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Topping D10 Balanced Review (USB DAC)

I don't really understand Linux, but I think you've set the resampling rate in the PulseAudio output settings and it's being output to USB at that value.
ALSA seems to have a mechanism to resample everything to the same frequency (48000 by default) when software mixing is enabled.
MOTU M4 supports 44.1 ~ 196kHz input, but since there is no display, the actual operation is unknown. Maybe it's working at a resampled frequency.

In the case of PCM / DSD playback, Windoes installs the XMOS ASIO driver provided by Topping to perform native playback. If you use only PCM, you can play it in WASAPI mode without a driver.
You don't need to rely on the DAC to tell you what is happening. You can check exactly the situation in pulseaudio using the following command in the terminal while audio is playing:
Code:
pacmd list-sink-inputs

This shows (among many other things):
  • the source sample rate and bit depth
  • whether resampling is in use
  • the module used for the resampling
Edit: the output sample rate and bit depth of a device can be seen with the following command. You may need to substitute your card's number for card0 and same for sub0:
Code:
cat /proc/asound/card0/pcm0p/sub0/hw_params

Running the commands show that the Motu M4 output sample rate is the same as the source sample rate, while the D10 output sample rate is always the pulseaudio default sample rate; the source audio is resampled to the pulseaudio default sample rate before it is sent to the D10. Both devices use the same set pulseaudio configuration.

If you review the gitlab link in my first post, the speculation is that the ALSA query used to retrieve the DAC's supported sample rates doesn't work properly in certain circumstances, leaving the default sample rate as the only option. It doesn't happen for all DACs using pulseaudio or ALSA. I can't say whether it happens for DACs other than the D10 because I haven't looked for them. It does seem to happen to others using the D10 with pulseaudio as I've found 4 different examples as you can see from my first post.
 
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@rtraideo ... great post. Do you use pCp/squeezelite also??
Any idea if there is an equivalent command as "pacmd list-sink-inputs" for pCp/Squeezelite?? Needed similar commands especially when you use a dongle dac without any kind of display capability.

Just FYI on my pCp ... I usually just use "cat /proc/asound/card0/stream0" for info. But thanks for your tips I am looking around "/proc/asound/card0/pcm0p/sub0" and interesting info there indeed.
 
@nsfgp For the above I'm just using standard Debian Bullseye on a desktop PC with a few different playback applications: mpd, Rythmbox, Firefox (for Bandcamp and Youtube listening).

In my audio system I use mpd direct to ALSA. It's running headless on an ancient Alix 2d2 single board computer. It's pretty much worked without a hiccup for 14 years.

I don't know much about pCp/Squeezelite.
 
First post. I purchased this DAC to use with my Debian Linux system. I've just set it up and have encountered an issue that others may encounter.

The issue is that the sampling rate used by the DAC refuses to change from the default sample rate configured in pulseaudio. All other sample rates are resampled. This happens even if the avoid-resampling = TRUE option is used in /etc/pulse/daemon.conf

The DAC is functional and sounds fine. It's just a mental irritation that everything is unnecessarily resampled.

It could be an issue specific to the D10 based on other reports I have found, all of which describe a similar issue when using pulseaudio with the D10s or D10b:

I'm aware there are workarounds involving not using pulseaudio. Unfortunately, I listen to a lot of stuff through various applications like Firefox, Rythmbox and MPD. Pulseaudio is what makes it possible for different applications to share the same sound devices in Linux.

I'm not going to put much effort into finding a resolution because while the D10 was on the way I decided I wanted a built in front panel volume control and have another DAC coming to replace it. I just thought I would post my findings here in case someone else encountered a similar problem. Maybe someone has had a similar problem and resolved it, but none of the above links offer a satisfactory resolution in my view.
Just an update here, I received a 3rd type of DAC and had the same issue as the D10. So this issue doesn't seem specific to the D10. I suspect it could be that a Linux kernel or pulseaudio update has broken something that used to work.

I can no longer edit my original post to include this information or I would do so.

I'm investigating a resolution. If I find one I will post it here.
 
Just an update here, I received a 3rd type of DAC and had the same issue as the D10. So this issue doesn't seem specific to the D10. I suspect it could be that a Linux kernel or pulseaudio update has broken something that used to work.

I can no longer edit my original post to include this information or I would do so.

I'm investigating a resolution. If I find one I will post it here.
What was the third DAC, if you don't mind saying?
 
What was the third DAC, if you don't mind saying?
It's the RME ADI-2-DAC FS, latest revision.

I'm going to see if going back to an earlier Linux kernel resolves the issue. I am using a later kernel from Debian Bulleye backports that I installed in an attempt to resolve some unrelated glitches (lock-ups, crashes) with the Motu M4 that I ended up returning.
 
@rtraideo File some bugs, etc. first while you are tinkering with your system ... does not make much sense without the developer acknowledging/involving with any fixes required.
Was reported over a year ago already. The fact that the report involved the D10 is part of why I originally thought it was specific to the D10. I included this link in my first post.

I rebooted and selected the standard Bullseye kernel version at start up. It now works correctly! When resampling occurs, "resample method" shows something other than copy. So the below indicates resampling is not occurring. Also the DAC display is showing the native sample rate of the source track.

44.1khz:
Code:
$ pacmd list-sink-inputs
1 sink input(s) available.
    index: 2
    driver: <protocol-native.c>
    flags:
    state: RUNNING
    sink: 2 <alsa_output.usb-RME_ADI-2_DAC__51100960__7B56222641ACDC8-00.analog-stereo>
    volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
            balance 0.00
    muted: no
    current latency: 68.21 ms
    requested latency: 210.00 ms
    sample spec: s16le 2ch 44100Hz
    channel map: front-left,front-right
                 Stereo
    resample method: copy

192khz:
Code:
$ pacmd list-sink-inputs
1 sink input(s) available.
    index: 3
    driver: <protocol-native.c>
    flags:
    state: RUNNING
    sink: 2 <alsa_output.usb-RME_ADI-2_DAC__51100960__7B56222641ACDC8-00.analog-stereo>
    volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
            balance 0.00
    muted: no
    current latency: 249.56 ms
    requested latency: 210.00 ms
    sample spec: s24-32le 2ch 192000Hz
    channel map: front-left,front-right
                 Stereo
    resample method: copy

It still doesn't occur with all pulseaudio. My internal sound always works without resampling regardless of the kernel version. I think the Motu M4 worked correctly too, but I can't be 100% certain of that now. It may be that I didn't notice the regression when I began using the new kernel version. I've returned the M4 so I can't go back and confirm.

Anyway, the issue has arisen in some Linux kernel beyond the 5.10 LTS version that is used by Debian Bullseye. Could be useful information for someone wanting to resolve this issue with pulseaudio.
 
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Cool @rtraideo good job. Now you should report all your findings/etc. back to the developer and make sure they can port whatever permanent changes into new builds going forward. And they should grant you a VIP perpetual license for helping them :)

My non-windows stuff is mainly in the pi/Squeezebox/LMS/pCp/Material skin/Squeezelite-x eco system. Those folks are very responsive and works closely with hardware vendors like Hifiberry, etc. to adapt to new hw/drivers when needed. Just saying.
 
Cool @rtraideo good job. Now you should report all your findings/etc. back to the developer and make sure they can port whatever permanent changes into new builds going forward. And they should grant you a VIP perpetual license for helping them :)

My non-windows stuff is mainly in the pi/Squeezebox/LMS/pCp/Material skin/Squeezelite-x eco system. Those folks are very responsive and works closely with hardware vendors like Hifiberry, etc. to adapt to new hw/drivers when needed. Just saying.
I have a perpetual license because it's all free!

If the ecosystem you use has cracked the problem of enabling a bunch of applications (including Firefox) to use the same sound card simultaneously, I'm listening. I believe that's being left to pulseaudio because that's what it is designed to do.

In my dedicated music listening systems, mpd direct to ALSA device is trouble free and bit perfect. Has been since 2008.
 
Actually i use the Cambridge Dacmagic 100 and the Smsl Sh-9. I think it makes sense to upgrade to a balanced dac (4 instead of 2 V Rms). My question is: What would you do? A Topping DX7Pro? A good whole in one Solution but a bit weak unbalanced.

Then: Schiit Modius or D10 Balanced? The Topping has got the newer chip and is cheaper. But there is no power button. Which one has got the better sound quality?

Is it a problem that the D10 is always on if the TV is on? For instance: If i don‘t use the headphone for hours and the DAC is on: Does it waste a lot of energy?
 
Actually i use the Cambridge Dacmagic 100 and the Smsl Sh-9. I think it makes sense to upgrade to a balanced dac (4 instead of 2 V Rms). My question is: What would you do? A Topping DX7Pro? A good whole in one Solution but a bit weak unbalanced.

Then: Schiit Modius or D10 Balanced? The Topping has got the newer chip and is cheaper. But there is no power button. Which one has got the better sound quality?

Is it a problem that the D10 is always on if the TV is on? For instance: If i don‘t use the headphone for hours and the DAC is on: Does it waste a lot of energy?
Also consider the SMSL D-6. I doubt there will be audible differences between any of them. Buy the one whose features or country of origin best fit your preferences.
 
I think I will buy this one. Some questions: Is it ok to connect it with the TV? Is 0,5A enough or is it better to put it in a 0,9A Usb-Port? Does the E50 also power on automatically like the D10B or is it always necessary to push the remote or standby button? So of you only need a dac the D10 is enough and the E50 doesn‘t make sense? If you say you shouldn‘t connect it with the TV I will buy another one. But if it is fine the D10B is my first choice.
 
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I think I will buy this one. Some questions: Is it ok to connect it with the TV? Is 0,5A enough or is it better to put it in a 0,9A Usb-Port? Does the E50 also power on automatically like the D10B or is it always necessary to push the remote or standby button? So of you only need a dac the D10 is enough and the E50 doesn‘t make sense? If you say you shouldn‘t connect it with the TV I will buy another one. But if it is fine the D10B is my first choice.
The specs state 5V @ 0.5A
1659482476187.png
 
I have a question, just out of curiosity, regarding the D10s (RCA) or this D10b.

If somebody uses the KEF LS50 Wireless II and doesn't want to use wireless but Optical In&Out and additionaly wants to use an active subwoofer, let's say an ADAM Sub 7 (with RCA and balanced inputs).

Would the following setup work?
Windows PC, USB-> Topping D10b, Optical Out -> KEF LS50 WII, Optical In
and at the same time:
Topping D10b Balanced Out -> ADAM Sub 7 Balanced In (or the same with the D10s and RCAs)

Would this be a possible setup or is there an error? Can a DAC put out digital and analog at the same time with the exact same timings?

PS: I know you could just use the Sub Outs at the KEFs but that's not part of the question.
 
Should work just fine. I haven't tried it, but I have a D10 balanced, and I've fed the digital to a recording device with optical in and it looked to be the same signal.
 
I have a question, just out of curiosity, regarding the D10s (RCA) or this D10b.

If somebody uses the KEF LS50 Wireless II and doesn't want to use wireless but Optical In&Out and additionaly wants to use an active subwoofer, let's say an ADAM Sub 7 (with RCA and balanced inputs).

Would the following setup work?
Windows PC, USB-> Topping D10b, Optical Out -> KEF LS50 WII, Optical In
and at the same time:
Topping D10b Balanced Out -> ADAM Sub 7 Balanced In (or the same with the D10s and RCAs)

Would this be a possible setup or is there an error? Can a DAC put out digital and analog at the same time with the exact same timings?

PS: I know you could just use the Sub Outs at the KEFs but that's not part of the question.
The D10s/D10B can be used in two connection patterns of your choice.
Strictly speaking, there is a delay between OPT/COAX and analog, but I don't think you can tell by listening.

Note that the D10s/D10B don't have a volume control. You need to adjust the volume on the speaker side or the PC side.
 
Should work just fine. I haven't tried it, but I have a D10 balanced, and I've fed the digital to a recording device with optical in and it looked to be the same signal.
The D10s/D10B can be used in two connection patterns of your choice.
Strictly speaking, there is a delay between OPT/COAX and analog, but I don't think you can tell by listening.

Note that the D10s/D10B don't have a volume control. You need to adjust the volume on the speaker side or the PC side.
Thanks for the enlightenment guys! Not that i would own any of the 3 products but it was just a random thought when I read about the KEFs and I had to get an answer. :)
 
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