T+A DAC 200

[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.

 

[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...

 

[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...

 

[OE333]

10 hours ago, Andruha said:

This clipping problem can explain audible distortions I can hear from time to time. I thought the distortions in the recordings themselves and as D200 was just revealing them but I was wrong.

Also I wonder why nobody is mentioning the compression D200 seems to add to a recording... Like some sort of normalization.  It is noticeable on opera recordings and orchestral staff.

Is it possible to fix by a software update (firmware) or it is a hardware staff?

 

No, it can't explain any perceivable distortions. Maybe a slight reduction in dynamics, but no obvious distortion.

Also the DAC200 does not "compress" or "normalize" the the recording, as I will try to explain below:

 

Intersample overs occur, when (at recording or mastering)  the level of the recording is not properly adjusted and if amplitude peaks in the original music signal go above 0dBFS --- i.e. above the maximum level that can be encoded by the digital (PCM) code.

 

In case of a intersample over, the music signal can only be stored in digital form, because the peak excursions just fall in between  2 samples stored on the disc. By oversampling the original waveform with its original peak excursion is reconstructed (more or less) and this reconstructed peak now falls above 0dBFS. A signal above 0dBFS can not be converted by any DAC.

So intersample overs are in first place a problem of the recording which was done without a proper dynamic headroom for peak excursions. But obviously such recordings exist and we need a way how to deal with this problem. In principle there are two possibilities:

 

1.) clipping the signal at 0dBFS (which the DAC200 does)

2.) scaling the signal downwards in digital domain - i.e. lowering the volume of the recording before doing the oversampling

 

Both methods are a compromise and each one does have its disadvantages. So again: a good recording (and mastering) is the only correct way of dealing with the problem.

 

 

Now a look at the two possibilities from above:

 

Method no. 1 has the (obvious) problem of clipping the (hopefully relatively rare) extreme signal peaks of bad recordings resulting in a slight reduction of peak dynamics with such recordings.

 

Method no. 2 has the problem, that lowering the volume in digital domain inevitably brings a higher noise floor and a reduction in the resolution, especially of of low volume signals. And this will happen with all recordings - good or bad.

Moreover this method brings the problem, that the original samples stored on disc can not be kept untouched like it is possible with exact Bezier oversampling. Scaling will always slightly alter the original signal.

 

So no matter what we do, both methods constitute a compromise and the question arises, which compromise to choose.

 

For good quality recordings with proper headroom, the problem of intersample overs does not exist and method 1 (as in the DAC200) is clearly the better choice.

 

Also for digital sources with adjustable volume method 1 is the better choice - thank you @bogi for your hint.

 

For recordings where the original analog signal had peaks above the max. permissible level and in cases where no volume adjustment is possible, it can be discussed, which method is the better one.

 

So in my opinion, the DAC200 as it is offers the best possible compromise. But this is of course just my personal opinion and I know that there are good reasons to have an other opinion.

 

As I really don't want to start here a discussion about intersample overs, I would prefer to have a look at possible solutions to the problem. I would be very much interested in a solution, which gives the user the choice of how to deal with intersample overs - thanks again to @bogi for your suggestion of making this a user selectable option.

 

The good news:

As the whole digital signal processing in the DAC200 is software defined, no hardware changes will be required.

I am currently investigating, if we can simply add one or two more oversampling algorithms with scaled down coefficients, which would bring back the amplitude level of the intersample overs into the dynamic range of the DAC chips.

 

With such additional oversampling settings everybody may choose (even individually for each recording), which type of oversampling to use - either optimized for low noise and best possible signal integrity or optimized for improved peak handling.

 

I will need a bit time to see how this can best be implemented, but I think it can be done. I will report about our progress.

 

 

[OE333]

11 minutes ago, GoldenOne said:

A 'perfect' DAC would probably have both the option for applying the headroom prior to oversampling AND an indicator of when intersample clipping is occurring so the user knows if they should perhaps enable it or not.

 

That is a very good suggestion. I will discuss with my colleagues, if we can add such an ISO indicator...

[OE333]

15 hours ago, Hydrology said:

Hey @OE333 Ive been trying codes on the PDF you linked to but of all the codes Ive tried so far, only a POWER/TOGGLE command seems to work (for me).

I also tried learning the non-discrete codes from the DAC200's remote but am having difficulty getting both my Logitech Harmony Elite and my new Unfolded Circle Remote Two to learn and recognize the commands. Any ideas? Is the FM200 remote a hybrid of some kind?

Probably the most required codes I need would be discrete POWER ON, discrete POWER OFF, discrete USB input and discrete SPDIF 1 (COAX).

 

 

Here a few hints:

 

As far as I know the Logitech remotes support T+A devices. On the Logitech compatibility site the DAC200 is not yet listed, but you can find some other T+A devices like PA-1200R and DAC 8. The IR commands for all T+A devices are the same - so the Logitech codes for DAC8 and PA-1200R should also work for DAC200.

If that doesn't work, I would recommend to contact Logitech directly. Logitech could then contact T+A for assistance bringing the T+A codes to the Logitech IR code data base.

 

Learning remotes:

Most IR remotes send one specific IR telegram for a button press and repeat this command at a certain rate (normally every 100ms or so).

The T+A telegrams are a bit more complex. They consist of a START command, a specific button command (which is repeated every 120ms as long as the button is pressed) and after releasing the button an END-command.

Start and end commands are there to make the remote operation quicker and more responsive - for example the motorized volume potentiometers of T+A amps will immediately stop turning when the Volume-button is released. Without the STOP command the pot would keep on turning until the IR receiver detects that no more IR telegrams arrive which lasts 100...200ms and during this time the motor-pot would continue to turn and create a little "overshoot".

 

Unfortunately many remotes have problems to learn such more complex IR telegrams - they normally can only learn simple IR telegrams consisting of only one single repeated code.

 

Fortunately the START and STOP commands are not necessarily required. The T+A devices will work even when Start & Stop commands are missing. Learning the T+A codes without Start & Stop codes will normally work with all learning remotes. To capture the T+A IR telegrams without start & stop codes you can proceed as follows:

 

1.) start the learning process on the learning remote

2.) cover the T+A IR transmitter (at the front of the T+A remote) with one hand

3.) now press the button to be learnt on the T+A remote control (IR transmitter still covered)

4.) after a short time (one second or so) quickly remove your hand from the T+A IR transmitter

5.) the learning remote will now only "see" the repeated button commands (the start code was sent when the IR transmitter was still covered)

6.) release the button on the T+A remote when the learning remote control  signals that the code was correctly captured

 

Repeat the above process for all buttons that you need.

 

Note:

Sometimes it can happen, that the hand is removed just in the middle of a IR telegram. In such a case the learning process might not be successful. So when the learned code for a button does not work, just try the process above again - the chance to remove the hand just during a transmission is not very likely, because the length of the IR telegram is 10 times shorter than the pauses in between the telegrams. So the chance of removing the hand at the "wrong" time is just 1:10

 

 

 

[OE333]

2 hours ago, Hydrology said:

Many thanks @OE333 for all your help - I will go through your suggestions and report back!

 

Completely unrelated question, but about the 200 series as a whole - are the isolation feet on the underside of the units simply unscrewable? I know they twist with a gentle nudge and I have some other isolation footers I would like to try in their place, but I know sometimes with footers on equipment unscrewing them sometimes means a nut or washer comes undone inside the cabinet which I dont want to happen. Just thought I would check before proceeding...cheers!

Mark

 

 

The screws for the feet are screwed into threads in the bottom plate from the inside of the case.

If the feet are unscrewed, the screws will protrude from the bottom side as studs.

 

There is a certain chance that the screws might become loose when unscrewing (or even more when re-fitting) the feet.

If these screws loosen, they might touch the circuit board and cause short circuits.

Especially the screw at the rear left (close to the mains socket) might cause a short between live mains and the case.

 

So my strong advice: don't touch the feet - or if you really want to, open the case to check on the inside that no shorts exist before connecting the device to mains again.

 

[OE333]

Hi @Woolf,

 

yes, you will need USB for DSD256 from your streamer.

 

For the CD player I would recommend BNC or AES/EBU. Which of both sounds better, also depends on the player. So maybe try both and decide which is better.

 

[OE333]

9 hours ago, russellbobby said:

Sorry I havent read all 29 pages , but will the optional HDMI card pass SACD / DSD from an OPPO player?  I assume the answer is no but it would be a great feature if available.

Thanks

 

 

The DAC200 fully supports SACD/DSD via its (optional) HDMI inputs.  

 

I found this statement from OPPO:

If your A/V receiver or audio processor supports native DSD, your OPPO player can send DSD signal over HDMI to the A/V receiver or audio processor.

 

According to this there should be no problem.

 

 

[OE333]

1 hour ago, davidv100 said:

Hello,
I am a bit puzzled about what exactly is Dolby Atmos and Apple Spatial Audio.
How does this translate in bits and bit rate : Is this 24/192 or is there no link ?
Are these formats supported by DAC200 ?

I understand at least one thing : when listening to Apple Spatial Audio thru my DAC200 and Hifiman HE1000 V3 headphone, the sound will obviously not change direction when I turn my head. But even more obviously, I do not care at all about that.

So, my question is only about the sound quality :
1) what are these formats exactly ?
2) are they supported by DAC200 ?

Thank you very much

David  

 

Dolby Atmos / Apple Spacial Audio are multichannel formats.

You will need a multichannel Audio Processor to use such formats.

DAC200 is a stereo 2-channel DAC. It does not support multichannel playback.

 

[OE333]

@DonaldM:

The DAC200 is supported as native DSD sink in all Linux kernels 5.x and higher. I don't know which kernel version ifi is using in their streamers, but if they use an older kernel, then this kernel needs to be patched to work with DAC200.

If you have a contact person at ifi you can send me the contact information via PM. I will then try to contact ifi to check if this problem can be solved.

 

 

 

 

[OE333]

6 hours ago, Woolf said:

@OE333

Hi,

can You please give us some more detailed info about the internal filtering (upsampling?, if yes how much, what about DSD files).

Thanks in advance!

Wolfram

 

If the input signal is PCM, then the incoming stream is synchronously upsampled to 705,6 / 768 kSps (depending on wheather the sample rate of the incoming signal is a multiple of 44.1 or 48 kHz).

For this upsampling there is a choice of four different upsampling algorithms (FIR filter short, FIR filter long, pure Bezier Interploation, Bezier+IIR). Furthermore there is a non-oversampling mode in which the internal PCM upsampling is bypassed and the incoming stream is directly sent to the DAC stage. This NOS mode is intended for digital sources with internal upsampling and with high PCM outut sample rates.

 

DSD signals are are directly sent to the T+A True-1-Bit converter stage. For DSD signals only the internal resynchronisation (for jitter attenuation) is active, but there is no internal processing or updsampling for DSD. DSD signals in native or DoP format are accepted with sample rates from DSD64 up to DSD1024.

 

 

[OE333]

13 hours ago, Woolf said:

The reason I asked, I have three positions to up-/oversample (Roon nucleus plus, Auralic Aries G2.2 and the DAC200) and I am always wandering, where this job should be done?

According DSD: does it make sense to convert the PCM files in the Nucleus to DSD and send them as DSD to the DAC200?

 

 

I would send PCM files unaltered from the Nucleus to DAC200 and leave the PCM upsampling to the DAC200.

I wouuld recommend the following settings in DAC200: OVS = BEZier 1 or BEZier 2, FILTER=OFF.

 

DSD files could be upsampled in Nucleus for example to DSD256.

 

An other way to go would be to convert everything (PCM and DSD) to DSD256 or higher in your Nucleus and use the DAC200 as a pure DSD DAC.

 

 

What is better is of course always a question of personal taste and also depends on the other components of your system.

I would recommend to start with one of the above setups and then alter settings (only one at a time) and see if what works best for you.

 

 

13 hours ago, Woolf said:

And one more: To set any filter on the DAC200 has no influence to the DSD files?
 

 

That is correct.

 

 

[OE333]

21 hours ago, camott said:

Fantastic! When do you anticipate this being available? I will order the requisite programming cable asap.

 

I have included the HT Pas-Through in my latest firmware version. I am currently testing it and I think I can release it by the end of this week.

 

Please note:

T+A will include the HT Pass-Through in the next official firmware version V 1.20.

This official version will also be available in short time and it can be installed with the help of a MP200.

 

 

[OE333]

Just now, camott said:

 

Great news. Did T+A ever release a ready-to-go USB -> E2Link  programming adaptor to update firmware? Can't find it on the webshop. I can build my own but happy to spend a few more dollars to get a prebuilt one ... 

 

 

... it should be on the web-shop .   --  I will ask my colleagues to check why it is not there.

For the time being you can contact the T+A service dept. [e-service(at)ta-hifi.com] they will be able to give you more details on price and delivery options.

 

[OE333]

22 hours ago, camott said:

 what is the gain of the preamp section? I notice that when I switch to line out and use an external line stage with unity gain I have much less headroom. So just curious - can’t find gain number in any specs …

 

 

At volume setting "0" the gain for RCA in --> RCA out is 13.5 dB (= factor 4.73)

Gain for RCA --> XLR is 19.5 dB (factor 9.46)

 

At VOLume= -13  the RCA/RCA gain is almost zero (+0.5dB to be exact).

 

 

 

[OE333]

@camott:

 

My plan is to have the volume setting fixed @ -13  In HT Pass-Through mode  - so the overall gain of the DAC200 will be close to zero.

 

An other option would be to make the HTPT gain adjustable - but maybe this would be too much...

I see the danger that the HTPT volume in the DAC200 might get altered accidentally causing the front channels to be out of balance with the other channels.

 

For this reason I currently tend to have a fixed HTPT volume setting --- or do you see any reason to have it variable ?

 

[OE333]

12 hours ago, ericuco said:

If you are looking for a cable to upgrade DAC firmware, I made one a few years ago and was able to successfully do upgrade. PM me if you want to borrow my cable.

 

I guess your cable was made for a DAC8DSD ?

Please note that the DAC8DSD cable will not fit the DAC200.

The DAC8DSD had a RJ12 connector, the DAC200 programming input is RJ45.

[OE333]

14 hours ago, camott said:

 

Can't the pass-through mode effectively use the same path as when "line" is selected for pre output? which I assume bypasses the preamp stage entirely? or maybe "line" bypass is not available for RCA/analog input?

 

 

To bypass the pre amp for analog input signals is not possible. We need the pre-amo to generate a balanced signal for the XLR output from the unbalanced RCA input. This is why the pre-amp stage is always in the signal path for the RCA input.

 

 

15 hours ago, camott said:

I think it should be fixed at 0 gain, ideally passing preamp altogether as per above. All AV receivers have per channel balance/volume offset control. Adding that ability to the DAC200 will just complicate things imo.

 

Yes, I also think to make it as simple as possible with a fixed 0dB gain for HTPT mode.

 

 

[OE333]

On 1/16/2024 at 5:06 PM, droffen said:

@OE333

Can I ran the displays of MP200 & DAC200 (HA200) 12 hours a day turned on?

Or is it better to turn them off to avoid burn-in?

Same question for the meters?

 

In general all electronic displays are wear parts. They normally have a life time of several thousand operating hours and they will last longer, the more their brightness setting is turned down.

The normal life span (defined as the point where the pixel brightness is 30% down) of a white OLED display is about 20.000..30.000 operating hours. This translates to about 3,5 years at 24/7 operation and to about 20 years at a daily use of 4 hours.

 

To prolong the life span of the displays, the 200 series has a "temporary" display mode which turns on the display when a user interaction occurs and turns it off afterwards. With the "temporary" display mode turned on, the display will have a VERY log life time.

 

An other option is to not use the brightest display settings. Using a brightness setting 4  (instead of 7) will increase the display life by about a factor of 2.

 

A final note: if necessary the display can be exchanged quite easily by plug&play - no soldering or any tricky things.

 

 

The meter illuminations is done by LEDs with a very long life time - no need to turn them down.

[OE333]

35 minutes ago, camott said:

 

Hi,

 

Any progress/news on this front? I was able to build my own E-Link programming cable, all with parts from Amazon prime, and verify connectivity with the E_Link_UpdateTool so eagerly awaiting your extended firmware update on GitHub!

 

(Side note - I initially tried running the tool via a windows VM but it only successfully connected on 1 of every 5 or so attempts. Not sure why exactly but not surprising).

 

 

Hi @camott,

sorry that the new FW is not on Github yet, but I caught a bronchitis and was not feeling well during the last days - so I was not able to finish and upload the FW.  But it will be available within the coming days - promised.

 

The reason why the tool will not run reliably on a VM is that communicating with the DAC200 is extremely time-critical.

The tool mimics the T+A E-Link bus communication, which is similar to, but not identical to RS232. Some timings must be precise with a very small tolerance of only a few microseconds. This is not possible with a relatively slow and not just true real-time capable VM. This is by the way the reason why only some specific USB -> RS232 converters will function properly.

 

... and congrats to your self made programming cable. If it finishes the tool's connection check without errors, everything is good and programming should work without problems.

 

[OE333]

23 hours ago, Patu said:

Is there any way to display the volume level with bigger font? Now we have tons of useless (IMO) information on the display all the time. I mean, not useless, but something you don't need to see frequently. Mostly you need to see the volume level when using the DAC200 as preamp also. Now the number is so small that I can't see it from the sofa. Luckily the volume bar helps a little but it's not as accurate. If this is not possible currently, maybe add the possibility in future firmware update, to change the input text to volume level or just show the volume level number with bigger font when you switch it, for a short time? Thanks!

 

 

Good idea. 👍

I will think about it and try to integrate it into the next firmware revision.

 

 

[OE333]

On 2/10/2024 at 5:06 PM, AlwaysListening said:

 

I doubt I'm alone in my appreciation for the time you take to post such detailed information to the forum - thank you.

 

As I edge closer to retirement and plan my final audio system, a source with pre-amp capability connected to active speakers is my firm preference. The DAC 200 would be top of the list if there was volume control via Roon.

 

While I understand the current T+A firmware situation, I simply don't want a return to juggling a tablet and device remote while listening to music.

 

Volume control via roon should be possible.

I will have a closer look at this.

 

I wonder if there are more users out there who would like to see this implemented ???

[OE333]

18 hours ago, Apollo said:

Definitely not for me.

if I can avoid digital volume control, I absolutely will.

And T+A 200 dac is fully equipped to so in my book. Nothing as quick and handy as a decent remote control.

 

This feature would permit the control of the analog (i.e. relay) volume control in the DAC200 from the roon control APP.

[OE333]

On 3/5/2024 at 4:36 PM, Woolf said:

Exactly, the filter settings work, afaik, on all files but not on DSD.

 

 

#Very well answered by @Woolf

 

The Filters in DAC200 are only applied to PCM streams.

DSD always goes directly to the converter stages.