• 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 PA5 Review (Amplifier)

xrk971

Member
Audio Company
Joined
Nov 19, 2021
Messages
68
Likes
269
Location
Metro Wash. DC
We knew that from the performance and from the PFFB traces visible on the bottom of the PCB.

From Post 1078.


index.php
 

pma

Major Contributor
Joined
Feb 23, 2019
Messages
4,602
Likes
10,768
Location
Prague
This is why I think it is important for an amp to be able to reproduce sound without imparting too much phase shifts below 300kHz. When I design a discrete Class A or Class AB amp, I look at the phase shift and optimize the design in LTSpice to minimize phase shifts below 300kHz. In listening tests, this seems to improve the amplifier’s ability to provide better imaging and soundstage (all subjective measures).
What kind of music files do you use to keep phase shifts unchanged up to 300kHz? What kind of speakers, microphone? Are you saying that the amp must keep the phase up to 300kHz, however the signal source may be 10 - 100 x degraded in this regard?
 

xrk971

Member
Audio Company
Joined
Nov 19, 2021
Messages
68
Likes
269
Location
Metro Wash. DC
Digital audio files do not need to be at 600kHz data rate to have 300kHz phase accuracy. Although there is 768khz rate dats files. Also much faster for DSD and other formats. Typical clocks that control ADC’s and DAC’s run at 12MHz and have typically 50ppm stability and jitter measured in the picoseconds range. More than enough to preserve phase accuracy to 3usec (300kHz).
 

DS23MAN

Active Member
Joined
Sep 18, 2019
Messages
161
Likes
190
Location
Hypex doorstep
We knew that from the performance and from the PFFB traces visible on the bottom of the PCB.

From Post 1078.


index.php
Please read the tech note on PFFB carefully, Ti states the THD improvement is max 5 db with pffb. So it is not only PFFB!
 

pma

Major Contributor
Joined
Feb 23, 2019
Messages
4,602
Likes
10,768
Location
Prague
Digital audio files do not need to be at 600kHz data rate to have 300kHz phase accuracy. Although there is 768khz rate dats files. Also much faster for DSD and other formats. Typical clocks that control ADC’s and DAC’s run at 12MHz and have typically 50ppm stability and jitter measured in the picoseconds range. More than enough to preserve phase accuracy to 3usec (300kHz).
And you seem to believe that the captured music has the same "phase accuracy" in microphone paths. Do you have an idea how their response looks like? And have you ever analyzed the "768kHz data files" for their spectral content? So where does it end?
 

Yaiba

Member
Joined
Mar 18, 2021
Messages
14
Likes
38
Just received and test it. I connected with SU-9 via XLR(female) to TRS cable and speaker JBL Stage A130, amplifier power more than enough for listen loud 80dBA(measured with UNIT-T UT353 sound level meter) at middle range (3m).

I'm not report subjective review as some may not want to hear it. But I promised to buy the next class AB amp that @JohnYang1997 will release soon. I respected your R&D result.

Anyway, the build of speaker binding post connector very loosen. I need to open cover to fix it from inside. Thank @genfreeciv for internal picture ,it made easier to find how to open it. It need to be more qc for this price level.
 

mikolaj

Member
Forum Donor
Joined
May 11, 2021
Messages
9
Likes
19
Any idea if this amp would work properly with speakers that has impedance down to 2.9 ohm? (focal aria 926)
Topping specification mentions load only to 4 ohm
 

KSTR

Major Contributor
Joined
Sep 6, 2018
Messages
2,730
Likes
6,100
Location
Berlin, Germany
Please read the tech note on PFFB carefully, Ti states the THD improvement is max 5 db with pffb. So it is not only PFFB!
Post Filter Feedback is a design technique, not a specific circuit. The passive TI example is just one simple variation with a lot of specific conditions, like not loosing too much circuit gain and simplicity of the compensation (2-pole).

The tough thing with heavy PFFB is the compensation scheme, you need a higher order feedback loop (3rd order minimum) to have still some gain for correction at 5kHz++ without running into stability issues. Running the main LC resonance at the highest possible frequency helps (~50kHz in this case).
 

KSTR

Major Contributor
Joined
Sep 6, 2018
Messages
2,730
Likes
6,100
Location
Berlin, Germany
Any idea if this amp would work properly with speakers that has impedance down to 2.9 ohm? (focal aria 926)
Topping specification mentions load only to 4 ohm
Only depends on the maximum level you can get before you run into current clamping or thermal issues. This is true for all amplifiers.
A dip to 2.9Ohm in a small frequency range is not going to be problem.
 

KSTR

Major Contributor
Joined
Sep 6, 2018
Messages
2,730
Likes
6,100
Location
Berlin, Germany
@xrk971,
so you design your power amplifiers with unrestricted bandwidths of 1Mhz and higher, basically capable for AM transmitter duty?
And you want to tell us that a reasonable bandwith restriction to, say ~40kHz (on both channels) is audible wrt soundstage and localization?
Also, you can hear the difference in DAC's imaging filter with the same magnitude response but one being minimum phase and the other linear phase?
And, what's the bandwidths of your tweeters (and the matching, not to forget)?

Where is the serious evidence of audibility of all this?
 

BoredErica

Addicted to Fun and Learning
Joined
Jan 15, 2019
Messages
629
Likes
900
Location
USA
Max the volume on the DAC. This gives the best signal to noise/hum/buzz ratio, notably when you run unbalanced interconnects.
Would you mind explaining why maxing the volume on the dac and turning down the amp makes a difference, and why this is more important for unbalanced interconnects? I always found this confusing. Do I turn the volume down on the dac, the amp, the preamp (if there even is one in the system), or digitally.
 

KSTR

Major Contributor
Joined
Sep 6, 2018
Messages
2,730
Likes
6,100
Location
Berlin, Germany
Because...


"Only have rca output on DAC. Is it possible to DIY a rca to TRS balanced cable? Will this work?
It should but you won't get maximum power unless the DAC can produce 2.5 volts (some can, most cannot)."
Fair enough.
 

KSTR

Major Contributor
Joined
Sep 6, 2018
Messages
2,730
Likes
6,100
Location
Berlin, Germany
Would you mind explaining why maxing the volume on the dac and turning down the amp makes a difference, and why this is more important for unbalanced interconnects? I always found this confusing. Do I turn the volume down on the dac, the amp, the preamp (if there even is one in the system), or digitally.
With unbalanced connections any mains leakage current that flows through the RCA cable shield causes a voltage drop which is part of the signal as seen by the receiving end.
Thus, the higher the signal level on the line the larger the ratio of signal to noise (hum/buzz).
With balanced, the problem is less severe but still there.

General rule for signal transmission with cables is: always use the highest possible signal levels. Of course not so high as to overload/clip the input of the amp before it has a chance to attenuate it.
 

BoredErica

Addicted to Fun and Learning
Joined
Jan 15, 2019
Messages
629
Likes
900
Location
USA
With unbalanced connections any mains leakage current that flows through the RCA cable shield causes a voltage drop which is part of the signal as seen by the receiving end.
Thus, the higher the signal level on the line the larger the ratio of signal to noise (hum/buzz).
With balanced, the problem is less severe but still there.

General rule for signal transmission with cables is: always use the highest possible signal levels. Of course not so high as to overload/clip the input of the amp before it has a chance to attenuate it.
Seems like the volume knob is an extra bonus for me then. I think I can go without a preamp and just use the volume knob on the amp! :)
 

SylphAudio

Active Member
Audio Company
Forum Donor
Joined
Jan 16, 2021
Messages
192
Likes
289
Location
Philippines
Please read the tech note on PFFB carefully, Ti states the THD improvement is max 5 db with pffb. So it is not only PFFB!
TI note only has a passive PFFB implementation. That's why I've said before that this level of performance can be only done with an active PFFB.
Looks like they've figured out how to implement active PFFB properly.

Take a look at this TI Support thread

I already read this thread before and I thought back then that this is cdsgames which is the designer of Allo
 

Attachments

  • Capture.PNG
    Capture.PNG
    131.7 KB · Views: 147

TNT

Active Member
Joined
Jun 24, 2020
Messages
238
Likes
157
To be clear, I am talking about amplifiers needing to have phase accuracy in one channel. I am just saying a recording should have a time base accuracy equivalent to a bandwidth of 300kHz (or 0.3MHz). Not a hard thing to ask for given that DAC clocks typically have a time base originating from a 12MHz Xtal oscillator. An amp should be able to reproduce a time series signal with phase accuracy relative to itself of at least 2usec. This can be confirmed with an auto correlation function.

It is a well known fact that people totally deaf in one ear can spatially locate sound sources in 3D space with their remaining ear. There have also been experiments where the pinna (external ear lobes) was taped over with a smooth funnel and the subject lost their ability to locate the directionally with one ear. (Perhaps try this experiment yourself). There are a lot of scientific papers on this. Just search “Human Pinna Sound Localization”. It’s no accident that the pinna is asymmetric so that directionality of the sound wave can be resolved through the phase differences imparted by the pinna, and the the brain applies real-time “bio-DSP” aka psychoacoustic processing, to determine the spatial location.

A 1mm path difference in the pinna can impart about a 3usec shift to a wave traveling at 342m/sec (speed of sound).

If you look at the paper here:

According to section V in the paper, the time domain of the pinna is 2usec to 300usec (0.3ms).

Look at Fig. 4 which shows the transformation of a sound pulse from a source from the side by a cast mold of a human pinna fitted with a microphone. We can see that it imparts a transformation of the pulse in the time domain.

View attachment 171186
More recently, there have been papers discussing the microsecond resolution of the human hearing:

Krumbholz K, Patterson RD, Nobbe A, Fastl H. Microsecond temporal resolution in monaural hearing without spectral cues? J Acoust Soc Am. 2003 May;113(5):2790-800. doi: 10.1121/1.1547438. PMID: 12765396.

Here is a very good review paper on sound localization by the human ear. Tons of references in there for those interested.


You can look for many more scientific publications discussing this.

This is why I think it is important for an amp to be able to reproduce sound without imparting too much phase shifts below 300kHz. When I design a discrete Class A or Class AB amp, I look at the phase shift and optimize the design in LTSpice to minimize phase shifts below 300kHz. In listening tests, this seems to improve the amplifier’s ability to provide better imaging and soundstage (all subjective measures).
Yet, there is virtually no spectral content above 15k from an orchestra at a normal listening position.
TI note only has a passive PFFB implementation. That's why I've said before that this level of performance can be only done with an active PFFB.
Looks like they've figured out how to implement active PFFB properly.

Take a look at this TI Support thread

I already read this thread before and I thought back then that this is cdsgames which is the designer of Allo
From the TI thread... "We are developing a hiend amp based on tpa3255 (under 400$ including linear PSU 250W" ..

This PSU is only 38x4= 152VA (~watt)...

Allo?

//
 

restorer-john

Grand Contributor
Joined
Mar 1, 2018
Messages
12,678
Likes
38,772
Location
Gold Coast, Queensland, Australia
Pretty sure the PA5 is a composite with some opamp being the master and the class-D chip being the power slave. Which makes a bit easier to squeeze out the last bit of loop gain even at higher frequencies, ~5kHz.

What about the 15kHz (45kHz BW) THD dip? An aberration in the AP?

1639046865447.png

That suggests an LF and HF separate FB network, perhaps trimmed for minimum THD at specific frequencies. There were a bunch of designs that utilized similar topologies in the 1980s, including some non-NFB, direct distortion cancelling offerings.
 

KSTR

Major Contributor
Joined
Sep 6, 2018
Messages
2,730
Likes
6,100
Location
Berlin, Germany
What about the 15kHz (45kHz BW) THD dip? An aberration in the AP?

View attachment 171266
That suggests an LF and HF separate FB network, perhaps trimmed for minimum THD at specific frequencies. There were a bunch of designs that utilized similar topologies in the 1980s, including some non-NFB, direct distortion cancelling offerings.
AP abberation? No ;-)
The decrease around 1W is the result of many factors but I don't think there is special trimming involved. Composite or passive PF feedback is always split in AF and HF sections, otherwise no compensation possible.

By the way, I could be wrong and the PFFB is passive (PPFFB) but perhaps higher order than 2nd. As long as the input (into the PPFFB) can be driven high enough (say 10Vrms with frontend OpAmps running from +-15V) the feedback factor can be increased significantly but limited of course by the 21dB "open loop" gain of the chip.
 
Top Bottom