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Topping DX1 II DAC & HP Amp Review (NTTY)

If I have a playlist with tracks having different sample rates and bit depths, should I expect pops and clicks? I would be using USB, Thanks.
If you play each track at its native sampling rate then yes. However, in Windows (and at least in Linux) by default, you have whats called a desktop or graph sampling rate, and this is fixed. So unless your music playback is using a dedicated audio device (for example foobar2000) or on Linux, you have your pipewire/pulseaudio settings configured to switch to the native sampling rate, then its all going to be resampled to 48khz (or whatever you've set it to), and in that case no.

If it bothers you, you could set your default sampling rate to whatever the highest sampling rate is of any of your music, and then the lower stuff will be upsampled to that by the operating system. Not ideal from a purist perspective. For most people using default settings, this is not going to be an issue.

The vast majority of my music is 44.1khz, I have some stuff which is 48khz, a small amount of 96khz stuff, and some DSD64 and DSD256 stuff just for curiosities sake really. I play back at the native sampling rate and just live with it. Its a nice little reminder that the sampling rate has changed.
 
If it bothers you, you could set your default sampling rate to whatever the highest sampling rate is of any of your music, and then the lower stuff will be upsampled to that by the operating system. Not ideal from a purist perspective. For most people using default settings, this is not going to be an issue.
It's also worth noting that if you don't use the DAC volume control, but adjust the volume after the DAC in the amplifier, the problem is practically nonexistent. Since the volume is rarely turned up to maximum, any sounds that do occur are muffled and don't annoying.
 
It's also worth noting that if you don't use the DAC volume control, but adjust the volume after the DAC in the amplifier, the problem is practically nonexistent. Since the volume is rarely turned up to maximum, any sounds that do occur are muffled and don't annoying.
I am curious if a proper preamp might buffer the noises?
 
A proper pre-amp buffers all sounds.
The volume control on it will lower all incoming sounds including the small ticks when switching sample rates.

The ticks (what I get from the posts) are not 'DAC volume control' dependent so when using the DAC also as pre-amp (no external volume controls) you might be bothered by this especially when the volume is lowered considerable.

When using it as a DAC followed by a (pre)amp with volume control it is not an issue in practice.
 
We've already discovered that using the DAC in preamp mode at about -2.5dB is better than using it in DAC mode. For my use cases, DAC mode is basically useless to me. It's too loud for some things. I like to have my operating system at 100%, then use the DAC for gain staging so I don't have to adjust the volume on my speakers. Besides they're old speakers (Edirol MA-10A) and seem to have a sweet spot in the middle, so I go anywhere from -24dB up to -2.5dB or so on the DAC depending on what I'm listening to. The click appears to be audible no matter what you do, and my previous interface (an Edirol UA-25) did not have this issue. My headphones, a pair of Sennheiser HD6 Mix sound incredible, and the headphone section has an independent gain setting when you switch between line out and headphone, which is pretty cool. You can also run both at the same time. This is a great device. The pops and clicks range anywhere from mildly annoying (for some people) to barely noticeable, it just depends on your individual use case and how sensitive you are to such things. With pipewire on linux, you can set the "session.suspend-timeout-seconds = 0" so that it keeps the device open, and you can force the sample rate, so that it never changes sampling rate, and the problem technically goes away. Windows users, I don't know, I would have to do more testing to get a better understanding. Having said that, people have reported the issue with optical input and output, and i've never tested those so I can't comment. Someone has mentioned that turning an optical device off, causes a loud pop when its playing back the input. These are minor issues in my opinion, but they're also probably not fixable. I have asked Topping if they can add a mute function on playback initialise/sampling rate changes, or if they've tried this already and it doesn't work, or cuts the beginning of the sound off too much.
 
I have asked Topping if they can add a mute function on playback initialise/sampling rate changes, or if they've tried this already and it doesn't work, or cuts the beginning of the sound off too much.
This really isn't the best solution, because the DAC then change the music and loses the initial part. And not everyone likes that. I had DACs that did this. I replaced them, in part, for this reason. Better to be click.

We've already discovered that using the DAC in preamp mode at about -2.5dB is better than using it in DAC mode.
Can you show me where we figured this out? Just because headphone outputs have a higher output signal and dynamic range doesn't mean they have a lower noise floor. The first post in this thread lists the noise floor N on the balanced outputs as -124.6 for the line output and -124.1 for the headphone output. The headphone output is excellent. But it's not quieter than the line output.
 
Hi everyone, I wanted to share a specific issue I'm experiencing with my Topping DX1 II and see if anyone has insights or recommendations.

My current setup is: Xbox Series X -> HDMI Audio Extractor (Toslink) -> Topping DX1 II. I having to types of behavior with the 2 options in "Optical Compatibility" with this device.

1) Mode 1:
When I turn on the Xbox, I get constant audio dropouts. The audio constantly stutters and cuts out for a few seconds at a time.This only lasts for the first 1 or 2 minutes. After those initial 2 minutes of dropouts, the audio is flawless for hours, with absolutely zero interruptions.

2) Mode 2:
If I switch the DAC to Mode 2, the audio works perfectly from the very first second. There are no start dropouts at all. But after every 1 to 2 hours of continuous playback, there is a very brief micro-stutter/dropout (about 1/4 of a second), which I assume is the DAC doing a quick re-sync or buffer flush.

Is there any way to fix this kind of error? Has anyone else experienced this specific behavior with Topping DACs? Thanks.
 
Little side by side with he Topping DX3 Pro+
 

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Firmware Update V15.86.22.86​

  • Added: Option to keep either the volume level or sample rate displayed on the screen.
  • Added: Left/right channel balance control.
  • Added: PEQ preview, supporting cloud PEQ presets and AutoEQ profiles.
  • Optimized: Smoother audio transitions when adjusting PEQ parameters, enabling or disabling PEQ, or switching between PEQ profiles.
  • Optimized: When “OPT Output Volume Follow” is enabled, muting the device also mutes the optical output.
  • Fixed: Audio was not automatically muted when entering the settings menu.
  • Fixed: PEQ could briefly become inactive when switching between USB and OPT inputs.
  • Fixed: Some known issues.
Update to Latest Firmware
 
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Wow. They've gained the speed and determination to perfect this little beast :) .
Sort of.
"Added: Option to keep either the volume level or sample rate displayed on the screen." - Only changes the online browser interface, not on the device itself!
 
Sort of.
"Added: Option to keep either the volume level or sample rate displayed on the screen." - Only changes the online browser interface, not on the device itself!
The main thing is to get started. Another 4-5 firmware updates and it'll work :) .
 
Sort of.
"Added: Option to keep either the volume level or sample rate displayed on the screen." - Only changes the online browser interface, not on the device itself!
There’s a setting in the menu of the device where you can choose between d-U and d-(3 horisontal bars). This works on mine.
 
There’s a setting in the menu of the device where you can choose between d-U and d-(3 horisontal bars). This works on mine.
After introducing this parameter, the DAC mode became practically meaningless. I used it only to display the bitrate. Now, to test the clicks when changing the bitrate, I switched the DAC to preamplifier mode and displayed the bitrate. After experimenting, I set the volume to 0 dB. This means I can reduce the volume on the DAC if I suddenly need to. And I display the bitrate. And I don't use the volume on the DAC unless I need to.
 
After experimenting, I set the volume to 0 dB. This means I can reduce the volume on the DAC if I suddenly need to. And I display the bitrate. And I don't use the volume on the DAC unless I need to.
As I explained in the review, you should reduce the volume by 2.5dB to get full resistance to intersample overs. No need to take the risk of distortion when the solution is that simple.
 
As I explained in the review, you should reduce the volume by 2.5dB to get full resistance to intersample overs. No need to take the risk of distortion when the solution is that simple.
These are unrelated things. My volume is set to 0 dB. And the attenuation in the PEQ is -4.8 dB. I do attenuate, but not there.
 
These are unrelated things. My volume is set to 0 dB. And the attenuation in the PEQ is -4.8 dB. I do attenuate, but not there.
Ok, I missed that info in your previous replies.
Is that -4.8dB attenuation relative to 0dBFS at the peak level of the curve you defined?
 
Is that -4.8dB attenuation relative to 0dBFS at the peak level of the curve you defined?
If I understand the question correctly, my curve only attenuates. It doesn't boost or have filters with high peaks. Therefore, -4.9 dB is simply attenuation for the same purposes you described above. This attenuation isn't for filters, but to eliminate TruePeak overloads.
 
If I understand the question correctly, my curve only attenuates. It doesn't boost or have filters with high peaks. Therefore, -4.9 dB is simply attenuation for the same purposes you described above. This attenuation isn't for filters, but to eliminate TruePeak overloads.
Ok, understood. It would be interesting to test if the results are the same (I guess yes, as I’d assume it’s done the same way, but I would run a quick check).
 
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