Which DACs bypass digital filtering?

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I have been unsuccessfully searching for a list of high-performance* commercial DACs that allow user to bypass digital filtering (NOS), leaving that task to a software music player such as HQ Player.

It would be interesting if we could compile such a list in this thread.

 

* - no TDA1543-based DACs please, only equipment that can take 24bit and sampling rates above 176.4/192kHz.

[semente]

I know that the following qualify:

 

- Phasure NOS1 DAC

- Teac UD-501 and UD-503

 

How about the Lampizators?

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1 hour ago, trappy said:

Aqua DACs all avoid it, I think, no?

 

I'll check, cheers.

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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? 

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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.

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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? 

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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.

[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.

 

What are the advantages of further upsampling DSD?

And potential disadvantages?

Is it better to do it with hardware or in a computer by software?

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3 hours ago, Jud said:

 

I have seen ads for some boards from China - don't know if they're fully populated and what the quality might be.

 

Since schematics and (IIRC) suggested parts lists are available,  another possibility is to pay someone local with decent electronics skills to build one for you.

 

Is it just a hardware thing or does it require programming? 

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17 hours ago, mansr said:

Digital filters are always more flexible than analogue ones. Especially for DSD64 where the noise is quite close to the audio band, a digital low-pass filter followed by remodulation at a higher rate can give an end result superior to using an analogue filter directly.

It's always possible for a poor implementation to do more harm than good. If the implementation is of reasonable quality, I see no downsides.

There is no difference in principle between hardware and software digital filters. A hardware filter is simply a network of multipliers and adders hardwired to perform one function. A computer uses exactly the same kind of arithmetic blocks to do its calculations, only here they are controlled by a program so the function is not fixed. That said, cheap DAC chips might use lower precision arithmetic in order to save silicon space (and power). Good DACs have 32-bit filters whereas computers can easily use 64-bit precision, although the benefit of that is marginal at best. The main advantage of software is the flexibility it offers.

 

What about jitter?

Doesn't it get increasingly higher with higher sample rates?

What's the "current" limiting threshold for DSD? 

[semente]

Thanks to everyone who participated.

The list of DACs hasn't grown much, though:

 

- Phasure NOS1 DAC

- Teac UD-501 and UD-503

- Aqua DACs

- Lampizator DACs

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7 minutes ago, AnotherSpin said:

 

Jud, would you share more details what micro-iDSD settings you use? Thank you.

 

 

Probably the analogue filters?

 

Filters
– PCM	Bit-Perfect Processing/Minimum Phase/Standard	Digital filters selectable
– DSD	Extreme/Extended/Standard Range	Analogue filters selectable
– DXD	Bit-Perfect Processing	Fixed analogue filter

[semente]

Apparently here's another one for the list:

 

http://www.audio-gd.com/R2R/R2R11/R2R11EN.htm

 

Audio-gd R2R 11

•True NOS , R-2R discrete ladder DAC
•Built in pre- and headhpone amp
•Built in 2 groups DA-8 R-2R modules with 450MHz ultra speed CPLD
•Unique noise process technology will remove any digital artifact
•DSD native , DXD, PCM up to 384KHz Support
•Amanero Combo 384 USB interface

[semente]

On 25/10/2017 at 6:03 PM, NoellEagan said:

Hi..i am a new user here. As per my knowledge 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.

 

The idea is to have a D/A only DAC, leaving the upsampling and the filtering to the music file player/processor.

[semente]

A quick update to the list:

 

- Phasure NOS1 DAC

- Teac UD-50x family

- Esoteric (some models?)

- Aqua DACs

- Lampizator DACs

- Audio-GD (some models)

- Spring AudioDACs (some models?)

- Denafrips DACs

- iFi micro (some models)

- Metrum Acoustics DACs

 

[semente]

6 hours ago, Miska said:

And of course, if it wasn't mentioned before:

 

- T+A DAC8 DSD

 

I wasn't sure about that one.

[semente]

I don't mind a bit of side-stepping but you're taking this a bit too much away from the topic.

It's nonetheless interesting so why no start a new thread. Maybe Chris can give a hand moving some of the last messages?

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Update:

- Aqua DACs

- Armature Asterion (same as Holo Spring)

- Audio-GD (some models)

- Denafrips DACs

- Esoteric (some models?)

- Lampizator DACs- Spring AudioDACs (some models?)

- Holo DACs (some models)

- iFi micro (some models)

- Lampizator DACs

- Metrum Acoustics DACs

- Phasure NOS1 DAC

- Teac UD-50x family

[semente]

- Aqua DACs

- Armature Asterion (same as Holo Spring)

- Audio-GD (some models)

- Denafrips DACs

- Esoteric (some models?)

- Lampizator DACs- Spring AudioDACs (some models?)

- Holo DACs (some models)

- iFi micro (some models)

- Lampizator DACs

- Metrum Acoustics DACs

- Phasure NOS1 DAC

- RME ADI-2 DAC

- Teac UD-50x family

[semente]

- Aqua DACs

- Armature Asterion (same as Holo Spring)

- Audio-GD (some models)

- Border Patrol

- Denafrips DACs

- Esoteric (some models?)

- Lampizator DACs- Spring AudioDACs (some models?)

- Holo DACs (some models)

- iFi micro (some models)

- Lampizator DACs

- Metrum Acoustics DACs

- Phasure NOS1 DAC

- RME ADI-2 DAC

- Teac UD-50x family

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1 hour ago, Summit said:

Totaldac has some NOS DACs and sound great.

Audio note has many NOS DACs as well.

MHDT has some like Atlantis, Orchid and Stockholm.

 

Thanks.

Asfar as I know, AN DACs are Redbook, and Redbook should be oversampled.

[semente]

- Aqua DACs

- Armature Asterion (same as Holo Spring)

- Audio-GD (some models)

- Border Patrol

- Denafrips DACs

- Esoteric (some models?)

- Lampizator DACs- Spring AudioDACs (some models?)

- Holo DACs (some models)

- iFi micro (some models)

- Lampizator DACs

- Metrum Acoustics DACs

- MHDT (some models)

- Phasure NOS1 DAC

- RME ADI-2 DAC

- Teac UD-50x family

- TotalDAC (some models?)

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1 hour ago, opus101 said:

I consider that more in the realm of transportable than portable

 

Sorry, wrong topic. My mistake.

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1 hour ago, opus101 said:

I consider that more in the realm of transportable than portable

 

This thread is about non-portable, NOS, filterless DACs that can take 24-bit and at least 175.4/192kHz.

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5 hours ago, butifull01 said:

Trinity DAC can bypass perfectly with LIANOTEC.

 

I don't feel like reading the manual.

 

Can the Trinity DAC run in a NOS & filterless mode?