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Topping L30 Headphone Amplifier Review

Veri

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Upsampling to max always sound smoother. And if you do it to DSD that even much smoother.
Funny how things only go one way, "smoother". While on a NOS DAC (a holo spring for example), setting it to oversampling or DSD mode of course makes it.. brighter. As opposed to the very smooth NOS mode! Almost like your very own bias is causing you to hear one way or another ...?
 
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Oh dear. Please stop. At least pretend to science it up a bit. This is how nonsense propagates and becomes received wisdom.
Sorry but then you are speaking about science but don't understand how delta sigma DAC works and what does it mean direct DSD. There are technical reasons why software upsampling and software delta sigma modulation can easily beat in quality the hardware oversampling and delta sigma modulation in DAC chip. Generally, with delta sigma DAC chips, for PCM source content it is possible to bypass first stages of hardware oversampling and for DSD source content it is possible to bypass complete oversampling and delta sigma modulation (direct DSD mode). For delta sigma DACs the PDM (Pulse Density Modulation) signal (not PCM signal) is their native type of signal which is finally converted into analog by low pass filter. DSD signal is one bit two level PDM signal so it is native type of signal for Delta sigma DAC, which (in contradiction with PCM signal) does not need to be complicatedly processed before it enters the D/A conversion stage itself (no oversampling, no modulation).

You and your colleagues are repeatedly attacking people who carry about sound quality. You have to educate yourself before you write a nonsense and call it "scientific". I am watching this "science" forum a week and I am seeing that it is rather a "science" kinder garden.
 

JohnYang1997

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Sorry but then you are speaking about science but don't understand how delta sigma DAC works and what does it mean direct DSD. There are technical reasons why software upsampling and software delta sigma modulation can easily beat in quality the hardware oversampling and delta sigma modulation in DAC chip. Generally, with delta sigma DAC chips, for PCM source content it is possible to bypass first stages of hardware oversampling and for DSD source content it is possible to bypass complete oversampling and delta sigma modulation (direct DSD mode). For delta sigma DACs the PDM (Pulse Density Modulation) signal (not PCM signal) is their native type of signal which is finally converted into analog by low pass filter. DSD signal is one bit two level PDM signal so it is native type of signal for Delta sigma DAC, which (in contradiction with PCM signal) does not need to be complicatedly processed before it enters the D/A conversion stage itself (no oversampling, no modulation).

You and your colleagues are repeatedly attacking people who carry about sound quality. You have to educate yourself before you write a nonsense and call it "scientific". I am watching this "science" forum a week and I am seeing that it is rather a "science" kinder garden.
You literally have no evidence suggesting that software oversampling improves performance. In real world testing, no matter what source sampling rate is performance stays roughly the same. Or in contrast, the best performance is at 44.1k and 48khz. Science is evidence based. We need evidence, not pseudoscience.
 
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An usual personal computer has much higher computational power than resource constrained DAC chip. It can perform upsampling and delta sigma modulation calculations in considerably higher quality in 64bit floating point precision than it is done in few dollars costing consumer DAC chips. Thus feeding the delta sigma DAC with 44.1k / 48k PCM signal, when it disposes with DSD direct mode, is the worst thing you can do IF you have the possibility to perform high quality software upsampling and software delta sigma modulation. It is about the possibility to substitute the majority of DSP which is performed in typical delta sigma DAC chip with software based solution.

High quality software based upsampling, dithering and delta sigma modulation can be performed in HQPlayer Desktop or Embedded in real time during playback, or PCM to DSD conversion can be performed offline in HQPlayer Pro and other professional tools like Saracon. There are also some free or lower cost alternatives (foo_dsd_asio for Foobar2000 or for example the solution of Jriver MC), which don't reach the quality level of HQPlayer's algorithms, but still can provide good result.

So it is no nonsense to upsample or convert PCM to DSD. And there is no base for a rule that it would be best to feed a DAC with 44.1k / 48k PCM signal.
 

JohnYang1997

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An usual personal computer has much higher computational power than resource constrained DAC chip. It can perform upsampling and delta sigma modulation calculations in considerably higher quality in 64bit floating point precision than it is done in few dollars costing consumer DAC chips. Thus feeding the delta sigma DAC with 44.1k / 48k PCM signal, when it disposes with DSD direct mode, is the worst thing you can do IF you have the possibility to perform high quality software upsampling and software delta sigma modulation. It is about the possibility to substitute the majority of DSP which is performed in typical delta sigma DAC chip with software based solution.

High quality software based upsampling, dithering and delta sigma modulation can be performed in HQPlayer Desktop or Embedded in real time during playback, or PCM to DSD conversion can be performed offline in HQPlayer Pro and other professional tools like Saracon. There are also some free or lower cost alternatives (foo_dsd_asio for Foobar2000 or for example the solution of Jriver MC), which don't reach the quality level of HQPlayer's algorithms, but still can provide good result.

So it is no nonsense to upsample or convert PCM to DSD. And there is no base for a rule that it would be best to feed a DAC with 44.1k / 48k PCM signal.
Still no evidence. In measurements we've done, native DSD doesn't provide better performance. In contrast, the DSD bypass has constraints. For example, you can't have high output mode in ak4499 which potentially reduces DNR performance.
 

SJ777

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So it is no nonsense to upsample or convert PCM to DSD. And there is no base for a rule that it would be best to feed a DAC with 44.1k / 48k PCM signal.
Evidence? Measurements? Facts?

Without these it's just waffle (and not the tasty kind!).
 

MechEngVic

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Upsampling to max always sound smoother. And if you do it to DSD that even much smoother.

We have ears to evaluate. :)
For God's sake. Why would you want to do that ? :facepalm:
You take a perfectly good 44.1 signal, you alter it by needlessly resampling it to 768, just to have a nice "768 KHz" display in your E30 ? :rolleyes:
That is not "Hi-Fi", sorry, that is something else.
You don't really believe that "just because it's 768 it will sound better", do you ? o_O

If you had actual 768 KHz music I wouldn't say anything (except that your ears are limited to 20 KHz in your younger years, which is the reason why 44.1 exists BTW : 44.1 / 2 = 22.05 and 22.05 > 20). But AFAIK such music doesn't exist yet at this point. Or maybe limited so some rare files.
100% of people here and elsewhere on Earth have mostly 44.1 music. In my case I have 1 to 2% of 48/88.2/96/176.2/192 KHz music, but no more.
For 44.1 music, 44.1 is the way to go : bit-perfect output and 44.1 on your DAC. That is "High Fidelity".
Analog to digital conversions made at higher than CD quality bit rates and depths show improvement in resolution, but the improvement has not been conclusively identified by ear in any of many listening tests. Maybe if loudspeaker technology improves someday, we'll be able to hear the difference.

A digital to digital conversion to a higher bit rate and depth can only be a lateral improvement at best. Would it be like increasing your "digital headroom"? Or is it like a digital gear reduction for greater smoothness and stability of torque? If there is evidence of it making an audible improvement, I'd like to read about it.

EDIT: Another thought: Do the "motors and gears" of digital reproduction gear run smoother at higher bit rates and depths? I'm trying to layman-out the possibilities.
 
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MechEngVic

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I finally tried the L30 in preamp mode hooked up to my main system. Chromecast audio>Soncoz la-qxd1>L30>Dynaco ST-70 Series II>Klipsch KLF-10's.

It sounded just like it sounds through headphones: Flat like a ruler and dead quiet.

But it just can't compete with my Nobsound NS-01P tube preamp with NOS Mullard 12ax7's and fancy-shmancy cap upgrades. The L30 has none of the glorious harmonic distortions and sparkles that, while wildly un-transparent, attract and addict!

The L30 will remain in headphone amp duty. I hated headphones... Until I got this amp.
 

MechEngVic

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Unfortunately, whilst perhaps enjoyable for you, that will be a complete waste of time.

There are many helpful experts on this forum who could provide objective information on the subject.
Well hopefully putting my thoughts out there on this forum will illicit a helpful expert's objective information (instead of a snide comment), making it a completely stupendous use of time.
 

w1000i

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You literally have no evidence suggesting that software oversampling improves performance. In real world testing, no matter what source sampling rate is performance stays roughly the same. Or in contrast, the best performance is at 44.1k and 48khz. Science is evidence based. We need evidence, not pseudoscience.
Rob Watt explain why DSD sound smoother, I don't remember the exact explanation but DSD have issue with timing and technically it faulty- but that is why is sound smoother.
 

JohnYang1997

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Rob Watt explain why DSD sound smoother, I don't remember the exact explanation but DSD have issue with timing and technically it faulty- but that is why is sound smoother.
You are quoting Rob? I have seen that video. He's a salesman. That's all I can say.
What he was talking about was the error being accumulated before correction happening. And the transients are delayed. Where this is completely false and irrelevant.
 
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My main sources are Flac up to 384khz (most being 192 or 96) and some DSD files. I don't expect 768khz to be a lot better (or even feel a difference), but thats what Topping is selling and the drivers don't output that in my experience (tested both Asio and Wasapi).

At this level of sound sampling the sound quality issues I get concern the album itself, how it was recorded, mastered, etc. Some albums sound incredibly good, and others sound as good as an MP3 version of it (too much compression when mastered, etc).
On the sound quality overall topic, only a few of the best recordings I have sound really better (usually recent recordings of acoustic music such as classic music or jazz).

I dont understand why Jose Hidalgo got so nervous about my question. I am simply checking that the devices I have are correctly setup.
The only thing that doesn't make me happy with the Topping experience is that the L30 headphones amp gets hot like crazy and it's output is barely enough for some headphones (such as the Beyerdynamic DT 990 PRO) when listening to albums with high dynamic range and high quality (>96khz, DSD, etc), making the sound too low. I also think that it lacks a bit of bass punch (compared to the Musical Fidelity V-CAN II), but the sound is overall very natural.

Back to the DAC, as I said, I think it sounds better than the Audio Fidelity V-DAC II I had). But the difference is barely noticeable. The main reason I am changing DAC is because I am changing computer and I will no longer have a coaxial out (and old DACs are less good over USB usually)

I tried several DSD files on Foobar2000 after following the setup recommended by Topping and in my opinion the same files sound the same or even better when I read them on Aimp (outputs 384 PCM via Wasapi), because the DSD sound sounds a bit too crisp, sometimes with subtle cracks and less bass (some of these DSD files where recorded from LPs which seem to explain this). But really, I would say that I feel no difference between DSD and PCM on those tests.

I am not an audiophile fanatic. I want my music to sound good, as close as possible as being on stage with the musicians. And I want the devices I buy to work properly. That's all :)
I don't expect to increase sound quality by magic like in the Hollywood movies when they can zoom into a garbage picture and find incredible details thanks to some magic algorithm. I want realism... even when that means confirming that an album you love was really poorly recorded / mastered (both ?)

Aimp remains my main player because it is a lot easier, useful and sexier than Foobar2000. In my opinion it is just the best option (at least on PC), and I wasn't able to find any quality difference that would justify changing to F2000.
 

boXem | audio

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Sorry but then you are speaking about science but don't understand how delta sigma DAC works and what does it mean direct DSD. There are technical reasons why software upsampling and software delta sigma modulation can easily beat in quality the hardware oversampling and delta sigma modulation in DAC chip. Generally, with delta sigma DAC chips, for PCM source content it is possible to bypass first stages of hardware oversampling and for DSD source content it is possible to bypass complete oversampling and delta sigma modulation (direct DSD mode). For delta sigma DACs the PDM (Pulse Density Modulation) signal (not PCM signal) is their native type of signal which is finally converted into analog by low pass filter. DSD signal is one bit two level PDM signal so it is native type of signal for Delta sigma DAC, which (in contradiction with PCM signal) does not need to be complicatedly processed before it enters the D/A conversion stage itself (no oversampling, no modulation).

You and your colleagues are repeatedly attacking people who carry about sound quality. You have to educate yourself before you write a nonsense and call it "scientific". I am watching this "science" forum a week and I am seeing that it is rather a "science" kinder garden.
Want science? Here is science:
Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications

I spotted one minor mistake in the paper, doesn't change anything to the conclusion
TL;DNR:
- since DSD is only 1 bit, it cannot be properly dithered
- since it cannot be properly dithered, quantization errors will appear
- since these errors cannot be eliminated, the PCM to DSD conversion process can be compared to digestion
 

w1000i

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You are quoting Rob? I have seen that video. He's a salesman. That's all I can say.
What he was talking about was the error being accumulated before correction happening. And the transients are delayed. Where this is completely false and irrelevant.
I had to take his word for that, he is engineering DAC for 30 years and he do it for fun. He is the most experience DAC designer to date.
 

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