# Which processors have FIR filtering? Must have toslink input...



## 2010hummerguy (Oct 7, 2009)

I am looking for processors that have FIR filtering. The XDP-4000X I have on the way does but I am also interested in others such as Audison, Helix, Mosconi, etc. I know that PPI, Soundstream and other miniDSP based models might not since miniDSP does not have FIR filtering except for in their miniSharc which requires separate DACs. Before I spend hours researching the net today, has anyone already considered this? Thanks!


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## thehatedguy (May 4, 2007)

There are none, not 12 volt native.


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## 2010hummerguy (Oct 7, 2009)

Bummer! From my research so far only the DRZ-9255 and XDP-4000X have FIR filtering and the 9255 requires workarounds to accept optical input. Interesting since phase is such a big issue in automotive environments.


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## 14642 (May 19, 2008)

Phase EQ is not the panacea that it appears to be. Alpine Imprint was an FIR filter. JBL MS-2 is an FIR filter.


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## thehatedguy (May 4, 2007)

Well, there were a couple...lol.


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## sqnut (Dec 24, 2009)

Architect7 said:


> Bummer! From my research so far only the DRZ-9255 and XDP-4000X have FIR filtering and the 9255 requires workarounds to accept optical input. Interesting since phase is such a big issue in automotive environments.


Phase is *NOT* an issue in a car environment because you can do so little about it. You're just not getting enough direct sound in a car. Meanwhile, response is a huge issue in a car. In a car phase is nothing more than timing, using same orders on HP/LP and some response tweaking around the xover points. To worry about phase beyond this in a car, is a waste of your time. 

31 bands of GEQ per driver from a processor is more than enough. Resolution is important. A processor allowing you to eq in +/- 0.3 db is better than one that allows +/- 1db, and so on.


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## 2010hummerguy (Oct 7, 2009)

sqnut said:


> Phase is *NOT* an issue in a car environment because you can do so little about it. You're just not getting enough direct sound in a car. Meanwhile, response is a huge issue in a car. In a car phase is nothing more than timing, using same orders on HP/LP and some response tweaking around the xover points. To worry about phase beyond this in a car, is a waste of your time.
> 
> 31 bands of GEQ per driver from a processor is more than enough. Resolution is important. A processor allowing you to eq in +/- 0.3 db is better than one that allows +/- 1db, and so on.


Cool, thanks for the info, that makes sense why we don't see FIR phase correct devices in cars anymore.

Funny enough, this is for a home application, hence my interest in it


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## lecuisinier (May 26, 2009)

waveflex caraudio


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## stochastic (Jan 24, 2012)

I was under the impression that FIR was the standard digital filter design because of its advantages over IIR. I just assume most did. Serves me right for assuming.

There is this option with toslink: http://aplaudio.com/conc2/apl1/


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## 14642 (May 19, 2008)

There are also drawbacks of FIR. Getting good resolution in the bass requires either partitioning the filter or many taps. The more taps, the longer the filter, the more inherent delay and the more processing power required. 

Without the partitioning, a bunch of the processing power is wasted on high frequencies where it isn't needed. For example, a 4096 tap filter would give you 19,980/4096 = 4.8Hz resolution, but half of that would be used between 10kHz and 20kHz. That's a pretty long filter. The MS-2 uses a partitioned filter with 512 taps below 1kHz and 512 above. That gives 2Hz resolution below 1k and 37Hz resolution above.


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## Raimonds (Jun 13, 2014)

Andy Wehmeyer said:


> There are also drawbacks of FIR. Getting good resolution in the bass requires either partitioning the filter or many taps. The more taps, the longer the filter, the more inherent delay and the more processing power required.
> 
> Without the partitioning, a bunch of the processing power is wasted on high frequencies where it isn't needed. For example, a 4096 tap filter would give you 19,980/4096 = 4.8Hz resolution, but half of that would be used between 10kHz and 20kHz. That's a pretty long filter. The MS-2 uses a partitioned filter with 512 taps below 1kHz and 512 above. That gives 2Hz resolution below 1k and 37Hz resolution above.


I would like to disagree with your point of view regarding FIR drawbacks. You could say that 10 years ago there was a massive processing power needed for FIR but even at that time not expansive FPGAs were available. Nowadays you do not have rights to say that FIR requires massive processing power.
The delay, introduced by FIR, completely depends on how you synthesize particular FIR.
If you use linear phase FIR – yes, you will have a delay equal to half of FIR impulse response length.
But, you will have no delay for minimum phase filter and, additionally, the phase correction of the minimum phase part of a loudspeaker transfer.
You can see this in example of correction of Quested monitors:
delay estimation and measurement

Instead of possible drawbacks, FIR filters have advantage that they do not have recursive feedbacks as IIR filters that are making seriously degradation to the processing quality.

But most important thing about FIR is that FIR is like a very sharp knife – you can do anything with it including cutting off your finger. And result of using of FIR completely depends on how accurately you prepare correction filter for it to create predistortions that will be compensated by distortions of loudspeaker, instead of creating (introducing) new distortions.
If you will try to equalize such „measured” curve, you will get into big trouble (1. attachment).

The way of measurement mic (or just mic) sound field perception shows more exact „problems” than our hearing and this circumstances causes appearance of problems that actually do not exist. This can be explained in such simple way. Let’s take loudspeaker stereo system on your desk, created of the best loudspeakers you can find. Feed same signal to both of them and listen to mono performance. Than move yourself from side to side. You will hear that mono image will move the same direction as you and nothing more. But if you put measurement mic in your position, it will show „problems” caused by interference from your two speakers. Same thing happens when you have just one speaker near the reflective surface. It means that you must measure performance of loudspeaker with consciousness of such an effect. It cannot be removed by simple smoothing of measured curve as it is in all ordinary analyzers.
Next picture shows true loudspeaker „problems” of the same loudspeaker in the same placement as in previous graph. The correction, that follows this curve, will give you excellent result (2. attachment)


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## 14642 (May 19, 2008)

Of course, all of the above is true. One of the marketing angles for FIR is the use of linear phase filters. The second point, which could be summarized by saying, "any form of EQ is only as good as the interpretation of the measurement and the application of the correction" is true as well. 

I think from a practicality standpoint, the discussion of recursive vs. nonrecursive is pretty esoteric. I've used both in the same car to arrive at the same correction results in the frequency domain and can't hear the difference.

Finally, throwing away the phase measurement and synthesizing using minimum phase is valid for correcting the response of a minimum phase system--but we don't listen to minimum phase systems. In that regard, the correction doesn't include phase correction and that feature is the one that often causes people to want FIR.

I guess my point is that all methods are valid, but the simple "one is better than the other" isn't really useful.


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## Alextaastrup (Apr 12, 2014)

It was interesting comment about Alpine Imprint using FIR filter. My impression of Imprint tuning results were simply bad, despite application of almost all possible tips, which could be found on internet (and many night hours, of course). The way how APL is processing signal - rather different, but final results amazed me from the first time. I suppose that one of the advantages - a rather big number of filters (4096). I desagree with the opinion that 31 band EQ is enough in caraudio. At least not in my case. Another advantage - in measuring the power intensity response curve instead of regular RTA...
The most people reading such threads do not need to go deep in the theoretical investigations, They just need a simple DSP unit, which does not destroy the sound. APL is in this case a good choice. As confirmation to this - two first places at Eurofinals in Zalzburg, March 2014 (car of Sergey Dubinin, installer: Alexander Martyanov).

In the post above there was only one model mentioned: APL1 - for two channels. There are other models available from the same manufacturer. For example APL3 designed for six channels. This will be my next step in the ongoing investment project.


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## sqnut (Dec 24, 2009)

Alextaastrup said:


> I disagree that 31 band EQ is enough in caraudio. At least not in my case. Another advantage - in measuring the power intensity response curve instead of regular RTA...
> The most people reading such threads do not need to go deep in the theoretical investigations, They just need a simple DSP unit, which does not destroy the sound. APL is in this case a good choice. As confirmation to this - two first places at Eurofinals in Zalzburg, March 2014 (car of Sergey Dubinin, installer: Alexander Martyanov).
> 
> In the post above there was only one model mentioned: APL1 - for two channels. There are other models available from the same manufacturer. For example APL3 designed for six channels. This will be my next step in the ongoing investment project.


Dude I said 31 bands per driver. It would be great if we could have 1/6 oct cause our hearing is basically setup at that level. It would be a steep learning curve from 1/3 to 1/6 though. I hope you know that you *don't *need to flatten every peak that you can measure in order to get good sound and stop plugging APL whatever it is. Auto tune doesn't win championships.


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## [email protected] (Nov 7, 2010)

sqnut said:


> Auto tune doesn't win championships.


Nope, it just sells rap records....


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## Bluenote (Aug 29, 2008)

Joey, auto-tune also sells Pop records ( A LOT of Pop records haha..)


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## cajunner (Apr 13, 2007)

that there is a European product now available on a consumer market level, using FIR filters and winning in competitions, is a decidedly "good" thing.

I like being made aware of these things.

I also like seeing FIR positives and negatives discussed more, as it is a fuzzy logic area in audio, in that most people logically fuzz the details about why or why not.

If phase correction isn't an inherent feature of an FIR-based DSP engine, then I like knowing that, I was under the mistaken impression that using FIR filters were giving you an automatic positive in that it didn't mess with phase, or the phase you got in, is what you got out.

If that is not a fact, and that FIR filters **** your phase, (excuse my french) then what about IIR filters, do they just dig deeper in the phase ****ing?

and if you can't hear it, why is it so special?


that's the main thing, I guess. Somebody has to claim to hear it, like rise time in amplifiers, or slew rate on op-amps.

can you hear it?


is it better than before, is it worth another 200 bucks in DSP re-working, or can you get just as good without it?


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## tonny (Dec 4, 2010)

I tested a pioneer odr processor with irr and fir filters on board in a car we did setup the car with both filter options one setting with fir filters and one without fir filters, the setting without fir filters sounded better to me! With the fir filters the sound was just if the dsp had to work to hard to get everything in line you could hear that! 
The car was tuned using a measure meant system for the same curve and loudness and with the same crossover points. 

Also I think in the car with the already strange phase curves off alone the speakers without a crossover with a irr filter and eq you can adjust the phase to get the different speakers better phase alined as with with a fir filter!


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## Raimonds (Jun 13, 2014)

Thanks, Andy,

„recursive vs. nonrecursive” 
– it depends how your IIR is created. If it is using a double precision and not a simplest s-z transform, there is nothing to say about IIR (serious DAWs in studio). But DSPs in car audio head units is much different story. Alexander Martyanov (his nick is „martyanov” here and he has beautiful build log here and he is two times EMMA European champion Sergey’s engineer) told me his experience with usage of head unit equalizers and this experience is not good for that equalizers. And proximity of Nyquist frequency for FIR is nothing in opposite to IIR.

FIR equalizer may have ordinary parametric equalizer interface and may be used just as parametric.
I got a lot of compliments from pro (concert, studio) clients (I was not specially asking them) how „light” is sound of such FIR parametric. And they asked „can we get such quality into DAW plagin” : )

Most important advantage of FIR is that you have ability to use virtually any (unlimited) number of different equalizers same time but still having just a one FIR engine. For example 1) accurate loudspeaker correction equalizer, 2) parametric 1., 3) parametric 2., 4) time domain delay equalizer, 5) Hi and Low timbre for taste, 6) parametric 3. for final tweak, 7) parametric 4. for ... and so on...
And no degradation by switching on new additional EQ.

„we don't listen to minimum phase systems” 

– 90% of problems in loudspeakers is minimum phase problems and can be ideally (amplitude and phase) corrected by minimum phase equalizer. Non minimum phase problems may be caused only by inaccurately tuned crossover. But let’s tune it as accurately as we can. Especially if we have possibility to „see” delays between different bands by use of TDA. And, if we are still not happy with such result, let’s use a frequency dependant delay correction. Sorry for returning to previous example of Quested monitors: delay estimation and measurement

„I hope you know that you don't need to flatten every peak”

- sorry, I did not read books and forums and now this : ), but it is true for 1. attachment picture in post #11. And it is not true for 2. attachment picture.

Just in few past years we have offers of dsp units with FIRs for car audio. But story is same as with professional BSS processors that offered FIRs much early (and it was much, much early offered by yours faithful servant). When I asked them how to use they FIRs, what to upload to them, the answer was – we don’t know.
Serious reason to use FIR is just in one case - when we have very accurate and detailed information (curve) of a loudspeaker performance. And 10 years experience shows that we can obtain this only by working in power domain (with some similarity with microwave engineering, where system parameters are distributed, not lumped and systems are described with much more poles then four, and we talk not about voltage and currant but about power – the loudspeaker is the same from a wave theory prospective)

And another serious reason to use FIR is because of FIR ability to follow any curve we like and as accurate as we like. We may spend hours of our time to try to recreate some curve by use of parametrics. And this will be just an approximation anyway, a raw approximation.

“If phase correction isn't an inherent feature of an FIR-based DSP engine, then I like knowing that, I was under the mistaken impression that using FIR filters were giving you an automatic positive in that it didn't mess with phase, or the phase you got in, is what you got out.”
-	FIR is allowing to do anything with phase independently regarding amplitude. And it is completely your responsibility what to do. My first suggestion is to use minimum phase filters that are doing phase correction based of properties of minimum phase systems.

“…like rise time in amplifiers, or slew rate on op-amps.” - it is about non linear behavior (non linear distortions) of the system that is creating new additional components in signal. It is some kind opposite to linear distortions that is not creating new components but is changing relationship of existing components of the signal.


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## cajunner (Apr 13, 2007)

Raimonds said:


> “If phase correction isn't an inherent feature of an FIR-based DSP engine, then I like knowing that, I was under the mistaken impression that using FIR filters were giving you an automatic positive in that it didn't mess with phase, or the phase you got in, is what you got out.”
> -	FIR is allowing to do anything with phase independently regarding amplitude. And it is completely your responsibility what to do. My first suggestion is to use minimum phase filters that are doing phase correction based of properties of minimum phase systems.
> 
> “…like rise time in amplifiers, or slew rate on op-amps.” - it is about non linear behavior (non linear distortions) of the system that is creating new additional components in signal. It is some kind opposite to linear distortions that is not creating new components but is changing relationship of existing components of the signal.



yes, I was unclear about my analogy.

my point is that there are some things we take for granted as having an audible effect and an accepted premise that we should be able to tell the difference between products that measure differently, but in real time we hardly can do this with accuracy.

when you say there is a big difference between FIR filters and IIR filters, I am sure people will interpret that as being a big audible difference, and I attempted to show that perhaps, it's an audible difference in the amount of significance, as slew rate might be in audible terms.


so it's just not clear whether an FIR filter is worth the amount of trouble it seems to take to implement them, and their promised superiority is not all that is claimed.


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## 14642 (May 19, 2008)

I'm not dissing FIR. In fact, I, along with a team of talented engineers, made two of them. One has been instrumental in winning MANY competitions. 

The APL stuff is really interesting and I'm a firm believer in a good autotune. From my perspective, it isn't the tool that's most important (filter type) it's the implementation of the tool. 

And "autotune" for turning a singer who can't sing into an operatic virtuoso is not the same autotune as an equalizer that corrects frequency response.

One of the beauties of sound in cars is that we don't have to consider the room as separate from the speakers. None of the reflections are late enough to necessitate that. We equalize room and car at the same time using the same measurement. Essentially, we EQ the power response of the speaker and all of the reflections as if they were minimum phase. The result is good performance with relatively simple tools applied well. The bad part is that there are some anomalies that cannot be equalized because it isn't really a minimum phase system.


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## 14642 (May 19, 2008)

Another note about this. An algorithm is simply a set of rules that a machine follows in order to achieve a desired result. The result can only be as good as the rules. Developing an autotune designed to work in big rooms with loudspeaker systems that include the full audible frequency range in one box and designed to achieve flat on-axis response with uniform directivity is NOT THE SAME as a bunch of speakers mounted in irregularly shaped baffles which are also the room's boundaries and which are mounted at varying distances to the listening and measurement position(s). A set of rules for equalizing one may not be effective in equalizing the other.

In my opinion, the initial Audyssey algorithm included two related mistakes. It allowed the machine to determine the crossover frequency between the sub and the mains based on a flat response target applied to the mains. In a car, a crappy 6" driver can play flat to 20Hz pretty easily. The result was no bass. The mistakes were: 1) assuming that the flat response target that we like in bigger rooms was appropriate for cars, 2) not realizing that a customer who pays $1,000 for a subwoofer and an amp to drive it wants it to play.

This mistake was later rectified, somewhat, with the addition of some selectable target curves. 

In developing the one I helped develop, the original algorithm was developed NOT to equalize high frequencies following the home audio mantra. I had many discussions about this with the engineers who explained until they were blue in the face why it wasn't valid. Finally, I took them all out to the car and let them listen to the difference. They were then willing to add it to the product.


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## t3sn4f2 (Jan 3, 2007)

Andy Wehmeyer said:


> Another note about this. An algorithm is simply a set of rules that a machine follows in order to achieve a desired result. The result can only be as good as the rules. Developing an autotune designed to work in big rooms with loudspeaker systems that include the full audible frequency range in one box and designed to achieve flat on-axis response with uniform directivity is NOT THE SAME as a bunch of speakers mounted in irregularly shaped baffles which are also the room's boundaries and *which are mounted at varying distances to the listening and measurement position(s)*.


And at greatly varying listening axises. Of which many are at the extremes of the drivers radiation pattern where the response is completely different than more uniform pattern nearer the on axis mark. Due to install restraints and non symmetrical listening positions.


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## Raimonds (Jun 13, 2014)

„The APL stuff is really interesting and I'm a firm believer in a good autotune.”
- how and where did you get “autotune” in APL stuff : ) ?
FIR is allowing us to use CAD of equalization – Computer Aided Design of equalization. But we must be the designers! And some kind of „templates”, obtained by measurements, are very helpful for this. 


It is very hard to describe the effect of usage of FIR correction in words and make some selling message. The experience of an offering of such solution to customers, is showing, that we must have some sounding example to show what we can get from detailed loudspeaker system correction by switching it on and off. And biggest audible difference is when we switch it off.
This can be achieved in some minutes if particular customer car audio system has line input or allowing easy way to directly connect to power amplifier inputs. Connect PC to car audio system, make measurement and correction filter files, upload them to processing plagin and you have sounding result just from computer. It is possible to do this in 5 minutes. If the particular car audio is not requiring a repair. If requiring – you can offer (and base it on measurement curves) your service to fix that.

First example. Attached picture is showing SPDR curve of BMW 528 F10 audio system that has very serious problem at 220 Hz. It is not because of some “out of phase”. It is because of pipe connecting midbase woofer volume to a volume in a car doorstep, to increase overall volume. But instead of getting advantages of an increased volume BMW 528 F10 gets huge resonance of that pipe. Where were BMW engineers looking when they did such thing?

Second example of blind tuning. Attached picture is showing TDA measurement of some competition car. Everything is good except of incorrect delay set (error for 5 ms) in base caused by blind tuning – setting of processor parameters based on measurements by ruler …

Third attached picture is showing TDA measurement of Audi Concert audio system. No comments.

„The bad part is that there are some anomalies that cannot be equalized because it isn't really a minimum phase system.”
- The residual is excessive part of phase (usually „all pass” phase response – frequency dependant delay) and can be corrected by introduction of frequency dependant delay in FIR filter. I must return again to example of Quested monitors.

„A set of rules for equalizing one may not be effective in equalizing the other.” 
– rules are very simple,
1)	remove any coloration of loudspeaker
2)	add any „character” you like, even „copy” the sound of some of yours preferable system or do „mastering” of competition CD to impress judges of that competition. But you must start from „uncolored” one.

It is not good idea to try to sell any defect as an effect.

FIR is not giving something superior by themselves. FIR is giving as a freedom to do very sharp things and the result will depend how well we understand the problems.
And most important thing is to correct the problems such way which do not introduce new additional ones. It is an engineering task and it requires advanced troubleshooting skills.


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## cajunner (Apr 13, 2007)

would it be fair to say that the 'rules' to creating an auto-tuned algorithm, when taking 2 channel stereo into account, doesn't change that much from what is needed in a home bookshelf situation, to a car components situation?

does the moving of the tweeter to a better acoustic location in the car, make the algorithm no longer suitable?

is the JBL MS-2 auto-tune, which is created by an FIR filter using 1024 taps, with more correction in the bottom half, or bass range, actually doing a better job of fixing problems in the car environment than it's big brother MS-8, due to the partition of taps?

is it more or less difficult today, to do a multi-channel DSP with FIR filter and all the time alignment and crossover functions built-in, than it was say, 5 years ago?

should we expect some raising of the bar, in regards to auto-tune units being released?

is the issue of cost, regarding things like SHARC processors and floating bits, whatever, being addressed, is the speed of the hardware now fast enough, so that there are few bottlenecks in processing speed at a reasonable cost, or is this still part of the pricing puzzle?

other than the tethering of miniDSP to REW or whatever, what is coming down the pipe that makes the MS-8 appear primitive, or dated in it's design details?

It would seem to me, the speed of correction has reached the point of inaudibility, and amplifier PWM, class D hardware can do control into the low megahertz. Is the processor not able to keep up, what is the problem of making a product that can do real-time correction at the driver's seat, windows up, windows down... whatever?

Not just a test sweep and corrective filter applied, but a continuously correcting, undetectable control on the response that damps all outside phenomena due to road noise or wind noise?


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## 14642 (May 19, 2008)

"would it be fair to say that the 'rules' to creating an auto-tuned algorithm, when taking 2 channel stereo into account, doesn't change that much from what is needed in a home bookshelf situation, to a car components situation?"

No.


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## 14642 (May 19, 2008)

"is the JBL MS-2 auto-tune, which is created by an FIR filter using 1024 taps, with more correction in the bottom half, or bass range, actually doing a better job of fixing problems in the car environment than it's big brother MS-8, due to the partition of taps?

is it more or less difficult today, to do a multi-channel DSP with FIR filter and all the time alignment and crossover functions built-in, than it was say, 5 years ago?

should we expect some raising of the bar, in regards to auto-tune units being released?"

No, the MS2 and the MS-8 both use FIR, but the implementation is VERY different. The MS-2 uses a fairly standard partitioned convolution. The MS-8 is much more complicated and uses a series of 8 filters per frequency band that can be any shape. 

I'd say that MS-8 and the APL stuff have raised the bar. 31-band graphic EQs implemented in a DSP are not adequate to do any real correction, but are straightforward for many users.


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## cajunner (Apr 13, 2007)

in comparison, would the commercial DSP units by the usual suspects, be considered on par, above par, or below par, (par being the MS-8) when it comes to creating a fixed response inside the confines of a vehicle?


can a newer DSP do things that current ones do not, or is there something more advanced coming down the pipe? Has the current state of the art, been improved upon?

Will there be a significant change by a car audio company, has there been any rumblings of technology yet to be released, that moves things forward?

How much better is a PONO player in the car, than using the standard 24 bit DAC's on 16 bit recorded Red book?


will this be too much information, if processed normally by DSP hardware, or will there need to be a jump in processing power to create a soundfield using the higher resolution of these new formats?


what is the future coming, is it more storage, better sound, cheaper entry level DSP, or some Bongiorno trinaural approach?


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## Alextaastrup (Apr 12, 2014)

cajunner, you forgot active noise reduction and surround sound 
Architect7 just asked about DSP's with FIR filters in his first post. 

Actually, discussing future development of DSP technology is rather interesting issue, which might be put in a new thread.

As I understand, APL team is using acoustic power intensity curve measured in a certain field opposite to regular frequency response taken in a single point. In general, measuring in one point is not a good idea for the car SQ due to many of factors well known to all of us. Coud it be a reason that APL sounds so good despite placement of a listener in the car environment?


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## sqnut (Dec 24, 2009)

Andy Wehmeyer said:


> I'd say that MS-8 and the APL stuff have raised the bar. 31-band graphic EQs implemented in a DSP are not adequate to do any real correction, but are straightforward for many users.


31 bands per driver in the hands of a good tuner should always get better results than the MS8 auto tune. Each car is very different in terms of things like interiors, placement, amt of sound absorption treatment etc etc for one algorithm to sound equally good across all cars. 

Eg something like Gary Summers car which has tons of interior damping would have a much lower roll from 5-15khz than a bone stock car with hard plastic everywhere and mids and tweets on pillars/dash. In the second case you could easily have a 5-7db roll off from 5-15 while in Garys car this may be just 2-3db. This is just one example.

The auto tune is at best a starting point more like a house curve. The auto tune is much better than just fiddling with TA and eq if you are not sure of what you're doing. But if you are comfortable with tuning, manual will take you further. Of course you can run the auto tune and then tune manually too.


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## thehatedguy (May 4, 2007)

31 bands is good when the problems lie in those filter bands. But if you have something between those areas, then you are crap out of luck...you can try touching it by messing the adjacent bands but at the expense of those bands.

But why do you need 31 bands per driver? Do you sub woofers play 20k? Do your tweeters play 20?


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## Airforceyooper (Sep 22, 2005)

I'm really new to all this stuff and maybe I'm wasting space on the forum, but I've been researching the MiniDSP stuff quite a bit and they've got a piece with FIR filtering. I don't fully grasp the difference yet, but I'm working on it. Here's the link to the unit and it comes in both a digital version or analog version.

https://www.minidsp.com/products/opendrc-series/opendrc-di

Here's information on the plugin for it, https://www.minidsp.com/products/plugins (You have to click the Open DRC Series tab)


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## 2010hummerguy (Oct 7, 2009)

Airforceyooper said:


> I'm really new to all this stuff and maybe I'm wasting space on the forum, but I've been researching the MiniDSP stuff quite a bit and they've got a piece with FIR filtering. I don't fully grasp the difference yet, but I'm working on it. Here's the link to the unit and it comes in both a digital version or analog version.
> 
> https://www.minidsp.com/products/opendrc-series/opendrc-di
> 
> Here's information on the plugin for it, https://www.minidsp.com/products/plugins (You have to click the Open DRC Series tab)


I was looking at the miniSharc actually and then paired with four DACs of my choice.

But I am now converting a DRZ-9255 to optical input. Should sound great. Even running analog from my computer to its AUX input sounds amazing  Not many PEQ bands but it is for home use so not as many needed. And FIR filtering built in with 96khz/24bit resolution.


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## BoomHz (Apr 20, 2007)

Architect7 said:


> I was looking at the miniSharc actually and then paired with four DACs of my choice.
> 
> But I am now converting a DRZ-9255 to optical input. Should sound great. Even running analog from my computer to its AUX input sounds amazing  Not many PEQ bands but it is for home use so not as many needed. And FIR filtering built in with 96khz/24bit resolution.


Your using the DRZ in home?


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## 2010hummerguy (Oct 7, 2009)

BoomHz said:


> Your using the DRZ in home?


Yessir, 3-way active to a pair of RAAL 140-15D dipole ribbons and Audible Physics Arians powered by a Clarion APA4300HX and an Adire Shiva MKII powered by a Bash 500S. It sounds amazing!


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## monawilliams (Jun 19, 2014)

I am looking for processors that have FIR filtering. The XDP-4000X I have on the way does but I am also interested in others such as Audison

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## Airforceyooper (Sep 22, 2005)

Architect7 said:


> Yessir, 3-way active to a pair of RAAL 140-15D dipole ribbons and Audible Physics Arians powered by a Clarion APA4300HX and an Adire Shiva MKII powered by a Bash 500S. It sounds amazing!


That's bad ass. We're always looking for ways to adapt home to car, but not the other way around.


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## 14642 (May 19, 2008)

31 bands per driver in the hands of a good tuner should always get better results than the MS8 auto tune. Each car is very different in terms of things like interiors, placement, amt of sound absorption treatment etc etc for one algorithm to sound equally good across all cars. 

No WAY. Not a chance. There are some systems that MS-8 can't tune, but for the ones it can (which is most), it's much more powerful than a 31-band.


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## 14642 (May 19, 2008)

thehatedguy said:


> 31 bands is good when the problems lie in those filter bands. But if you have something between those areas, then you are crap out of luck...you can try touching it by messing the adjacent bands but at the expense of those bands.
> 
> But why do you need 31 bands per driver? Do you sub woofers play 20k? Do your tweeters play 20?



Of course you don't. SO in practice, it's a big waste because the biquads for the out-of-band bands are still used, they're just set to unity. Makes no sense other than to present some slider controls so users can draw a nice curve.


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## 14642 (May 19, 2008)

OK, I'll backpeddle a bit on the 31-band EQ. 

31-band EQs are OK for shaping the response after it's been corrected. They aren't powerful enough to do much real correction, unless problems correspond closely to the placement of the bands.

You can fool yourself into believing that they are good enough by using a 31-band RTA, though. 

It's a little different for home audio systems that include well designed speakers that start out with a flat response. Then, the 31-band EQ can be used for a little room correction and to shape the response according to listener preference. Car audio systems don't work that way because:

1. The room is so small that speaker and room correction aren't as separate as they are in bigger rooms.
2. The speaker system INCLUDES the car as the baffle, which makes a huge difference in the speaker's response.


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## ErinH (Feb 14, 2007)

yea. the issue, at least IMHO, is tied to the RTA's being limited in resolution. so all the DSPs simply went after what the consumer saw on the screen... in 1/3 octave. with modern software/hardware such as REW, Holm, OmniMic, etc ,we are now able to better see the response. so we no longer are limited to adjust in 1/3 across all octaves. I do wish DSPs would allow more flexibility in how I set up my EQ frequencies. Even the more modern parametric based EQs use a fixed 1/3 octave as their standard set with variability within some band (ie; 1/6 or 1/12 octave). So, that allows me to adjust 63hz or change that to 50,56,70, or 76hz (just made up numbers for now). What stinks is the next band doesn't allow you much more room.. so if it's based on 80hz you've only got some play room in there... what happens when I have a few target areas that are within the same range but those processor bands aren't allowed to overlap? Or, additionally, what if I wanted to target a few modes but still needed 1 octave to do some general shaping? I wish companies could take their 31 bands for a subwoofer and allow me to change any of those bands to cover what I need ... because I don't need 2khz covered for a sub. The PS8 is the only DSP I know of that allows for this much flexilibity. I think the mosconi might as well. But, I still prefer my helix over these two. In short, nothing's perfect yet. 


Then there's the notion that a good speaker with good power response needs much less EQ than a crappy one. that's NOT to say that you can EQ a crappy response to be better... the issue here is that a good power response is good because the response in all axes is well maintained and fairly symmetric wrt each other axis of measurement. whereas the 'crappy' setup is adjusted as a whole, but you don't really understand just why it's crappy. if there's a peak in one axis that overrides a null there by 3dB and you EQ it down, you've still got a discontinuity between axes of measure that cannot be resolved via EQ. So, even if you have access to high resolution measurements, they're only going to do you so much good and in many cases can make things worse if you don't understand what the measurements are really telling you. I see examples of this all the time here where some one tells another person how to change their RTA curve, having never heard the setup. Matching left & right response via an RTA isn't the all-in-one solution. For that matter, I've found, at least with high frequency content, that a matching response results in near-side bias and I've since started trying to account for this.


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## sqnut (Dec 24, 2009)

thehatedguy said:


> 31 bands is good when the problems lie in those filter bands. But if you have something between those areas, then you are crap out of luck...you can try touching it by messing the adjacent bands but at the expense of those bands.
> 
> But why do you need 31 bands per driver? Do you sub woofers play 20k? Do your tweeters play 20?


Most processors give you 31 bands per driver. For a given channel, I use the eq in the bandpass the driver is playing plus an octave above and below to smooth out the response between two sets of drivers in the xover zone. 

Beyond this I just cut everything to the max. Typically this is enough eq to set up really good sound. Assuming of course the TA is correct and xovers are selected appropriately. The only time I felt 1/3 oct was not enough was whenever I used 12db or lower slopes. 

Of course the resolution is only 1/3 oct but the point is you don't need to flatten everything you can measure. I'm not dissing auto tune, it is usefull if one is not into tuning and you want decent sound.


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## sqnut (Dec 24, 2009)

Andy Wehmeyer said:


> No WAY. Not a chance. There are some systems that MS-8 can't tune, but for the ones it can (which is most), it's much more powerful than a 31-band.


The eq that sets the auto tune is more powerful than 31 bands, sure no denying that. But iirc the eq that lets you tweak manually once the auto tune is done is 31 bands per side. 

Us manual tuners, we just like to do it ourselves . In any case the toughest part of tuning is learning to hear with some degree of accuracy. In comparison using The ta, xover, eq etc is much easier to learn.


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## Alextaastrup (Apr 12, 2014)

sqnut,
Perhaps you did not read carefully the manual for APL processor, as it provides you with much more possibilities than just 31 band per drive for manual tuning. Result of this difference - two champion places at EMMA (2014) in the most difficult classes. That has changed my mind. With the APL1 or APL3 you will be a real designer of SQ in your car down to almost unlimited resolution. You can see and handle system response down to 4-5 Hz, isn't it amazing? It allows you to simply forget about RTA with 1/6 or 1/3 octave. I had before three different DSP, but only APL has changed drastically sound quality in the car despite of middle budget drives used for the front in the current setup. Nightmare of measurements is completely over for me (OK - right now) - I am enjoing music, which I suppose is the major goal for many of us here on this forum.


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## cvjoint (Mar 10, 2006)

Sorry to bump this from the dead, but this is a good topic. Can we revisit?

Can anyone compare the Alpine H800, MS8, and ALP1 correction methods? What is different, just the number of taps or something else? There is a lot of talk about FIR filters and their usefulness. Let's assume you want to use them, which unit does what? What are the differences? I'm only referring to the auto tune algorithm, not inputs and other stuff.


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## Elgrosso (Jun 15, 2013)

I'd be curious to learn more on the algorithms too.
APL uses the power response, so I guess it's one of its thing to compensate poor off axis, and maybe even xo points especially? (There is a specific method to kind of un-correct high directivity drivers)
Since it uses hundreds of measurements points, maybe it can also detect anomalies and select what's correctable or not?
But it works really well, it's surprising because it's not focused on the "ear points" at all.


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