There are setups that benefits from more correction than the average setup for sure. And setups that suffers from troubled acoustics is certainly among these.
The next thumbnail is from a correction that has a window of 12 cycles from 10 Hz to 400 Hz, then gradually goes down to 5 cycles in the top.
Too much as a general approach, but a setup that requires a lot of room correction is likely to benefit from such a huge amount of midrange correction. On one visit with difficult acoustics we ended up using close to 20 cycles in the midrange.
The next thumbnail is from a correction that has a window of 12 cycles from 10 Hz to 400 Hz, then gradually goes down to 5 cycles in the top.
Too much as a general approach, but a setup that requires a lot of room correction is likely to benefit from such a huge amount of midrange correction. On one visit with difficult acoustics we ended up using close to 20 cycles in the midrange.
Attachments
ShinOBIWAN said:I've downgraded, yep downgraded my audio interface. Previously I used a Lynx Aurora 16 DAC but the fall of the pound against the dollar now means this unit is rather expensive over here. A good time to sell then. Pure luck really but in this case I made quite a bit over my original second hand purchase price.
With the money I was in two minds as to whether to go really crazy and buy some very highend converters such as Universal Audio, Lavry, Mytek or be conservative and consolidate the money into other parts of the system. The crunch came this weekend after I compared a Mytek 8x192 to an old favourite of mine(had one before) the RME fireface 800 in a pro audio music store. I was surprised, happy and yet slightly disappointed all that same time. I think the differences we perceive in converters can often be down placebo or whatever else you like to call it because this little demonstration quickly reminded me that once you hit a good level then things start to get expensive very quickly with almost imperceivable sonic gains. In short I was straining to hear a significant difference between the almost legendary Mytek vs. the relatively affordable RME.
I already had Apogee DA16x, another good convertor, a couple of years ago and came to the same conclusion so I shouldn't have expected much change but you keep look for that upgrade that will take your system higher than before but in this case it wasn't the DACs'. Seems the ones in the RME are pretty darn good.
On the flipside of the my fortitous gain with the Lynx from the downturn of our economy I found out that the Euro was beating the hell out of the pound and goods from the EU were accordingly more expensive. Guess what? RME is manufactured in Germany. What was once an ~£800 interface was now £950, quite a leap for the same product. But I lucked out once again and found a second hand model on ebay for £630. That's more like it.
RME is no slouch so don't take this too literally but my thought of the day is don't obsess too much over DAC's if the unit your looking at is well engineered. You don't have to spend a fortune(however much you want to) to get top end results.
Also be frugal and always buy second hand if possible. I'm finding great saving doing this and you can put that into more important things besides audio.
An externally hosted image should be here but it was not working when we last tested it.
Sorry if that is OT, but I see that you use an HFX case for your PC.
Are you happy with it? Do you house only audio processing in that PC, or is it used as a complete HTPC?
If the latter, is the fanless cooling enough, or did you have to add fans ?
BTW, anyone using the Mini model of HFX?
Again, sorry for the OT!
Hi Shin,
can you take a look of my questions on this site ?
Because of being a prospect the thread will be later seen
as the discussion is going...
I have downloaded your selfwritten manual about accourate,
but the (for me) interesting part of Roomcorrection is missing.
I know how correct the room in stereo, but i have no idea to do
this for more channels.
can you take a look of my questions on this site ?
Because of being a prospect the thread will be later seen
as the discussion is going...
I have downloaded your selfwritten manual about accourate,
but the (for me) interesting part of Roomcorrection is missing.
I know how correct the room in stereo, but i have no idea to do
this for more channels.
mycall said:
Sorry if that is OT, but I see that you use an HFX case for your PC.
Are you happy with it? Do you house only audio processing in that PC, or is it used as a complete HTPC?
If the latter, is the fanless cooling enough, or did you have to add fans ?
BTW, anyone using the Mini model of HFX?
Again, sorry for the OT!
I use the HFX case for amps which is what you see in the photo but yes, also there is another HFX housing the PC. Its not a dedicated box but a jack of all trades. The fanless cooling works well but I also bought the optional fan kit which turns on should the PC get too hot.
For Acourate the steps are the same as stereo but use your left channel as reference ie.
Do DRC for left and right first
Do center WRT left.
Do rear right WRT left
Do rear left WRT left
DRC sub on its own and set level on the panel amp
M-Audio 1010LT is not suitable.
Depends. For 1080p H.264 video with multichannel DRC you might be struggling but everything else will work within an optimised system.
One of the 3.33Ghz dual core Intel CPU's is fine on everything. For extra insurance the E0 stepping of this CPU clocks easily to 3.8Ghz and often upto 4Ghz with good cooling.
ShinOBIWAN,
Considering that your thumbnail was an actual measurement from the listening position it was quite impressive. But there seems to be methodical differences that makes this feature less relevant in Audiolense. A group delay correction handles this part pretty well already.
To the free lunch I was questioning;
Provided that the interchannel matching is done with wise moderation, it will not always look as good as in the sample you showed us. And by wise moderation I mean not correcting @ 10 kHz for 20 ms for instance. Many setups will require high frequency correction with very long durations to achieve the kind of matching you showed us and then I think it's better to settle for less.
Luckily for us, the hearing is also frequency dependant to a certain extent. The midrange is suspect no 1 when the image is disturbed in spite of a good frequency balance and a properly timed first arrival.
But if the channel balance is much off, I would expect that a substantial amount of time domain correction is still required to get the kind of result you posted. And then I wonder: Why settle for similarity when both the step responses can be made much better?
_____
I was rather surprised to find out that the step respone you showed was from a corrected speaker. By the look of the response I assumed that you had simply done an interchannel balancing on an uncorrected measurement.
An IR view may tell a more precise story, but the step response looks more minimum phase than linear to me. Are you not correcting any of the time domain smearing fro the room boundaries? Are you doing the DEQX thing? (Full *anechoic* correction paired with frequency correction in the sweet spot)
I can understand why interchannel balancing can make a difference with such a correction.
Audiolense was subject to an exceptionally favourable review in the Norwegian magazine Fidelity a couple of months ago. The Group Delay Correction was the icing of the cake to the reviewer. A GDC with the kind of step responses that I have shown above. An English translation of the review is available for download from the webshop.
Among the more serious Audiolense users there is a rather uniform opinion that GDC is quite superior to a minimum phase correction. And then you will get a pretty good interchannel balance as well.
Considering that your thumbnail was an actual measurement from the listening position it was quite impressive. But there seems to be methodical differences that makes this feature less relevant in Audiolense. A group delay correction handles this part pretty well already.
To the free lunch I was questioning;
Provided that the interchannel matching is done with wise moderation, it will not always look as good as in the sample you showed us. And by wise moderation I mean not correcting @ 10 kHz for 20 ms for instance. Many setups will require high frequency correction with very long durations to achieve the kind of matching you showed us and then I think it's better to settle for less.
Luckily for us, the hearing is also frequency dependant to a certain extent. The midrange is suspect no 1 when the image is disturbed in spite of a good frequency balance and a properly timed first arrival.
But if the channel balance is much off, I would expect that a substantial amount of time domain correction is still required to get the kind of result you posted. And then I wonder: Why settle for similarity when both the step responses can be made much better?
_____
I was rather surprised to find out that the step respone you showed was from a corrected speaker. By the look of the response I assumed that you had simply done an interchannel balancing on an uncorrected measurement.
An IR view may tell a more precise story, but the step response looks more minimum phase than linear to me. Are you not correcting any of the time domain smearing fro the room boundaries? Are you doing the DEQX thing? (Full *anechoic* correction paired with frequency correction in the sweet spot)
I can understand why interchannel balancing can make a difference with such a correction.
Audiolense was subject to an exceptionally favourable review in the Norwegian magazine Fidelity a couple of months ago. The Group Delay Correction was the icing of the cake to the reviewer. A GDC with the kind of step responses that I have shown above. An English translation of the review is available for download from the webshop.
Among the more serious Audiolense users there is a rather uniform opinion that GDC is quite superior to a minimum phase correction. And then you will get a pretty good interchannel balance as well.
Out of curiosity have you heard such a correction? With DRC the variables involved lead to the sensible conclusion that no one thing works in all situations but given the right conditions and for me, its a step above and over the tradition thinking on DRC.
The only real problem is its difficult to create an algorithm that will analyse and usefully correct as an automated function. From what I understand the problems are widespread and the solutions varied which doesn't lend itself well to such automation. Instead it requires a trained mind and a fair amount of time. I think Uli mentioned he spent around 2 hours tweaking my own filters for such a result.
That particular measurement wasn't made by me nor is it from my system. Its one Uli performed whilst setting up Acourate for one of his friends.
You'd have to ask him for specifics in that particular case. I used it as an illustrative example.
As already mentioned I can't comment on the example but in my own case I used the traditional Acourate method of driver correction or anechoic correction as you explain it and then measurement from listening position. Afterward the linearised crossover filters and the uncorrected room measurements where sent to Uli where they were worked on to achieve what we're discussing here.
How is group delay correction different to the FIR correction filters that acourate is a capable of generating?
Re: my Idea of a cheapo XO
No Ideas?
otter17 said:Hallo,
here is my Idea of an "el cheapo XO":
I want to use foobar2000 as mediaplayer to play back my .flac-files with the following two plugins:
- ChannelDivider
- foo_convolve
That way I will have:
- lossless files
- FIR filter with correct phase
- flattened, room corrected response
Here is a nice DRC HowTo:
http://www.file-upload.net/download-1350058/DRC_Guide_v1.0.pdf.html
The Audiotrak Prodigy 7.1 Hifi ist the Soundcard I want to use. The two reasons for that are:
- number of channels (8 out)
- internal routing with DirectWire 3.0
I have a Behringer ECM 8000 mic and currently a EMU 1212m here.
The speakers are 2-Way DIY Widerange in Tractrix Horns with 12" Bass:
An externally hosted image should be here but it was not working when we last tested it.
(my Audio-PC in the middle)
Within the next days I want to buy a 4-Way T-Amp from 41Hz.com:
- Amp9 with Tripath TAA4100A
This should work pretty good, shouldn´t it?
Thank You!
No Ideas?
ShinOBIWAN said:
We have a misunderstanding. Of course what you showed works. But if the channels are very asymmetrical to begin with, the contribution from the interchannel correction has a substantial amount of time domain correction in it. That's just the nature of the beast. On the other hand - if the speakers are fairly symmetrical to begin with - there is less work to do for the interchannel correction.
I can understand that. Ref my comment about "wisely used"
OK
Am I reading what I think I am reading?
So Acourate doesn't have this function after all!!!
I am stunned. You have stated a few times that Acourate has functionality that Audiolense doesn't match, and then you produce an example of such that doesn't even exist.
Re: Re: my Idea of a cheapo XO
I don't know how the channeldivider works. But AFAIK, Foo Convolve only can handle two channels.
If that's enough for your system - and the channel divider (XO functionality I assume) follows the Foo Convolve chances are good that this will work pretty well.
I don't know about the sound card you're proposing. But it is always a good idea to browse the www for opinions and also check if anyone is experiencing driver related problems. Driver problems happens from time to time with multichannel setups.
otter17 said:
No Ideas?
I don't know how the channeldivider works. But AFAIK, Foo Convolve only can handle two channels.
If that's enough for your system - and the channel divider (XO functionality I assume) follows the Foo Convolve chances are good that this will work pretty well.
I don't know about the sound card you're proposing. But it is always a good idea to browse the www for opinions and also check if anyone is experiencing driver related problems. Driver problems happens from time to time with multichannel setups.
BerntR said:
No misunderstanding here. I said basically what you've just said - no one solution is viable for all situations.
Am I reading what I think I am reading?
So Acourate doesn't have this function after all!!!
I am stunned. You have stated a few times that Acourate has functionality that Audiolense doesn't match, and then you produce an example of such that doesn't even exist.
Come on Bernt, I had you pinned as an reasonable sort of guy and you come up with this. Look at my post that sparked this exchange:
The open ended nature of Acourate is like a calculator with functions that allow an almost limitless scope of filter manipulation. Uli knows best how to bend his software to creative ends but one example where this went to furthering the sound was matching the pulse responses of the left and right channels after room correctionie. they both perform close to identical in the time domain. Can Audiolense provide functionality to do this?
From here:
http://www.diyaudio.com/forums/showthread.php?postid=1705439#post1705439
Acourate has this functionality because of its open ended nature. If it didn't do what I'm talking about then why would I by advocating for it? Do you see your faulty logic here?
The filters are sent to Uli because he's the one who's smart enough to figure out what needs to be done to achieve a result of matching pulse responses. He uses Acourate to achieve this. Theoretically anyone could do this with a copy of acourate but its takes someone above a casual user level to do it competently. So as it stands Uli offers the invitation to any Acourate owners to send their files and have the computations done by him. He's offers great support like that.
Foo_convolve does nothing more than loading a wave-file. I am pretty sure it it independent from any channel numbers.
ChannelDivider is a FIR filter generator for 2- and 3-way speakers for foobar2000.
My idea is the following:
audiofile (flac) -> foobar -> foo_convolve -> channeldivider -> soundcard -> DirectWire -> outputs -> amp -> speakers -> voila!
ChannelDivider is a FIR filter generator for 2- and 3-way speakers for foobar2000.
My idea is the following:
audiofile (flac) -> foobar -> foo_convolve -> channeldivider -> soundcard -> DirectWire -> outputs -> amp -> speakers -> voila!
otter17 said:
You seem to know this software far better than I do. I wish you the best of luck. If you make all the parts do their parts you have a very good chance of experiencing pretty good sound quality.
I hopy you will come back and tell the world about your results. A multichannel Foobar system with FIR sound correction and digital filters is probably an interesting solution to quite a few.
Hi,
I have a setup with the following config:
SB3 -> Meridian 501 (volume and input switching) -> PC w/Echo Audiofire 8 (doing stereo 3-way XO and DRC) -> Rotel RMB-1066 -> DIY speakers.
I have connected everything, done XO and DRC with Audiolense (won't get into the Accurate vs Audiolense discussion, but I have to say that Audiolense was very easy to use), and loaded the filter into ConvolverVST in Console. All the channels are correctly played, but playback is VERY choppy.
The PC is used for nothing else that hosting Console, and CPU load is below 1%. I have tried to alter the ASIO buffer setting, but without any luck.
The PC has been used with several other soundcards, so could there be a driver problem with one of the other drivers, or is this a common problem with an easy fix?
Appreciate any help.....
I have a setup with the following config:
SB3 -> Meridian 501 (volume and input switching) -> PC w/Echo Audiofire 8 (doing stereo 3-way XO and DRC) -> Rotel RMB-1066 -> DIY speakers.
I have connected everything, done XO and DRC with Audiolense (won't get into the Accurate vs Audiolense discussion, but I have to say that Audiolense was very easy to use), and loaded the filter into ConvolverVST in Console. All the channels are correctly played, but playback is VERY choppy.
The PC is used for nothing else that hosting Console, and CPU load is below 1%. I have tried to alter the ASIO buffer setting, but without any luck.
The PC has been used with several other soundcards, so could there be a driver problem with one of the other drivers, or is this a common problem with an easy fix?
Appreciate any help.....
I think you set the bar pretty high by this one:
Then came the channel matching "test" as a potential proof of evidence.
And now this:
I really didn't expect to find out that the designer was the core part of the solution. That sounds like a prototype to me. I was taking it for granted that you, being a super-user, could easily do this yourself.
The corrections I posted was made with standard Audiolense. The only exception was the transfer from IR to step response view.
ShinOBIWAN said:
Then came the channel matching "test" as a potential proof of evidence.
And now this:
I really didn't expect to find out that the designer was the core part of the solution. That sounds like a prototype to me. I was taking it for granted that you, being a super-user, could easily do this yourself.
The corrections I posted was made with standard Audiolense. The only exception was the transfer from IR to step response view.
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