Hi minidsp,
Im see in your post 187 in this thread that for 3way can use:
1 x miniDIGI to input your source
2 x miniDSP to process your stereo 3way
is this true for 4way too? This make me little confused because in your "Jumper Chart - 3 or 4 way Configuration" looks like you must use
2 x miniDIGI
2 x miniDSP
What is now true?
If 1 x miniDIGI + 2 x miniDSP is not posible what will happen if you I2S signals from miniDIGI connect(parallel) to both miniDSP boards and use 2way plugin + 2.1plugin? In one board 2 way plugin configure for stereo TW and MID and on other 2.1 plugin for MID-BASS and SUB?
Can this be done if I can't use 3 board combination with 4way PEQ plugin?
Im see in your post 187 in this thread that for 3way can use:
1 x miniDIGI to input your source
2 x miniDSP to process your stereo 3way
is this true for 4way too? This make me little confused because in your "Jumper Chart - 3 or 4 way Configuration" looks like you must use
2 x miniDIGI
2 x miniDSP
What is now true?
If 1 x miniDIGI + 2 x miniDSP is not posible what will happen if you I2S signals from miniDIGI connect(parallel) to both miniDSP boards and use 2way plugin + 2.1plugin? In one board 2 way plugin configure for stereo TW and MID and on other 2.1 plugin for MID-BASS and SUB?
Can this be done if I can't use 3 board combination with 4way PEQ plugin?
Im get some answers from minidsp support team... So if someone will search this things:
For 4way stereo we can use 1 x miniDIGI + 2 x miniDSP but limitation is that you can't control both miniDSP together. You will need to connect to one miniDSP, make a modification, then move the USB cable to the other miniDSP to modify the settings...
So there is another question for this 4 way setup:
1.)Which is better option:
-Using "4Way PEQ plug-in" on each DSP and one DSP is for left and another for right side.
-Using "Stereo crossover PEQ plug-in" on one board for TW and MID and "2way PEQ 21 plug-in" on another board using for MIDBASS and SUB(mono)?
2.)Will there be in second option difference in latency between two boards, because use different plugins?
For 4way stereo we can use 1 x miniDIGI + 2 x miniDSP but limitation is that you can't control both miniDSP together. You will need to connect to one miniDSP, make a modification, then move the USB cable to the other miniDSP to modify the settings...
So there is another question for this 4 way setup:
1.)Which is better option:
-Using "4Way PEQ plug-in" on each DSP and one DSP is for left and another for right side.
-Using "Stereo crossover PEQ plug-in" on one board for TW and MID and "2way PEQ 21 plug-in" on another board using for MIDBASS and SUB(mono)?
2.)Will there be in second option difference in latency between two boards, because use different plugins?
Dejan,
In answer to your questions:
Using 4way PEQ plug-in for both speakers is the only option that would work since 4 way plug-in has the architecture of 1 x in, 4 x out. You couldn't do 1x 4way plug-in on one board and 2nd plug-in on the other. How would you get 3way for both L&R? It wouldn't work.
As for your questions about 1 x miniDIGI + 2 miniDSP, it's possible but requires special custom connector and configuring both miniDSP in I2S slave mode and the miniDIGI in I2S Master mode using a special custom configuration for the miniDSP. This configuration also requires custom jumper cables so if you want this system, you need to warn us ahead of your configuration.
Hoping this makes sense
In answer to your questions:
Using 4way PEQ plug-in for both speakers is the only option that would work since 4 way plug-in has the architecture of 1 x in, 4 x out. You couldn't do 1x 4way plug-in on one board and 2nd plug-in on the other. How would you get 3way for both L&R? It wouldn't work.
As for your questions about 1 x miniDIGI + 2 miniDSP, it's possible but requires special custom connector and configuring both miniDSP in I2S slave mode and the miniDIGI in I2S Master mode using a special custom configuration for the miniDSP. This configuration also requires custom jumper cables so if you want this system, you need to warn us ahead of your configuration.
Hoping this makes sense
Hi,
Thanks for your answers about this 4way configuration.
I think that you are not understand what Im mean in second option:
So example:
I have special configured 1 x miniDIGI + 2 miniDSP.
So is possible this:
miniDSP1: run "Stereo crossover PEQ plug-in" which will be used for runing tweeters and mids.
miniDSP2: run "2way PEQ 21 plug-in" which will be used for midbass and mono subwoffer.
Is this configuration possible with this special configuration? If is I preffer it because you can setup tw and mid in same time without reconnecting device to usb port and restarting application... With 4way plugin will be configuring setup nightmare...
I hope that stereo 4way users will get some plugin which will allow configuring both devices in same time... I think it is not to complicated handle both HID device at same time...
Yes, I indeed misunderstood what you were trying to do.
Sure this configuration with the Stereo crossover would work as well.
However from our experience of designing speakers, we're not sure if that would be easier than a 4way plug-in though.. When building/tuning a speaker, getting one speaker, time alignment, crossover frequencies to work is the first step. Getting the stereo field combining to work is the second step. Not the other way around.
As for the idea of allowing control of 2 miniDSP, you'll realize that it's not as bad as you make it sound as a lot of our end users did. Most setup being symetric, a symetric processing should be applied. Getting 2 miniDSPs to be recognized and controlled brings up a lot of configuration/software design issues which the software wasn't designed for. So no, it's not that simple when you look at the real engineering challenges.
Hope this makes sense.
Sure this configuration with the Stereo crossover would work as well.
However from our experience of designing speakers, we're not sure if that would be easier than a 4way plug-in though.. When building/tuning a speaker, getting one speaker, time alignment, crossover frequencies to work is the first step. Getting the stereo field combining to work is the second step. Not the other way around.
As for the idea of allowing control of 2 miniDSP, you'll realize that it's not as bad as you make it sound as a lot of our end users did. Most setup being symetric, a symetric processing should be applied. Getting 2 miniDSPs to be recognized and controlled brings up a lot of configuration/software design issues which the software wasn't designed for. So no, it's not that simple when you look at the real engineering challenges.
Hope this makes sense.
amp-sub system, and main speakers via miniDSP
Is this feasible?
Or if not, what needs to change to allow the use of the existing quality (Rythmik) servo controlled amplifier-sub system, and the main speakers via miniDSP:
Source: a computer (prefer to use my Mac Mini)
> to a USB soundcard - prefer to use my existing M-Audio Audiophile-USB
Simultaneously outputting:
analog > a top quality (servo controlled) sub plate amp
digital > miniDSP > analog > amp for main speakers
Thanks
Is this feasible?
Or if not, what needs to change to allow the use of the existing quality (Rythmik) servo controlled amplifier-sub system, and the main speakers via miniDSP:
Source: a computer (prefer to use my Mac Mini)
> to a USB soundcard - prefer to use my existing M-Audio Audiophile-USB
Simultaneously outputting:
analog > a top quality (servo controlled) sub plate amp
digital > miniDSP > analog > amp for main speakers
Thanks
Otto88,
In answer to your question, I don't see any reason why this setup wouldn't work. The main argument I had in few post about an hybrid system (i.e. part of the system being processed by miniDSP, the other by a passive/external crossover) was just time alignment. i.e. you'll have to account time taken by minidsp kit (1.5 to 2.5ms depending on filters used) to process your signal.
Now, the problem will only occur if you have both processed and un-processed speakers in the same enclosure, with the un-processed signal ahead (in time) of the processed signal and with no way of changing their physical alignment. In your case, because these 2 systems are separate units, simply use space (distance) to make sure that the rythmic is not ahead of the processed miniDSP. You can have the miniDSP ahead (in time) of the rythmic since you can delay this one after in the time alignment, but not the other way around.
Anyway, that's how I try to explain this story.. It's all about time alignment.
hope this explanation makes sense....
In answer to your question, I don't see any reason why this setup wouldn't work. The main argument I had in few post about an hybrid system (i.e. part of the system being processed by miniDSP, the other by a passive/external crossover) was just time alignment. i.e. you'll have to account time taken by minidsp kit (1.5 to 2.5ms depending on filters used) to process your signal.
Now, the problem will only occur if you have both processed and un-processed speakers in the same enclosure, with the un-processed signal ahead (in time) of the processed signal and with no way of changing their physical alignment. In your case, because these 2 systems are separate units, simply use space (distance) to make sure that the rythmic is not ahead of the processed miniDSP. You can have the miniDSP ahead (in time) of the rythmic since you can delay this one after in the time alignment, but not the other way around.
Anyway, that's how I try to explain this story.. It's all about time alignment.
hope this explanation makes sense....
Thanks Tony
I understand the principle. Two questions:
If I have the facts correct, 1 ms equates to (343 m/ 1000) ie 0.343 m.
So 1.5 - 2.5 ms is 0.52 – 0.86 m. 1.7 – 2.8 feet
*Way more than I would have guessed.
“You can have the miniDSP ahead (in time) of the Rythmiks, since you can delay this one after in the time alignment; but not the other way around”
If I understand you correctly, the speakers processed by miniDSP should have their acoustic centres (“luckily” they are fullrangers, hence only 1 acoustic centre) 0.52 – 0.86 m closer to the listening position than the unprocessed subs’ acoustic centres?
I understand the principle. Two questions:
If I have the facts correct, 1 ms equates to (343 m/ 1000) ie 0.343 m.
So 1.5 - 2.5 ms is 0.52 – 0.86 m. 1.7 – 2.8 feet
*Way more than I would have guessed.
“You can have the miniDSP ahead (in time) of the Rythmiks, since you can delay this one after in the time alignment; but not the other way around”
If I understand you correctly, the speakers processed by miniDSP should have their acoustic centres (“luckily” they are fullrangers, hence only 1 acoustic centre) 0.52 – 0.86 m closer to the listening position than the unprocessed subs’ acoustic centres?
Yes, that's correct in practice although I just want to add comments:
- Time alignment in an enclosure matters to prevent destructive wavefront at crossover points. Thinking about it, in a sub + stereo , I'm not sure how much this is be an effect. Imagine the physical distance you already have between your sub and 2 L&R speakers (not even including a miniDSP processing). They may/may not already be time aligned in your system... so what I just say may/may not apply...
- the distance is just ballpark idea based on theory (50% wrong in most cases as you start real world measurements). You won't know for sure how you're supposed to locate your sub and time align your miniDSP until you do a simple impulse + frequency response to see the effect at the crossover points.
Hoping this makes sense.
Thanks Tony
(crossing as high as c 130 Hz, there'd be one 'sub' per channel)
For impulse + frequency response, I was thinking of HOLMImpulse HOLM Acoustics
do you use/ can you suggest an easy to use/ free software package?
Cheers
(crossing as high as c 130 Hz, there'd be one 'sub' per channel)
For impulse + frequency response, I was thinking of HOLMImpulse HOLM Acoustics
do you use/ can you suggest an easy to use/ free software package?
Cheers
Sorry, but no. Not the application it was designed for unless you get your custom board to connect to one of the I2S for example....
Have a look at I2S tech note on our download section for more info.
That's correct, so far only going up to 20Hz. Greater care needs to be applied when designing IIR filters below 20Hz. And although we're doing all calculations in double precision, it's not just a matter of allowing users to enter values below 20Hz,
We'll maybe consider in the future putting the engineering resources (time=money) to see what's required to make it happen.
) ) it will be played in output or just "deleted"? Another words - if I play test tone 15-19hz, will it play something? or just silence...
A little mis-understanding here:
What b-force was talking about was the fact that the PEQ settings in miniDSP plug-in start from 20Hz. The board rolls of much lower than 20Hz (5Hz HPF). The reason why 20-20kHz is mentioned is like any other piece of audio out there, referenced to AES measurement standards, linearity is calculated from 20Hz to 20kHz.
If your audio source has some sub 20Hz signal (like most sources), then surely miniDSP will reproduce it.
Hope this makes sense,
What b-force was talking about was the fact that the PEQ settings in miniDSP plug-in start from 20Hz. The board rolls of much lower than 20Hz (5Hz HPF). The reason why 20-20kHz is mentioned is like any other piece of audio out there, referenced to AES measurement standards, linearity is calculated from 20Hz to 20kHz.
If your audio source has some sub 20Hz signal (like most sources), then surely miniDSP will reproduce it.
Hope this makes sense,
I have a question for mobile use, but this may apply to other applications. Would you ever consider offering dual inputs (4 channels) for those of us that already have external crossovers? This would be used as a dedicated equalizer. Being able to use a parametric and 30 band graphic EQ at the same time is what I had in mind. This would be a much more affordable option than converting a Behringer DEQ2496 for mobile use.
Will a 96kHz SR version of the miniDSP be available in the future? There seems to be a wide range of debate (scientific or otherwise) concerning the merits of 96kHz sampling over 48kHz.
I have a Pioneer SC-05 AVR capable of processing 192kHz/24b source material that outputs either an analog signal thru a 192kHz/24b Wolfson WM8740 DAC or a down-coverted digital SPDIF output. Assuming there would be a future miniDSP with 96kHz capability throughout the signal path, which AVR connection to this miniDSP in your opinion would have better SQ - analog input using the higher resolution DAC/ADC interface or direct digital SPDIF input with its 48kHz SR?
Thanks,
Darrell
I have a Pioneer SC-05 AVR capable of processing 192kHz/24b source material that outputs either an analog signal thru a 192kHz/24b Wolfson WM8740 DAC or a down-coverted digital SPDIF output. Assuming there would be a future miniDSP with 96kHz capability throughout the signal path, which AVR connection to this miniDSP in your opinion would have better SQ - analog input using the higher resolution DAC/ADC interface or direct digital SPDIF input with its 48kHz SR?
Thanks,
Darrell
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