• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

Topping B100 Amplifier Review

Rate this amplifier:

  • 1. Poor (headless panther)

    Votes: 23 5.8%
  • 2. Not terrible (postman panther)

    Votes: 22 5.6%
  • 3. Fine (happy panther)

    Votes: 77 19.5%
  • 4. Great (golfing panther)

    Votes: 272 69.0%

  • Total voters
    394
See the spectral impedance graph from Amir’s review; almost half the spectrum covered at or around 3.7ohms for LS50 Metas.

View attachment 393944

-Ed
Almost half? It's only 3.7-ish ohms between 200 to 300Hz from what I'm looking at.
 
Last edited:
Almost half? It's only 3.7-ish ohms between 200 to 300Hz from what I'm looking at.
Ah apologies, you’re right—the part under 20hz is irrelevant
 
Ah apologies, you’re right—the part under 20hz is irrelevant
Yeah, the scaling on the x-axis can be a bit misleading if you aren't careful.
 
I watched that video, then tracked down the track they were playing. Not quite my cup of tea, but it has some nice things going for it. The track has a ton of subbass starting at around 30Hz. This might be what's drawing all that power. Strangely with my headphone setup I actually had to turn the volume down compared to something with a much lower DR because the pulsing subbass was so irritating. I put my HPA into high gain and tried to see how high I could get it without pain, and the result was...not very high. So I'm not sure how the Harbeth people were pumping 600 watts and putting their heads near the speaker.
It's definitely the bass transients pulling the (peak!) power. I can't recall if they measured SPL in that video. I do remember the track as it's in my test track playlist. It's "Laptevinmeri". I don't think 600 W low frequency peak into a 12" bass driver is unrealistic. Keep in mind that this is low frequency and that we are not very sensitive to hearing such content which explains how they are able to talk to each other and that we can hear their voices in the video.

I've only played the track a handful of times on my system. I've not bothered to listen to it with headphones or IEM. I could easily imagine it distorting, prompting you to turn it down?
E70V has 1.3μV(A) of self-noise.

L70 in Low gain (+6dB) will amplify that to 2.6μV(A).

L70 has 0.7μV(A) of self-noise on top.

Incoherent summing gives us 2.7μV(A) going into the B100 (no, you don't just add them :))

B100 in Medium gain will amplify that to 8.9μV(A).

B100 has 1.0μV(A) of self-noise on top.

-> ~9μV(A) of noise on the B100 output.

At 4Ω, the B100 can output 100W max (=20Vrms), so the maximum achievable playback DR is 127dB(A).

That is, if you leave the L70 on max volume and attenuate using the DAC.

If you leave the DAC on max and attenuate with the L70, then SNR/DR should improve slightly.
You are not showing all of the steps. When you have calculated the voltage after amplification why can't just add them. -Why wouldn't you? It's not out of phase or anything.

BTW if you don't know the characteristics of the noise itself (frequency band) much of this is mostly of academically value, isn't it?
 
It's definitely the bass transients pulling the (peak!) power. I can't recall if they measured SPL in that video. I do remember the track as it's in my test track playlist. It's "Laptevinmeri". I don't think 600 W low frequency peak into a 12" bass driver is unrealistic. Keep in mind that this is low frequency and that we are not very sensitive to hearing such content which explains how they are able to talk to each other and that we can hear their voices in the video.

I've only played the track a handful of times on my system. I've not bothered to listen to it with headphones or IEM. I could easily imagine it distorting, prompting you to turn it down?

You are not showing all of the steps. When you have calculated the voltage after amplification why can't just add them. -Why wouldn't you? It's not out of phase or anything.

BTW if you don't know the characteristics of the noise itself (frequency band) much of this is mostly of academically value, isn't it?
Generally the whole thing is academic considering I am happy with the sound and cannot find any flaws during my listening.

-Ed
 
Generally the whole thing is academic considering I am happy with the sound and cannot find any flaws during my listening.

-Ed
It is.
But when summing noise quantities without respect to which frequencies the noise consists of it get's a bit too illogical for my liking. At least with my current knowledge on the area. Which is lacking! :D

Maybe this is how it's done professionally. I don't know. Hopefully staticV3 will answer.
 
I've only played the track a handful of times on my system. I've not bothered to listen to it with headphones or IEM. I could easily imagine it distorting, prompting you to turn it down?
You were right! I was playing them on HD6XX, adjusted to Harman. It did not sound particularly distorted, just uncomfortable. But of course the subbass measures with plenty of distortion. So then I tried my Salnotes Zero, with negligible subbass distortion, and I was able to crank them up way higher without the discomfort.
 
Not like you can have impedance/phase plots on hand for everyone's speakers.

That's why audiosciencereview, erin's audio corner and stereophile exist. Why posting what is apprently technical stuff (with calculations) if you don’t use measurements at your disposal ?

Taking for granted what could be an carfully designed experiment with private parameters can be seen as not resonnable at all.

By the way, I really don’t understand the « good old days with true amplifiers » trend of some individuals of this thread. Sure, if you had large income, you could buy japanese Accuphase, american McIntosh (« japan » and « american » because, apparently, « chinese » is a crucial parameter of technical abilities, or maketing positionnement). But for low income individual like me (even today), the choice of an amplifier or a speaker was purely russian roulette. I didn’t miss that good old times at all and I praise topping in publishing measurements and do the best they can do to invent the best electronic topologies for a resonable price, even though 16 bits resolution is perfectly fine with me.

ryanmh1 : "I'm assuming running without an active crossover and subwoofer."

My bad, I wasn't clear. My point was that the concept of "speaker sensitivity" that you use (and as it is used generally in forums) has no signification (especially when you have to choose an amplifier) if you don't precise the bandwith in which this sensitivity is mesured. If you don't precise the bandwith and the area of the drivers are the same, "higher sensitivity" has the exact same meaning as "less bass".

By the way, you don't need a subwoofer to alleviate a high EPDR. The only thing you need is an equalized vigorous room mode (20 dB are not uncommons) if this room mode is near the EPDR point.

In short : to chose an amplifier, you have to take into account the loundness that you want (which is tied with the dimensions of your room), the bandwith you want to reproduce (with or without a subwoofer), the impedence and electrical phase of your speakers and the acoustical modes of your room. If you choose carfully (and the measurement of audioscience review and others allow you to do that), you can be perfectily fine with a B100. Abstract caculations about a mythological "speaker sensitivity" are surely not enough.
 
When you have calculated the voltage after amplification why can't just add them. -Why wouldn't you? It's not out of phase or anything.
Because they're incoherent noise sources. They don't just perfectly add. Their relative phase is random.

BTW if you don't know the characteristics of the noise itself (frequency band) much of this is mostly of academically value, isn't it?
The noise spectrum of all three Topping devices, in the audible range, is uniform (white noise).
 
Because they're incoherent noise sources. They don't just perfectly add. Their relative phase is random.


The noise spectrum of all three Topping devices, in the audible range, is uniform (white noise).
How then do you add them? And also I wonder if the characteristics (frequencies inherent) in the noise is uniform why you can't add the voltage after amplification?
For example this one:

L70 in Low gain (+6dB) will amplify that to 2.6μV(A).

L70 has 0.7μV(A) of self-noise on top.

Why is that 2.7 uV and not 3.3? Incoherent summation, like you say it is, would have you to know each frequency phase to know which would add and which would subtract from the summed one? Or does white noise have a characteristic from where you can derive a factor in this kind of arithmetic?

Lots of questions but they are of honest character I promise. And quite useful if I can turn it into an excel formula to use in sound chain noise calculations.
 
How then do you add them? And also I wonder if the characteristics (frequencies inherent) in the noise is uniform why you can't add the voltage after amplification?
For example this one:

L70 in Low gain (+6dB) will amplify that to 2.6μV(A).

L70 has 0.7μV(A) of self-noise on top.

Why is that 2.7 uV and not 3.3? Incoherent summation, like you say it is, would have you to know each frequency phase to know which would add and which would subtract from the summed one? Or does white noise have a characteristic from where you can derive a factor in this kind of arithmetic?

Lots of questions but they are of honest character I promise. And quite useful if I can turn it into an excel formula to use in sound chain noise calculations.

With noise and distortion, you square, sum and root.
 
Yep, I would consider it IF it was those things I mentioned and IF I were in the MARKET for AMPs

Those NAD 2200s will be buried with you.
 
I watched that video, then tracked down the track they were playing. Not quite my cup of tea, but it has some nice things going for it. The track has a ton of subbass starting at around 30Hz. This might be what's drawing all that power.

That's another myth, or an anthropomorphic projection if you will. Power doesn't increase automatically on the opposite way of frequency in the bass and subclass region. You have to know the resonant frequency of the driver to make such assumption. This is because a driver has two functioning modes :
  1. Under the resonant frequency, the amplifier sees the stiffness of the driver suspension. It's called "control by stiffness"
  2. Above the resonant frequency, the amplifier sees the inertia of the mobile mass of the driver. It's called "control by the mass".
Stiffness of the suspension generally easily controllable by the amplifier : the power draw is reasonable even with a high excursion.
But with inertia, you can reach a considerable need of power very quickly.

In a case of subwoofer, if you have a high pass filter to 80hz and a resonant frequency around 40hz, it's probable that the power requirement will be reasonable.
If you have a low resonant frequency, your driver will be controlled by the mass and then the power requirement will be huge.
 
How then do you add them?
Here's the formula:
Screenshot_20240922-112747_Chrome.png
Source

And also I wonder if the characteristics (frequencies inherent) in the noise is uniform why you can't add the voltage after amplification?
Because the phase difference between the summands is random.
 
Here's the formula:
View attachment 394016
Source


Because the phase difference between the summands is random.
Not to take this totally off topic, and thank you for nudging me in the right direction, but using this formula got me 4.8 uV for the two levels; 2.6 and 0.7 uV. After checking the formula in excel was okay I checked with this known page for all sorts of technical sound related stuff, which also gave 4.8 uV.
You got 2.7 uV.

Where's the mistake or misunderstanding?

1727017915386.png

1727017934230.png

 
Not to take this totally off topic, and thank you for nudging me in the right direction, but using this formula got me 4.8 uV for the two levels; 2.6 and 0.7 uV. After checking the formula in excel was okay I checked with this known page for all sorts of technical sound related stuff, which also gave 4.8 uV.
You got 2.7 uV.

Where's the mistake or misunderstanding?

View attachment 394054
View attachment 394055
You are mixing up dB (which is a ratio) with voltages. The calculator only works for "adding" dB, not voltages.
 
You are mixing up dB (which is a ratio) with voltages. The calculator only works for "adding" dB, not voltages.
Of course. Sqrt(2.6^2+0.7^2) = 2.7 uV. The formula stated voltage as well and got me confused there for a moment. It's of course referring to dBV with "voltage levels"..

1727019774753.png
 
@ you three distortion and noise adding experts (nice discussion btw.): If the power supply would have had less distortion at 60 Hz, 120 Hz, and 180 Hz - would the SINAD of the B100 be even better than 120 dB? I know, it’s only academically, because everything over 116 dB is fully transparent anyway, but if the power supply would have had the LA 90 48 Volt Power Supply-Qualities of minus 145 dB at 60 Hz:
the B100 would have killed the LA90 discrete by even more than 0.5 dB margin?
1727020528739.png
 
Back
Top Bottom