T+A DAC 200

[OE333]

10 hours ago, Vangelis said:

Is the fuse inside the DAC200 easily accessible? Anyone know the spec on the fuse?

 

Which fuse do you mean ? There is more than 1 fuse inside the DAC200.

 

The 2 mains fuses (one for analog and one for digital pwr supply) depend on the mains voltage - so we also need to know the mains voltage spec. of your device (100V, 115V, 127V, 230V)

[OE333]

Just a short status report:

It seems a few members have meanwhile installed the 1st version of the extended DAC200 firmware.

I have received some reports of successful installations and I hope everything is working well.

 

I am currently working on the second version of the extended firmware for DAC200.

This 2nd version will have the home theater pass-through as an additional feature.

 

The following extra features will then be available with the extended DAC200 firmware:

• storage of filter/OVS and Invert settings for each input
• 2nd USB input
• separate Phones/PreAmp volume&balance
• screen saver for OLED display
• Home Theater Pass-Through for the analog input

 

[droffen]

Hi OE333,

I really hope this will somehow come to the official firmware for the HA200 as well …

[OE333]

7 minutes ago, droffen said:

Hi OE333,

I really hope this will somehow come to the official firmware for the HA200 as well …

 

Well, I think some of the features like Home Theater Pass-Through and separate Phones/PreAmp Volume do not make sense for the HA200.

The other features could become part of the standards firmware, but I think this will to a great deal depend on the feed back on the new features and if enough users express demand for it...

[droffen]

5 minutes ago, OE333 said:

 

Well, I think some of the features like Home Theater Pass-Through and separate Phones/PreAmp Volume do not make sense for the HA200.

The other features could become part of the standards firmware, but I think this will to a great deal depend on the feed back on the new features and if enough users express demand for it...

Exactly everything else besides these 2 features is interesting for me. 

[jrsub]

6 minutes ago, OE333 said:

The other features could become part of the standards firmware, but I think this will to a great deal depend on the feed back on the new features and if enough users express demand for it...

As a DAC 200 user, the second USB input and HT pass-through would come in handy for me in a standard release.  

[Vangelis]

It is a U.S version 115 V. I only see one fuse in the chassis by the transformer. Where is the second fuse?

Thank you

[OE333]

On 10/31/2023 at 10:21 PM, Vangelis said:

It is a U.S version 115 V. I only see one fuse in the chassis by the transformer. Where is the second fuse?

Thank you

 

 

There are two mains fuses ion the DAC200 (see picture below).

The value for the fuses is printed on the PCB (or on a sticker next to the fuse).

 

According to my records the value of the fuse for the analog power supply for 115V mains voltage is   T 630mA L 250V~

For 230V this fuse is T 315mA L 250V~

 

The input mains fuse (TR5 type) is T 2A L 250V~ for all mains voltages.

 

 

[SPAZ]

Shoutout to OE333. Today, I got out of lazy mode and soldered the homemade cable and performed the custom firmware update. I am happy to confirm that the 2nd USB port is working as expected. This awesome for me! Thanks again

[OE333]

@SPAZ:

 

Congratulations - I'm glad to hear that everything works as expected.

Have fun with your 2nd USB input !

 

[e.Latte]

Is the DAC 200 a true non-oversampling DAC? I was informed that some DACs are not true oversampling DACs. I do not fully understand what would make a DAC a true non-oversampling DAC. Thank you in advance for any help or advice on this l. 

[Miska]

11 hours ago, e.Latte said:

Is the DAC 200 a true non-oversampling DAC? I was informed that some DACs are not true oversampling DACs. I do not fully understand what would make a DAC a true non-oversampling DAC. Thank you in advance for any help or advice on this l. 

 

If you run it at DSD, yes it is. PCM side will have at least S/H oversampling.

[Hydrology]

Would anybody have a complete set of remote codes for the 200 series?

Some of the buttons toggle through different inputs and ideally to keep the system as family friendly as possible I would like to set up macros that use discrete codes.

[OE333]

4 hours ago, Hydrology said:

Would anybody have a complete set of remote codes for the 200 series?

Some of the buttons toggle through different inputs and ideally to keep the system as family friendly as possible I would like to set up macros that use discrete codes.

 

 

Here you can find some information about T+A IR remote codes:

https://www.ta-hifi.de/en/support/support-accessories/support-programmable-remote-controls/

 

You can use the above link as a starting point and check if your programmable remote is able to generate these codes.

 

The discrete codes from this list T+A IR Codes will work for the 200 series - even though this is not mentioned in the document.

 

If you have any questions or need help, please let me know.

 

[GoldenOne]

If anyone was curious about measurements I've posted some here: https://goldensound.audio/2023/11/22/ta-dac200-measurements/

The PCM section is 'fine', but the DSD section measures significantly better, and sounds SIGNIFICANTLY better than the PCM section IMO.
If you own this DAC and not using it with HQPlayer DSD you're doing yourself a disservice

[OE333]

Dear @GoldenOne

thank you very much for sharing your measurements and your profound comments on the DAC200.

 

I would like to give some additional information on the following passages from your review:

 

"And also worth noting, the DAC section is galvanically isolated from the digital section, and we can see a SI8660BA isolator chip on the board just to the right of the DAC section. Interestingly, it seems that the clock signals themselves also run through this chip, rather than having the clock source on the DAC side of the isolation barrier. I unfortunately can’t say for certain given as I cannot look at the underside of the board, but there doesn’t seem to be a clock source close to the DACs nor any sort of controller that would facilitate a PLL, so it seems that the PLL and clock itself are on the other side of the isolation barrier."

 

It is correct, that the DAC clock is generated on the "digital" side of the isolation barrier. This is done to keep the noise generated by the digital clocking section out of the DAC and analog section of the DAC200.

To avoid possible jitter, that might be caused by distance and by the SI8660 isolatior chip, the clocking signals are separately transferred to the DAC section via impedance controlled PCB traces and all relevant signals are re-synchronized directly at the DACs. To keep the DAC clock jitter free, a special high frequency pulse transformer (instead of integrated isolator chips) is used for its galvanic isolation.

We found that this technique keeps digital noise effectively out of the analog section and it preserves extremely low DAC clock jitter.

 

 

"... these features are all controlled by an ARM microcontroller by STM."

 

It may look like, but the DAC200 is not controlled by the ARM processor, it is controlled by a MSP3430 micro controller which is located on the front panel PCB.

The ARM controller on the signal board serves as USB input and DSP for the audio signals. This USB receiver is a proprietary T+A development, combining (UAC 3 compliant) USB reception, digital signal processing and generation of the input and control signals for the 1-bit DSD DAC on a single chip. This avoids extended digital circuitry and keeps electro-magnetic noise at a minimum.

 

Intersample overs

Yes, the DAC200 may show some clipping when intersample overs occur. This clipping could easily be avoided by adding some headroom in the oversampling filters. But of course, adding headroom will (as you correctly state in your comments) decrease the signal/noise ratio. 

According to my understanding intersample overshoot can only occur with audio signals close to the upper end of the pass-band and with amplitudes close to 0dBFS. Such signals are very rare in real life and I am really not sure, if avoiding some (minor) clipping at 20kHz/0dBFS is worth sacrificing S/N over the whole frequency band.

So, if it is the general opinion, that intersample overs constitute a severe problem, I will be happy to discuss this matter with my former colleagues at T+A to see if it makes sense to introduce a few dBs of additional headroom in the oversampling filters.

 

[bogi]

55 minutes ago, OE333 said:

So, if it is the general opinion, that intersample overs constitute a severe problem, I will be happy to discuss this matter with my former colleagues at T+A to see if it makes sense to introduce a few dBs of additional headroom in the oversampling filters.

 

Not every album is mastered up to 0dB. I think the right approach is to adjust volume level at digital source (player) side and not to compromise SNR for example for classical music or jazz recordings playback.

 

How many people are using players/transports which don't allow volume control, like old fashion CD players? I have no idea, but on this forum people are using mostly software players, often running also room eq and/or other DSP in them.

[davidv100]

1 hour ago, OE333 said:

Dear @GoldenOne

thank you very much for sharing your measurements and your profound comments on the DAC200.

 

I would like to give some additional information on the following passages from your review:

 

"And also worth noting, the DAC section is galvanically isolated from the digital section, and we can see a SI8660BA isolator chip on the board just to the right of the DAC section. Interestingly, it seems that the clock signals themselves also run through this chip, rather than having the clock source on the DAC side of the isolation barrier. I unfortunately can’t say for certain given as I cannot look at the underside of the board, but there doesn’t seem to be a clock source close to the DACs nor any sort of controller that would facilitate a PLL, so it seems that the PLL and clock itself are on the other side of the isolation barrier."

 

It is correct, that the DAC clock is generated on the "digital" side of the isolation barrier. This is done to keep the noise generated by the digital clocking section out of the DAC and analog section of the DAC200.

To avoid possible jitter, that might be caused by distance and by the SI8660 isolatior chip, the clocking signals are separately transferred to the DAC section via impedance controlled PCB traces and all relevant signals are re-synchronized directly at the DACs. To keep the DAC clock jitter free, a special high frequency pulse transformer (instead of integrated isolator chips) is used for its galvanic isolation.

We found that this technique keeps digital noise effectively out of the analog section and it preserves extremely low DAC clock jitter.

 

 

"... these features are all controlled by an ARM microcontroller by STM."

 

It may look like, but the DAC200 is not controlled by the ARM processor, it is controlled by a MSP3430 micro controller which is located on the front panel PCB.

The ARM controller on the signal board serves as USB input and DSP for the audio signals. This USB receiver is a proprietary T+A development, combining (UAC 3 compliant) USB reception, digital signal processing and generation of the input and control signals for the 1-bit DSD DAC on a single chip. This avoids extended digital circuitry and keeps electro-magnetic noise at a minimum.

 

Intersample overs

Yes, the DAC200 may show some clipping when intersample overs occur. This clipping could easily be avoided by adding some headroom in the oversampling filters. But of course, adding headroom will (as you correctly state in your comments) decrease the signal/noise ratio. 

According to my understanding intersample overshoot can only occur with audio signals close to the upper end of the pass-band and with amplitudes close to 0dBFS. Such signals are very rare in real life and I am really not sure, if avoiding some (minor) clipping at 20kHz/0dBFS is worth sacrificing S/N over the whole frequency band.

So, if it is the general opinion, that intersample overs constitute a severe problem, I will be happy to discuss this matter with my former colleagues at T+A to see if it makes sense to introduce a few dBs of additional headroom in the oversampling filters.

 

Thank you very much OE333 for this very comprehensive technical answer.

 

I am curious to know your take about GS’s statement that DSD is so much better than PCM on DAC200.

 

My point is that I have always wanted to avoid adding a computer and HQP in my audio chain (as a personal choice for simplicity). Should I reconsider this if DSD is so much better (I am purely questioning PCM to DSD conversion, not about the other features of HQP like oversampling).

 

Thank you very much.

[GoldenOne]

2 hours ago, OE333 said:

Dear @GoldenOne

thank you very much for sharing your measurements and your profound comments on the DAC200.

 

I would like to give some additional information on the following passages from your review:

 

"And also worth noting, the DAC section is galvanically isolated from the digital section, and we can see a SI8660BA isolator chip on the board just to the right of the DAC section. Interestingly, it seems that the clock signals themselves also run through this chip, rather than having the clock source on the DAC side of the isolation barrier. I unfortunately can’t say for certain given as I cannot look at the underside of the board, but there doesn’t seem to be a clock source close to the DACs nor any sort of controller that would facilitate a PLL, so it seems that the PLL and clock itself are on the other side of the isolation barrier."

 

It is correct, that the DAC clock is generated on the "digital" side of the isolation barrier. This is done to keep the noise generated by the digital clocking section out of the DAC and analog section of the DAC200.

To avoid possible jitter, that might be caused by distance and by the SI8660 isolatior chip, the clocking signals are separately transferred to the DAC section via impedance controlled PCB traces and all relevant signals are re-synchronized directly at the DACs. To keep the DAC clock jitter free, a special high frequency pulse transformer (instead of integrated isolator chips) is used for its galvanic isolation.

We found that this technique keeps digital noise effectively out of the analog section and it preserves extremely low DAC clock jitter.

 

 

"... these features are all controlled by an ARM microcontroller by STM."

 

It may look like, but the DAC200 is not controlled by the ARM processor, it is controlled by a MSP3430 micro controller which is located on the front panel PCB.

The ARM controller on the signal board serves as USB input and DSP for the audio signals. This USB receiver is a proprietary T+A development, combining (UAC 3 compliant) USB reception, digital signal processing and generation of the input and control signals for the 1-bit DSD DAC on a single chip. This avoids extended digital circuitry and keeps electro-magnetic noise at a minimum.

 

Intersample overs

Yes, the DAC200 may show some clipping when intersample overs occur. This clipping could easily be avoided by adding some headroom in the oversampling filters. But of course, adding headroom will (as you correctly state in your comments) decrease the signal/noise ratio. 

According to my understanding intersample overshoot can only occur with audio signals close to the upper end of the pass-band and with amplitudes close to 0dBFS. Such signals are very rare in real life and I am really not sure, if avoiding some (minor) clipping at 20kHz/0dBFS is worth sacrificing S/N over the whole frequency band.

So, if it is the general opinion, that intersample overs constitute a severe problem, I will be happy to discuss this matter with my former colleagues at T+A to see if it makes sense to introduce a few dBs of additional headroom in the oversampling filters.

 

Thank you very much for this! It's rare that manufacturers are this responsive to third party testing and I really appreciate it when you guys are involved in community discussions.

I've amended the post with corrections on the parts you noted. Many thanks.

On the note of intersample overs, in isolation/for a single tone, then yes they will usually only occur with high frequency high level signals. However with full spectrum content that isn't the case and unfortunately intersample overs are frustratingly common.

The most common cause is simply where tracks have clipped already to a significant extent even before you consider reconstruction, but many other tracks also have intersample overs.
A good way to have a look is to open some tracks in adobe audition as that displays the waveform with reconstruction, OR, use HQPlayer set to -1dB for a while and see how frequently the volume knob turns red (indicating clipping).

Generally -3dB alleviates 95% of intersample overs (minimum phase filters tend to require more headroom however). Though someone a while ago shared with me a really good paper that I've unfortunately not been able to find again (it may have been @Miska that shared it or he may have a copy?).
In the paper they demonstrated that actually intersample overs have no limit to how high they can get, with an interesting artificial test that could be best described as deliberately inducing a sort of 'oscillation' which then produced higher and higher intersample peaks the longer the signal duration prior to cutting off. Which explains why even with -3dB headroom a very small number of songs actually will still clip. (Personally I'd say -3dB is an ideal value to have for internal headroom however. If something is still clipping beyond that then the producer/mastering engineer needs a talking to)

 

It'd be great if the DAC did have at least an option/setting to mitigate intersample overs. As mentioned they are incredibly common, and my personal experience has been that I've never found a situation where sacrificing 3dB of dynamic range in exchange for alleviating that issue was NOT the preferable choice vs letting the intersample clipping remain. 

[OE333]

Thank you very much for this additional information. I will discuss this with T+A and we will dive into this matter more deeply and I will report on our findings...

 

[GoldenOne]

1 hour ago, OE333 said:

Thank you very much for this additional information. I will discuss this with T+A and we will dive into this matter more deeply and I will report on our findings...

 

I've attached some files, a 'source' and an 'upsampled A' file. This is a snippet from "Get Away" from the Electronic Soundtrack album Code Lyoko by the Subdigitals

 

It's an example of a track with some pretty huge intersample peaks. 
After upsampling with HQPlayer, and applying -6dB headroom, we can see that there is signal content reaching upto -1dBfs. Meaning it's an intersample peak of +5dB!

 

Just putting a line at the -6dB mark to see what goes over that there's a significant quantity of other parts over that too:

 

 

The exact level of intersample peaks will depend on the particular filter's design ofc, but yeah, unfortunately it's something that does happen in real music plenty.

 

This was a list from Splice of a few popular songs and the true peak values they found within them:

 

Travis Scott and Drake’s “Sicko Mode” (2.4 dBTP)
Dua Lipa’s “Levitating” (1.8 dBTP)
Doja Cat’s “Say So” (0.8 dBTP)
Carly Rae Jepsen’s “Call Me Maybe” (0.8 dBTP)
Mariah Carey and Boyz II Men’s “One Sweet Day” (0.7 dBTP)
Bruno Mars’ “Uptown Funk” (0.4 dBTP)

 

So generally, -3dB headroom seems to be plenty for the vast majority of scenarios, that's definitely been my experience anyway. But having none at all definitely means that a large number of tracks will produce clipping/distortion that can easily be avoided

[davidv100]

4 hours ago, davidv100 said:

Thank you very much OE333 for this very comprehensive technical answer.

 

I am curious to know your take about GS’s statement that DSD is so much better than PCM on DAC200.

 

My point is that I have always wanted to avoid adding a computer and HQP in my audio chain (as a personal choice for simplicity). Should I reconsider this if DSD is so much better (I am purely questioning PCM to DSD conversion, not about the other features of HQP like oversampling).

 

Thank you very much.

To @OE333 
I forgot to mention I already proudly own a DAC200 and I am very happy with it, using PCM.
So, my question is more precisely : should I be even happier with converting PCM to DSD upfront (using HQP) ?

PS: For the sake of being complete, I am adding that GS has also posted his analysis of DAC200 on Head-Fi, with some additional context. For example GS is praising the excellent Pre-Amp stage of DAC200, confirming what we already have been discussing above. So, this is also nice... 

[SPAZ]

Hi David,

 

I'm not OE333 but I can give you some answers. Personally, the DAC200 still sounds great with music in PCM mode but using HQPlayer and upconverting to DSD is a total game changer sound quality wise. I did the comparisons and I couldn't believe the differences. 

[OE333]

5 hours ago, davidv100 said:

Thank you very much OE333 for this very comprehensive technical answer.

 

I am curious to know your take about GS’s statement that DSD is so much better than PCM on DAC200.

 

My point is that I have always wanted to avoid adding a computer and HQP in my audio chain (as a personal choice for simplicity). Should I reconsider this if DSD is so much better (I am purely questioning PCM to DSD conversion, not about the other features of HQP like oversampling).

 

Thank you very much.

 

I would recommend to try it.

Install HQP, use the free of charge test mode, convert to DSD256 or higher and listen.

There will be a difference.

If it is worth the additional complexity or if you want to stay with your DAC in PCM mode (which also gives a very high sound quality and is absolutely not bad !) only you can decide...

 

[davidv100]

41 minutes ago, OE333 said:

 

I would recommend to try it.

Install HQP, use the free of charge test mode, convert to DSD256 or higher and listen.

There will be a difference.

If it is worth the additional complexity or if you want to stay with your DAC in PCM mode (which also gives a very high sound quality and is absolutely not bad !) only you can decide...

 

If @GoldenOne has not yet done all the rushes for his upcoming video, I hope he will be mentioning how you replied, and how you listened to the community for further improvements and building parallel software upgrades. According to me, your commitment is unseen in the audio community. Plus DAC200 is an incredible all-in-one devise including both a pre-amp and a head-phone amp. I hope GO/GS will mention this.
 

Kudos to both of you, @OE333 & @GoldenOne