T+A DAC 200

[SPAZ]

I get all that but I would like to have two concurrent connections that can do higher DSD rates since I have a PC connected and also a Holo Audio red.  Having two USB connections would be great too. I think for most of the use cases one connection would be enough so I get that.

[e.Latte]

On 9/7/2023 at 3:39 AM, OE333 said:

 

Volume (and all other DAC200 settings) can be controlled from a computer via RS232. If this is an option for you I can give you some advice on how to do that - please let me know.

 

 

I have a question that is about volume control. I use a computer as my music server and I also have Roon and HQ Player installed. I use an NAA adapter that plugs into the DAC 200 usb in. With this current setup I cannot control the volume. Will RS232 allow me to control volume using this setup?

 

 

[StreamFidelity]

13 hours ago, e.Latte said:

With this current setup I cannot control the volume. Will RS232 allow me to control volume using this setup?

 

There is an alternative. I have my preamplifier (in the SDV 3100 HV) fixed at 65dB. 

 

I control the volume via Roon. This passes the volume on to the HQPlayer. This is very comfortable and sounds very good. 

 

By the way, the digital volume control has some advantages. No channel imbalance and no noise (unless the amplifier is noisy). And no bit loss, because the HQPlayer upconverts everything to at least 64Bit before. 6dB correspond to 1Bit. 

 

You can also drive the power amplifier directly. But be careful. I once forgot to deactivate the Windows system chime. The system wanted to alert me to something and my ears were ringing. But the sound was very good. 😂

[bogi]

1 hour ago, StreamFidelity said:

By the way, the digital volume control has some advantages. No channel imbalance and no noise (unless the amplifier is noisy). And no bit loss, because the HQPlayer upconverts everything to at least 64Bit before. 6dB correspond to 1Bit.

 

That's not so easy. Noise level remains the same with digital volume control - it's the quantization noise.
Then, no bit loss is valid only for digital domain, when processing is performed at higher bit resolution than the audio content resolution. Lowering of digital volume level by 6dB approximately looks like shifting audio content one bit right, so the highest bit will be always zero and thus not used for audio content. So with every further volume lowering by 6dB you effectively shift audio content by one bit again. If you have enough bits in digital domain, no information is lost. So far OK.

 

Then, it is important to consider analog domain in the picture. Resolution of DAC is restricted to say 20 bits. Let's your audio content is 16bit recording.

16 bit recording at 0dB digital volume setting:
x ...16 bits usable for audio content
o ... remaining bits still within 20bit DAC resolution
- ... bits out of DAC resolution (not usable for audio content with given DAC)

xxxxxxxxxxxxxxxxoooo------------

When digital volume is lowered by 24 dB, 4 upper bits are no more used and 16 bit audio content is shifted to lower bits:

0000xxxxxxxxxxxxxxxx------------

You see there are no remaining bits within 20 bit DAC resolution for 16bit recording. When you would lower digital volume yet more, part of input digital audio content could not be reflected at DAC output because is would be lost below DAC noise floor.

 

With 24bit recording it is a bit more complicated. Audio content is 24 bit, but it's questionable what's the real resolution of the recording. No A/D or D/A converter is able of 24 bit resolution (although DACs are going close). But the same principle works with 24bit content too. You loose more and more bits of original recording when you output digital volume control processing result to devices with restricted resolution.

 

Digital volume control has advantages and disadvantages. It is wise to use it in a range considering your real DAC resolution. Digital volume control in player is the safe way to avoid intersample overflows in DAC. Some of you may experience that lowering digital volume level and increasing amp volume level change sound a bit. It may have more reasons. One of them is that DACs usually provide lowest distortion somewhere at -10 dB. It may be good idea to find an optimal digital volume level for your chain for example somewhere between -3 and -12dB according to your listening test and then to use analog volume control for adjustments.

 

I have HQPlayer set for volume adjustment in range from -12dB to 0dB, so I am not getting lower that -12 dB with digital volume control. Of course, I am using analog volume control of my headamp.

 

[OE333]

There have been quite a few questions during the past days. Mainly about (Volume) control and about an additional I2S input for DAC200 and HA200.

I will try to give some answers to both topiccs in separate posts following.

[OE333]

Question1 - Control of DAC200/HA200 from a PC

As already mentioned, it is possible to control the DAC/HA 200 via a serial RS232 connection.

The RS232 control interface is used internally by T+A for production and QA tests in the factory but it could be used to implement a remote control of the DAC from a PC. Unfortunately there is no official T+A documentation of the RS232 interface pulicly available.

 

For further information I have made a short list (see attachment) of commands implemented in the DAC that could be used to control the DAC.

 

As you can see, the commands are plain ASCII text commands. These commands can be sent from any computer with the help of a terminal program (eg. "Putty" or Windows "Hyperterm").

Sending for example the command VOL 50 will set the volume control of the DAC to 50%.

In the same way all the other functions (input switching, output ON/OFF etc.) can be controlled.

 

Currently there is no APP with graphical user interface available for controlling the DAC - just the ASCII text cammands from a terminal.

It would be quite easy to make a simple graphical user interface for the basic control of the DAC.

Question: Is there a demand for such a GUI control application ????

 

Control over USB instead of RS232

Technically it would be possible to send the commands over USB instead of a separate RS232 connection. To achieve this the USB receiver firmware needs to be extended to handle control commands. Such an extention of the firmware might be quite expensive and I don't think that it will be implemented in the near futrure.

 

DAC200_RS232_Commands.pdf

[Shadorne]

1 hour ago, bogi said:

One of them is that DACs usually provide lowest distortion somewhere at -10 dB. It may be good idea to find an optimal digital volume level for your chain for example somewhere between -3 and -12dB according to your listening test and then to use analog volume control for adjustments

Digital volume attenuation is a failure because it fails to respect that analog electronics perform best at line level. No digital volume attenuation can compare to well designed analog. It is always best to run all your equipment at the volume level where it performs best - therefore run most devices at their full line level output of the design (usually where it performs best overall). Finally a good preamp can attenuate the signal at the last stage prior to the final power amp output stage. 
 

This is good headroom management. The headroom of your setup is governed by your weakest link in the chain so everything should be run at the optimal setting (just like selecting the correct gear for a car maximizes the performance output of your engine).
 

High quality analog attenuation from a dedicated preamp in the last stage makes an audible difference in my experience. At normal listening levels of -40dB attenuation you are losing a lot of audible details when passing such a sub-optimal signal into the analog stage.  Like trying to move a car from a stop in 3rd gear - it can be done but performance suffers.

 

[barrows]

1 hour ago, bogi said:

 

That's not so easy. Noise level remains the same with digital volume control - it's the quantization noise.
Then, no bit loss is valid only for digital domain, when processing is performed at higher bit resolution than the audio content resolution. Lowering of digital volume level by 6dB approximately looks like shifting audio content one bit right, so the highest bit will be always zero and thus not used for audio content. So with every further volume lowering by 6dB you effectively shift audio content by one bit again. If you have enough bits in digital domain, no information is lost. So far OK.

 

Then, it is important to consider analog domain in the picture. Resolution of DAC is restricted to say 20 bits. Let's your audio content is 16bit recording.

16 bit recording at 0dB digital volume setting:
x ...16 bits usable for audio content
o ... remaining bits still within 20bit DAC resolution
- ... bits out of DAC resolution (not usable for audio content with given DAC)

xxxxxxxxxxxxxxxxoooo------------

When digital volume is lowered by 24 dB, 4 upper bits are no more used and 16 bit audio content is shifted to lower bits:

0000xxxxxxxxxxxxxxxx------------

You see there are no remaining bits within 20 bit DAC resolution for 16bit recording. When you would lower digital volume yet more, part of input digital audio content could not be reflected at DAC output because is would be lost below DAC noise floor.

 

With 24bit recording it is a bit more complicated. Audio content is 24 bit, but it's questionable what's the real resolution of the recording. No A/D or D/A converter is able of 24 bit resolution (although DACs are going close). But the same principle works with 24bit content too. You loose more and more bits of original recording when you output digital volume control processing result to devices with restricted resolution.

 

Digital volume control has advantages and disadvantages. It is wise to use it in a range considering your real DAC resolution. Digital volume control in player is the safe way to avoid intersample overflows in DAC. Some of you may experience that lowering digital volume level and increasing amp volume level change sound a bit. It may have more reasons. One of them is that DACs usually provide lowest distortion somewhere at -10 dB. It may be good idea to find an optimal digital volume level for your chain for example somewhere between -3 and -12dB according to your listening test and then to use analog volume control for adjustments.

 

I have HQPlayer set for volume adjustment in range from -12dB to 0dB, so I am not getting lower that -12 dB with digital volume control. Of course, I am using analog volume control of my headamp.

 

When considering digital volume control, it is also important to own up to the fact of how much dynamic range is actually available in room.  Even in a very quiet room when playing music at very high levels (say 120 dB peaks), one is only going to have, perhaps, around 12, maybe 14 bits of real, audible dynamic range.  So considering what happens to 16, 24, or more bits is of no concern in the real world.  Properly implemented digital volume control (as it is in HQPlayer, for example), in a system with appropriate gain, has essentially no real world compromises.

[barrows]

5 minutes ago, Shadorne said:

Digital volume attenuation is a failure because it fails to respect that analog electronics perform best at line level. No digital volume attenuation can compare to well designed analog. It is always best to run all your equipment at the volume level where it performs best - therefore run most devices at their full line level output of the design (usually where it performs best overall). Finally a good preamp can attenuate the signal at the last stage prior to the final power amp output stage. 
 

This is good headroom management. The headroom of your setup is governed by your weakest link in the chain so everything should be run at the optimal setting (just like selecting the correct gear for a car maximizes the performance output of your engine).
 

High quality analog attenuation from a dedicated preamp in the last stage makes an audible difference in my experience when at normal listening levels because at typical attenuations of -40dB you are losing a lot of details.

 

If you need to attenuate the level by 40 dB for typical listening levels your system has poor gain matching and is subject to higher noise floor because of such.  Lowering the gain of the amplifier would be a good idea here as doing so would lower the noise floor.

[Apollo]

Question related to the (use of the) HDMI module.

 

95% of the time, I will use the DAC200 with Usb+Nos2 oversampling (which means no oversampling, but also no filtering)

 

The remaining 5% I will cast audio/video to a Chromecast TV, connected to HDMI module.  There's also a connection from the DAC200 HDMI out to my EPSON projector.

 

Is there any potential harm when starting HDMI audio (and video) whilst the DAC200 is in Nos2 position?

 

Reason for asking as there is only information on the (small) display of the DAC200, it is easy to forget or make a mistake with filter (OVS) setting when switching input.

 

My request would be to have the Nos1/Nos2 modes disabled when switching to HDMI. 

 

Dirk

 

 

 

 

[bogi]

3 minutes ago, barrows said:

Even in a very quiet room when playing music at very high levels (say 120 dB peaks), one is only going to have, perhaps, around 12, maybe 14 bits of real, audible dynamic range.

 

I am not so much focused on loudspeaker topics since I am listening on headphones, but I am surprised by those numbers.

Maybe you mean an usual not much treated room with quite high amount of reflections, where real acoustic pressure level cannot lower so much and so quickly like in open space.

[Kalpesh]

Bob Katz published a list of good CDs to be played with volume settings varying from +2 to -9 dB with regards to his K20 system ; as we all know no single volume setting can match the dynamic compression out there.

However, using adequate eQ (in room response) with at least 2 options for treble options, most recordings can be played within a 0 to -12 range.

I set my preamp internal gain (+8.5) so that at -2 HQP volume, I'm -8 with regards to Katz's 0 when preamp volume setting is at unity gain. Most records play fine in that ball park. Need to lower volume ? I lower digitally in HQP. Need to crank ? I turn up the analog preamp volume up to 8 dB (max) so that I get Katz's 0 dB. Very very very very scarcely I have to add 1 or 2 dB via gain

[Shadorne]

2 minutes ago, barrows said:

If you need to attenuate the level by 40 dB for typical listening levels your system has poor gain matching and is subject to higher noise floor because of such.  Lowering the gain of the amplifier would be a good idea here as doing so would lower the noise floor.

It depends. Like the car analogy - no use having a big engine on narrow tires.
 

I need to attenuate the level by 40 dB for typical listening levels because my system has enough headroom it can play up to 120 dB SPL cleanly. 
 

The fact that most speakers can’t be tested above 95 dB SPL (without fear of damage) illustrates another way in which headroom can be compromised (like narrow tires on a muscle car).

 

Headroom is what you hear with live music and live instruments. It is one of the characteristic audible differences of a playback system that lets you know it isn’t real music or real instruments. Drum sets and pianos can be whisper quiet and also do 110 dB unamplified, not to mention trombone, trumpet etc. - headroom is the “life” in live music and of course the visual experience of watching the musicians perform.
 

 

[Kalpesh]

11 minutes ago, bogi said:

 

I am not so much focused on loudspeaker topics since I am listening on headphones, but I am surprised by those numbers.

Maybe you mean an usual not much treated room with quite high amount of reflections, where real acoustic pressure level cannot lower so much and so quickly like in open space.

https://en.wikipedia.org/wiki/Audio_bit_depth 12 BITS 74 DB so 36 dB noise floor/110 dB peak...

[OE333]

Question 2 - Additional I2S input for the DAC200/HA200

 

First of all some basic facts regarding the I2S inteface:

I2S (= "Inter IC Sound") was developed as an internal interface between audio circuits (DSPs, codecs, ADCs, DACs). This interface was never meant to be used as an external interface between devices.

 

Standardized external audio interfaces are S/P-DIF, TOSLINK, AES-EBU, HDMI, USB, etc. These interfaces are standardized and in general provide good compatibility between devices which use there interfaces.

 

Some companies have started to (mis-)use I2S as an external interface between a digital source (e.g. SACD player) and DAC.  The reasons for this might be that

1.) the audio data needs to be encoded and packaged into a S/P-DIF, HDMI, USB stream and be decoded at the receiving end

2.) some of the above external interfaces are not capable of very high sample rate PCM or DSD signals

3.) additional costs for encoder/decoder, license fees (eg. HDMI, HDCP)

 

So some companies may have chosen to directly send the internal I2S data straéam to the external DAC. This saves expenses for electrical components and the (really very expensive) license fees.

 

A common claim is that I2S is sonically superior due a very "short" and direct signal path and the elimination of encoding/decoding hardware.

Well, I am a fan of short signal paths, but in case of I2S I have my doubts. I completely agree with the arguments of @barrowsabove. I2S is a synchronous inteface needing to transport the clocking signals from the source device to the sink. I think, who has ever seen clocking signals in the multi-Megahertz region after travelling over a 1-2m distance and over some plugs, cables and inteconnects would not think any more that I2S over external cables is a very good idea...

 

Audio over an asynchronous interface such as USB with clocking sgnals generated directly at the D/A converters is technically (and commonally sonically) superior.

 

Anyway - I2S as an external interface exists and it is used by some source devices. My personal advice would be that if the source has some other interface like USB or HDMI which supports the wanted audio formats, always use this interface.

 

Are there any source devices which ONLY have an I2S output ?

In this case it would of course be desirable to have an I2S input option for the DAC/HA200. As I have stated earlier, it would be possible to create a I2S input module for the DAC/HA200 which could be used instead of the HDMI option. So this would be a hardware add-on - a pure software based solution is not possible.

 

So my question: is there a need for such a I2S input board and for which source devices shall it be used.

The latter question is quite relevant, because there exists no standardization for an exteral I2S inteface. Different sockets/plugs (HDMI, RJ45..) are used, the pinnings differ from manufacturer to manufacturer and also the electrical parameters of the interface are different.

 

So, these things will have to be discussed.

 

An other interesting question was to have an additional USB input on the DAC/HA200. I will come to this question in my next post.

 

 

[Kalpesh]

6 minutes ago, Shadorne said:

It depends. Like the car analogy - no use having a big engine on narrow tires.
 

I need to attenuate the level by 40 dB for typical listening levels because my system has enough headroom it can play up to 120 dB SPL cleanly. 
 

The fact that most speakers can’t be tested above 95 dB SPL (without fear of damage) illustrates another way in which headroom can be compromised (like narrow tires on a muscle car).

 

Headroom is what you hear with live music and live instruments. It is one of the characteristic audible differences of a playback system that lets you know it isn’t real music or real instruments. Drum sets and pianos can be whisper quiet and also do 110 dB unamplified, not to mention trombone, trumpet etc. - headroom is the “life” in live music and of course the visual experience of watching the musicians perform.
 

 

my system is 120 dB impulse pressure capable too ; yet, it's tuned so that -8 at preamp is median setting for optimum snr 

[OE333]

Additional USB input for the DAC/HA200

 

The DAC/HA200 comes as standard with two USB inputs, one of which (USB-SYS) is reserved for the MP200 streamer.

 

Technically it would be possible to make the USB-SYS input available for general use if no MP200 is connected. In this case it would be possible to connect two USB source devices to the DAC/HA200.

 

I am not sure if T+A would approve such modification but if there is a broad demand for it I could discuss it at T+A and see what I can achieve...

[StreamFidelity]

I keep the volume of the preamp high for this reason:

"So usually higher you can set the amp volume, less attenuation you have and thus higher dynamic range."

 

I combined the digital volume of the HQPlayer with the analog volume of the preamplifier as recommended here:

For me, this solution is very comfortable and the volume control from HQPlayer is very good. I do not hear any deterioration, quite the opposite. 

 

For those who prefer analog volume control, this is just as well. T+A offers a solution for this. Very good customer service. 🙂

 

[Shadorne]

35 minutes ago, barrows said:

 Even in a very quiet room when playing music at very high levels (say 120 dB peaks), one is only going to have, perhaps, around 12, maybe 14 bits of real, audible dynamic range

Good point but headroom is what allows those 14 bits to cleanly reach our ears. Every compromise starting with a speaker that distorts at 95 dB SPL to digital attenuation can ultimately compromise what is heard on certain recordings.

[Miska]

2 hours ago, OE333 said:

A common claim is that I2S is sonically superior due a very "short" and direct signal path and the elimination of encoding/decoding hardware.

Well, I am a fan of short signal paths, but in case of I2S I have my doubts. I completely agree with the arguments of @barrowsabove. I2S is a synchronous inteface needing to transport the clocking signals from the source device to the sink. I think, who has ever seen clocking signals in the multi-Megahertz region after travelling over a 1-2m distance and over some plugs, cables and inteconnects would not think any more that I2S over external cables is a very good idea...

 

What I've measured, those external I2S are invariably worse in terms of jitter performance due to such clocking issues. It could be made correct by having the clock go opposite direction from DAC to the source. And then some I2C/SPI control protocol to control the clock selection from the source. But unfortunately the common way to have external I2S implemented has the clocking wrong way around with it's negative effects.

 

In best case DAC runs the clock input through PLL similar way as for S/PDIF and AES/EBU. Which results in only slightly worse clocking performance than with internal clocks.

 

[Miska]

3 hours ago, barrows said:

When considering digital volume control, it is also important to own up to the fact of how much dynamic range is actually available in room.  Even in a very quiet room when playing music at very high levels (say 120 dB peaks), one is only going to have, perhaps, around 12, maybe 14 bits of real, audible dynamic range.  So considering what happens to 16, 24, or more bits is of no concern in the real world.  Properly implemented digital volume control (as it is in HQPlayer, for example), in a system with appropriate gain, has essentially no real world compromises.

 

For example typical HQPlayer modulator output use case has dynamic range of over 180 dB, depending on case can be also well over 200 dB. Thus the dynamic range will exceed what ever the analog stages following the DAC are capable of by good margin.

 

[Asdfgh]

7 hours ago, OE333 said:

Additional USB input for the DAC/HA200

 

The DAC/HA200 comes as standard with two USB inputs, one of which (USB-SYS) is reserved for the MP200 streamer.

 

Technically it would be possible to make the USB-SYS input available for general use if no MP200 is connected. In this case it would be possible to connect two USB source devices to the DAC/HA200.

 

I am not sure if T+A would approve such modification but if there is a broad demand for it I could discuss it at T+A and see what I can achieve...

Thank you for all the info. Being able to utilise the USB-SYS port with any computer to control the DAC 200’s analog volume control from Roon would be a fantastic feature! 

[SPAZ]

9 hours ago, OE333 said:

Additional USB input for the DAC/HA200

 

The DAC/HA200 comes as standard with two USB inputs, one of which (USB-SYS) is reserved for the MP200 streamer.

 

Technically it would be possible to make the USB-SYS input available for general use if no MP200 is connected. In this case it would be possible to connect two USB source devices to the DAC/HA200.

 

I am not sure if T+A would approve such modification but if there is a broad demand for it I could discuss it at T+A and see what I can achieve...

That would be amazing. That would suit my needs even better than the I2S.

[stereousa]

Hello,

I keep hearing how good the DAC200 is when HQPlayer sends DSD streams to its USB. What about PCM playback or DSD streams from Foobar2000? Anybody has done this here?  My setup would be a silent desktop connected to the DAC via USB. I control the Foobar2000 playback using MonkeyMote. Also use Spotify Connect and sometimes Remote into the desktop for Tidal playback, since Tidal Connect cannot control desktop.
 

Thanks!

[cpcat]

15 hours ago, barrows said:

When considering digital volume control, it is also important to own up to the fact of how much dynamic range is actually available in room.  Even in a very quiet room when playing music at very high levels (say 120 dB peaks), one is only going to have, perhaps, around 12, maybe 14 bits of real, audible dynamic range.  So considering what happens to 16, 24, or more bits is of no concern in the real world.  Properly implemented digital volume control (as it is in HQPlayer, for example), in a system with appropriate gain, has essentially no real world compromises.

Except how do you turn the volume up and down in real time? 😉

 

Oh, ok, on the web interface.  Duh, sorry.