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Topping E30 DAC Review

Jimbob54

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companyja

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Not a single word about how it sounds? Measuring does not tell the whole story IMHO.
Hi mate, there are things where measurements will not tell the whole story, such as speakers and headphones. However, for DACs and AMPs, the standard suite of measurements will pretty much tell you what to expect. If you want subjective impressions, you can definitely find them at head-fi, SBAF (probably, I haven't been here forever) and other places where they discuss such things. Otherwise, from an engineering standpoint, there are only so many factors you can design for and that are known to be audible, and they all have their own thresholds of audibility. Luckily, analyzers are far more sensitive than the best human ears and the measurements you see, as long as they are competent, just mean that to the end-user, the amplifier or DAC will sound the same, or transparent.\

This is why I'm disappointed that there have been several pages here of users discussing audible differences between....AKM filters. Using descriptions of things like pre-ringing and post-ringing in a general sense and applying it to a low-pass filter is just the kind of audiophile connect-the-dots that creates confusion and is used to confirm untested subjective impressions. The ringing, lack of ringing, and otherwise, as long as we're cutting around 22-24khz, is going to concern frequencies close to 22-24khz. The frequencies in the audible range are unaffected. The pre or post ringing is not going to be audible to a human because the frequencies that will ring are near the nyquist limit for the sampled signal. Unless you're using a 22.05khz sample rate, the shape of the filter will be irrelevant in terms of pre or post-ringing. So a short delay cannot create "more poweful notes" or whatever - the "notes" themselves are unchanged, exactly the same. The AUDIBLE difference comes from the speed of the filter. You can easily test this yourself - use this simple tone generator, set the tone to 16000hz or higher depending on how good your hearing is atm, and play it fairly loudly so that you can clearly hear it. Then switch the filters. The fastest filters 1 and 3 will be the loudest (although the difference is incredibly slight), the slower filters will be quieter, and the super slow filter number 5 will sound like multiple tones, not a pure sine wave - this is presumably because it aliases frequencies above nyquist back into the audible range, thus producing audible information that was not in the original signal - needless to say, this is typically unwanted if we're talking about high fidelity reproduction of audio.
 
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When I listen to hifi, I always fall for equipment with a tight bass, warm midband and airy treble (some prefer to call it sweet), but how do you measure that? Isn’t this exactly where measuring equipment falls short? The small nuances, which makes hifi a fun hobby.
 

companyja

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When I listen to hifi, I always fall for equipment with a tight bass, warm midband and airy treble (some prefer to call it sweet), but how do you measure that? Isn’t this exactly where measuring equipment falls short? The small nuances, which makes hifi a fun hobby.
You can definitely talk about these differences in speakers and headphones. We are especially not that great at measuring headphone frequency responses reliably - and if you look at a headphone mesuarement, there are all these peaks and valleys and shapes on the frequency response - that determines how dark or bright the headphone is, and also elusive things like soundstage are correlated to the frequency response but can't be 100% determined just from looking at the graph. This is where the fun in the hobby lies. For equipment such as DACs or AMPs, you can have extremely reproducible measurements because you're not dealing with microphones and fittings and acoustic properties of the room you're measuring in and larger unit-to-unit variances - if you look at frequency response measurements of an amplifier or a DAC, you're 99% of the time going to see a flat line going from 20hz-20khz - if this is flat, then the DAC or amp is not coloring the sound, so it can't be darker, warmer, less or more bass or airier treble, it's simply rendering the file as-is. Additionally, if the linearity and distortion figures are low enough, there won't be additional garbage information coming to you as a bi-product of digital-to-analog conversion, meaning you're only going to hear what's in the original recording. As long as these parameters align, there's nothing to suggest that the piece of gear can sound any different to another piece of gear that measures similarly.

This is at least my understanding of it, and I understand that this takes away some of the "magic" out of the hobby, it's not like going into a hi-fi store in the 70s and seeing 5 different amplifiers and recievers from Japan and being lucky to have any basic measurements from the COMPANY themselves, not an indepentent evaluator, and just sitting down and listening to hear if you like the sound or not. Thanks to analyzers, we've been able to pinpoint the parameters that influence the fidelity of the sound and measure them more precisely than the human ear. If there are parameters that we are not measuring that influence the sound, firstly someone would have to prove the difference exists with proper ABX testing, and then spend a long while isolating what exactly is making the difference. The reason I personally am confident in the testing methodology, and people here generally are, is that I haven't seen research that confirms that measurably transparent gear can be discerned by a listener in an experimental ABX test. Remember, people will hear differences in cables, different surfaces they put the amplifier on, 44.1khz and 192khz files, etc - we are very easily fooled with biases and our audio memory is way worse than we'd like to admit. In an incomplete testing, we can hear all kinds of things we want to hear unless we take a proper listening test.

I'm still very much excited about headphones and speakers, but as long as I have enough power and a transparent enough DAC, I am absolutely confident that I don't need to touch anything with the source chain.
 

pieterv1

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I've been running the E30 for a while now as my first DAC for TV & music. I paired it with the Breeze Audio BA25W linear psu (build quality is awesome btw).

My whole amp setup + DAC is powered through one smart plug. As much as I love how the E30 performs sound-wise, I feel like it's auto power-on feature is somewhat broken... When music is already playing and then I turn on power to my amp & DAC, it won't automatically turn on. So each time we're going to watch TV or listen to music, we need to make sure that nothing is already playing before turning on the amp. This wouldn't be such a big deal if the remote worked a little better :)

Ideally for my setup, the E30 would turn on automatically as soon as it receives 5v DC. I already emailed Topping if they could add this in future firmware, but they're not planning to do so. Maybe others could try contacting them about this as well?

Leaving the DAC powered on all the time is not an option for me with the linear psu.
 
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pieterv1

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Struggling to run this device with an 12v trigger switched outlet.
  • Auto power mode works as described in the manual.
    • When the device is in standby, a source playing will turn the dac on.
  • When I first power the device on, it STAYS in standby mode whether or not a signal is being fed.
    • If I plug and unplug the signal cable, the device will turn on and start playing music.
@JohnYang1997 this seems like a bug?
Same here - I'm running it in a setup that's powered on/off by a smart plug (see previous post). For me it'd be best if the Topping would just power on when it receives 5v DC.
 

Rottmannash

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I've been a little COVID bored, so I started tinkering with my home audio. I have an aging AVR (Onkyo TX-NR1010) and a nice set of (also aging) Martin Logan electrostatics in a 5.1 config. Nothing too crazy, Aerius i, Cinema Center, Scripts in the back. It's always been really wonderful sounding to me. Many years ago I ripped all my CDs to a lossless format and have used a Sonos Connect to play my library. I've owned DVD Audio and SACD players, so I had a handful of those disks. I like the idea of higher resolution digital, but the multi disk players I owned (NAD & Denon) both broke in less than two years, so I kind of gave up on it. Neither format made it anyway. Thanks to this board, I decided to try an alternative to Sonos in my system. I had an Intel i5 based mini-pc gathering dust so I loaded Volumio on it and attached an E30. I ripped my DVD Audio discs and graduated to higher resolution audio. I have 3 SACD disks and I think I'm going to have those ripped for me rather than adding to my junk equipment collection.
I have to say, I don't believe I can hear a difference between Sonos and the E30 on regular CD audio, though I'm still listening. So far, I don't really think I hear much difference with the higher resolution stereo content either but I have so little content. The quality of the recording and engineering seems to make much more difference than the digital resolution and the DAC. I must also confess I'm addicted to the Onkyo's multi-channel stereo mode. It really does a great job of spreading out the sound stage and adding depth. I will write an update after I do more critical listening.

Two things I do want to report and praise.
The e30 certainly sounds great and functions flawlessly on a intel based Volumio installation.
What it offers is an amazing amount of flexibility in high resolution PCM and DSD formats. It is low cost, measures exceptionally and sounds great. Many thanks to this board, Amir and his reviews.

Next, Volumio is the bomb! What a terrific application. I do hope that someday there will be multi-channel 5.1/7.1 DACs and Volumio will be enhanced to support multi-channel content and DACs. In the meantime, I remain addicted to the Onkyo multi-channel mixing magic. I can also play the few 5.1 tracks I have directly through the Onkyo's DAC. Clearly, engineering good 5.1 content is much harder than Stereo. There are so few fine examples. A shining example remains the Eagles Hell Freezes Over DTS. I wish there were more multi-channel content of that quality.
Try David Gilmour's recent DVD at Pompeii. It's in DTS and sounds truly incredible.
 

Rottmannash

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I've been trying Filter 5 and it seems to lessen bass "boominess" to some extent. I am running a SONY amp that has bass enhancement, which I like to keep turned on but with some music it does overcompensate. Filter 5 appears to help with that but I need to do more testing.
Pretty sure the engineers on this site would laugh at these comments if they weren't busy designing DACs. The filters have no effect on bass or midrange for certain and only marginally in audible higher frequencies.
 
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Do I need any driver if I am going to use E30 under Linux? I believe not, and it should work out the box, but since I see many post regarding drivers, just curious.
 

norcalscott

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Pretty sure the engineers on this site would laugh at these comments if they weren't busy designing DACs. The filters have no effect on bass or midrange for certain and only marginally in audible higher frequencies.
Good to know - as I mentioned, this was from a very quick test and anything I heard was very subtle and likely not related to the filter. I am a noob with DAC filters - this is the first external DAC I've owned that has filters.

edited to be more clear
 
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this simple tone generator, set the tone to 16000hz or higher depending on how good your hearing is atm, and play it fairly loudly so that you can clearly hear it. Then switch the filters. The fastest filters 1 and 3 will be the loudest (although the difference is incredibly slight), the slower filters will be quieter, and the super slow filter number 5 will sound like multiple tones, not a pure sine wave - this is presumably because it aliases frequencies above nyquist back into the audible range, thus producing audible information that was not in the original signal - needless to say, this is typically unwanted if we're talking about high fidelity reproduction of audio.
I will try. It's a double-blind-deaf test!
 

solderdude

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Do I need any driver if I am going to use E30 under Linux? I believe not, and it should work out the box, but since I see many post regarding drivers, just curious.
I am using the E30 with Linux Mint. Works fine out of the box. Did not need to install anything.
 

Balle Clorin

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It's minimal pop which happened in many DACs from cs4398 to ak4497 all need anti-pop circuit to function properly. It's only a issue if you directly connect the output to a poweramp which you wouldn't do for older DACs. The power amp amplifies the small pop to a much larger signal hence the pop. It will never be heard if a preamp is used. Even then it's an acceptable level of small pop that's improved over D70 and some older DACs.
And it's easily measurable using oscilloscope. I'll try to measure the level of the pop and share here.
So what DAC must I avoid then , wich ones are pop-less? I will run DaC directly to power amp and control the volume upstream of the DAC
 

companyja

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I dunno if it's been noted here but it seems like they silently revised the E30 from the AK 4118 reciever for SPDIF to a CS8416 chip. I assume there is a shortage of AKM chips that necessitated this change. I don't use SPDIF with my E30 but it might be of note to someone looking to buy it for this connection; I doubt the performance gets audibly worse or better but it probably does change the measurements on optical/coax some.
 
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