T+A DAC 200

[Zaphod Beeblebrox]

2 hours ago, Miska said:

No so much, especially with typical processed loudness wars RedBook. A bit of hard digital clipping commonly prominent on modern recordings is enough to trigger it.

Sorry, I had not included hard digital clipping in the universe of real (band limited) music, my bad. Yes of course hard digital clipping can look very much like a square waveform!  Here is the right channel of track 4 from 2008 Magnetic Death (sorry Metallica fans) which is notorious for its place in the loudness wars. Blue is original, Red is upsampled 16x. The levels are different because of the gain reduction to avoid clipping due to inter-sample-overs.

 

2 hours ago, Miska said:

In addition, it is right where many metal dome tweeters have +20 dB resonance peak.

 

I'm not fond of having constant 22.05 kHz whine, especially at +20 dB levels.

But then again 'ringing forever' and 'constant whining' are not true either unless the whole of the music track looks like above.

Here is a snapshot from few seconds later and no 'ringing' and it quite easy to guess which of the two one would rather listen to.

 

I will stop here perhaps further discussions need to happen outside of this thread.

[kennyb123]

45 minutes ago, Zaphod Beeblebrox said:

Here is the right channel of track 4 from 2008 Magnetic Death (sorry Metallica fans) which is notorious for its place in the loudness wars.

Kudos to you for posting graphs to support your arguments - with the source being real music!  

[Miska]

1 hour ago, Zaphod Beeblebrox said:

I will stop here perhaps further discussions need to happen outside of this thread.

 

You will need to use logarithmic Y-scale, in dB's at the corner points.

 

P.S. And I can already see ringing in the original, some work for apodizing filter obviously.

 

[luisma]

On 2/18/2023 at 4:05 AM, OE333 said:

 

I will post some more detail here as soon as I find the time to write the guide. Please give me a couple of days.

Hi, no rush or urgent but did you have a chance to look into the RS233 firmware update? Thank you, that will be awesome if it can be done that way.

[Jud]

7 hours ago, Miska said:

Both extremes are both right and wrong at the same time.

 

I do like this. 🙂

[taipan254]

7 hours ago, Zaphod Beeblebrox said:

Magnetic Death (sorry Metallica fans)

Death Magnetic.

 

Apology accepted.

 

=)

[OE333]

On 4/12/2023 at 5:15 PM, pavi said:

 

does the t+a dac200 perform equally well at dsd512 & dsd1024? 
 

 

Miska has already brilliantly answered your questions - I personally use DSD512.

DSD1024 would give a still better out-of-band noise performance but at the price of a very high work-load on the HQP server...

[OE333]

On 4/12/2023 at 11:01 AM, Nkam said:

... any FFT to show which harmonics are dominant?

I saw the specs and the THD is low enough.  
just which harmonics are there.  
 

thanks in advance! 

 

Here some FFT measurements showing the harmonics of the DAC200 at the RCA and XLR outputs.

 

The PCM measurement was taken with a 192/24 bit input stream, the DSD measurement with a DSD256 stream.

The harmonics do not change significantly with the data rate, just the out-of-band noise decreases with increasing sample rate.

 

The absolute THD value for PCM is 0.0015%, and for DSD it is 0.0006%.

 

Measurement1:  THD for PCM 192/24

 

 

 

 

 

Measurement 2:  THD for DSD256

 

 

 

 

 

And here a measurement of the in-band noise floor:

 

 

[pavi]

On 4/12/2023 at 3:07 PM, Miska said:

 

This depends on how you want to weight things, which aspects are most important for your case. I use it at DSD256 or DSD512.

 

 

Since DAC 200 is not R2R NOS DAC, it doesn't have such point. If you want to send PCM there, use 32-bit output which is best food for it's internal DSP.

 

 

thank you @Miska

[pavi]

1 hour ago, OE333 said:

 

Miska has already brilliantly answered your questions - I personally use DSD512.

DSD1024 would give a still better out-of-band noise performance but at the price of a very high work-load on the HQP server...

thank you @OE333

[Nkam]

1 hour ago, OE333 said:

 

Here some FFT measurements showing the harmonics of the DAC200 at the RCA and XLR outputs.

 

The PCM measurement was taken with a 192/24 bit input stream, the DSD measurement with a DSD256 stream.

The harmonics do not change significantly with the data rate, just the out-of-band noise decreases with increasing sample rate.

 

The absolute THD value for PCM is 0.0015%, and for DSD it is 0.0006%.

 

Measurement1:  THD for PCM 192/24

 

 

 

 

 

Measurement 2:  THD for DSD256

 

 

 

 

 

And here a measurement of the in-band noise floor:

 

 

 

thanks so much! 
 

much appreciated. 
 

nice.   Right on the cusp.  I kinda like that for harmonics 

and the noise floor is nice and low around -140db! 
super !

 

it would have been cool to see the FFT above 20khz and a bit further down like to -150db 
 

something like this attached. 
 

it’s my Marantz HD DAC1 

 

[Miska]

9 hours ago, OE333 said:

Here some FFT measurements showing the harmonics of the DAC200 at the RCA and XLR outputs.

 

I got similar results in my measurements, but even better at DSD256.

 

Here's 1k tone at 705.6k PCM input:

 

And here's the 1k tone at DSD256:

 

[Nkam]

On 3/13/2023 at 1:15 PM, OE333 said:

 

 

This is indeed an intersting question.

 

First: the facts:

The DAC200 uses 4 PCM1795 converter chips for converting PCM signals to analog. The PCM1795 consists of a 8x oversampling filter followed by a ΣΔ DAC.

In case of the T+A devices the 8x oversampling inside the PCM1795 is completely bypassed and the digital signal is directly fed to the ΣΔ  modulator.

If one of the DAC200's PCM oversampling filters (FIR1, FIR2, Bezier1, Bezier2) is selected, the oversampling is done in an external DSP (using T+A's own oversampling algorithms) , the internal filters of the PCM1795 are never used.

 

If the oversampling is switched OFF and NOS1 or NOS2 mode is selected, the external DSP is also bypassed and the incoming PCM stream is directly routed to the ΣΔ  modulator stage of the DACs without any signal processing before the modulator stage.

 

So, as a first result we can state that in the DAC200 all PCM oversampling filters are bypassed in case of NOS mode.

 

The question now is: does the ΣΔ  modulator stage perform oversampling or not ?

 

This question is a bit philosophical and the answer depends on weather you regard the output signal of the ΣΔ  modulator as a highly oversampled 1 bit digital signal (in this case it performs a kind of oversampling), or if you regard the modulators output signal as an analog signal having an average value representing the analog output value - in this case the modulator is DAC delivering an analog output voltage which only needs some averaging (analog low-pass filtering) to get rid of the unwanted high frequency noise and to deliver the wanted analog signal average.

 

Some more details about this topic are given in this paper: https://www.beis.de/Elektronik/DeltaSigma/DeltaSigma.html

 

 

More than this consideration it might be interesting to look at the output signal of the DAC200 and find out if it behaves as would be expected from a NOS DAC.

 

 

1.) Outut signal from the I/V stage in NOS2 mode when the DAC200 is fed with a digital step signal @fs=44.1 kHz (blue trace):

 

 

 

The output signal after the I/V stage of the DAC. It rises instantly at the digital signal step, without any trace of interpolation. A NOS R2R converter would behave exactly the same way.

 

Please note: This signal is the signal before the final analog reconstruction filter. It is an internal signal and can not be measured at the output jacks of the DAC200.

 

----------------------------------------------------------------------------------------------------------

 

2.) The same signal after the analog reconstruction filter, as it appears at the DAC200 output:

 

 

The output signal after the analog output filter slews a bit slower than the signal in the first measurement. This slower slew corersponds to the 120 kHz cut-off frequency of the DAC200's analog output filter (in "WIDE" mode) - it is NOT a consequence of digital interpolation.

 

----------------------------------------------------------------------------------------------------------

 

For comparison:

This is the DAC200 output signal with the same digital step input signal with oversampling filter "FIR1":

 

 

 

This output waveform shows the much slower response due to the interpolation performed by the digital FIR oversampling filter.

 

 

 

The presented step measurements clearly show the Non Oversampling behaviour of the DAC200.

The output is exactly what would be expected from a NOS DAC and the presented output signals are indistinguishable from the output signals of a NOS  R2R DAC fed with the same digital input signal.

 

Imho the DAC200 is very well suited to operate with high quality external oversamplers such as HQ player, because inside of the DAC200 no oversampling/interpolation takes place and the quality of the externally oversampled signal is preserved in all detail.

 

 

 

pardon to ask again for specifics. 
 

so that also means that the noise shapers work in the T+A DAC?

 

because now with my ΔΣ Cirrus DAC I get the Nyquist reconstruction.  But I don’t get the benefit of the increased dynamic range of the noise shaper.  
 

 

thank you once more.  

[bogi]

3 hours ago, Nkam said:

The question now is: does the ΣΔ  modulator stage perform oversampling or not ?

That's what I called as 2nd oversampling stage in this post and the following two.

[bogi]

3 hours ago, Nkam said:

This question is a bit philosophical and the answer depends on weather you regard the output signal of the ΣΔ  modulator as a highly oversampled 1 bit digital signal (in this case it performs a kind of oversampling), or if you regard the modulators output signal as an analog signal having an average value representing the analog output value - in this case the modulator is DAC delivering an analog output voltage which only needs some averaging (analog low-pass filtering) to get rid of the unwanted high frequency noise and to deliver the wanted analog signal average.

Delta sigma modulators in current DAC chips have still digital output. That output may be yet re-arranged to n bit unary form and then converted to analog using n equally weighted elements (they are usually realized in the form of switched resistor DAC or switched capacitor DAC). So the D/A conversion itself (which in fact is a kind of low pass filtering) is implemented behind digital output of delta sigma modulator.

[Nkam]

On 4/13/2023 at 7:09 AM, Zaphod Beeblebrox said:

Sorry if this is OT:

 

This argument is no different than what has been used by the proponents of minimum phase and apodizing filters on the 'dreaded, terrible' pre-ringing of linear phase filters which cannot really be seen nor heard with true music signals but can be easily simulated with artificial test signals (impulse, step, 22.05kHz signal with exactly two samples/cycle)  and does not have much empirical evidence to support it.

 

Yes, in theory any abrupt change in the frequency domain will lead to ringing in the time domain and vice-versa, some of which can be mitigated with windowing. However, ringing at the Nyquist is a phenomenon that is overblown for multiple reasons.

• First the Nyquist is 22.05kHz for CD rates far beyond human audible range.
• There is rarely any significant energy near the Nyquist with real music.
• With true music signals where it has already gone through a bunch of processing and filtering, it is very hard to demonstrate/show  ringing near Nyquist after oversampling with a good quality brick wall filter.

There is also a nice blog post here (it talks about other things too but the main focus is the ringing):  Archimago's Musings: MUSINGS: Digital Interpolation Filters and Ringing (plus other Nyquist discussions and "proof" of High-Resolution Audio audibility)

 

ps: PGGB 256 does away with the concept of ultra long filters and taps, but it is still equivalent to using a near ideal sinc filter.

 

 

 

To be honest I don’t have a clue as to what ‘ ringing’ sounds like. 
 

and the diffence I hear from min phase and linear phase is that , the min phase sounds like instruments are separated from each other too far.  Like they are kind of playing in space alone , if that makes sense. 
Yes there is a bit more ‘ attack’ of the separate instruments this way to my ear. 
 

however a linear phase filter sounds like there is the instruments separately and you can also hear the harmony in between the instruments as a whole sound. 
 

i say harmony, because, let’s say there are two guitars in a room.  Without too much music theory , I can have one guitar play the root note and the other play the fifth and it will create harmony.    The actual instruments will vibrate sympathetically to each other physically, when close enough and loud enough. 
when you hear this, you will hear a ‘ wall of sound’ which is the harmony and also the separate guitars.  
if you have both guitars play dissonance that ‘ wall of sound’ will not exist.  The interments will actually sound ‘ alone’.    
 

i have heard a few DACs with this signature and when I play a min phase filter.  
sometimes it’s a ‘ cool’ different sound.   But some DACs fail to reproduce the ‘ wall of sound’ harmony part. 
 

John Siau also said to me that the audibility of the pre ringing in a myth. 
 

i don’t know what to believe because there are so many conflicting views.  And that’s great actually because that how we figure things out.  By constantly searching.  Not just agreeing blindly. 
 

i can’t hear ringing.    What does it sound like?

 

 

many thanks 

[Nkam]

18 minutes ago, bogi said:

Delta sigma modulators in current DAC chips have still digital output. That output may be yet re-arranged to n bit unary form and then converted to analog using n equally weighted elements (they are usually realized in the form of switched resistor DAC or switched capacitor DAC). So the D/A conversion itself (which in fact is a kind of low pass filtering) is implemented behind digital output of delta sigma modulator.

 

pardon but all that flew over my head. Or most of it. 
speak two year old language to me about this stuff please 

I am not being snarky or rude.  I really don’t understand it but want to learn. 
 

Much obliged. 

[bogi]

@NkamI will try to be simpler. PCM1795 (like other older Burr Brown chips) may have 1 bit modulator output (not multibit like is nowadays more typical). Miska most probably knows that exactly.

But let start it simpler with the case of 1 bit 2 level modulator output. Then such an output signal is in principle the same as DSD signal. You raised a philosophic question if DSD signal is analog or digital. It is usually considered to be digital. I mentioned also other interpretations like it is on border of analog and digital and even an opinion of @tailspn(the debates about 10 years ago) that it is in fact analog. But generally DSD is considered to be a digital signal. So I would 1 bit output of delta sigma modulator consider to be digital signal too.

DAC in narrower sense is a circuit appearing behind digital output of delta sigma modulator. You see it also on PCM1795 block diagram

 

You see that 'Current Segment DAC' is a separate block for each channel. The modulator module name 'Advanced Segment DAC Modulator' can be misleading since you find the term DAC here. The modulator block does not contain DAC. The whole DAC chip is described as Advanced Segment, Stereo Audio Digital-to-Analog Converter. So the modulator block could be also named 'Delta Sigma Modulator of Advanced Segment DAC'.

 

In my previous post I pointed to a fact that most of current D/A chips contain delta sigma modulator with multibit output. Then the technique of n equally weighted elements is used to get better output precision (SNR and dynamic range). In the case of multibit delta sigma DACs it is yet more visible that the modulator output is digital and D/A conversion happens behind it.

[Nkam]

6 minutes ago, bogi said:

@NkamI will try be simpler. PCM1795 (like other older Burr Brown chips) may have 1 bit modulator output (not multibit like is nowadays more typical). Miska most probably knows that exactly.

But let start it simpler with the case of 1 bit 2 level modulator output. Then such output signal is in principle the same as DSD signal. You raised a philosophic question if DSD signal is analog or digital. It is usually considered to be digital. I mentioned also other interpretations like it is on border of analog and digital and even an opinion of @tailspn(the debates about 10 years ago) that it is in fact analog. But generally DSD is considered to be a digital signal. So I would 1 bit output of delta sigma modulator consider to be digital signal too.

DAC is narrower sense is a circuit appearing behind digital output of delta sigma modulator. You see it also on PCM1795 block diagram

 

You see that 'Current Segment DAC' is a separate block for each channel. The modulator module name 'Advanced Segment DAC Modulator' can be misleading since you find the term DAC here. The modulator block does not contain DAC. The whole DAC chip is described as Advanced Segment, Stereo Audio Digital-to-Analog Converter. So the modulator block could be also named 'Delta Sigma Modulator of Advanced Segment DAC'.

 

In my previous post I pointed to a fact that most of current D/A chips contain delta sigma modulator with multibit output. Then the technique of n equally weighted elements is used to get better output precision (SNR and dynamic range). In the case of multibit delta sigma DACs it is yet more visible that the modulator output is digital and D/A conversion happens behind it.

 

i didn’t know I raised a philosophical question if DSDZ is analog or digital.  I just said DSD sounds more soft to me.  If I remember correctly.
 

i need to read up on how DACs work.  I just can’t find any good documentation on it yet on the internet.  
 

thanks so much for trying to explain to me 

[bogi]

1 minute ago, Nkam said:

i didn’t know I raised a philosophical question if DSDZ is analog or digital.

You raised a philosophical question if delta sigma modulator output in DAC chip is considered to be digital or analog:

58 minutes ago, bogi said:

This question is a bit philosophical and the answer depends on weather you regard the output signal of the ΣΔ  modulator as a highly oversampled 1 bit digital signal (in this case it performs a kind of oversampling), or if you regard the modulators output signal as an analog signal having an average value representing the analog output value - in this case the modulator is DAC delivering an analog output voltage which only needs some averaging (analog low-pass filtering) to get rid of the unwanted high frequency noise and to deliver the wanted analog signal average.

I attempted to explain that 1 bit 2 level modulator output is in principle the same as DSD signal which is generally not considered to be analog. I used this to argue that delta sigma modulator output in current DAC chips has to be considered digital and that the D/A stage in narrower sense follows the modulator.

Your thinking was probably inspired by the general delta sigma modulator description you mentioned. Yes, both modulator input and output can be analog too, but in current DAC chips delta sigma modulator is implemented as digital circuit.

[Nkam]

19 minutes ago, bogi said:

You raised a philosophical question if delta sigma modulator output in DAC chip is considered to be digital or analog:

I attempted to explain that 1 bit 2 level modulator output is in principle the same as DSD signal which is generally not considered to be analog. I used this to argue that delta sigma modulator output in current DAC chips has to be considered digital and that the D/A stage in narrower sense follows the modulator.

Your thinking was probably inspired by the general delta sigma modulator description you mentioned. Yes, both modulator input and output can be analog too, but in current DAC chips delta sigma modulator is implemented as digital circuit.

 

i think what I was asking was if the T+A DAC since it has a PCM ΔΣ DAC , if it can upsample using external filters such as HQplayer and have the same results as a NOS dac. 
 

[bogi]

27 minutes ago, Nkam said:

i think what I was asking was if the T+A DAC since it has a PCM ΔΣ DAC , if it can upsample using external filters such as HQplayer and have the same results as a NOS dac.

Miska already answered that question here: https://audiophilestyle.com/forums/topic/63998-ta-dac-200/page/10/#comment-1235019

No matter what was done in the first stage (for example upsampling with external filters) there is no way to skip that 2nd oversampling stage with PCM input. The target fs of the 2nd oversampling stage is ΔΣ modulator operating frequency which is ~ 10MHz range.

 

With DSD input both oversampling stages (including that second one) can be skipped.

[Nkam]

26 minutes ago, bogi said:

Miska already answered that question here: https://audiophilestyle.com/forums/topic/63998-ta-dac-200/page/10/#comment-1235019

No matter what was done in the first stage (for example upsampling with external filters) there is no way to skip that 2nd oversampling stage with PCM input. The target fs of the 2nd oversampling stage is ΔΣ modulator operating frequency which is ~ 10MHz range.

 

With DSD input both oversampling stages (including that second one) can be skipped.

 

and what happens in that stage? 
 

 

for example when I use HQplayer in PCM mode with one of my ΔΣ DACs I get the below. Which is different from the DACs internal filter.  
the other is using the internal filter.  
you can tell which is which l because it’s blatantly obvious I’m using the Sinc M filter 

 

so what is missing more?

 

legit question.  
 

thanks again for all the info!

 

[bogi]

3 minutes ago, Nkam said:

and what happens in that stage?

I assume you are asking to that Zero Order Hold (or Sample And Hold) stage which I referred as 2nd oversampling stage.
2 things happen here:

1) No correct interpolation algorithm is used. Although the 1st stage has more impact on sound than the 2nd, the result is increased noise floor in comparison with the case when new interpolated value would be computed by a suitable algorithm.
2) Since the 2nd stage is unfiltered, repeating images of audio band at multiples of last filtered fs are coming to delta sigma modulator input and then they appear also on modulator output. See my answers to Jud one week ago. They are source of possible intermodulation distortion.

With DSD input and HQPlayer upsampling to say DSD256, HQPlayer interpolation filter algorithms are used up to the destination fs ~ 10MHz. Audio band images (which are result of every upsampling algorithm) are digitally filtered by HQPlayer so they cannot become source of intermodulation distortion.

About the graphs, ask Miska for answer.

[Miska]

@The Computer Audiophile Can you move this to some other thread, it is so much off-topic here.

 

1 hour ago, Nkam said:

and what happens in that stage? 

 

PCM samples are converted into SDM and the resulting data can be then sent to the actual conversion stage that converts digital data into analog signal.

 

1 hour ago, Nkam said:

for example when I use HQplayer in PCM mode with one of my ΔΣ DACs I get the below. Which is different from the DACs internal filter.  
the other is using the internal filter.  
you can tell which is which l because it’s blatantly obvious I’m using the Sinc M filter 

 

so what is missing more?

 

Higher rate digital filters to completely remove images, and better modulator.