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Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project

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#141
LTD said:
Yes it is. Inside there is a board like this http://www.minidsp.com/images/documents/Product%20Brief-miniSHARC.pdf programmed with DIRAC. This board has I2S inputs/outputs. The idea is to extract the board from the DDRC-22D and to connect it to your system as I explained before. The question is whether it is possible or not.


Thanks
Click to expand...

Currently I only have the I2S input but no output. I can't see the advantage of using I2S with this minidsp product since it sample rate converts the input to 96KHz so there will be no difference between an SPDIF or I2S connection. Also this minisharc setup may not be setup for I2S even though it exists on the expansion connector.

cheers
 
Last edited:
#142
steph_tsf said:
Please add the necessary analog inputs and loopbacks enabling realtime optimizations during the setup using pink noise audio :

- need to measure the transfer function of the loudspeaker, in realtime
- need to measure the transfer function of the correction, in realtime
- need to measure the global transfer function (correction + loudspeaker), in realtime

Three audio signals to monitor :
- pink noise
- DSP filter output
- measurement mike output

Three transfer functions to monitor :
- correction = DSP filter out / pink noise in (gain and phase)
- overall = measurement mike / pink noise in (gain and phase)
- loudspeaker alone = measurement mike / DSP filter out (gain and phase)

Need to get rid of the massive delays introduced by the ADC's and DAC's.
An easy way is to provide analog loopbacks. Doing so you don't need to fiddle with ADC's and DAC's delay compensations. Or course you'll remain in charge of taking the FIR filters delays in account.

If there are enough MIPS you can specify a target transfer function (in gain and in phase), and attain it by running a LMS (Widrow-Hoff) during the setup phase. This would lead to a purely FIR filter correction, say three times a 8k FIR filter in a 3-way system. Or possibly, if you want to spare MIPS : a 8k FIR filter for the bass, a 1k FIR filter for the midrange, and a 128 FIR filter for the treble.

Need to do the bass setup, the midrange setup, and the highs setup, all separately.

Finishing with a global setup (all channels working), providing some global equalization / pre-processing for shaping the deep bass range, say from 10 Hz to 100 Hz using a Linkwitz Transform (IIR-based deep-bass equalizer requiring the 32-bit precision), hopefully with variable Fc in function of the volume setting.

This would be unique and powerful, while remaining elegant and simple, conceptually speaking.

Need to ensure a decent communication (using USB) between the DSP and the PC (running Windows) for guiding the user, through the setup, in a clever way, letting the user learn and understand what the system is doing. The system needs to export three measurement curves to the PC, at the same time (see above). If you want to spare DSP MIPS, the system needs to export three kinds of audio frames (see above) to the PC, letting the PC calculate and display the three transfer functions.

In 2015, a dedicated audio DSP platform must provide this.
Click to expand...

There is a balanced microphone input so it is possible to measure the frequency response or impulse response of the speaker system. It is also possible to internally measure the frequency response of the filters to check their validity. Some of the other suggestions you mentioned can be accommodated in this design.

For long FIR filter lengths used for bass or room correction in the bass region it is not necessary to use 192K sample rate. 48K sample rate or 4 to 1 decimation would be sufficient. This would get you the 8K FIR for both channels.

cheers
 
#145
Tranquility Bass said:
Currently I only have the I2S input but no output. I can't see the advantage of using I2S with this minidsp product since it sample rate converts the input to 96KHz so there will be no difference between an SPDIF or I2S connection. Also this minisharc setup may not be setup for I2S even though it exists on the expansion connector.

cheers
Click to expand...

May be you are right, anyway, at this moment I don't know yet whether Is possible to connect the miniSHARC board between the output of your input selector and you DSP using SPDF lines. I would appreciate if you answer me about this and if affirmative, how do you do it?

Thanks in advance
 
#146
LTD said:
May be you are right, anyway, at this moment I don't know yet whether Is possible to connect the miniSHARC board between the output of your input selector and you DSP using SPDF lines. I would appreciate if you answer me about this and if affirmative, how do you do it?

Thanks in advance
Click to expand...

Currently on the Digital Audio IO the board has the following digital inputs:-

S/PDIF - TOSLINK
S/PDIF - Coax
S/PDIF - AES/EBU
I2S - PSAudio standard

the following digital outputs:-

S/PDIF - TOSLINK
S/PDIF - Coax

It is possible to do loop through but then this would take up the only S/PDIF input and output channel on the SHARC DSP. I didn't envisage that loop-through would be a requirement since it was assumed that all of the dsp needs would be catered for by this board.

But there is a way around this with the IO expansion port currently used for the Amanero USB board.

cheers
 
#147
Tranquility Bass said:
Currently on the Digital Audio IO the board has the following digital inputs:-

S/PDIF - TOSLINK
S/PDIF - Coax
S/PDIF - AES/EBU
I2S - PSAudio standard

the following digital outputs:-

S/PDIF - TOSLINK
S/PDIF - Coax

It is possible to do loop through but then this would take up the only S/PDIF input and output channel on the SHARC DSP. I didn't envisage that loop-through would be a requirement since it was assumed that all of the dsp needs would be catered for by this board.

But there is a way around this with the IO expansion port currently used for the Amanero USB board.

cheers
Click to expand...
Ok, I understand.

Your setup is fantastic but may be with a modular design, no so integrated, would permit more flexibility in order to introduce user improvements easily. Perhaps, next time.

Thanks a lot for your answers.
 
#148
LTD said:
Ok, I understand.

Your setup is fantastic but may be with a modular design, no so integrated, would permit more flexibility in order to introduce user improvements easily. Perhaps, next time.

Thanks a lot for your answers.
Click to expand...

If you need DIRAC functionality then it is probably better to work out what it does and then implement it on the board rather than go through another bit of hardware and sample rate down and up conversion stages.

cheers
 
#149
Digital inputs

Are multiple digital coax inputs possible, with switching between sources?

I understand volume control is in the digital domain, so this device seems to be capable as an ADC for my analogue preamp, with source switching for digital sources (CD, DAB tuner, streamer), and multiple DAC ouputs to power amps.

Just reassure me my understanding is correct, and that multiple digital inputs are catered for ..

cheers, Maurice
 
#150
Carrera2S said:
Are multiple digital coax inputs possible, with switching between sources?

I understand volume control is in the digital domain, so this device seems to be capable as an ADC for my analogue preamp, with source switching for digital sources (CD, DAB tuner, streamer), and multiple DAC ouputs to power amps.

Just reassure me my understanding is correct, and that multiple digital inputs are catered for ..

cheers, Maurice
Click to expand...

Currently there is only one coax S/PDIF input. How many did you want ?

regards
 
#151
Digital inputs

Currently I envisage needing to switch between 3 digital inputs .. both current digital devices have coax, one has toslink as well.

So 3 would be a minimum I think - I haven't settled on a streamer yet, but most seem to have coax or AES/EBU XLR.

This would make your device compulsory !

cheers, Maurice
 
#153
Carrera2S said:
Currently I envisage needing to switch between 3 digital inputs .. both current digital devices have coax, one has toslink as well.

So 3 would be a minimum I think - I haven't settled on a streamer yet, but most seem to have coax or AES/EBU XLR.

This would make your device compulsory !

cheers, Maurice
Click to expand...

It seems that you can never have a enough inputs so I will look at doing a super IO board with 4 of each type of input. ie 4 x Coax, 4 x S/PDIF, 4 x TOSLINK. But it will be a very long board and will now require a 3U case to accommodate all of the boards.

cheers
 
#159
Hsmeets said:
I know this comes late in the game but what about pluggable modules so people can choose their own config? the modules can be USB,analog line or phono, spdif, aes or toslink.
Click to expand...

In engineering it's usual a controversy, both HW and SW, between a monolithic architecture or an open architecture (with pluggable modules). May be because I am engineer in computers I always prefer an open architecture. In this specific case, imho:

  • Inputs
  • ADC
  • Input selector (pre-amp)
  • DSP
  • DAC
  • Outputs
 
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