T+A DAC 200

[GoldenOne]

40 minutes ago, davidv100 said:

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

It will definitely be mentioned!

This is exactly the kind of thing I wish more manufacturers did and I will always make a point of noting when a manufacturer has been particularly responsive be it to questions or criticism.

[Andruha]

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?

 

[stereousa]

Quote

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

 

Probably because all the folks are enjoying the DAC and this issue seems to not bother them? BTW, I don’t get why only orchestral staff should be able to hear this compared to non-orchestral staff. Are you saying that people performing in orchestras have better hearing ability?

[bogi]

On 11/23/2023 at 2:07 PM, GoldenOne said:

It'd be great if the DAC did have at least an option/setting to mitigate intersample overs.

If at all, then please an option, not fixed setting, for people with fixed 0dB level digital sources. Because for people with software players that would be redundant processing, which is better to avoid.

[Andruha]

6 hours ago, stereousa said:

 

Probably because all the folks are enjoying the DAC and this issue seems to not bother them? BTW, I don’t get why only orchestral staff should be able to hear this compared to non-orchestral staff. Are you saying that people performing in orchestras have better hearing ability?

I mean stuff not staff :) Orchestral recordings have a very high dynamic range and so opera recordings...

[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.

 

 

[GoldenOne]

Definitely would be nice to have as a user selectable option.

As @OE333 mentioned it's a problem that SHOULDN'T technically exist and is the result of production/mastering simply having things too loud. 
In an ideal world DACs wouldn't have to account for it since it shouldn't be a problem that exists in the music itself, though unfortunately it is.

I suppose one could argue though that tracks having more headroom is no different than the DAC doing it internally and in theory the loss in dynamic range etc should be identical regardless, but still would be better for the tracks to just not have the issue since then they could be pushed riiiiight to the edge of clipping but no further, rather than the DAC having to apply a fixed attenuation that may be more than what is needed for many tracks.

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. But Just having the option would be awesome

[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...

[GoldenOne]

31 minutes ago, OE333 said:

 

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

That'd be amazing if that was doable!

[Shadorne]

Would be great to start a thread on T+A HQP settings.
 

Ok. I like Sinc-long filter with ASDM7EC-Super modulator on HQPlayer feeding DSD256 to T+A DAC200.  This is for my Redbook collection which dwarfs everything else (about 600+ albums). I highly prefer DSD over any PCM with DAC200.
 

After thorough testing the above is the best spatial, transient and timbre to my ears. Nothing else sounds quite right to me (although lots of fantastic sounding options - just none are quite as “perfect“ to my taste which is Sinc-Long.

 

Any suggestions what else to try? What is the most rigorous linear phase sync filter (in keeping with audio engineering tradition since the 80’s) that is recommended?

 

Sinc-long seems to have more space and better soundstage than Sinc-L - is there any other Linear phase pure sinc function filter to try? (apodizing or minimum phase filters are audibly degraded to my ears/taste - I have given up on them)

 

thanks anyone who cares to indulge me….or share their favourites. 
 

I find running at -4dbfs helps clear up some harshness distortion on loud tracks like U2 Walk On (DAC200 does not handle intersample overs so a little digital headroom is a good thing)

 

 

 

 

[Hydrology]

On 11/15/2023 at 4:58 PM, OE333 said:

 

 

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.

 

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).

 

[stereousa]

@Shadorne, that is a great idea about HQP settings. This would be similar to the website that has Foobar2000 for dummies, Part 1 and 2. That website finally made me super comfy with Foobar2000. If we have something like that for HQP, more folks would dive in.

[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

 

 

 

[Miska]

16 hours ago, Shadorne said:

Any suggestions what else to try? What is the most rigorous linear phase sync filter (in keeping with audio engineering tradition since the 80’s) that is recommended?

 

My personal all-round favorite is poly-sinc-gauss-long, it has good time-frequency balance for wide variety of content. One may still like to go one way or the other from there depending on source genre and personal sensitivity/preference.

 

16 hours ago, Shadorne said:

Sinc-long seems to have more space and better soundstage than Sinc-L - is there any other Linear phase pure sinc function filter to try? (apodizing or minimum phase filters are audibly degraded to my ears/taste - I have given up on them)

 

I personally don't like non-apodizing for RedBook content, because so much of RedBook content is infested with error signals make them sound harsh and shimmery due to distorted highs.

 

16 hours ago, Shadorne said:

I find running at -4dbfs helps clear up some harshness distortion on loud tracks like U2 Walk On (DAC200 does not handle intersample overs so a little digital headroom is a good thing)

 

I assume you are doing DSD output from HQPlayer? In such case, inter-sample overs are not an issue.

 

[Shadorne]

@Miska Thank you, and as usual you are highly responsive. Much appreciated! 
 

I am aware of your preferences and I feel I understand as to what for you constitutes improved Transients, Space and Timbre. I can see how the amplifier/speakers/room setup may affect preference for one modulator/filter combo over another and how the DAC itself is going to affect the choice of settings - this is before adding personal taste to the equation.

 

To my ears processing Redbook to DSD256

 

1) apodizing adds slight compression (creates impression of transients because of a more punchy sound in a similar way to all compressors). 

2) minimum phase filters affect timbre and softens transients

 

3) the combination 1 and 2 can in some cases balance but there remains an overall shift in emphasis more towards lower frequencies (so the shimmer and harshness is also gone)

 

4) traditional long sync linear phase still extracts the most out of a recording and is likely (theoretically) the most accurate reconstruction filter (but it does, as you correctly point out, equally present all of the recording issues too - like poor anti-alias filtering from studio downsampling or AtoD)

 

My ears prefer linear phase traditional sync very long sharp cut off filters, as timbre, transients and spatial cues sound most correct or natural (to my taste) especially for vocals. It may be simply what I am used to hearing and I have learnt to adjust for recording issues.

Of course modern music (younger) listeners are no longer exposed to much in the way of natural sounds as everything has been digitally processed (I can hear auto-tune on most modern vocals but most younger listeners may not even realize it having never been exposed to real voices on their favourite music) I can understand that most listeners will not gravitate towards “old school” filters and modulators.

 

On the DSD256 modulator side of things (converting Redbook), I assume noise shaping plays the starring role, and I am amazed at how much the noise floor affects the sense of space. Most digital on most DACs seems trapped largely in the plane of the speakers and audibly I have the impression that sounds emanate more from individual speakers. The ASDM7EC-Super really does magic in freeing the sound from the speakers and making sounds emanate from in between and outside the speakers  - it also retains good depth. A superb modulator to my ears, YMMV, depending on the setup.

 

Question: what would be the most rigorous old school long sync filter that I should try? (I know this will have lots of pre and post ringing but this has never bothered me, possibly because I don’t hear above 14.5KHz anyway).

Thank you again for your tireless efforts to clarify and give honest to the best of your knowledge answers to the endless questions from the community! I know that your online presence helps market HQPlayer but nevertheless your commitment is commendable if not outright incredible.

 

 

 

 

 

 

 

[Hydrology]

5 hours ago, OE333 said:

 

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.

 

 

 

 

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

 

[Miska]

2 hours ago, Shadorne said:

1) apodizing adds slight compression (creates impression of transients because of a more punchy sound in a similar way to all compressors). 

 

But there is no compression (mathematically). Rather you may get some slight expansion because the transient distortion is now gone.

 

2 hours ago, Shadorne said:

2) minimum phase filters affect timbre and softens transients

 

It makes transient attacks faster due to lack of pre-ringing. But use linear phase apodizing filters if you don't like minimum phase.

 

2 hours ago, Shadorne said:

3) the combination 1 and 2 can in some cases balance but there remains an overall shift in emphasis more towards lower frequencies (so the shimmer and harshness is also gone)

 

You may get that impression, because the highs are now cleaner without the glare and shimmer. If you are used to listening all the "digital hash" it may sound a bit darker at first. But if you focus on the attack leading edges and the actual high frequency details, you may notice that it is now much cleaner and smoother when the distortion is gone.

 

2 hours ago, Shadorne said:

4) traditional long sync linear phase still extracts the most out of a recording and is likely (theoretically) the most accurate reconstruction filter (but it does, as you correctly point out, equally present all of the recording issues too - like poor anti-alias filtering from studio downsampling or AtoD)

 

Well, mathematically, poly-sinc-gauss group is much more accurate reconstruction filter, since it has over 100 dB more stop-band attenuation.

 

2 hours ago, Shadorne said:

My ears prefer linear phase traditional sync very long sharp cut off filters, as timbre, transients and spatial cues sound most correct or natural (to my taste) especially for vocals. It may be simply what I am used to hearing and I have learnt to adjust for recording issues.

 

That is likely the case. But I personally prefer to have as much production side digital errors corrected as possible. Those just don't belong to the original analog signal that once entered the recording system.

 

2 hours ago, Shadorne said:

Question: what would be the most rigorous old school long sync filter that I should try? (I know this will have lots of pre and post ringing but this has never bothered me, possibly because I don’t hear above 14.5KHz anyway).

 

I don't think there's such thing as old school long filter, since old school filters are like horrible 100 - 150 taps Remez designs.

 

But if you want to get as very plain long sinc, sinc-Ll or it's sister sinc-long. But of course it means it comes with limitations.

 

[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.

 

[Allan F]

3 minutes ago, OE333 said:

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.

 

Excellent advice, as usual! 🙂

[Hydrology]

So, utilising Roon to do all upsampling to, say, DSD256 (for stability), are there any preferred settings that yield the best results?

I understand there are those who will say HQPlayer is the way to go - not for me at this stage, but if there is ideal upsampling settings in Roon that get me some of the way there, I'm prepared to try!

[Woolf]

@OE333 hi, 

I am using two digital sources (Transports) with the DAC200, an Auralic Aries G2.2 (is on the way) and an Ayon CD-T II CD-transport (without the DSD module). For the streamer I think to use the USB connection because I would prefer the DSD formats up to DSD256 and I think therefor i will need USB?.

As someone who knows the DAC200 and the quality of its connections best, what would You suggest would be the "best sounding" connectors at the DAC for the Ayon CD-T II: AES/EBU, BNC, SPDIF or is it possible to connect the  I2S in any way?

Thanks for Your great support on this forum.

[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.

 

[russellbobby]

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

[davidv100]

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  

[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.