Which DACs bypass digital filtering?

[mansr]

2 minutes ago, Jud said:

I don't know the OP's reason(s).  To me, the goal of accuracy and a DAC that works in the way the OP is asking about can be related.  If we hypothesize that we have interpolation filtering and/or SDM in software producing more accurate results than can be obtained in the DAC's internal processing, then bypassing the internal processing would make sense.

What matters is the accuracy of the software plus hardware available. If the best performing solution involves some hardware processing, that should not be seen as a weakness.

[Jud]

7 minutes ago, mansr said:

What matters is the accuracy of the software plus hardware available. If the best performing solution involves some hardware processing, that should not be seen as a weakness.

 

Yes, best accuracy at whatever price point we're talking about.  Even inexpensive DACs seem to cost more than some fairly good upsampling and SDM software.

[semente]

On 2017-5-28 at 7:24 AM, Norton said:

 

20 hours ago, Jud said:

 

 

Most NOS DACs have components that serve as an analog low pass filter in order to do the reconstruction of the digital bitstream to analog music.  With Redbook input they thus reproduce the situation with the very earliest CD players, whose intermodulation distortion levels led to the use of the phrase "digital sound" in a negative way.  Soon afterward, even before the advent of the DAC as a separate piece of equipment, 8x oversampling chips became typical in order to significantly reduce intermodulation distortion.

 

Edit: If oversampling is applied, filtering is necessary to avoid aliasing and consequent harmonic and intermodulation distortion.

 

But isn't the use of analogue filters compulsory to leave out noise-shaping high frequency trash? 

[semente]

17 hours ago, jabbr said:

This is my understanding:

(I'm going to limit the discussion to DSD in this post and can discuss PCM separately if desired -- just ask)

 

The "sound" is contained in the digital recording. The goal of the reproduction system (DAC + Amp + Speakers) is to accurately product the "sound".  During the playback process, the "sound" is mixed with "noise". In a DSD (SDM) bitstream, the "sound" is directly contained in the "analogue" part of the bitstream, the "noise" is contained in the "digital" clock that is used to transport the stream from one place to another. The function of the DAC is to separate the analogue sound from the digital noise.

 

This is really really simple, so if you don't understand what I've written above, go back and reread, because understanding this is essential to understanding the process. The last sentence, in particular, accurately and specifically describes the function of the DAC.

 

In DSD/SDM the digital noise is contained in the carrier clock (BCLK) as well as its harmonics. The BCLK is necessary to interface the analogue signal with the digital system and the goal of the DAC is to remove all vestiges of the BCLK from the analogue signal without disturbing the signal itself. This where upsampling and filters come into play.

 

Let's say we allow everything to pass including the carrier BCLK -- we can't hear it right? Speakers can't reproduce it right? What's the big deal? That's where intermodulation distortion comes in: high frequency noise interacts with the electronics to produce measurable, audible and very harsh sounding distortion in the audible band.

 

One might consider a "brickwall" filter which would allow the analogue signal to pass and cut off everything above what we define as either 44 kHz or 96 kHz or whatever we define as the upper limit of the analogue signal we want.

 

Well it turns out that these "brickwall" filters also have distortion that extends below the cutoff frequency: the brickwall filters aren't perfect. So a much much better idea is to use a gentle filter at the corner frequency but in order to get the gentle filter to effectively filter out the digital noise we need to "noise shape" which is where the upsampling comes into place: the upsampling increases the frequency of the digital carrier clock (BCLK) thus increasing the frequency separation between the analogue signal and the digital noise and thus improving the ability of the gentle filter to remove the noise. Viola'

 

Now 99% of PCM starts out as SDM/DSD and ends up as SDM/DSD to the same argument applies with the added complexity of where, when and how to convert between SDM and PCM.

 

 

 

Thanks.

I would also be interested in reading the PCM discussion if you feel like writing about it.

[semente]

7 hours ago, mansr said:

What matters is the accuracy of the software plus hardware available. If the best performing solution involves some hardware processing, that should not be seen as a weakness.

 

Could you elaborate a bit on the hardware processing for PCM and DSD? 

[Jud]

2 hours ago, semente said:

 

But isn't the use of analogue filters compulsory to leave out noise-shaping high frequency trash? 

 

An analog filter is necessary, as confirmed by @mansr.  I would say noise-shaped high frequency trash, since that's what noise shaping does, push what you don't want to higher frequencies.

[Jud]

2 hours ago, semente said:

 

Could you elaborate a bit on the hardware processing for PCM and DSD? 

 

I too would like to see information on superior hardware processing (DAC chips, FPGAs, shift registers, whatever) at prices competitive with available software, if @mansr knows of anything that has been implemented in consumer, pro, or DIY equipment.

[Norton]

51 minutes ago, Jud said:

 

I too would like to see information on superior hardware processing (DAC chips, FPGAs, shift registers, whatever) at prices competitive with available software, if @mansr knows of anything that has been implemented in consumer, pro, or DIY equipment.

 

I'm listening now to HQP via NAA into my Esoteric DAC with (as per the TEACs) its onboard upsampling and filtering options set to off.  Sounds good, but is bettered to my ears just fed by my BDP, letting the Esoteric do the up sampling and filtering (although admittedly not when using the DSD conversion option)

 

In terms of superior hardware processing, I borrowed  a DAVE a few weekends back.  The results were superb, better than any digital  I've heard to date.   However it's an expensive  item and I'm interested in how close I could get instead with a DAC specifically tailored to HQP.

 

I can't see though that comparing software vs. DAC prices is meaningful.  The software is just upsampling and filtering, surely the analogue conversion, power supply and output  stage of a DAC make up 50% plus of eventual SQ?

 

 

 

[Jud]

34 minutes ago, Norton said:

 

I'm listening now to HQP via NAA into my Esoteric DAC with (as per the TEACs) its onboard upsampling and filtering options set to off.  Sounds good, but is bettered to my ears just fed by my BDP, letting the Esoteric do the up sampling and filtering (although admittedly not when using the DSD conversion option)

 

In terms of superior hardware processing, I borrowed  a DAVE a few weekends back.  The results were superb, better than any digital  I've heard to date.   However it's an expensive  item and I'm interested in how close I could get instead with a DAC specifically tailored to HQP.

 

I can't see though that comparing software vs. DAC prices is meaningful.  The software is just upsampling and filtering, surely the analogue conversion, power supply and output  stage of a DAC make up 50% plus of eventual SQ?

 

 

 

 

I tend to think of a DAC as one third digital (filter) design, one third analog circuit design, and one third parts quality.  So if one can get a DAC with reasonable analog circuit design and parts quality, and substitute software for the cost of digital filter design, cost efficiencies for you as a consumer might be realized.

 

I'm guessing the DAVE's filtering, implemented in an FPGA, might be pretty well simulated by someone who knew what he or she was doing with the excellent iZotope software and @mansr's SDM, available in Audirvana+ for I think $79 or so.

[rickca]

4 minutes ago, Jud said:

 

I tend to think of a DAC as one third digital (filter) design, one third analog circuit design, and one third parts quality.  So if one can get a DAC with reasonable analog circuit design and parts quality, and substitute software for the cost of digital filter design, cost efficiencies for you as a consumer might be realized.

 

I'm guessing the DAVE's filtering, implemented in an FPGA, might be pretty well simulated by someone who knew what he or she was doing with the excellent iZotope software and @mansr's SDM, available in Audirvana+ for I think $79 or so.

I agree you can simulate a filter in software, but those 3 elements aren't really independent.  There's an overall architecture/design that brings it all together that determines the ultimate performance of the DAC.

[Jud]

Just now, rickca said:

I agree you can simulate a filter in software, but those 3 elements aren't really independent.  There's an overall architecture/design that brings it all together that determines the ultimate performance of the DAC.

 

I don't disagree.  But I've got a $375 DAC that uses commodity DAC chips, nothing special, and I use software upsampling with it to get better sound at far less expense than would otherwise be possible for me.  For anyone willing to do a little DIY, there's Miska's DSC1 with very good parts quality for not much over $400 and some of your time (balanced configuration under $1000).  Miska says it's the best DAC he's heard.  Let's say he's a little proud of his design and there are some very expensive DACs that beat it.  Still an interesting proposition.

 

I'm never averse to trying to gain and use a little knowledge to get something better for less expense.

[jabbr]

3 hours ago, semente said:

Thanks.

I would also be interested in reading the PCM discussion if you feel like writing about it.

 

PCM is conceptually different from SDM (DSD) in that the analogue signal itself has been converted to a digital signal (numbers). DSD stream does not directly contain numbers -- the analogue signal remains present being at the low frequencies while the digital carrier is present at the higher frequencies.

 

PCM is a stream of numbers -- in this case the DAC has to take the numbers and convert to an analogue voltage, rather than (simply) filtering out the high frequency carrier/noise. Classically an R2R resistor ladder is used, and the analogue output filter removes the remaining aliasing noise.

 

I am not going to focus on the implementation details of A/D or D/A conversion as implemented by various chips or software, rather the benefit of upsampling. I'd say that upsampling is even more important for PCM than DSD for the following reason:

 

In PCM, the aliasing noise has its largest component at the sampling frequency and this is what needs to be filtered out. Thus the brickwall filter. The "problem" with analogue filters is that they typically have effects not only on the frequencies above the corner frequency but also at frequencies lower than the corner and so with a corner frequency of 44.1 kHz, the output filter clearly has effects on the analogue signal itself. Not good. By upsampling PCM, this is the same as "noise shaping" DSD in that the aliasing frequency is pushed higher and similarly the analogue output filter may then have less effect on the desired signal. 

 

Of course if the DAC first converts the PCM signal to SDM, then the issues described for DSD apply.

 

I am using the term "digital noise" not in any sort of perjoritave sense that "digital is bad" rather to express from the point of view of the amplifier, speakers etc, this these frequencies, which are artifacts of the digitization process, are what need to be removed by the output filter. I could simply call this noise, but the term "digital noise" is intended to reflect the fact that the principal component is at the digital clock frequency. By upsampling whether PCM or DSD, this clock frequency is increased without changing the signal frequencies thus reducing the effects of the required output filter on the analogue signal itself.

 

Hope this is helpful, and I hope this post hasn't caused anyone to choke on their cheerios  

[jabbr]

6 minutes ago, Jud said:

 

I don't disagree.  But I've got a $375 DAC that uses commodity DAC chips, nothing special, and I use software upsampling with it to get better sound at far less expense than would otherwise be possible for me.  For anyone willing to do a little DIY, there's Miska's DSC1 with very good parts quality for not much over $400 and some of your time (balanced configuration under $1000).  Miska says it's the best DAC he's heard.  Let's say he's a little proud of his design and there are some very expensive DACs that beat it.  Still an interesting proposition.

 

I'm never averse to trying to gain and use a little knowledge to get something better for less expense.

 

Many equations are known and algorithms available and have been implemented in hardware for a considerable time. FPGA is a variant of hardware. Not sure who first implemented upsampling in software but at least for our purposes I credit @Miska for developing the HQPlayer software implementation for DSD (it does PCM also) and @PeterSt for the XXHighEnd for PCM, and the so named NOS1 as its intended DAC. I think price point contains many factors but given what they do the software packages are terrific bargains.

[rickca]

9 minutes ago, Jud said:

But I've got a $375 DAC that uses commodity DAC chips, nothing special, and I use software upsampling with it to get better sound at far less expense than would otherwise be possible for me.

Yes I have a micro iDSD too.  

[mansr]

6 hours ago, semente said:

Could you elaborate a bit on the hardware processing for PCM and DSD? 

Perhaps it's best to study a few examples.

 

Below is a block diagram of the TI DSD1793 chip used in iFi DACs. The PCM1795 in the TEAC UD-501 is similar.

 

 

This shows a standard PCM path with 8x upsampling followed by sigma-delta modulation. A filter bypass mode allows direct input to the modulator at 384 kHz. The datasheet reveals that this is in fact a hybrid design where the modulator, a 3rd order 5-level design, operates on the low 18 bits only. The output is combined with the high 6 bits to form a 66-level code which forms the input to the actual D/A conversion stage.

 

The DSD path is separate and does not involve any digital processing. The D/A stage supposedly consists of a shift register style FIR filter similar to Miska's design but with different weights on each position. The datasheet is vague but supports this idea. Four different filter choices are available.

 

Now look at AKM's top range. Several variants with similar design are available. They show up in devices from TEAC (UD-503), Linn, ESOTERIC, Marantz, and others. This diagram is from the datasheet of the AK4497:

Again, a fairly typical PCM path. The main difference compared to the DSD1793 is the addition of a digital attenuator (DATT). A filter bypass mode is available. I can't find any information on the sigma-delta modulator, but I would assume it is a multi-level design. The SCF (switched capacitor filter) blocks convert the modulator output to analogue.

 

DSD handling is quite different from the TI chip. In the default mode of operation, DSD input is low-pass filtered before going through the same digital volume control and sigma-delta modulator as PCM. As far as I can tell, the modulator is operated at the same rate as the DSD input, i.e. no resampling is performed. A bypass mode allows sending DSD data directly to the SCF without going through the modulator. Oddly, some versions of this block diagram show the digital low-pass filter always being active while others (the figure above) indicate that this too is skipped in the bypass mode. I don't know which is correct.

 

Finally, the ESS Sabre series found in various products ranging from the Audioquest Dragonfly to the Benchmark DAC3.

This is different from what most manufacturers do. PCM input is upsampled using a two-stage FIR filter to a maximum of 1.536 MHz. These filters are fully programmable and can also be bypassed entirely. The FIR filter is followed by volume control and an IIR filter. After this comes "THD Compensation" which I have no idea what it does. Next the data goes through an ASRC which upsamples further to a rate supposedly in the vicinity of 40 MHz (even a datasheet I'm not supposed to have doesn't say). Finally, there are the usual sigma-delta (which I assume is what hides behind the Hyperstream label) and D/A stages.

 

Like in the AKM chip, DSD input is digitally low-pass filtered and subjected to the same processing as PCM. Apparently unique to ESS is that even DSD is upsampled further, and there is no option to disable this.

 

From these examples we can see that each manufacturer has chosen a different approach. The TI design favours simplicity while ESS relies on heavy processing. AKM falls somewhere in the middle. All achieve excellent performance figures. In a final product, the surrounding electronics, notably clocking and analogue output drivers, matter far more than the DAC chip itself.

[mansr]

22 minutes ago, jabbr said:

PCM is conceptually different from SDM (DSD) in that the analogue signal itself has been converted to a digital signal (numbers). DSD stream does not directly contain numbers -- the analogue signal remains present being at the low frequencies while the digital carrier is present at the higher frequencies.

This is a common misconception. I really ought to do a proper write-up explaining how these things actually work.

22 minutes ago, jabbr said:

In PCM, the aliasing noise has its largest component at the sampling frequency and this is what needs to be filtered out. Thus the brickwall filter. The "problem" with analogue filters is that they typically have effects not only on the frequencies above the corner frequency but also at frequencies lower than the corner and so with a corner frequency of 44.1 kHz, the output filter clearly has effects on the analogue signal itself. Not good. By upsampling PCM, this is the same as "noise shaping" DSD in that the aliasing frequency is pushed higher and similarly the analogue output filter may then have less effect on the desired signal.

Aliasing is not noise and has nothing to do with noise shaping. Also, your filter frequencies are off by a factor 2.

[Jud]

2 minutes ago, jabbr said:

 

Many equations are known and algorithms available and have been implemented in hardware for a considerable time. FPGA is a variant of hardware. Not sure who first implemented upsampling in software but at least for our purposes I credit @Miska for developing the HQPlayer software implementation for DSD (it does PCM also) and @PeterSt for the XXHighEnd for PCM, and the so named NOS1 as its intended DAC. I think price point contains many factors but given what they do the software packages are terrific bargains.

 

The software packages also provide more flexibility than commonly found in hardware.  (Of course this allows you to screw up, too.) The iZotope SRC in A+ is used in recording studios (costing $300-$700) and offers adjustable parameters.  A+ also provides a choice of mansr's modulators.  Miska offers quite a variety of filters and modulators.  And Peter does offer custom filters as well as his own single one (I've never been motivated to try a custom filter with XXHE - I like Peter's).

[jabbr]

3 minutes ago, mansr said:

 

 In a final product, the surrounding electronics, notably clocking and analogue output drivers, matter far more than the DAC chip itself.

+1 this is true because implementations up to the analogue point are uniformly high. 

[jabbr]

5 minutes ago, mansr said:

This is a common misconception. I really ought to do a proper write-up explaining how these things actually work.

Not a misconception at all. Are you able to defend yourself in English?

[mansr]

3 minutes ago, jabbr said:

Not a misconception at all. Are you able to defend yourself in English?

Which part of my English do you have trouble understanding? Do I need to dumb down my vocabulary for you?

[jabbr]

6 minutes ago, Jud said:

 

The software packages also provide more flexibility than commonly found in hardware.  (Of course this allows you to screw up, too.) The iZotope SRC in A+ is used in recording studios (costing $300-$700) and offers adjustable parameters.  A+ also provides a choice of mansr's modulators.  Miska offers quite a variety of filters and modulators.  And Peter does offer custom filters as well as his own single one (I've never been motivated to try a custom filter with XXHE - I like Peter's).

I don't frankly fool around with filter parameters, at least yet. I'm sure there are software libraries to come with new filters for new applications. I have no doubt that HQPlayer could be copied with enough time and energy, but the ability to load room correction and other kernels being processed in the SDM domain is terrific. You could develop these kernels in external packages like acourate (iirc).

[jabbr]

1 minute ago, mansr said:

Which part of my English do you have trouble understanding? Do I need to dumb down my vocabulary for you?

Specifically what is the "misconception" and don't use equations or drift off into irrelevant hardware diagrams. Are you able to write plain English --- if find no evidence that you have that capability. 

[jabbr]

37 minutes ago, mansr said:

 

Aliasing is not noise and has nothing to do with noise shaping. Also, your filter frequencies are off by a factor 2.

Try not to argue by authority -- I don't consider that you've copied/reimplemented what Miska did years ago anything that grants you authority in my book.

 

"Aliasing is not noise" -- I never stated that. I have defined the term "noise" explicitly. Never said that aliasing has anything to do with noise-shaping.

 

Perhaps you are having a problem reading & understanding? 

[mansr]

9 minutes ago, jabbr said:

Try not to argue by authority -- I don't consider that you've copied/reimplemented what Miska did years ago anything that grants you authority in my book.

 

"Aliasing is not noise" -- I never stated that. I have defined the term "noise" explicitly. Never said that aliasing has anything to do with noise-shaping.

 

Perhaps you are having a problem reading & understanding? 

Got any more insults while you're at it?

[semente]

1 hour ago, mansr said:

Perhaps it's best to study a few examples.

 

Below is a block diagram of the TI DSD1793 chip used in iFi DACs. The PCM1795 in the TEAC UD-501 is similar.

This shows a standard PCM path with 8x upsampling followed by sigma-delta modulation. A filter bypass mode allows direct input to the modulator at 384 kHz. The datasheet reveals that this is in fact a hybrid design where the modulator, a 3rd order 5-level design, operates on the low 18 bits only. The output is combined with the high 6 bits to form a 66-level code which forms the input to the actual D/A conversion stage.

 

The DSD path is separate and does not involve any digital processing. The D/A stage supposedly consists of a shift register style FIR filter similar to Miska's design but with different weights on each position. The datasheet is vague but supports this idea. Four different filter choices are available.

 

Now look at AKM's top range. Several variants with similar design are available. They show up in devices from TEAC (UD-503), Linn, ESOTERIC, Marantz, and others. This diagram is from the datasheet of the AK4497:

Again, a fairly typical PCM path. The main difference compared to the DSD1793 is the addition of a digital attenuator (DATT). A filter bypass mode is available. I can't find any information on the sigma-delta modulator, but I would assume it is a multi-level design. The SCF (switched capacitor filter) blocks convert the modulator output to analogue.

 

DSD handling is quite different from the TI chip. In the default mode of operation, DSD input is low-pass filtered before going through the same digital volume control and sigma-delta modulator as PCM. As far as I can tell, the modulator is operated at the same rate as the DSD input, i.e. no resampling is performed. A bypass mode allows sending DSD data directly to the SCF without going through the modulator. Oddly, some versions of this block diagram show the digital low-pass filter always being active while others (the figure above) indicate that this too is skipped in the bypass mode. I don't know which is correct.

 

Finally, the ESS Sabre series found in various products ranging from the Audioquest Dragonfly to the Benchmark DAC3.

This is different from what most manufacturers do. PCM input is upsampled using a two-stage FIR filter to a maximum of 1.536 MHz. These filters are fully programmable and can also be bypassed entirely. The FIR filter is followed by volume control and an IIR filter. After this comes "THD Compensation" which I have no idea what it does. Next the data goes through an ASRC which upsamples further to a rate supposedly in the vicinity of 40 MHz (even a datasheet I'm not supposed to have doesn't say). Finally, there are the usual sigma-delta (which I assume is what hides behind the Hyperstream label) and D/A stages.

 

Like in the AKM chip, DSD input is digitally low-pass filtered and subjected to the same processing as PCM. Apparently unique to ESS is that even DSD is upsampled further, and there is no option to disable this.

 

From these examples we can see that each manufacturer has chosen a different approach. The TI design favours simplicity while ESS relies on heavy processing. AKM falls somewhere in the middle. All achieve excellent performance figures. In a final product, the surrounding electronics, notably clocking and analogue output drivers, matter far more than the DAC chip itself.

 

Thanks, this was really helpful.