DSP speaker/room correction

[esldude]

I posted a poll to gauge interest in a forum dedicated to discussing DSP speaker and room correction approaches.

 

It was poorly named, "poll on additional sub-forum".

 

Only later did it occur to me sub might be read as subwoofer.

 

So if you are interested in the poll on DSP speaker and room correction please look here:

 

http://www.computeraudiophile.com/f8-general-forum/poll-additional-sub-forum-16842/

[esldude]

Heck with lots of views of this thread, I would hope more people would have voted in the poll.

[sandyk]

Perhaps not everybody agrees that further processing of a digital signal before it gets to a DAC is desirable, and not too many have direct to digital speakers ?

[esldude]

Perhaps not everybody agrees that further processing of a digital signal before it gets to a DAC is desirable, and not too many have direct to digital speakers ?

 

Not sure what the direct to digital speaker comment is about. Not needed to use DSP. Maybe a dedicated forum would let one learn why it shouldn't be a problem.

[esldude]

Pre-dac processing if done well, and plenty of software does, wouldn't leave objectionable artifacts.

[sdolezalek]

Actually, there is a lot of discussion about this among the high-end pre/processor manufacturers. For example Bryston has chosen to leave DSP speaker and room correction completely out of their latest SP3 processor on grounds that they believe it interferes too much with the design characteristics of the speakers. Classe has taken a middle ground approach, with some EQ capabilities in the SSP800. On the other hand folks like Lexicon, Marantz, Pioneer, etc. (most of whom also do a lot of video processing) seem to have no fear in lots of audio processing as well.

 

BUT, if the audio processing is going on in your pre-amp, your are almost completely wasting your DAC, because in order to do that processing the now analog signal has to go through another A/D and then D/A conversion process, very likely not improving sound fidelity. So clearly you want all processing done at the pre D/A conversion stage AND you probably want that D/A conversion to happen as close to the individual speaker driver as possible (think about digital crossovers).

 

Also given what we are discovering about the differences between digital music players (Audirvana, Pure Music Amarra, JRiver, etc.) it appears that you don't want a lot of other processing going on simultaneously with the music decoding that could mess with the "bit-perfect" player. So the design constraints become really interesting (where and when do you do what and given that you only want a single D/A conversion, exactly where does that occur?). Mix in multi-channel sound and the complexities become overwhelming.

[esldude]

Exactly the sorts of issues a dedicated DSP forum would be good for discussing.

 

The Tact I have been using is one with the digital amp. The signal is never analog until it hits the output stage of the power amp.

 

I agree that one needs care in not causing problems for the speaker. But even partial correction being very conservative in regards to speaker issues creates clearly audible improvements in the results.

 

If one uses software to do the processing before a digital result is sent to the DAC, one sidesteps the issues you raise with added AD/DA conversions. One more benefit of computer based audio.

 

I am not familiar with the issues regarding bit-perfect players and other processing, and not to sound like an echo, but sounds like a good reason to have a dedicated forum for such issues.

[firedog]

Exactly the sorts of issues a dedicated DSP forum would be good for discussing.

 

The Tact I have been using is one with the digital amp. The signal is never analog until it hits the output stage of the power amp.

 

I agree that one needs care in not causing problems for the speaker. But even partial correction being very conservative in regards to speaker issues creates clearly audible improvements in the results.

 

If one uses software to do the processing before a digital result is sent to the DAC, one sidesteps the issues you raise with added AD/DA conversions. One more benefit of computer based audio.

 

I am not familiar with the issues regarding bit-perfect players and other processing, and not to sound like an echo, but sounds like a good reason to have a dedicated forum for such issues.

Yes, but TACT DRC works at a max of 24/96 so analog input or higher res gets converted.

[Kal Rubinson]

Yes, but TACT DRC works at a max of 24/96 so analog input or higher res gets converted.

 

Yes but the questions are (1) whether the down-conversion is audible, (2) whether the A/D/A conversions are audible and (3) whether the sound improvements due to EQ are more substantial than the effect(s) of the conversion(s).

[esldude]

Yes but the questions are (1) whether the down-conversion is audible, (2) whether the A/D/A conversions are audible and (3) whether the sound improvements due to EQ are more substantial than the effect(s) of the conversion(s).

 

Yes Kal has outlined it well.

 

Plus only the early Tact gear was limited to 96k. The rest is 192k.

[Guidof]

Yes but the questions are (1) whether the down-conversion is audible, (2) whether the A/D/A conversions are audible and (3) whether the sound improvements due to EQ are more substantial than the effect(s) of the conversion(s).

All 3 are excellent points, but in particular point 3!

 

Guido F.

[Regnad]

I use a DSPeaker Antimode for bass correction so everything gets A/D/A, no downside to me, and much cleaner bass.

[Guidof]

I use a DSPeaker Antimode for bass correction so everything gets A/D/A, no downside to me, and much cleaner bass.

 

It is also possible to avoid even this A/D/A by feeding the DSPeaker Antimode a digital signal via Toslink. I do this, and use the DSPeaker internal DAC which I found to be more transparent than the Cambridge Audio 840-C I was using previously.

 

Guido F.

[Panelhead]

It is also possible to avoid even this A/D/A by feeding the DSPeaker Antimode a digital signal via Toslink. I do this, and use the DSPeaker internal DAC which I found to be more transparent than the Cambridge Audio 840-C I was using previously.

 

Guido F.

 

 

Do you you the dac in the DSP as a crossover also? Looking for options to combine DAC/DSP/Crossover in one box.

[Guidof]

Do you you the dac in the DSP as a crossover also? Looking for options to combine DAC/DSP/Crossover in one box.

No, I run the main speakers full range and cross over the sub at about 31 Hz using the sub's own crossover. Because my sub is an active model and is thus fed directly from the RCA outputs of the Dual Core, (while its balanced outputs feed my preamp, amp, and main speakers) it is not possible to use the Dual Core crossover function.

 

Regards,

 

Guido F.

[firedog]

My point about the TACT doing DRC at 24/96 was twofold:

 

a) if you have 176 or 192 files and think they sound better played back at native rate;

b) It is possible to do DRC inside a computer these days through software and do it at the native rates; I'd prefer this approach to one using much lest robust computing built into the DAC/pre - both for computing power and the ability to pretty much isolate the computing/computer from negatively effecting SQ

[esldude]

Again, the last few posts illustrate why a dedicated DSP forum would be valuable to lots of folks on CA.

 

The ins and outs of the various DSP approaches will clearly enhance the enjoyment of music and this wonderful hobby.

 

This DSP stuff isn't exactly simple, but it isn't rocket science. Seeing what others have done, and sharing the results makes it much less intimidating to engage with it. Exactly the kind of thing good forums accomplish. The results are clear and beneficial.

 

Come on folks, if you are registered with CA, vote in the polls. If you aren't registered, there is lot to learn here, and all of you have something to contribute. Register and then vote.

[esldude]

My point about the TACT doing DRC at 24/96 was twofold:

 

a) if you have 176 or 192 files and think they sound better played back at native rate;

b) It is possible to do DRC inside a computer these days through software and do it at the native rates; I'd prefer this approach to one using much lest robust computing built into the DAC/pre - both for computing power and the ability to pretty much isolate the computing/computer from negatively effecting SQ

 

Yes, but the latter Tact units will do the higher rates natively. IRC, they convert all inputs to 192 or 176 to work with them. It has been awhile, but I ran files through and recorded the digital output. The residuals from the processing were well below 24 bit levels.

 

Just doing subjective listening, I preferred sending 44.1 files thru the Tact vs upsampling with good software to 192 and sending it to the Tact. To my ears, over a highly resolving system the differences were small either way.

 

I share your concerns over various hi-rez approaches. Yet again, another reason some DSP dedicated discussion would prove fruitful.

 

Finally, I have a simplified description of my system. I use Soundlab electrostats. They aren't exactly lacking in resolution. They make it difficult to listen to all, but the finest conventional speakers without feeling like the presentation is slow, blunted, and foggy.

[esldude]

My point about the TACT doing DRC at 24/96 was twofold:

 

b) It is possible to do DRC inside a computer these days through software and do it at the native rates; I'd prefer this approach to one using much lest robust computing built into the DAC/pre - both for computing power and the ability to pretty much isolate the computing/computer from negatively effecting SQ

 

I know there are discussions about whether or not extra computing negatively effects sound quality. When you are talking DSP you also would do well to think about all the ways it can positively improve sound quality. Just for instance, picking numbers out of the air, if you take a 1% hit from processing going on and gain 15% improvement you are still coming out ahead.

 

Now I think the main benefit of doing it all in a general purpose computer is the ability to change/upgrade capabilities as improvements occur without re-buying key expensive parts of your system.

 

Main benefit of hardware dedicated to DSP is less equipment and sometimes an easier to deal with way of getting the DSP done vs. using a computer with lots of user interaction needed.

 

And some systems are in between the two extremes. If a person knows how much they want to get involved vs plug n play listening they can make an informed decision. Without a good number of examples from other people that is pretty hard to do. You are looking at myriad choices. A forum would let those already using DSP describe what they are doing and we could all learn from each other as well as being a resource for those wishing to get started with the benefits of DSP in a modern high performance music system.

 

Those of us doing some DSP know the benefits are large and substantial. I think if it were more commonly known and experienced there would be a booming market for it. The market and suppliers are growing. I do believe it is poised to really take off now.

[DigiPete]

. . . . A forum would let those already using DSP describe what they are doing and we could all learn from each other as well as being a resource for those wishing to get started with the benefits of DSP in a modern high performance music system.

 

Those of us doing some DSP know the benefits are large and substantial.

I think if it were more commonly known and experienced there would be a booming market for it.

The market and suppliers are growing. I do believe it is poised to really take off now.

 

 

I couldn't agree more.

[wgscott]

I

 

Only later did it occur to me sub might be read as subwoofer.

 

I guess I should have read this first, before I tried to be funny.

[wgscott]

What I would like to be able to do, within OS X, is plug in a microphone, run some software, and generate one or a series of correction plug-ins that I could use with Audirvana, without having to get a Ph.D. in applications of Fourier analysis.

[Boris75]

What I would like to be able to do, within OS X, is plug in a microphone, run some software, and generate one or a series of correction plug-ins that I could use with Audirvana, without having to get a Ph.D. in applications of Fourier analysis.

 

+1. I have a Behringer ECM8000 and have been wondering how I could get someone to prepare a calibration file for this mic.

[esldude]

Calibration file for Behringer ECM 8000

 

Behringer ECM8000 calibration

 

Some info on generic calibration values in these threads.

 

I guess which kind of file format depends upon which software you are using. Your best bet might be to purchase a calibrated ECM 8000.

 

Also, some talk of the type capsule having changed and response differs between older and current units.

[Boris75]

Many thanks. It looks like I need to buy a calibrated one or send mine to a sound lab if I really want to use it for active room correction.