touchscreen combined with a Raspberry Pi or something would be pretty cool as well. Digital audio is great for car audio in several ways...



Yes, I have all Apple stuff and would certainly prefer to stay within their ecosystem, but do you think AirPlay is robust enough for in-car use? I sometimes have issues with it in my house, and my Apple TV is hardwired via ethernet to my router.


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## Durgesh (Sep 18, 2014)

Air play is wireless streming which may degrade the signal. Same with bluetooth.

I have been using use home media players with digital optical out since a decade for direct digital signal to DSP. I have used them to bitstream multichannel signals also...dts- dobly 5.1 in car. 

Now I have Popcorn Hour A500. It has both optical and coaxial out. I think both optical and coaxial work simultaneously. I control it with an app on my android phone. You don't need a screen for display. Files can be browsed over phone and played. Works fine. Boot time is around 30-45 seconds. 

Cons: I have to turn on my phone's wifi hotspot every time I sit in the car. IOS app not available.









Sent from my SM-G950F using Tapatalk


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## Hanatsu (Nov 9, 2010)

Airplay is not compressing audio, it's WiFi:



> What is the audio quality for this type of transmission?
> In contrast to Bluetooth connections, which sometimes lead to downsampling and loss of quality, AirPlay is transmitting “CD quality” audio (16bit / 44.1kHz). This can easily be verified with the Audio Midi Setup tool that is located in the Utilities folder in your app directory.
> 
> For streaming content Apple is using the in-house Apple Lossless Audio Codec (ALAC). Here data density is permitted for up to 120 megabit per second. No matter what format the audio originally coded, your Mac or iOS gadget will turn it into a lossless audio stream. Only so-called “Hi-Res” Audio formats used by pros and audiophiles with higher resolution (e.g. 24 Bit / 192kHz) will be down-sampled to 16 Bit / 44.1kHz.


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## Hanatsu (Nov 9, 2010)

subterFUSE said:


> Yes, I have all Apple stuff and would certainly prefer to stay within their ecosystem, but do you think AirPlay is robust enough for in-car use? I sometimes have issues with it in my house, and my Apple TV is hardwired via ethernet to my router.


No idea, no experience with the Airplay. It looks a little bulky judging from the pictures.


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## Holmz (Jul 12, 2017)

The player needs to work in a 4 wheel drive. So it needs to be a bit rugged.


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## subterFUSE (Sep 21, 2009)

How is volume controlled when using the HD USB Input?


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## subterFUSE (Sep 21, 2009)

Let's say you have the APL1012, and you have tuned it.

You get in the car and decide that the time alignment is a little off and you want to adjust it. Can the delays be adjusted in real-time while listening to the system? Or is the process a trial and error situation where you enter different delay numbers, and then have to listen and see if they improve the tune?

That's one thing I like about the Helix, is I can sit and listen while adjusting delay groups.


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## Hanatsu (Nov 9, 2010)

subterFUSE said:


> Let's say you have the APL1012, and you have tuned it.
> 
> 
> 
> ...




You can set delays while the system is on like any other DSP. Gotta do it from C5 software though.


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## an.vamv (Apr 24, 2017)

subterFUSE said:


> For car, do you recommend the Rasberry Pi over the Volumio Mini86?


subterFUSE , for car i have tested Volumio on three platforms: Raspberry RPi3, Asus Tinkerboard and Sparky but not mini86 so i can not comment about it. These three platforms are compact and are powered with 5V. The solution offered by Volumio with their mini86 is much bulkier and needs 12V so it might be trickier to get constant voltage regulation from the car's battery. From my tests , I liked the sound of ASUS Tinkerboard better but it had problems working with my Helix DSP HEC HD Audio USB card(that i had back then) plus its processor produced much more heat and needed good cooling. Sparky has more pronounced bass but i consider the sound of the Rpi3 more balanced. However the most decisive factor was the extensive support the RPi3 platform has from millions of users and the number of applications and add-on boards that one can find. Another problem one has to tackle is how to safely shut down the platform of his choice. If the power is suddenly cut off from the battery when switching off engine this is not a safe shut down, it is like switching off your PC suddenly by pulling the plug and this eventually might corrupt the OS that is installed on the SD card.For RPi3 platform there are some UPS add on boards that once they detect power cut off they automatically switch to battery power and execute a shut down command for safe shut down . Last but not least RPi3 is the cheapest of them all.


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## an.vamv (Apr 24, 2017)

subterFUSE said:


> How is volume controlled when using the HD USB Input?


By using the volume pot that Raimonds sends with the APL unit . He dispatches the unit with four pots : volume, sub volume, balance and presets as well as a ribbon cable that connects all these pots to the APL unit.


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## an.vamv (Apr 24, 2017)

Holmz said:


> The player needs to work in a 4 wheel drive. So it needs to be a bit rugged.


Holmz, I have been having my RPi3 player for over a year in a sports car with stiff suspension and under very hot temperatures during the summer with over 50-55 Degrees Celcius inside my trunk and no external ventilation. I have not had any problem so i consider it rugged enough.


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## Holmz (Jul 12, 2017)

an.vamv said:


> Holmz,
> I use APL’s USB audio input for sending digital audio up to 192Khz/24bit over USB from my Raspberry Pi3 .
> The RPi3 is placed in the trunk powered by a 12-5V DC-DC regulator . I have attached an SSD and a 128GB flash disc to it . I have installed Volumio which is a free Linux based operating system tuned exclusively for music playback. The RPi3 becomes a headless digital audio player which is controlled over Wi-Fi from my android mobile phone or my android Dashboard Multimedia System.
> I would suggest for better quality to send direct digital audio to APL instead of going into your head unit’s digital in and then from its RCA output to APL, as like this you have one DA-AD conversion less . In case your APL does not have USB input there are SPDIF output piggyback boards in the market for Raspberry which you can use to send direct coax digital to APL.


Ok thanks... I will see if that works with an iPad as the controller.


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## subterFUSE (Sep 21, 2009)

I am about to order one of these.

Should I get the 48 kHz or 96 kHz model?


My source will be the Sony RSXGS9 head unit via analog RCA cables.


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## Holmz (Jul 12, 2017)

subterFUSE said:


> I am about to order one of these.
> 
> Should I get the 48 kHz or 96 kHz model?
> 
> ...


My reasoning for the 48kHz was:
1) I doubt I hear well up high
2) I doubt that the filter banks are long enough to work well for a subwoofer.

Of course searching for the truth would be best.
Raimonds should be the one to opine on the matter.


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## subterFUSE (Sep 21, 2009)

So less taps at 96 kHz?

Is it half the taps?

4096 seems to be a bare minimum number of taps for any chance of EQ in the sub range. More would be better.


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## Holmz (Jul 12, 2017)

subterFUSE said:


> So less taps at 96 kHz?
> 
> Is it half the taps?
> 
> 4096 seems to be a bare minimum number of taps for any chance of EQ in the sub range. More would be better.


Same number of taps, so it is more like the bin's frequency width at 96k is twice as wide as the bin's width at 48k. (I would need to break out the book, but I a man pretty sure it is conceptually correct.)

The ideal (I think) would be to resample the sub channel to 1 or 2 k-samples/sec and then use 128 bins (or something on that order).

In any case I figured 48k was enough for me.


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## subterFUSE (Sep 21, 2009)

So at 96 kHz you need 2 times as many taps for the same frequency resolution at 48 kHz?


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## Holmz (Jul 12, 2017)

subterFUSE said:


> So at 96 kHz you need 2 times as many taps for the same frequency resolution at 48 kHz?


I'd have to crack open the books.

But I am pretty sure that for an identical filter response, the higher sample rate will need twice the number of taps.

Whether this is a problem, is only manifested as one approaches DC.
Which as where people generally also have the most issues with group delay as wild ported boxes are used... and where the ears have least amount of sensitivity to group delay.

Basically I am saying that do not know the answer, but these were my general thoughts.

If I was you and wanted some facts, I would suggest input from Raimonds is probably a good to get.


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## Raimonds (Jun 13, 2014)

subterFUSE said:


> So at 96 kHz you need 2 times as many taps for the same frequency resolution at 48 kHz?





Holmz said:


> I'd have to crack open the books.
> 
> But I am pretty sure that for an identical filter response, the higher sample rate will need twice the number of taps.
> .


Hi, yes it is so!


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## subterFUSE (Sep 21, 2009)

Raimonds said:


> Hi, yes it is so!




So does this mean the 96 kHz version has the same number of taps as the 48 kHz version?

So the 48 kHz version can process deeper bass?


Sent from my iPhone using Tapatalk


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## Holmz (Jul 12, 2017)

subterFUSE said:


> ...
> So the 48 kHz version can process deeper bass?


I believe the answer is not exactly "it is so"...

But rather that if the group delay is wildly swinging, then the frequency bins are closer, so it better follows with a finer/smaller "frequency (bin) resolution".

So maybe not deaper, but better when the group delay is wild...(?)


I could imagine that someone wanting 96k and also using a sealed box or IB, in a home setting, and if using hi-def digital, or old school LPs... then 96k might make more sense.

And I could also imagine someone with a ported box using 48k in either a home or car. Particularly if they use CDs that have a 22kHz bandwidth, which is less than the 24kHz that a 48k-samples/sec DSP yields.


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## Raimonds (Jun 13, 2014)

subterFUSE said:


> So does this mean the 96 kHz version has the same number of taps as the 48 kHz version?
> 
> So the 48 kHz version can process deeper bass?
> 
> ...


It is not about deeps. All the bass will be processed in high quality
It is about "sharpness" of details on equalizer`s curve in very very low frequency range.
They will be slightly smoothed when you will be close to limits.
Some tips should be used in Workshop to achieve all the sharpness available from particular number of taps/coefficients.


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## Gabriel (Nov 7, 2017)

Raimonds said:


> It is not about deeps. All the bass will be processed in high quality
> It is about "sharpness" of details on equalizer`s curve in very very low frequency range.
> They will be slightly smoothed when you will be close to limits.
> Some tips should be used in Workshop to achieve all the sharpness available from particular number of taps/coefficients.


Now would we need a line driver as the Rasp-Pi has such a low output voltage? I know whenever I had CARPC in past a line driver was necessary! Are class d amps that much more efficient in converting the heat (energy) into power?


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## Raimonds (Jun 13, 2014)

Gabriel said:


> Now would we need a line driver as the Rasp-Pi has such a low output voltage? I know whenever I had CARPC in past a line driver was necessary! Are class d amps that much more efficient in converting the heat (energy) into power?


APL1012 has sufficient gain (-10dBV mode) to work even with low signal level sources. You should not need additional line driver.
But it is highly advisable to use a digital signal path from source to APL1012.


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## subterFUSE (Sep 21, 2009)

My system that I am building will use the Sony RSXGS9 as the source. This means I need to use the analog RCA outputs from the source.

Does the APL1012 have adjustable input gain?

Why is digital preferred?

The Sony RSXGS9 has an optical out but it is lower quality than the analog out, and the digital output has no volume control. It is 0 dB FS always. No headroom for processing.


Sent from my iPhone using Tapatalk


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## ErinH (Feb 14, 2007)

subterFUSE said:


> Does the APL1012 have adjustable input gain?


I'm curious about this, too.


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## Jscoyne2 (Oct 29, 2014)

This thing is incredibly cool but unfortunately, due to the complicated tuning process and price. Im not sure how many people can actually afford/have the know how to use this.

Sent from my XT1710-02 using Tapatalk


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## Hanatsu (Nov 9, 2010)

Technically you can set gain in the input EQ file from workshop.


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## Holmz (Jul 12, 2017)

Jscoyne2 said:


> This thing is incredibly cool but unfortunately, due to the complicated tuning process and price. Im not sure how many people can actually afford/have the know how to use this.
> ...


With all I have read about the woes of most DSP tuning attempts, and with Jazzi and Skizer guides being praised as a golden light... it seems like "the know how" does not come naturally.

Unfortunatly I will not be able to tell how complicatied it is, in comparison to any other DSP.


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## Hanatsu (Nov 9, 2010)

Jscoyne2 said:


> This thing is incredibly cool but unfortunately, due to the complicated tuning process and price. Im not sure how many people can actually afford/have the know how to use this.
> 
> Sent from my XT1710-02 using Tapatalk


Let's say you're new to tuning audio systems, I'd say this is in fact easier than a conventional DSP (not counting those auto setup ones).

The EQ process is automated and very accurate. Delay values are shown directly in TDA and no need to fiddle around with asymmetrical crossover settings etc. Since crossover settings are applied with the EQ filter it sort of "thinks for you". 

The DSP in a system are among the most vital parts for a great SQ system and if you got the money - it's definitely worth it. You get really powerful software combined with a really nice FIR-based hardware unit.


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## Hanatsu (Nov 9, 2010)

Holmz said:


> With all I have read about the woes of most DSP tuning attempts, and with Jazzi and Skizer guides being praised as a golden light... it seems like "the know how" does not come naturally.
> 
> Unfortunatly I will not be able to tell how complicatied it is, in comparison to any other DSP.


Mastering the setup part of a conventional DSP can be complicated. You absolutely need to know how to measure properly, average results, re-measure with speakers playing together etc. In the very least it's more time consuming. I honestly setup the APL 1012 in perhaps 20-30min with good results the first time I used it and I had never used the software until that time. (I just used my old T/A values though).


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## Holmz (Jul 12, 2017)

Hanatsu said:


> Mastering the setup part of a conventional DSP can be complicated. You absolutely need to know how to measure properly, average results, re-measure with speakers playing together etc. In the very least it's more time consuming. I honestly setup the APL 1012 in perhaps 20-30min with good results the first time I used it and I had never used the software until that time. (I just used my old T/A values though).


I am still working through getting parallels up and running on OSX, but I might switch to Fusion.

In any case I do not have the mic yet.

What do you recommend for the measurements Hanatsu?


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## subterFUSE (Sep 21, 2009)

So I have the APL1, and I always have to use a basic USB mic with Workshop.

For some reason, I have a hard time getting workshop to work with my ASIO soundcard.


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## Hanatsu (Nov 9, 2010)

Holmz said:


> I am still working through getting parallels up and running on OSX, but I might switch to Fusion.
> 
> In any case I do not have the mic yet.
> 
> What do you recommend for the measurements Hanatsu?


I don't trust USB mics, from what I've seen some have noise issues related to the actual USB interface. The ECM8000 is popular. Requires a calibration though. I use a Sonarworks XREF20 now, it came with a cal file - although Raimonds calibrated it for me so it's good for 30-40kHz or so.

MiniDSP got their UMIK, also comes with cal files, Dayton got one too but I got no experience with those.


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## Hanatsu (Nov 9, 2010)

subterFUSE said:


> So I have the APL1, and I always have to use a basic USB mic with Workshop.
> 
> For some reason, I have a hard time getting workshop to work with my ASIO soundcard.
> 
> ...


Weird, ASIO works for me. Got a Scarlett 2i2 external preamp.


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## subterFUSE (Sep 21, 2009)

I have a Tascam 1608.

I use 5 Behringer ECM8000 mics with Smaart and SysTune for my normal tuning. But since I can’t get the Tascam to work with Workshop, I use a UMIK-1 for that. Works fine.


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## Holmz (Jul 12, 2017)

I see the Behringer ECM8000 mic, but there must be come phantom power and then USB connector to get it into the computer...
What else do I need?

Or do I shove it into the mic input?
With some a phantom power and then Neutrik to 1/8" connector?


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## dgage (Oct 1, 2013)

For quick measurements, I also use the Scarlett 2i2 mic preamp though for more extensive measurements I pull out the MOTU 1248 mic preamp.


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## Raimonds (Jun 13, 2014)

subterFUSE said:


> My system that I am building will use the Sony RSXGS9 as the source. This means I need to use the analog RCA outputs from the source.
> 
> Does the APL1012 have adjustable input gain?
> 
> ...


The input gain is adjustable in hardware - +4dBu, -10dBV
The Volume controle is working as gain as well. Becouse it is close to the input in the units structure.
The gain of input equalizer stage can serve this function as well (as Hanatsu mentioned).
The post crossover gain (pot on PCB) is also avilable to adjust overall gain of the system.

The digital is prefered becouse it reveals all the beauty of APL1012.
It removes two converters and pretty long (usually) cables (with respective noise) 
which require advanced engineering skills to make connection right way to exploit balanced properties of it.
But, if your particular setup requires to use an analog in, you are welcome to use it.
It is almost best available.

There is no problems with full scale digital signals because the Volume control is right after the input.




Jscoyne2 said:


> This thing is incredibly cool but unfortunately, due to the complicated tuning process and price. Im not sure how many people can actually afford/have the know how to use this.


It is time to ask your friendly car audio shop for help in that.

Regarding measurement mic and sound card.
Yes, one of my mics is ECM8000 for almost 20 years ...
It requires phatom power but no problem.
Such very small sound card serves all the needs:
https://www.thomann.de/intl/lv/the_tbone_micplug_usb.htm?ref=prod_rel_190404_0
There are lot of other mics offered.
Any of them should work, especially, if the correction file is available.


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## Holmz (Jul 12, 2017)

dgage said:


> For quick measurements, I also use the Scarlett 2i2 mic preamp though for more extensive measurements I pull out the MOTU 1248 mic preamp.


Can I use the "Focusrite Scarlett Solo USB Audio Interface (2nd Gen)"?

Should the USB not work, then it looks like could also come out of the Headphone jack to a 1/2"to 3.5mm and then into the mic port on the MacBook...


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## Bnlcmbcar (Aug 23, 2016)

ifi Audio has 2 products that will tackle any computer USB port noise gremlins.

The iDefender 3.0:
http://ifi-audio.com/portfolio-view/accessory-idefender3-0/

The iSilencer 3.0:
http://ifi-audio.com/portfolio-view/accessory-isilencer3-0/


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## dgage (Oct 1, 2013)

Holmz said:


> Can I use the "Focusrite Scarlett Solo USB Audio Interface (2nd Gen)"?
> 
> Should the USB not work, then it looks like could also come out of the Headphone jack to a 1/2"to 3.5mm and then into the mic port on the MacBook...


The Scarlett Solo would work just fine. I took a quick look at the specs and they were basically the same. I bought the 2i2 because sometimes I do a hardwired loopback to compare the signal output (if analog) to the mic measurement.


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## Jscoyne2 (Oct 29, 2014)

So. Whats the total price on this thing to have a fully running set up? Someone said earlier $1800 euros so thats $2k and then how muxh for tda and whatever else?

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## Holmz (Jul 12, 2017)

Jscoyne2 said:


> So. Whats the total price on this thing to have a fully running set up? Someone said earlier $1800 euros so thats $2k and then how muxh for tda and whatever else?


One could always start out with the software and see if it was to their liking?
It is more significant in cost than just sticking a toe in the water... more like the whole foot. But it could be a way to ease into it. Maybe it would also work for miniDSP FIRs?

As far as the DSP part....
I found three or four units that were FIR based.
All pass filters go a ways towards addressing phase, as well as speakers that are well behaved and crossovers that do not have wild phase excursions. However FIR based DSPs are techicnically a more straight forward and better approach than IIR based DSPs.

... whether one can actually hear the difference, is another discussion.
There are undoubtedly good IIR based systems, and also systems running no DSP. So a FIR based DSP is not exactly required.


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## Jscoyne2 (Oct 29, 2014)

Holmz said:


> One could always start out with the software and see if it was to their liking?
> It is more significant in cost than just sticking a toe in the water... more like the whole foot. But it could be a way to ease into it. Maybe it would also work for miniDSP FIRs?
> 
> As far as the DSP part....
> ...


Doesn't answer the question

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## crackinhedz (May 5, 2013)

Jscoyne2 said:


> Doesn't answer the question
> 
> Sent from my XT1710-02 using Tapatalk


Message Raimonds, he's not opposed to offers. Great guy to work with!

I already had TDA, Workshop etc so I cannot speak on the price, but again, talk to Raimonds.


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## Holmz (Jul 12, 2017)

Jscoyne2 said:


> Doesn't answer the question


If you reread it, you find it does answer a lot towards the question.
2) understanding whether there a particular issue that a FIR would work toward fixing.
1) working towards using the software, before committing to a FIR-based DSP... or with a current system.

These can be somewhat decoupled, and if the software indicates that nothing is to be gained over the current system, then spending up on a different DSP may not yield much.

As I was starting from scratch, it was an easier commitment than for someone who already has a Helix for example.

I cannot say it is a good or bad idea for anyone else unless I know what their goals and issues are, so as to determine whether it could address their needs.

These are the questions that could precede the question of whether it is the best value approach towards those meeting those needs.

(Or if your system is already stunning, then it will not make 1800 euros worth of difference, or whatever the price is... it will make no difference)


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## Spartak802 (Jul 18, 2016)

Holmz said:


> Or if your system is already stunning, then it will not make 1800 euros worth of difference, or whatever the price is... it will make no difference


Sorry for misunderstanding... If you are glad for your existing setup and have 100% satisfaction with the carsound quality, why you spend your time on this forum instead of having pleasure by listening your stunning system?

APL technology has changed mind of many enthusiasts despite their pervious experience in the audio World. Carsound is not an exclusion.

Sure, 1800 euros is a lot of money and one has to think about before investing in new components... But let listen first to what APL technology could make before final decission. 

APL1012 for sure is not for newbees, but with the help and support from Raimonds, nothing will be impossible even for them.

But again, perhaps it is simply a wrong timing to change the stunning system. Especially if you made it by yourself and now it is time to enjoy it. Fully understanding you in this point. Good luck!


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## subterFUSE (Sep 21, 2009)

The APL1 instantly makes my soundstage wider in my car with Helix DSP Pro2, and I am an expert tuner of the Helix.

Before the APL1, my stage boundaries are pillar to pillar.

With the APL1, instantly the stage goes to the inside edges of the side mirrors, outside of the pillars a good 6 inches on each side.

I don’t know why or how, but it certainly works. 


Sent from my iPhone using Tapatalk


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## Holmz (Jul 12, 2017)

Spartak802 said:


> Sorry for misunderstanding... If you are glad for your existing setup and have 100% satisfaction with the carsound quality, why you spend your time on this forum instead of having pleasure by listening your stunning system?
> 
> APL technology has changed mind of many enthusiasts despite their pervious experience in the audio World. Carsound is not an exclusion.
> 
> ...


If you think that we are "understanding each other in this Point", Then My post must have been poorly worded...

1) I already got an APL - it appears that I must think it was worth having.
2) as the APL is my first DSP, then maybe it is occasionally for noobies? (or... One has to start somewhere, so why not with this DSP?)

With that in mind, perhaps my thoughts directed to Jscoyne2 may be perceived as being in the spirit of a helpful discussion towards his questions? (I can only try to give my perspective of reasoning on the subject, which I may have failed to do in a clear fashion)


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## Hanatsu (Nov 9, 2010)

Jscoyne2 said:


> So. Whats the total price on this thing to have a fully running set up? Someone said earlier $1800 euros so thats $2k and then how muxh for tda and whatever else?
> 
> Sent from my XT1710-02 using Tapatalk



All software are included.


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## crackinhedz (May 5, 2013)

General question; Since I am using a ported subwoofer my port tune is 33Hz, so I have the SSF on the amp set around 27Hz.

When using Workshop, the measurement I assume will reflect the percieved bass slope and try to compensate to match an EQ curve, which usually keeps a flat response to 20Hz, thus defeating the purpose of the ssf? So am I wrong in thinking when making my own custom EQ curves, I should build a high pass about 25Hz into the curve so APL doesnt try to boost below my ssf? 


Also, the APL1012 the clipping lights on the output panel - what exactly does this indicate (aside the obvious), and how to fix should it be clipping?


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## Raimonds (Jun 13, 2014)

crackinhedz said:


> General question; Since I am using a ported subwoofer my port tune is 33Hz, so I have the SSF on the amp set around 27Hz.
> 
> When using Workshop, the measurement I assume will reflect the percieved bass slope and try to compensate to match an EQ curve, which usually keeps a flat response to 20Hz, thus defeating the purpose of the ssf? So am I wrong in thinking when making my own custom EQ curves, I should build a high pass about 25Hz into the curve so APL doesnt try to boost below my ssf?
> 
> ...


You should direct the Workshop not to touch (equalize) the curve in your SSF region. It means - set to "flat" that region with some gain. That is avilable by use of "low frequency limit" parameter in Workshop which is setting the gain for LF region and making curve flat.
You should add the post crossover gain in case of clipping at outputs of crossover`s filters (leds on APL1012 front panel). It is available by use of the potenciometer on PCB and can be set from 0 to +18dB. The default setting is about +10dB.


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## Bnlcmbcar (Aug 23, 2016)

Please correct me if I’m misunderstanding. But from what I gather, the measuring technique has one painting the mic across the front of the dash while sweeps are run... 

Does this mean that the final APL 1012 tune is a viable 2 seat tune? Or is it still 1 seat orientated?


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## Raimonds (Jun 13, 2014)

APL measurement/tune are not seat (place) orientated in general.
But there is option to add some kind of a final tune (after everything is done) for particular position/seat in case if some drivers have an explicit directivity which must be taken into account.


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## Bnlcmbcar (Aug 23, 2016)

Raimonds said:


> APL measurement/tune are not seat (place) orientated in general.


As in when your done with the workflow described in the manual, and performed measurements as uniform as possible, the soundstage will sound equivalent whether one is sitting the the driver seat or passenger seat of a car?


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## Durgesh (Sep 18, 2014)

Bnlcmbcar said:


> As in when your done with the workflow described in the manual, and performed measurements as uniform as possible, the soundstage will sound equivalent whether one is sitting the the driver seat or passenger seat of a car?


Stage will also depend on distance (time alignment ) values you input relative to seating position.

Sent from my SM-G950F using Tapatalk


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## Hanatsu (Nov 9, 2010)

crackinhedz said:


> General question; Since I am using a ported subwoofer my port tune is 33Hz, so I have the SSF on the amp set around 27Hz.
> 
> 
> 
> ...



If Fb is at 33Hz I’d remove the highpass. My sub is tuned at 32 and I never used any filters on it. Otherwise you can create the filter in workshop after measuring.


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## captainobvious (Mar 11, 2006)

Bnlcmbcar said:


> As in when your done with the workflow described in the manual, and performed measurements as uniform as possible, the soundstage will sound equivalent whether one is sitting the the driver seat or passenger seat of a car?





You have to keep in mind that you are setting time alignment to a particular seat in the dsp. So if you set your time alignment setting up for the drivers seat and you run the measurements and corrections it will NOT sound equivalent between the driver and passenger seats because of your time alignment settings.


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## Bnlcmbcar (Aug 23, 2016)

captainobvious said:


> You have to keep in mind that you are setting time alignment to a particular seat in the dsp. So if you set your time alignment setting up for the drivers seat and you run the measurements and corrections it will NOT sound equivalent between the driver and passenger seats because of your time alignment settings.


Thanks that makes more sense.


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## VincMartel (Mar 21, 2017)

What if I tune with an APL with no time alignement? Will I get a decent 2 seats system? 

Probably too simple but just asking.


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## subterFUSE (Sep 21, 2009)

VincMartel said:


> What if I tune with an APL with no time alignement? Will I get a decent 2 seats system?
> 
> Probably too simple but just asking.




No. Not unless you have path lengths within about 8 inches or so.


Sent from my iPhone using Tapatalk


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## captainobvious (Mar 11, 2006)

You can get great tonality, but don't expect to get accurate imaging.


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## robi17 (Aug 11, 2018)

this is very nice and informative discasion


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## seafish (Aug 1, 2012)

robi17 said:


> this is very nice and informative discasion


The occasion for a "discussion" is over since you are merely a chatbot.


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## Durgesh (Sep 18, 2014)

Hanatsu said:


> Sorry for the wait. A basic workflow is already included in the pdf file.
> 
> https://drive.google.com/open?id=1sMad-ujMQbrxfMOzGlNTeZpGExsYpMcx
> 
> Edit: Dropbox don't work for some reason...


Hi Hanatsu

Could not follow the 5th step of the work flow in the manual made by you.

.......5. Add your target curve by use of the unit’s input EQ. Use the 5th unit field of C5 software to manage that......

Pls elaborate this (step by step if possible).

Thanks

Sent from my SM-G950F using Tapatalk


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## Hanatsu (Nov 9, 2010)

Durgesh said:


> Hi Hanatsu
> 
> Could not follow the 5th step of the work flow in the manual made by you.
> 
> ...


It chooses whatever "mode" you got on the switch to send filters to. If you import a fir file of a fullrange measurement you can use it as "overall EQ". You can import different curves you choose in workshop to use. 0 and 6-15 are available iirc. 1-5 are used by the 10 outputs of the unit. Just import it as you would with any fir filter.


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## crackinhedz (May 5, 2013)

Anyone doing a video tutorial? Step by step workflow process, Workshop/TDA/C5

I know I can stumble my way to success, but its always helpful to see someone else go at it.


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## Durgesh (Sep 18, 2014)

Hanatsu said:


> It chooses whatever "mode" you got on the switch to send filters to. If you import a fir file of a fullrange measurement you can use it as "overall EQ". You can import different curves you choose in workshop to use. 0 and 6-15 are available iirc. 1-5 are used by the 10 outputs of the unit. Just import it as you would with any fir filter.


Do we need to apply target curve (Mp1) in the workshop to the initial/raw measurements for each way/speaker. 

And then import the measuments in respective ways of C5.

Sent from my SM-G950F using Tapatalk


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## Hanatsu (Nov 9, 2010)

Durgesh said:


> Do we need to apply target curve (Mp1) in the workshop to the initial/raw measurements for each way/speaker.
> 
> And then import the measuments in respective ways of C5.
> 
> Sent from my SM-G950F using Tapatalk



No. Target curve is set in C5. I’ll try do a better explaination later


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## aholland1198 (Oct 7, 2009)

Someone want to do me a solid and see if the iPhone will work with the usb input?! Raimonds didn’t know. 


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## Raimonds (Jun 13, 2014)

Dear Friends, yes, please, let's test is iPhone working with HD USB input.
It works with W10, MAC OS version 10.6.4 and above,Linux with UAC2 compliant kernel.
But iPhone?


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## subterFUSE (Sep 21, 2009)

Whoever tests it will need an Apple USB Camera Adapter.


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## LumbermanSVO (Nov 11, 2009)

subterFUSE said:


> Whoever tests it will need an Apple USB Camera Adapter.


This is the part: https://www.apple.com/shop/product/MK0W2AM/A/lightning-to-usb-3-camera-adapter

BTW, I use these, with USB to Ethernet dongles, to put iPads on a wired networks.


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## aholland1198 (Oct 7, 2009)

I have the adapter. I’ve used it for my helix usb input for two or so years. Just need to k ow if it works on this dsp. 


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## Durgesh (Sep 18, 2014)

Hi

Finally installed and tuned my 1012 which was purchased in April. 

Extremely happy with the results. The sound is so much detailed now. Centre staging is almost perfect. Never heard such sound before. And this is just with basic tune and left-right TA by ear.....TA and crossover check between each way still to be learnt and done with Tda.

Thank you Hanatsu, LumbermanSVO , An.vamv and all others who have shared the setup procedure and guide. 

And many thanks to Raimonds...the mastermind behind this unit.


Sent from my SM-G950F using Tapatalk


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## subterFUSE (Sep 21, 2009)

Does anyone have the APL1012 and is using a car audio head unit as their source, with analog inputs into the APL?

I’m considering the 1012 with a Sony RSX-GS9 as the source, running analog RCA from the Sony.

I’m curious about the input voltage range on the APL1012.


Sent from my iPhone using Tapatalk


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## Hanatsu (Nov 9, 2010)

subterFUSE said:


> Does anyone have the APL1012 and is using a car audio head unit as their source, with analog inputs into the APL?
> 
> I’m considering the 1012 with a Sony RSX-GS9 as the source, running analog RCA from the Sony.
> 
> ...


I currently use it with analog input from a P99. Never bothered to check the output levels so I run mine on 58/62 volume on my P99 for some small headroom, no distortion on the output of APL. Can't say exactly how far you can push it though. I would assume it handles a 4V input.


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## thehatedguy (May 4, 2007)

Getting mine going too.

Need to check back and read this thread where I last left it.


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## crackinhedz (May 5, 2013)

```

```
Waiting for warmer weather to install mine, still hoping someone makes a video tutorial of the process. Not that I couldn't fumble my way through, but helpful visual tips along the way by those more savy would help tremendously. I still have trouble figuring out TDA :blush:


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## captainobvious (Mar 11, 2006)

crackinhedz said:


> Waiting for warmer weather to install mine, still hoping someone makes a video tutorial of the process. Not that I couldn't fumble my way through, but helpful visual tips along the way by those more savy would help tremendously. I still have trouble figuring out TDA :blush:





It is significantly more advanced/difficult than using the APL1, which was already very difficult for many people. Just a forewarning :blush:


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## captainobvious (Mar 11, 2006)

aholland1198 said:


> Someone want to do me a solid and see if the iPhone will work with the usb input?! Raimonds didn’t know.
> 
> 
> Sent from my iPhone using Tapatalk





Raimonds said:


> Dear Friends, yes, please, let's test is iPhone working with HD USB input.
> It works with W10, MAC OS version 10.6.4 and above,Linux with UAC2 compliant kernel.
> But iPhone?







Sorry for the delay in response. The answer to this question is yes. You can connect an iPhone/iPad device with the camera connection kit part to extract digital signal sent over usb into the unit and play audio files with the player app on your device.


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## hdrugs (Sep 7, 2009)

Does it has 2nd order crossovers? Or only runs in 4th and 8th


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## Holmz (Jul 12, 2017)

hdrugs said:


> Does it has 2nd order crossovers? Or only runs in 4th and 8th


I am pretty sure it runs FIR, so I am a bit confused with the 2nd order crossover question.


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## hdrugs (Sep 7, 2009)

Ive never used fir filters

So they always go by 4th order?


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## Holmz (Jul 12, 2017)

hdrugs said:


> Ive never used fir filters
> 
> So they always go by 4th order?


Better google FIR and IIR.
They are different beasts.

A large tap fir is more like a 20 or 100th order.

The phase of a FIR can be zero phase, which is more interesting than the steepness.


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## captainobvious (Mar 11, 2006)

hdrugs said:


> Does it has 2nd order crossovers? Or only runs in 4th and 8th





No, the unit uses 4th order or 8th order crossovers only.


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## Raimonds (Jun 13, 2014)

hdrugs said:


> Does it has 2nd order crossovers? Or only runs in 4th and 8th


Thanks for the question!
Filters can be any kind that you can image and create as the unit is FIR machine.
C5 program`s Crossover tab offers 4th and 8th order as mostly usable crossover filters.
But you can use Parametric EQ to create even 2nd order filters by use of respective HP and LP filters. Please find pictures attached!
You can create any kind of your custom curve by use of any tool you like that exports FIR filters in wave format files. But it will be not easy to manage all that.


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## captainobvious (Mar 11, 2006)

subterFUSE said:


> Does anyone have the APL1012 and is using a car audio head unit as their source, with analog inputs into the APL?
> 
> I’m considering the 1012 with a Sony RSX-GS9 as the source, running analog RCA from the Sony.
> 
> ...



John- 



The manual states “-10dBV” “+4dBu” for input.

There are 2 jumper positions on the PCB for input sensitivity to switch between these. There are also 2 other methods for adjusting in a different way. The volume potentiometer is inserted close to the input in the PCB design so it can act as a sort of input gain adjustment as well. Finally, you can adjust the "EQ Zero Level" on the actual individual workshop measurements to lower the gain as well. Although this last method is making the adjustment in software, not hardware- so if you were to put the APL1012 in "bypass" mode, that reduction in gain would not be there, causing you to have more volume in bypass mode vs enabled.


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## sareea (Oct 19, 2016)

I just went through this post.
Great info ! 
Really interesting


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## oabeieo (Feb 22, 2015)

There goes another 3hrs of my life , I missed this completely 
I can see this definitely does more than the opendrcda8 
Which I’m about to purchase (maybe I need to rethink) 

This might be good for what I want to do 

Is this LPCM?


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## Holmz (Jul 12, 2017)

oabeieo said:


> There goes another 3hrs of my life , I missed this completely
> I can see this definitely does more than the opendrcda8
> Which I’m about to purchase (maybe I need to rethink)
> 
> ...


It was a chin scratcher for me too.
One can, in theory, remove reflections with a FIR.


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## oabeieo (Feb 22, 2015)

subterFUSE said:


> For rear fill, you need to mix the left and right inputs together into a mono signal, with one of the inputs polarity inverted.
> 
> So, Left + and Right - mixed to the left rear speaker.
> 
> ...


Isn’t tjat the same as hooking up just the positives of a 2ch amp to a speaker? 
Like echoing backgrounds, or is that what hanatsu is saying quadraphonic 






Hanatsu said:


> Can always physically connect speakers in 70’s fashion ”quadrophonic mode”. I’ve done it without mono mixing before, I swapped channels and inverted one. Can’t think of a good way of doing it on this DSP.
> 
> I’m going with the ”signal difference” thing. Psuedo rear ambience or whatever it’s called
> 
> Edit; (As spl said above, tapatalk keeps bugging can’t see all posts...)


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## oabeieo (Feb 22, 2015)

Holmz said:


> It was a chin scratcher for me too.
> One can, in theory, remove reflections with a FIR.


LPCM (basically fixed point) like the apl1 

Floating is IEEE


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## Holmz (Jul 12, 2017)

oabeieo said:


> LPCM (basically fixed point) like the apl1
> 
> Floating is IEEE


"Chin scratcher" was referring to APL vs the others.


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## oabeieo (Feb 22, 2015)

Holmz said:


> "Chin scratcher" was referring to APL vs the others.




Yeah I was referring basically to this 10ch vs the apl1


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## captainobvious (Mar 11, 2006)

This 10 channel can be thought of as 5 APL1's in a single chassis pcb design with some additional (great) dsp features.


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## LumbermanSVO (Nov 11, 2009)

captainobvious said:


> This 10 channel can be thought of as 5 APL1's in a single chassis pcb design with some additional (great) dsp features.


Yeah, C5 adds some pretty neat DSP features. You can also get more info about what Workshop is doing. Fir example, this is a graph of an FIR file created with Workshop:


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## oabeieo (Feb 22, 2015)

No video yet....uugh ! 

I’m installing the 1012 now , C5 doesn’t seem to load up right I get license error 
Can’t connect to dsp
Stuck can’t do anything as of now

Is there a way I can load the files using cmd or anything, I’ll just use rephase files if I have to, although would be nice to get it to work. Are fir on a sd card somewhere? Do I absolutely have to connect using C5? 


Emailed Raimonds......waiting for response. 
Sorry a little grouchy just worked a 20hr shift trying to get this to connect and nothing.
Well 10 of it on Apl. 



Also what is this about 

Ch 1,2 =1w
3,4=2w
5,6=3w
7,8=4w
9,10=5w

Is this mean channel 9,10 is 2.83v (1w) x5 output on analog rca that’s like 11.5v am I understand this right,


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## oabeieo (Feb 22, 2015)

Anyone want to make a little $$$$ to provide phone support? 

This waiting on emails is no fun I think if someone could just talk to me and walk me through some of the setup I could breeze through this 
I need to get this thing working stat


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## tonynca (Dec 4, 2009)

Man you have too many DSP. minidsp now this


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## crackinhedz (May 5, 2013)

Here is the user guide:



> How the "units" are used:
> Example: in a typical 3-way car setup:
> Unit 1 (1-way): Subwoofer L/R
> Unit 2 (2-way): Midbass L/R
> ...


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## LumbermanSVO (Nov 11, 2009)

oabeieo said:


> No video yet....uugh !
> 
> I’m installing the 1012 now , C5 doesn’t seem to load up right I get license error
> Can’t connect to dsp
> ...


Was the software installed to a USB drive? That's the best place to install it so it can be used on multiple computers. Was the license file sent to Raimonds after the install? If not, that needs to be done, and he will send a file back, and then you'll be good to go. His DRM is pretty strong on these, presumably because a previous business partner screwed him over.

As far as the channels go, you are looking at the crossover page, correct? The 1w, 2,w, etc,.. refers to have many ways your setup is. 1 being the subs and working your way to higher frequencies as you go up in numbers. 

I've sent you a PM with my number for phone support. Do you have TeamViewer installed and an internet connection good enough to support it?


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## oabeieo (Feb 22, 2015)

LumbermanSVO said:


> Was the software installed to a USB drive? That's the best place to install it so it can be used on multiple computers. Was the license file sent to Raimonds after the install? If not, that needs to be done, and he will send a file back, and then you'll be good to go. His DRM is pretty strong on these, presumably because a previous business partner screwed him over.
> 
> As far as the channels go, you are looking at the crossover page, correct? The 1w, 2,w, etc,.. refers to have many ways your setup is. 1 being the subs and working your way to higher frequencies as you go up in numbers.
> 
> I've sent you a PM with my number for phone support. Do you have TeamViewer installed and an internet connection good enough to support it?



Aaahhhh thank you!


I will go grab my pc and call in 10min 

No no team viewing and my shop internet is so slow it’s on a old t1 that is on a CAN to 5 diffrent states with a server in Seattle. I’m lucky to see 1mbps more like 500k 

But yes thank you!


It’s not for me , I have a guy from the forums (here) that sent me his car 2000miles away to do his install, and boy oh boy we went to the 9s with this thing, it’s going to be kick ass when it’s done.

Needless to say he and I are on the clock and I want to do like the best job ever on tuning his car , and any info just getting fluent with apl is so much appreciated, he’s getting it shipped back his home state soon and I’m running short on time. 

Guys thank you for any help.


Ramionds is great , he’s helping too , I hope he can understand the amount of hours I have into this build and the urgency of getting it done promptly. 
And hope he knows I do appreciate his help. 


Okay the “ways” I get it now. I like the terminology “channels” a lot better , but it don’t matter now I get it, i was like what is this broken English or something lol. Now it makes sense...... 


I was running some sims on rephase , this thing has decent power in fir .....very decent, more than minis HDs not quite a opendrc but for 10ch this thing is a powerhouse. I definitely worried about doing everything in fir, eq and phase , 
Hope I don’t run out of coefficients at 96k tuning horns. 

The IB should need very little eq it should play mostly flat on its own , no box to cause problems. 

I’m very curious how this thing sounds!


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## crackinhedz (May 5, 2013)

Any pics or a build log? Would love to see the final install!

Hope to install my APL 1012 within the next month, been sitting boxed up for almost a year need to get this badboy going. Dont suppose you have the time to make a video on the 1012 process? Or maybe some screen pics of the settings used/not used. Just to wrap my head around it. ?


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## oabeieo (Feb 22, 2015)

crackinhedz said:


> Any pics or a build log? Would love to see the final install!
> 
> Hope to install my APL 1012 within the next month, been sitting boxed up for almost a year need to get this badboy going. Dont suppose you have the time to make a video on the 1012 process? Or maybe some screen pics of the settings used/not used. Just to wrap my head around it. ?




TONS of pics 

I’m going wait for him to post a build log out of respect. I don’t want to be that guy (yet) his car his build, if he wants to share , although it’s pretty badass he should definitely at least show off them kicks! Got 60hrs just on kicks alone. 

This thing is going to be dope as ****


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## oabeieo (Feb 22, 2015)

What kind of latency numbers you guys been seeing ? 

I’m using a vxi amp for rears and it has a limited amount of delay on the outputs. 
And it’s not running through the apl , so I wil have to align with delay , 

Are you guys under 30ms


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## LumbermanSVO (Nov 11, 2009)

I haven't had latency issues with the APL1, haven't tested latency on the 1012 yet.

I have an APL1 on the mains in my home theater and have zero latency issues. If you don't have an answer later today, I'll swap it with my 1012 and see what I get. I suspect you have nothing to worry about.


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## crackinhedz (May 5, 2013)

Oabeieo, have any success with the tuning? If so how did you like the process and outcome?


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## oabeieo (Feb 22, 2015)

crackinhedz said:


> Oabeieo, have any success with the tuning? If so how did you like the process and outcome?


I’m super excited to try it 

Raimonds is mad at me for some reason

(I think he’s still mad at me for poking fun at the apl1, which was rather duschy of me , the 1012 is bass ass little dsp , I’m super excited to hear it 


I’ve waited three days now for him to send the license key, he won’t respond my emails now. Or he’s taking his sweet time,

Kinda sucks because this single job is like half my months pay and if I can’t deliver by end of month I’m ****ed! Would be so bad if I didn’t have 10 defendants that live off my paychecks. 


So I’m waiting, I hope he sends it soon. I’m I really think I’m going to like it.


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## Holmz (Jul 12, 2017)

oabeieo said:


> ...
> Kinda sucks because this single job is like half my months pay and if I can’t deliver by end of month I’m ****ed! Would be so bad if I didn’t have 10 defendants that live off my paychecks.
> ...


Were those 10 defendants convicted of guilt by association?


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## oabeieo (Feb 22, 2015)

Defendants dependents

Potato potato 

Lol


Heheheh 



All I have to say about now is John is a ****n stud!


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## rton20s (Feb 14, 2011)

oabeieo said:


> Defendants dependents
> 
> Potato potato
> 
> ...


----------



## High Resolution Audio (Sep 12, 2014)

I read this thread to the end. It sounds like an interesting product, but complicated software.


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## LumbermanSVO (Nov 11, 2009)

I work in the live event world with video gear, the software looks very similar to that. For example, download and run NovaLCT or dbstar. It’s pretty clear to me that Raimonds designed the software with the pro market in mind. Because of this, there is a steep learning curve. The bonus is that once you learn it, it can literally take just a couple minutes to measure, add filters, and send those filters to to the unit.

Also, if you want to change house curves, you don’t have to remeasure, just apply that curve to the original measurement, then send it to the unit.

The software looks intimidating, but is pretty easy to work with once you learn it.


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## oabeieo (Feb 22, 2015)

rton20s said:


>


Oh yes the mullet and everything..lol hahaha 

Isn’t that rick flare or something. Didn’t he become governor of Minnesota


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## oabeieo (Feb 22, 2015)

We ended up going a different direction, 
I got it to work, and it would crash. I got the 1st measurements done and it crashed 
I was trying to load crossovers and gave up. It just would boot me off 

John had informed me that the presets are eq only so that along with the crashing 
And the hourly rate we decided go with a minidsp , it does what the guy wants 
By having presets that can turn off channels, and has Dirac live so it’s still a win win 
And he has another device upstream also that has tons of fir ability so any residual group delay can be dealt with quite easily. I promised I wouldn’t spill the beans, but think million tap filters if one wanted. (Jel) 

I am super impressed with the TDA software, that works so good. Way better than I thought it would I am definitely thinking about buying a copy for me. 
I like tda very much. So much easier than reading impulses. 

I wish I coulda heard it tho. Dang it. And this guy will have more fun with the mini being all his amps work off one dsp instead of having the apl and that another dsp for his rears. 
And not having to worry about fir delay between the dsps . That makes tuning no fun, and if the apl has impulse offsets longer than 20.21ms the rears just wouldn’t have worked anyway. 

On the 1012 tho the crossovers have a cutoff frequency for both the sides of the crossover that can’t be altered? Meaning, if you cross the sub at 100 the mid is 100 and no way to enter it diffrent forced complimentary crossover.

Makes me wonder if he does forward-backward filtering on that, it would give better stopband attinuation , the inband amplitude would carry the error which could be eq if I remember reading about backward running fir. Possibly saves taps too....so where he wanted overlapping speakers wouldn’t have had to do the peq option on crossovers. Not that that’s horrible, just one more thing that made this not the best option for this install , although would have still worked in a roundabout way. Just seemed like a lot of things to manage and work just to turn off a speaker pair or change a crossover slope or time align to another dsp. 
But I digress, 

Very curious about this dsp tho. Still has a lot of good things going on, for a traditional multi-way it could be one of the best. Definitely top 3 in my book 

Another day I’ll get to work on one. Maybe I’ll buy one for a project.


I think the install drive was messed up I don’t know. But it all worked out well and worked out better for this particular setup


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## captainobvious (Mar 11, 2006)

oabeieo said:


> We ended up going a different direction,
> I got it to work, and it would crash. I got the 1st measurements done and it crashed
> I was trying to load crossovers and gave up. It just would boot me off
> 
> ...





The defined/locked in crossover junctions is a good thing. The APL1012 corrects each drivers response to perfectly (or as close as can be achieved) align its acoustic response with that electrical filter so you have essentially beautifully aligned crossovers between drivers. There would be no reason to underlap or overlap if you have perfectly summed response. Just use the EQ for that


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## LumbermanSVO (Nov 11, 2009)

I ran some REW sweep through mine to look at the phase response around the crossovers, it's REALLY cool to see perfectly flat phase through the crossover. If you use a LR crossover instead, you get the predictable phase shift. But if you then turn on Group Delay correction, the flat is perfectly flat again. I'll post up some screenshots tonight.


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## captainobvious (Mar 11, 2006)

LumbermanSVO said:


> I ran some REW sweep through mine to look at the phase response around the crossovers, it's REALLY cool to see perfectly flat phase through the crossover. If you use a LR crossover instead, you get the predictable phase shift. But if you then turn on Group Delay correction, the flat is perfectly flat again. I'll post up some screenshots tonight.



Yes I was surprised by that too. Also show the before and after FR correction of two drivers at the crossover point. It's quite good.


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## oabeieo (Feb 22, 2015)

captainobvious said:


> The defined/locked in crossover junctions is a good thing. The APL1012 corrects each drivers response to perfectly (or as close as can be achieved) align its acoustic response with that electrical filter so you have essentially beautifully aligned crossovers between drivers. There would be no reason to underlap or overlap if you have perfectly summed response. Just use the EQ for that



Steve he had like duplicate channels , horns and tweeters for example ,
Definitely don’t want to cross horns at 3k 

And that’s not all. There were a lot of speakers installed.
So that was my original point, like 2 seperate full systems of speakers in the front of this car plus all kinds of rears. 

Although in fir crossovers, you don’t have to have the same frequency either, you can actually build filters that overlap and they sound amazing. 
(Remember Steve, it’s linear phase so it don’t matter. The “crossover” is a amplitude only thing, all you do is make sure the driver is flat before applying the crossover and it playes perfect in time with the inband. The whole “crossover” mentality gets thrown out the window really. It opens a lot of doors 

You should listen to a overlapping filter , bring the speakers down like 3 or 6 dB so your output isn’t doubled. Than for an oactave and let them interact and blend at -6db(I think that’s where pos puts them) , than shut them down fast like 48db is nice.
You can find them in rephase, there’s a whole bunch of unique filter designs , there one oactave overlapping, 1/2 oactave , 2 oactave , Ramionds told me you can use rephase (I never got that far) 

He said the .fir ad .wav are the same and apl will read both 

So if it’s 24bit you would export 24bitLpCM.wav in rephase 
Throw it in a bank (oops I mean unit not bank) and enjoy. 

But in that instance you get the blending than steep slopes. 
So the outband interaction between speakers is almost non existent and the spekEkers are a oactave (or whatever) away so bye bye crossover interaction. 
It’s like my most favorite filter for a tweeter or a horn to a mid.
I don’t care much for them in the LF I like plain old LR4 or LR8 

Check it out tho


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## rton20s (Feb 14, 2011)

oabeieo said:


> All I have to say about now is *John is a ****n stud!*





oabeieo said:


> Oh yes the mullet and everything..lol hahaha
> 
> Isn’t that rick flare or something. Didn’t he become governor of Minnesota


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## oabeieo (Feb 22, 2015)

rton20s said:


>



Oooh yes yes. lol 
I remember now. Man I used to watch wwf (e) like every episode 
Back in the day , I can’t believe I forgot that. 
That was before super h came on the scene if I remember right 
That was the good ole days with Andre the giant and rick flare, and the hulk 

Wasn’t Vince McMahon a wrestler (actor) also back than


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## CDT FAN (Jul 25, 2012)

oabeieo said:


> Oh yes the mullet and everything..lol hahaha
> 
> Isn’t that rick flare or something. Didn’t he become governor of Minnesota


You're probably think of Jesse Ventura.

https://www.google.com/search?q=jes...UIEigC&biw=1440&bih=797#imgrc=k2jVrSIANf1QOM:


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## oabeieo (Feb 22, 2015)

CDT FAN said:


> You're probably think of Jesse Ventura.
> 
> https://www.google.com/search?q=jes...UIEigC&biw=1440&bih=797#imgrc=k2jVrSIANf1QOM:




That’s right....it is Jesse 

Man he turned into like a anti government, government worker. He went way out there.


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## LumbermanSVO (Nov 11, 2009)

rton20s said:


>


I'll never look like that, I scared my hairline away a long time ago! 

Here are a couple REW files to look at.

Index of /APL

The no loopback file is what I saw the other day when I was looking at things. I wanted to test delay through the unit as I fiddled with the crossovers, and it turned up some interesting phase stuff. Should I just ignore the phase in the loopback sweeps?

Tomorrow I'll toss the unit into by bedroom system and take some measurements of the actual audio using these crossovers. That system is a TV, a low end Yamaha receiver, some Acoustic Research bookshelf speakers, and a pair of W15GTI's running off a Crown XLS1002.


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## crackinhedz (May 5, 2013)

Well after my 1012 sitting in the closet for over a year, finally had the opportunity to get installed! 

Being that I have a 7" CarPC, the windows 10 resolution was giving me fit getting the GUI to work without crashing everytime. Finally found a fix by using "overide high dpi" option in the C5 exe properties. Still slight scaling issue, but very useable and no crashes. Also, for anyone having program installation issues, ensure using "Administrator" privilege, also to install into the root level of your drive, C: or a flash drive. Might save you headaches. Also once you install the program, you must send Raimonds the siteID file so he can make and send your License file.

Anyway, the C5 GUI is not too complicated, once you get to tinkering around, theres really not much to figure out. Its layed out well in three simple tabs (there are a couple other tabs, but they are very limited to use). The downside, some of the buttons are not explained in the instructions, and for a basic amateur like me its a little intimidating on why/how/when to use. But thats what the DIYMA community is for, and of course Raimonds is very helpful.

I have not had any time yet to measure etc, but set basic crossovers and got music playing just to know I could. Will spend some time another day working on the tuning.


Question for those with experience, I am using:

Sub (1 way)
Midbass (2way)
Midrange (3way)
Tweeter (4way)


So in crossover I have:

Number of ways= 4

f1/2= 60 (sub/midbass)
f2/3= 300 (midbass/midrange)
f3/4= 3500 (midrange/tweeters)
f4/5= ???

Not sure what I put here as I don't want to cap my tweeters? Do I just put 20,000? But I think only lets me input up to 18,000?


Thanks!


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## LumbermanSVO (Nov 11, 2009)

You shouldn't need to put anything in f4/5


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## crackinhedz (May 5, 2013)

Thanks, wasn't sure I could leave blank.


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## crackinhedz (May 5, 2013)

Anyone with the 1012 using optical have any trouble when turning volume down to 0 (zero)? If I mute or turn to 0, the audio cuts off and nothing when I turn volume up. I have to physically unplug the optical cable and plug back in. Im assuming its some sort of signal sensing, and at 0 must turn off the optical...

my old Mosconi 6to8v8 I believe had a button checked that kept optical on no matter what.. never had this issue. I dont see a comparable solution in the settings anywhere. 

I am using volume control from my steering wheel & carPC, not the volume knob included with 1012.

Thanks for any help in advance


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## Durgesh (Sep 18, 2014)

Hi
Anyone faced clipping issues while working in C5. 

It sometimes happens when I change delays or press the mute button while the music is playing. The associated speaker goes into nasty clipping/popping. The only way to stop is to turn off the car key. 

It has happened twice in a couple of days with my midbass driver. It has happened before also with other drivers.

I am using digital input with approx 50% volume of my digital source.

Sent from my SM-G950F using Tapatalk


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## crackinhedz (May 5, 2013)

No, not any sound distortion for me. As you might see my previous post, I do lose sound after mute or turning vol to zero. I have to physically unplug the optical cable to get sound back. 

Try unplugging your cable instead of turning car off, see if that fixes?


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## Durgesh (Sep 18, 2014)

Earlier was using optical input now coaxial. I have seen this clipping with both.

The clipping is so bad that the driver would damage if don't shut off the system immediately. Removing the digital cable is going to take some time.

It's as if the driver has been given some 1000 watts.

Its something with c5. This only happens when I change settings while music is playing. 

Otherwise the music plays clean and loud enough without any sort of distortion or clipping.

Sent from my SM-G950F using Tapatalk


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## Durgesh (Sep 18, 2014)

Attaching c5 main tab screen shot. 

Will try reducing unit 2 channel gain in c5 and increase from the amp.









Sent from my SM-G950F using Tapatalk


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## captainobvious (Mar 11, 2006)

Durgesh said:


> Earlier was using optical input now coaxial. I have seen this clipping with both.
> 
> The clipping is so bad that the driver would damage if don't shut off the system immediately. Removing the digital cable is going to take some time.
> 
> ...



I have had the same thing happen. It seems to happen while trying to change settings rapidly. And that was with analog in and out. This is not an issue isolated to digital input/output usage. It's like while you are making tuning changes, it seems like the unit blasts to max output, without filters. Extremely dangerous for the drivers. All you can do is turn the volume down quickly and restart system to fix it. I would check with Raimonds to see what he says.


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## Durgesh (Sep 18, 2014)

Yes..Exactly when changing things rapidly in c5. Like quickly muting one channel after another. Changing 
before the previous task has fully processed by c5 and uploaded to the processor. 

But with me this happens with digital inputs both optical and coaxial. I use analogue inputs only for taking measurements.

Sent from my SM-G950F using Tapatalk


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## Raimonds (Jun 13, 2014)

Hello Friends!
Good health, good spring time, good tunes : )

Thank you Durgesh for question and issue reporting!

Each function (button press) needs time to execute.
It is very hard to implement the "ENABLE OFF" on every button on Graphic User Interface of the program.
So, if you working rapidly, you may start the new function (press button) before previous is completely executed.
One of them may be executed with error in such case and that actually happened in the Durgesh`s example.

If you need to change lot of settings, it is recommended to switch of "Direct Upload to the Unit".
Make your changes, switch the "Direct upload to the unit" on and press "Update All".
Please observe the speed of updating. It maximal available for your PC.
Please try to use your working speed less then previously observed when the "Direct upload to the unit" is on.

Good luck,
Raimonds


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## Durgesh (Sep 18, 2014)

Alright will try as per your suggestions. 

Thanks

Sent from my SM-G950F using Tapatalk


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## Durgesh (Sep 18, 2014)

Also I would like to add that I am using a powered usb hub. On the usb hub I have audio interface with xlr mic and Apl1012.

The usb hub is connected to a usb 2.0 port on my laptop (my usb 3.0 port is broken and I only have two ports).

Slow usb speed could be an issue.



Sent from my SM-G950F using Tapatalk


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## Durgesh (Sep 18, 2014)

Hi

Can anyone pls explain the use of Group Delay correction. What to look for and how to set the the time span in the crossover tab when GD cor is enabled. And and then how to verify that the entered time span is correct.

Or

Should I simply save IR (.wav) and import the .wav file as master in EQ3 and apply to all the units .

My crossovers are 80/ 350/4000. 

I have spent almost 7-8 days but couldn't understand this part.

Sent from my SM-G950F using Tapatalk


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## High Resolution Audio (Sep 12, 2014)

Subbed for reference.


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## Hanatsu (Nov 9, 2010)

Durgesh said:


> Hi
> 
> Can anyone pls explain the use of Group Delay correction. What to look for and how to set the the time span in the crossover tab when GD cor is enabled. And and then how to verify that the entered time span is correct.
> 
> ...


I believe GD cor is automatic. It's calculated from the filters used and applied as cascaded phase correction on top of the other filters.

Skickat från min SM-G960F via Tapatalk


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## Holmz (Jul 12, 2017)

@Hanatsu i thought you were in Sweden, but you flag shows GDR... are you travelling?
And how are you doing with the recent dramas?


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## Hanatsu (Nov 9, 2010)

I sit behind a VPN since we have china-net here. Things are as usual otherwise, gonna be a boring summer with everything closed down.

Skickat från min SM-G960F via Tapatalk


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