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Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project
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Old 23rd November 2014, 10:24 AM   #41
googlyone is offline googlyone  Australia
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Tranquility, looking at the ESS "bit perfect" volume control caught my attention.

The literature you posted a link to is presented in a manner that, in my mind, tells half the story.
--> Using a 16bit DAC to implement D/A and volume is asking a lot of it. The obvious imitation being overall dynamic range.

--> Their paper makes a lot of using a 32bit internal DAC. Yes, I get it, mathematically at 35dB attenuation, a 32bit resolution set of data representing this will have bags of remaining dynamic range.

Here are a few questions the paper avoids:
- The precision (and related Spurious Free Dynamic Range) will be nothing like 32 bits. "32" is great marketing, but the number is not really relevant.
- Interestingly they did allude to the noise floor being -135dB, which is a much more relevant number, and not bad at that.
- The datasheet states S/N 135dB and THD -120dB, both of which are absolutely creditable and realistic.
- For the record, 135dB is about 22 bits.

Do the sums for a decent 24bit DAC, and you will find fairly similar results.
- A 24 bit DAC running at -35dB will at worst have errors of 3-4 ppm (let me go on record that I think talking in "PPM" is misleading), or something way down below -100dB.

What is the intent of the above rant?
- If you implement a volume control in either a 24 or 32bit DAC, then at any sensible level you will have masses of resolution to implement the volume function.
- For decent modern DACs, the SFDR of the DAC will support a volume control down to a good low level.
- You could do this for yourself in the DSP quite happily.
- Oh, I guess the message is I think that the SABRE data you pointed to is overstated: "Don't believe the hype"

As an aside:
- I have measured the performance of a number of DACS (CS4398 and a range of predecessors and Analogue Devices CODECS). These were all "24 bit" DACS, fed 24 bit data from a DSP. This is in a system using digital domain volume implementation. I didn't record the measured results, however the harmonics and overall signal to noise were low enough that I concluded the digital volume was not a problem with these devices and moved on.
- That said, the purist in me led to a number of tests on the PGA2310, 2320. I built a set of DACS using these on the output of a CS4398. It is extremely hard to measure the distortion at these low levels, but I did see evidence that the PGA2320 was increasing harmonic levels. The difficulty is that you need to be very specific about the load impedance you look at.

Last edited by googlyone; 23rd November 2014 at 10:27 AM.
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Old 23rd November 2014, 11:00 AM   #42
TNT is offline TNT  Sweden
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The ESS chip is a 24 bit payload processing capable circuit with a 32 bit internal processing depth structure. So, it's not a "32bit DAC", even if they unfortunately use this term themselves...

Ther is no relevance to speculate what any 32 bit would represent in the analogue domain (e.g. spurious, noise floor etc) as the extra bits (32-24) is used for internal computing/calculations made solely in the digital domain. Rounding/estimation errors of such calculations is relevant to speak of as they would manifest themselves as distorsion / noise in the analogue domain i.e. after the conversion of the resulting 24bit stream. Now, ESS claim that there is no such errors present in their digital "volume" implementation. I understand it as they are shifting the whole payload 24 bit word up or down in the 32bit space, affecting the output only by the static aspects of the low level performance of the 24 bit conversion - at full level, i.e. no attenuation, the volume function do nothing. I think it is very transparent and at least is for me means a system improvement as I have been able to remove a pre-amplifier. Its the net system result that counts

//

Last edited by TNT; 23rd November 2014 at 11:03 AM. Reason: Spelling...
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Old 23rd November 2014, 11:21 AM   #43
Tranquility Bass is offline Tranquility Bass  Australia
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Quote:
Originally Posted by googlyone View Post
Tranquility, looking at the ESS "bit perfect" volume control caught my attention.

The literature you posted a link to is presented in a manner that, in my mind, tells half the story.
--> Using a 16bit DAC to implement D/A and volume is asking a lot of it. The obvious imitation being overall dynamic range.

--> Their paper makes a lot of using a 32bit internal DAC. Yes, I get it, mathematically at 35dB attenuation, a 32bit resolution set of data representing this will have bags of remaining dynamic range.

Here are a few questions the paper avoids:
- The precision (and related Spurious Free Dynamic Range) will be nothing like 32 bits. "32" is great marketing, but the number is not really relevant.
- Interestingly they did allude to the noise floor being -135dB, which is a much more relevant number, and not bad at that.
- The datasheet states S/N 135dB and THD -120dB, both of which are absolutely creditable and realistic.
- For the record, 135dB is about 22 bits.
Yes at the output it may have 22 bits resolution and anything below that is buried in the noise floor. No different than analog processing with the same noise floor. The important bit is the internal arithmetic is 32 bits to start with and not 24 bits. This means that a 32 bit number attenuated by 16 bits or 90 dB still has 16 bits LSB to play with instead of 8 bits for a 24 bit DAC.

cheers
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Old 23rd November 2014, 12:17 PM   #44
googlyone is offline googlyone  Australia
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Agreed on all the above - and I would expect the DSP to be operating at well in excess of 24 bits.

Just remember that once that once you have attenuated the 32 bit data by 16 bits (90 odd dB) there is only 30dB above the noise floor. Oh, and the actual signal is so low you need specialised test gear to even find the signal is there!!!

I have used 32 bits in some software DSP I have done - purely because the ALU operates at 32 bits. I would have been happy with a few bits less - but what the hey. The real challenge hits you in implementing filters (particularly IIR) where some fo the coefficients become very close to one and also very small. Having those few extra bits and the right structure of maths is important then. Much less in the DAC.
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Old 25th November 2014, 10:53 PM   #45
Nonlinear is offline Nonlinear  United States
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Tranquility, this is almost exactly what I've been looking for; you can never have too much DSP bandwidth when using FIR filters for mulitway crossovers and room correction. I've had a miniSHARC or a Najda on my list as the core of my DSP-based crossover solution, but your product could be a much more integrated fit for what I'm looking for.

Quick questions and comments:

1. Would the I2S input support DSD (double DSD, in fact)? I believe the PS Audio I2S interface does support it?

2. I would not need/use the phono preamp section; I will already be using the PS Audio Nuwave Phono Converter which has the I2S outputs that I'd use for double-DSD

3. Would LOVE to see an HDMI input that would route PCM audio through the DSP for eq/crossover/etc.

4. What are your thoughts re: the SW and UI? Something similar to how miniDSP/miniSHARC is handled?

5. I like the idea of including AD and DA on the base offering, then also having expansion connectors to allow an upgrade path. I know I would like the option of tweaking with DAC's in the future, but want to get up and running right away with a base package.

Major kudos!
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Old 26th November 2014, 03:34 AM   #46
Tranquility Bass is offline Tranquility Bass  Australia
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Quote:
Originally Posted by Nonlinear View Post
Tranquility, this is almost exactly what I've been looking for; you can never have too much DSP bandwidth when using FIR filters for mulitway crossovers and room correction. I've had a miniSHARC or a Najda on my list as the core of my DSP-based crossover solution, but your product could be a much more integrated fit for what I'm looking for.

Quick questions and comments:

1. Would the I2S input support DSD (double DSD, in fact)? I believe the PS Audio I2S interface does support it?
I did not know that PS Audio piped DSD down the I2S interface. Do you have anymore details on this ?

Currently the design handles DSD processing via the USB audio interface. It can process DSD64,DSD128 and DSD256 although this has only been simulated and not actually tested.

Quote:
2. I would not need/use the phono preamp section; I will already be using the PS Audio Nuwave Phono Converter which has the I2S outputs that I'd use for double-DSD
Should be no problems. See above.

You may like to try the built-in phono preamp because it makes use of the on-board Sabre ADC which would probably be a better spec device than what is in the Nuwave Converter. Also it would eliminate the conversion step from PCM to DSD inside the Nuwave converter as well as the conversion back to PCM which is necessary for processing by the DSP.

EDIT: I just read the manual for the Nuwave Phono converter and it states that the Phono EQ is done in the analog domain whereas I plan to do it in the digital domain which is much more accurate as well as handling a multitude of different EQ curves with software changes.

Quote:
3. Would LOVE to see an HDMI input that would route PCM audio through the DSP for eq/crossover/etc.
Well there is already a HDMI input to handle I2S audio so I'm not sure if this would be suitable for your needs.

Quote:
4. What are your thoughts re: the SW and UI? Something similar to how miniDSP/miniSHARC is handled?
I will start with something basic and then add to it based on requests

Quote:
5. I like the idea of including AD and DA on the base offering, then also having expansion connectors to allow an upgrade path. I know I would like the option of tweaking with DAC's in the future, but want to get up and running right away with a base package.

Major kudos!
Currently there are no digital outputs specifically for running external DAC's but there is an external digital interface which is being used for the USB Audio interface board. It could easily be configured to drive external DAC's but you would have to forfeit the USB Audio interface. Perhaps another board design without the DAC's and a header might be more appropriate.

cheers

Last edited by Tranquility Bass; 26th November 2014 at 04:00 AM.
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Old 26th November 2014, 05:28 PM   #47
Tytte71 is offline Tytte71  Norway
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I discovered an issue today with my current DSP. Maximum time delay is limited to 10ms. I need twice to implement a double bass array system delaying the back woofers according to the length of the room divided by 343m/s. I also checked MiniDSP and it seems like their standard IIR platforms are limited to 9ms. MiniSharc however can delay each channel by up to 3000ms, way more than my need.
Just for info with regards to your development and variations in applications.
Another thing I would be happy to see is the possibility of using custom biquad parameters on IIR filters to enable use of any filter type (like e.g. allpass, Linkwitz transform and others I haven't though of).
Cheers,
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Old 27th November 2014, 12:13 AM   #48
Nonlinear is offline Nonlinear  United States
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Quote:
Originally Posted by Tranquility Bass View Post
I did not know that PS Audio piped DSD down the I2S interface. Do you have anymore details on this ?
I don't have much specific detail, but PS Audio does specify that DSD output is supported through their "I2S interface protocol" connector in both standard (DSD64) and double-data rate (DSD128). My plan was to use this with I2S inputs an a miniSHARC

Quote:
Originally Posted by Tranquility Bass View Post
You may like to try the built-in phono preamp because it makes use of the on-board Sabre ADC which would probably be a better spec device than what is in the Nuwave Converter. Also it would eliminate the conversion step from PCM to DSD inside the Nuwave converter as well as the conversion back to PCM which is necessary for processing by the DSP.
I will most certainly try out this feature, it sounds promising. If it has sufficient performance, I could possibly replace the Nuwave Phono Converter (NPC). The A/D section of the NPC uses a Burr-Brown PCM4222. They run it as a core DSD converter directly from analog, then if the user wants PCM output instead of DSD, they convert in digital domain. That's one reason I'm looking for DSD via I2S input in your system.

Quote:
Originally Posted by Tranquility Bass View Post
Well there is already a HDMI input to handle I2S audio so I'm not sure if this would be suitable for your needs.
So your product will have a true HDMI input, separate from the PS Audio I2S interface connector? The PS Audio I2S interface is not a true HDMI interface (no handshaking, etc), it just uses a physical HDMI connector to carry the I2S signals.

Quote:
Originally Posted by Tranquility Bass View Post
Currently there are no digital outputs specifically for running external DAC's but there is an external digital interface which is being used for the USB Audio interface board. It could easily be configured to drive external DAC's but you would have to forfeit the USB Audio interface. Perhaps another board design without the DAC's and a header might be more appropriate.
I would only be looking to use other/future DAC's that use I2S inputs. My initial system was going to use BuffaloIII DAC's, but since that uses the ES9018, I would look at a different upgrade path for the long term.

I'm very much looking forward to hearing about your prototype testing, this is a good looking product for my project!
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Old 27th November 2014, 10:11 AM   #49
Tranquility Bass is offline Tranquility Bass  Australia
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Quote:
Originally Posted by Tytte71 View Post
I discovered an issue today with my current DSP. Maximum time delay is limited to 10ms. I need twice to implement a double bass array system delaying the back woofers according to the length of the room divided by 343m/s. I also checked MiniDSP and it seems like their standard IIR platforms are limited to 9ms. MiniSharc however can delay each channel by up to 3000ms, way more than my need.
Just for info with regards to your development and variations in applications.
Another thing I would be happy to see is the possibility of using custom biquad parameters on IIR filters to enable use of any filter type (like e.g. allpass, Linkwitz transform and others I haven't though of).
Cheers,
Will be able to easily accommodate the delays that you require

Also no problems with the custom biquads

cheers
david
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Old 27th November 2014, 10:29 AM   #50
Tranquility Bass is offline Tranquility Bass  Australia
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Quote:
Originally Posted by Nonlinear View Post
I don't have much specific detail, but PS Audio does specify that DSD output is supported through their "I2S interface protocol" connector in both standard (DSD64) and double-data rate (DSD128). My plan was to use this with I2S inputs an a miniSHARC
Don't worry I will work it out

Quote:
I will most certainly try out this feature, it sounds promising. If it has sufficient performance, I could possibly replace the Nuwave Phono Converter (NPC). The A/D section of the NPC uses a Burr-Brown PCM4222. They run it as a core DSD converter directly from analog, then if the user wants PCM output instead of DSD, they convert in digital domain. That's one reason I'm looking for DSD via I2S input in your system.
Currently the analog input board only supports MM cartridges. You would need a head amp or steup transformer for MC cartridges. I was going to add MC cartridge support but ran out of space for an extra relay and impedance selection switch.

Quote:
So your product will have a true HDMI input, separate from the PS Audio I2S interface connector? The PS Audio I2S interface is not a true HDMI interface (no handshaking, etc), it just uses a physical HDMI connector to carry the I2S signals.
I believe that the use of HDMI requires a license and payment of royalties or something along those lines. Perhaps there is a chip that simplifies it.

Quote:
I would only be looking to use other/future DAC's that use I2S inputs. My initial system was going to use BuffaloIII DAC's, but since that uses the ES9018, I would look at a different upgrade path for the long term.

I'm very much looking forward to hearing about your prototype testing, this is a good looking product for my project!
Yes no need for external DAC boards

cheers
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