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Topping E50 Review (Balanced DAC)

Blew

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I found curious that people who deny to perform software upsampling are fans of MQA down and upsampling DSP.
I don't claim to be any type of expert here, but as I understand it, upsampling does not correct errors in the time domain like MQA does. Upsampling can have advantages for the filtering process but is not a panacea for reconstruction of 44.1KHz recorded digital audio. My subjective testing in listening to MQA encoded 44.1KHz audio vs straight PCM is MQA definitely sounds better. You could argue that a better DAC would rectify this for the PCM but the point of MQA is to enable this without requiring an expensive DAC with more sophisticated reconstruction filters.
2) Ultrasonic noise added by MQA filters - see Archimago's measurements comparing Topping D90SE built-in filter and MQA filters:
"As you can see, I've created a 96kHz file with content starting at 17.5kHz rolling-off by 45kHz. Obviously music should not have high levels out to 40+kHz, but you never know and I'm trying to be illustrative here. The shape is intentional so we can identify the pattern across the frequencies."
Archimago is known, and freely admits, to being biased against MQA due to the lack of transparency and general business practices. The measurements shown are of audio "content" from 17.5kHz to 45kHz. MQA is not designed to encode this, as it's designed to encode music within the audible spectrum. So this test appears designed to make MQA look bad due to his personal bias against it. I wouldn't place too much importance on it.
It is easier to trust propaganda words than to try to understand things.
There appears to be propaganda on both sides of the argument, as demonstrated above.
 

Jobblin

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I don't claim to be any type of expert here, but as I understand it, upsampling does not correct errors in the time domain like MQA does. Upsampling can have advantages for the filtering process but is not a panacea for reconstruction of 44.1KHz recorded digital audio. My subjective testing in listening to MQA encoded 44.1KHz audio vs straight PCM is MQA definitely sounds better. You could argue that a better DAC would rectify this for the PCM but the point of MQA is to enable this without requiring an expensive DAC with more sophisticated reconstruction filters.

Archimago is known, and freely admits, to being biased against MQA due to the lack of transparency and general business practices. The measurements shown are of audio "content" from 17.5kHz to 45kHz. MQA is not designed to encode this, as it's designed to encode music within the audible spectrum. So this test appears designed to make MQA look bad due to his personal bias against it. I wouldn't place too much importance on it.

There appears to be propaganda on both sides of the argument, as demonstrated above.
So, MQA is "correcting errors in the time domain", deciding what part of an audio signal constitutes "music" and somehow subjectively sounds "better" (definitely)?
I thought this was Audio Science Review and not Snake Oil Corner?
 

audiofun

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Thank you sir.
But I can't find it on Google.
that's what chinese ICs are. poor/no documentation. you can't really find useful info on web.
You can contact the company and order a few samples.
 
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audiofun

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@Hee here's some information I got from google:


matches exactly what it is.

you may need to use google translate to read it.
 

Hee

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@Hee here's some information I got from google:


matches exactly what it is.

you may need to use google translate to read it.
Bravo! Thank you.
 

Blew

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So, MQA is "correcting errors in the time domain", deciding what part of an audio signal constitutes "music" and somehow subjectively sounds "better" (definitely)?
I thought this was Audio Science Review and not Snake Oil Corner?
This may help explain it:

Let's not derail this thread into yet another anti-MQA rant. If you don't like MQA then don't listen to it.
 

Jobblin

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This may help explain it:

Let's not derail this thread into yet another anti-MQA rant. If you don't like MQA then don't listen to it.
Certainly don't want to derail, but feel it important that subjective fallacies are shot down at ASR.
Hugh Robjohns, the author of that piece and whom I greatly respect, rowed back on this article later on...
 

bogi

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but as I understand it, upsampling does not correct errors in the time domain like MQA does.
Apodizing filters can. They don't preserve original samples but reconstruct the whole waveform. HQPlayer is capable of that.
Please note "don't preserve" is meant as principle. Of course, if the same value as in original content is computed then such sample seems like preserved. But the principle is that every original sample is re-computed in order to restore the original waveform. And of course in the case of upsampling (interpolation) new samples are added between the original ones.
The measurements shown are of audio "content" from 17.5kHz to 45kHz. MQA is not designed to encode this, as it's designed to encode music within the audible spectrum.
MQA advertise to compress high frequency band of original recording and to store it in lower frequency band on encoding side and then on decoding (playback) side to recover that information. Archimagos measurements show how that original high frequency content is really 'restored'. Instead of restored original high frequency content what you get are images of original band below fs/2.
 
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bogi

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So as I understand it now:

"(DSD Over PCM) DoP is a method for transporting DSD audio over USB ports that do not have a DSD driver. Each consecutive set of 16 DSD bits is stored as PCM bits in the lower 16 bits of a 176.4/24 sampling rate. An 8-bit DoP header is added to each sample. In order to reassemble the PCM bits into a continuous DSD stream, the DAC at the playback end must be DoP compliant." (PCmag)

USB_DSDviaPCM_1v0.jpg


To summarize DSD-guide.com:

The normal DSD64 sample rate of 1-bit at 2.8224MHz has a data rate of 2.8224Mbits/sec. This is equivalent to 16-bit PCM at 176.4kHz but 8-bits are used to mark the PCM package as DSD so 24-bit 176.4kHz. The DAC requires 2x 16-bit streams, to be able to form a stereo stream and as such some compare it to a 16-bit 88.2kHz PCM file, though the differences in technology mean there are some heated opinions on which one is better and if it matters. There are some advantages to a continuous stream over blocks that need to be put together.
There is some built in safety there that if the package gets partially dropped or accidentally interpreted as PCM it will produce a low-volume, high-frequency 88kHz tone that is inaudible and won't harm most hardware. Though most will filter this out.

  • Packages the RAW DSD stream in a (24-bit 176.4 kHz) PCM package that gets unpacked (but not converted!) back into DSD in the DAC.
JaccoW, I know what DoP is. It is a very ineffective way how to pack DSD bits into PCM envelope. The content you posted: "Each consecutive set of 16 DSD bits is stored as PCM bits in the lower 16 bits". Notice "as PCM bits". They are not real PCM bits, they are only stored in the same way as PCM bits would be stored. DSD content stored in PCM envelope is not PCM content. As I wrote yesterday, you need to understand difference between binary coded PCM sample and unary coded DSD bitstream.

Packages the RAW DSD stream in a (24-bit 176.4 kHz) PCM package that gets unpacked (but not converted!) back into DSD in the DAC.
Please note "unpacked (but not converted) back into DSD in the DAC". Read it this way: Original 16 bits chunk of DSD data was packed (not converted to PCM) into 24bit 176.4kHz PCM package on player side and then on DAC side it is unpacked (not converted back to DSD, because it never was PCM).

I hope you start to understand at least. :) DoP is yet more ineffective way of packing/unpacking DSD bits on 32 bit interfaces like XMOS. Only 16 bits of DSD data are transferred in one 32bit PCM quasi 'sample'. Therefore 176.4/192k capable interface can transfer only DSD64 in DoP mode but DSD128 in the so called native DSD mode, where that packing/unpacking is not needed.
 

Grooved

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This may help explain it:

Let's not derail this thread into yet another anti-MQA rant. If you don't like MQA then don't listen to it.
Don't want to derail the thread (too much things to say), so I will try to only keep the main point in one answer.

I tested it a lot, and yes they some it's interesting, other not, and I say interesting because on these tracks, it's different, not better, it brings a positive effect on some points, and a negative on other.
But more important, it looks like you didn't realize that they are using it like in the patents, and that most of the tracks are done by auto-conversion, from a master already done, and that if it was used correctly, it would be during the creation of the first master, listening through the MQA encoder, and by doing that, the artist/engineer would potentially change something in the settings too not make it sound like what you are listening.
So the biggest problem is simply that it can't be "what the artist wanted you to listen" as used right now, and some artists/engineers have already confirmed that. And even MQA says, to you as a customer, it's what they wanted, it's funny that at the same time, they're saying because they are saying at the same time to the label "don't offer your best masters to the customers" ;)

I can take any master, add a bit of distorsion, or change a bit the EQ, change a bit transients,... let you hear both and you may say that the new one is sounding better, but it was not what was wanted first. There are enough people on this earth that like to hear with distorsion.
When you compare both, you are comparing one audio file, and the same one+processing, like with a FLAC and transparent DAC, but then a processing to change the sound before going into your amp.

They only one that mastered through the MQA encoder are 2L label, but they work with them and got the encoder, so they are more biased than someone not being linked with them, even if at least they used it more like described in the patents.
If it was possible to all record/mix/mastered with MQA for any studio and it would bring a better sound, I guarantee you that all artists/engineers would push for it (except it's almost impossible for a cost reason). But MQA prefers described it like it's used like that, while it's not, so their marketing claims are false.

It's not being biased to know and tell that, it's the truth.
 

Blew

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But more important, it looks like you didn't realize that they are using it like in the patents, and that most of the tracks are done by auto-conversion, from a master already done, and that if it was used correctly, it would be during the creation of the first master, listening through the MQA encoder, and by doing that, the artist/engineer would potentially change something in the settings too not make it sound like what you are listening.
Yes this is probably the case in the majority of the Tidal MQA library, but doesn't appear to be so for new releases and recent remasters. These are the versions that I personally appreciate the most and provide a good enough reason for me to want MQA support in the DAC. Regardless, as I mentioned before, the MQA versions of 44.1KHz CD albums sound subjectively better than the original CD. So it's a feature that I'd rather have than not have.

I don't understand the objection to including it in a DAC. If you don't like MQA then you don't have to listen to it. You can still play lossless PCM as well as you could without MQA support in the DAC.
So the biggest problem is simply that it can't be "what the artist wanted you to listen" as used right now, and some artists/engineers have already confirmed that. And even MQA says, to you as a customer, it's what they wanted, it's funny that at the same time, they're saying because they are saying at the same time to the label "don't offer your best masters to the customers" ;)
I don't really care what the original artist intended. I care what I prefer the sound of. The vast majority of music produced since the mid 90's has been engineered with terrible dynamic range compression anyway and the original artists don't seem to have cared about record companies doing that to their recordings. Music is engineered that way for commercial reasons, not because it sounds better, so why should we suddenly care what the original artist wants us to hear now? Record labels have been pulling the original high dynamic range recordings of albums off the shelves and replacing with terrible remasters for years, so not offering their best masters to customers is nothing new for them!
I can take any master, add a bit of distorsion, or change a bit the EQ, change a bit transients,... let you hear both and you may say that the new one is sounding better, but it was not what was wanted first. There are enough people on this earth that like to hear with distorsion.
When you compare both, you are comparing one audio file, and the same one+processing, like with a FLAC and transparent DAC, but then a processing to change the sound before going into your amp.
If it sounds better then I'm all for it.
If it was possible to all record/mix/mastered with MQA for any studio and it would bring a better sound, I guarantee you that all artists/engineers would push for it (except it's almost impossible for a cost reason). But MQA prefers described it like it's used like that, while it's not, so their marketing claims are false.
I very much doubt "all artists/engineers" would push for something that sounds better over something that sells better! As mentioned above, they've been pushing for the latter for many years now, and we as customers who appreciate good quality audio have suffered because of it.
 

Blew

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Certainly don't want to derail, but feel it important that subjective fallacies are shot down at ASR.
Hugh Robjohns, the author of that piece and whom I greatly respect, rowed back on this article later on...
Yes he makes good points, but he doesn't refute the time domain benefits. He's only expressing frustration with the lack of transparency on how it works, which is understandable. The rest of what he says is already known. MQA is not truly lossless, is unable to be tested and validated, the file size compression isn't as important nowadays, the provenance/authentication isn't robust, and there is noise in the ultrasonic spectrum (which we can't hear anyway). Most of that is secondary to the main reason for wanting MQA support in a DAC - because it sounds better

For me MQA is just a cost-effective way of reconstructing digital audio in a DAC, so I'd rather the DAC have it than not have it.
 

yanm

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This may help explain it:

Let's not derail this thread into yet another anti-MQA rant. If you don't like MQA then don't listen to it.
Thanks, very interesting article. I agree with you that MQA support in DAC is a welcome addition - especially if you're on Tidal where there is no pure non-MQA (AFAIK). However, MQA has, in my opinion, more or less serious shortcomings:
  • MQA seems a purposely opaque technology => no possibility to test it in an objective way => against the raison d'être of ASR,
  • MQA on 44.1/24 streams is likely transparent but MQA on 44.1/16 ones is unlikely to be fully transparent though*, and
  • no pre-processing of the stream possible before feeding the DAC.
I am personally using the MQA capability of my E50 and I am happy to have the possibility (I am a Tidal customer after all)... And, as someone having worked in the watermarking research field in a past life, I can appreciate the concept :)

( * For 24-bit audio, one remains with 120dBFS dynamic range if the 4 least significant bits are sacrificed for MQA folding and authentication. In contrast, for 16-bit audio, the same sacrifice reduces the dynamic range down to 72dBFS! As far as I know, MQA has an algorithm to automatically detect the noise floor and what can be transparently sacrificed... and while dithering can help some, I am not convinced at all about using MQA for 44.1/16 audio. This is actually my main concern with Tidal: they have virtually replaced their lossless CD-quality tier with 44.1/16 MQA! )
 
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yanm

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Back the topic... What is the track record of Topping in supporting and enhancing released products?

I am asking because they are releasing tons of new products and seems to have their energy focused on that. On the other hand, support and maintenance of 'legacy' products is often much more effort than one may think...

In particular for the E50, I would love that sampling rate, bit depth, and volume are shown in a cyclical manner (as long as there is no changes in volume). In addition, the auto standby feature is not really working (not sure it is the E50 or the Node fault though).
 
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Is there anyway to display bitrate rather than volume on the E50.

I know this can done by switching to DAC mode but would prefer to keep it as Preamp.
 

Aldoszx

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Back the topic... What is the track record of Topping in supporting and enhancing released products?

I am asking because they are releasing tons of new products and seems to have their energy focused on that. On the other hand, support and maintenance of 'legacy' products is often much more effort than one may think...

In particular for the E50, I would love that sampling rate, bit depth, and volume are shown in a cyclical manner (as long as there is no changes in volume). In addition, the auto standby feature is not really working (not sure it is the E50 or the Node fault though).
Indeed, the auto stand by is not working as it should do.
 

visekop

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Hello everybody!
E50 works very poorly via optical input. Constant noise (clicks and crackles). Replacing the toslink cable does not help. Switching operating modes (PRE or DAC) does not solve the problem. Resetting to factory settings does not help. As I see from the reviews on the Internet, this problem with optics is very "popular" with Topping. Very bad. It is not possible to use the DAC via the optical input. Has anyone been able to solve such a problem? Maybe I'm doing something wrong?
 

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Toku

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Hello everybody!
E50 works very poorly via optical input. Constant noise (clicks and crackles). Replacing the toslink cable does not help. Switching operating modes (PRE or DAC) does not solve the problem. Resetting to factory settings does not help. As I see from the reviews on the Internet, this problem with optics is very "popular" with Topping. Very bad. It is not possible to use the DAC via the optical input. Has anyone been able to solve such a problem? Maybe I'm doing something wrong?
What is the device to connect with OPT?
Only pure 2ch stereo signals can be handled by DACs such as the E50.
Such a symptom occurs when Dolby Surround etc. are enabled.
 
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