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AIYIMA A70 Stereo Amplifier Review

Rate this amplifier:

  • 1. Poor (headless panther)

    Votes: 16 3.9%
  • 2. Not terrible (postman panther)

    Votes: 41 10.0%
  • 3. Fine (happy panther)

    Votes: 195 47.3%
  • 4. Great (golfing panther)

    Votes: 160 38.8%

  • Total voters
    412
The graph is really expanded scale. The comments about corner frequency are spot on.
The 150 - 600hz range is about 3 times higher frequency range than of most sub outputs. Something around 40 - 200Hz is common and gives usable results.
I do not like double sub filtering. If using a a subwoofer built in frequency range adjustment, feed it with a full range input.
But even set to 150Hz, this sub out does not cut off soon enough. Almost every speaker will have full output below 150hz The overlap frequency range will yield very poor results.
 
I do not plan to use this sub output at all.

However, just for my learning, feedback on the below is welcome

I have a pretty big space and want max SPL from OG KEF LS50s, within the bounds of SQ and safety.

So, just because they "can" output below mid-bass frequencies, does not mean I want them to.

So, I will be HP filtering them at 200Hz maybe higher with testing

Choosing a woofer/subwoofer solution that has good SQ with high power right up to 300-500Hz

The overlap frequency range will yield very poor results.

Using real room correction / subwoofer integration (to address elsewhere), not part of any speaker amp.
 
I have never heard of any manufacturer putting a 3db per octave filter for subwoofer crossover so very much doubt this is the case that @AIYIMA did that here?
The 150/600hz indication were endpoint ones, where those frequencies were starting to near the relative noise floor and not onset of filtering frequencies!?
At no point did I mention 3db per octave.

See section starting with “ Frequency Response of a 1st-order Low Pass Filter”


Particularly the discussion on cut off point.
 
I did not mean 0dB, I meant the 150/600Hz are not the points where the filters start, but where they have finished
No, they are not finished. They have reached -3dB. SInce this is the cutoff freqeuncy, Erin doesn’t show the rest of the curve where it drops doen below -3dB at 12db/decade - right up to high frequencies.

50Hz and 250Hz might be better labels, where the filters' impact has begun?
“Where it has begun” is not how the cutoff frequency of a filter is defined: -3dB is. This is basic electronics. If you don’t understand this, and you don’t understand dB, you should really stop guessing.
 
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Two other items, the blue curve above is 150 Hz, the yellow curve is 600 Hz. Another reason not to use this sub out into a sub with frequency controls is there appears to be a subsonic filter on the sub out. If the sub also filters subsonic the bottom end may get attenuated.
Most subs have a subsonic filter built in. When people get carried away with output levels it can blow amp or damage the voice coils. The filter protects your sub and reduces warranty claims.
 
Anyone who did want to use this sub out, would only do so to amp + passive.

With an active sub you'd want a plain fullrange main input signal.

There's never been anything but a 100% consensus on that fact from anyone in the thread.
 
If you were to use this amplifier from a single-ended (RCA) source, but have a passive/switch conversion to XLR connector first, would you get the superior balanced input performance out of the amplifier? In other words, is the balanced performance of this amplifier an inherent part of the balanced inputs? The definite problem with this would be the gain structure of the setup with a typical 2V RCA signal.
 
If you were to use this amplifier from a single-ended (RCA) source, but have a passive/switch conversion to XLR connector first, would you get the superior balanced input performance out of the amplifier? In other words, is the balanced performance of this amplifier an inherent part of the balanced inputs? The definite problem with this would be the gain structure of the setup with a typical 2V RCA signal.
The benefit of balanced connections has nothing to do with the input to the amp.

They are about the connection between the source and the amp. The idea is that if noise is picked up between source and amp, it is picked up equally on both the positive and negative inputs. On the input of the amp, when the negative input is subtracted from the positive input, the noise picked up on the cable cancels out;

See section "Making a difference" in this article.

You can get most (but not all) of the benefit using a passive conversion cable IF it is wired to provide a pseudo balanced connection as shown in this diagram. (most RCA to XLR connectors are not wired this way and it is difficult to work out which are - you may need to make your own)

But see this thread also:



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with this connector you send the ground twice - both for the shield, and the negative input of the XLR.

The reason you don't get the full benefit of a full balanced connection is the impedances are not matched. On a balanced output - you will get both negative and positive outputs with the same impedance - eg 100ohm

With an RCA output the pin will have an impedance, but the ring is ground - essentially 0 impedance. On a rough back of the envelope calculation I did this results in a reduction in noise rejection from around 40dB to 30dB.
 
They are about the connection between the source and the amp. The idea is that if noise is picked up between source and amp, it is picked up equally on both the positive and negative inputs. On the input of the amp, when the negative input is subtracted from the positive input, the noise picked up on the cable cancels out;
They are also about separating the audio signal from ground so that ground noise (leakage currents, loops etc.) doesn't get mixed in with the audio. In a domestic setting this is the main benefit as there usually isn't that much interference, and cable runs are relatively short.
You can get most (but not all) of the benefit using a passive conversion cable IF it is wired to provide a pseudo balanced connection as shown in this diagram. (most RCA to XLR connectors are not wired this way and it is difficult to work out which are - you may need to make your own)
This is true for proper balanced inputs - taking signal as the difference between the hot and cold, then splitting phase to drive the halves of the bridge. Unfortunately with some amps we've seen the halves of the bridged amp effectively treated as separate single ended inputs, one having hot/ground as input and the other having cold/ground. If you drive that with the RCA-XLR cable you can't get the full power output of the amp, but you should at least get the ground noise rejection. I don't know how they implemented it in this amp.
 
They are also about separating the audio signal from ground so that ground noise (leakage currents, loops etc.) doesn't get mixed in with the audio. In a domestic setting this is the main benefit as there usually isn't that much interference, and cable runs are relatively short.

This is true for proper balanced inputs - taking signal as the difference between the hot and cold, then splitting phase to drive the halves of the bridge. Unfortunately with some amps we've seen the halves of the bridged amp effectively treated as separate single ended inputs, one having hot/ground as input and the other having cold/ground. If you drive that with the RCA-XLR cable you can't get the full power output of the amp, but you should at least get the ground noise rejection. I don't know how they implemented it in this amp.

Ground noise and common mode noise are the same thing.
 
Ground noise and common mode noise are the same thing.
Thanks, I know how a proper balanced connection works and the reason for it. The purpose of my question is whether, with this amplifier, one could use a single-ended source into the balanced input (via pseudo-balanced cables) to avoid the poor performance of the single-ended input without issue...whether that may depend on a specific amplifier's implementation or not. Some amplifiers do have issue connecting this way, perhaps the only way to know is test but without the proper equipment getting a definitive answer is hard.

I did not have the time to read a 35 page thread and thought maybe someone had tried this obvious solution to the worse single-ended input performance.
 
What is worse about the single ended input?
 
What is worse about the single ended input?
It's in the review...10 dB worse SINAD: 95dB for balanced so roughly 16-bit, versus 85 dB for single-ended. Possibly due to the higher gain needed for the single-ended input which matches the typical 6dB, but the performance difference is more than 6dB especially if you look at the noise versus frequency graphs in detail. It would really need to be tested to know I suppose...
 
It's in the review...10 dB worse SINAD: 95dB for balanced so roughly 16-bit, versus 85 dB for single-ended. Possibly due to the higher gain needed for the single-ended input which matches the typical 6dB, but the performance difference is more than 6dB especially if you look at the noise versus frequency graphs in detail. It would really need to be tested to know I suppose...
Exactly - the 6dB is from the needed gain difference. So you can't gain that back since the supplied voltage from an RCA output will still only be 2V - you'll still need higher gain (or the volume up higher which has the same effect) to get to the same volume.

However the further 4dB lost - which is unusual may well be differences in the specific input circuit. You may get some of that improvement.


However, in reality - the difference will most likely be inaudible.
 
Can anyone confirm if it is possible to run this amplifier as bridged mono? Thank you
 
Hello fellow audio enthusiasts,

I've been quite impressed with the AIYIMA A70, particularly its implementation of PFFB which yields excellent measured performance for its price point. However, through logical deduction and analysis, I believe there are still some cost-driven compromises in its design that can be overcome to unlock its full potential.

My primary concern started with thermal performance. Using a high-precision lab-grade stabilized power supply (0-60V, 0-20A), I conducted a simple comparison.

Operating at 36V:
IMG_4864.png
Operating at 48V: ( included AC adapter 48V 5A )
IMG_4865.png

The temperature difference, even under idle or low-load conditions, was significant enough to convince me that operating at 36V is the optimal choice for thermal stability and long-term reliability, without sacrificing the necessary power headroom for my setup (XLR balanced connection).

Based on this, I've developed a two-phase upgrade plan focused on addressing the most critical bottlenecks in a logical sequence. I wanted to share this plan to invite discussion and critique from this knowledgeable community.


Phase A: The Foundational Upgrade (High-Impact, Low-Risk)

This phase targets the two most fundamental components that define the amplifier's core performance: the op-amps and the main power supply capacitors.

  1. Op-Amp Swap to OPA1612:
    • Objective: To minimize noise and distortion at the very first stage of the signal path. The goal here is not to "color" the sound, but to achieve the highest possible fidelity by eliminating the inherent noise floor and distortion limitations of the stock op-amp. The OPA1612 is, by the numbers, one of the best choices for this mission.
  2. Power Capacitor Swap to Rubycon ZLJ Series (2200µF/63V):
    • Objective: To maximize transient response by drastically lowering the ESR of the main power caps. A powerful external supply is only as good as the internal "last-mile" delivery system. The ZLJ series is renowned not for being an "audio" capacitor, but for its outstanding technical specifications: extremely low impedance/ESR and high ripple current tolerance, which are precisely what's needed to support the instantaneous current demands of the TPA3255, especially for low-frequency reproduction.
I plan to implement Phase A first.


Phase B: The Micro-Refinement Upgrade (Advanced)

If Phase A proves successful, I will consider these more intricate modifications. These are aimed at optimizing the environment in which the core components operate.

  1. MLCC Bypass for Main Capacitors: Add a small (e.g., 1µF) Multi-Layer Ceramic Capacitor (MLCC) in parallel with each new Rubycon ZLJ capacitor. This is to filter out very high-frequency noise that the larger electrolytic caps are less effective at handling.
  2. MLCC Power Supply Bypassing for OPA1612: Add a 0.1µF MLCC directly to the power supply pins of the OPA1612, ideally on the underside of the PCB for the shortest possible path. This is to ensure the cleanest possible power is delivered directly to the op-amp, maximizing its performance potential.
  3. Input Coupling Capacitor Swap: Replace the input coupling capacitors with high-quality film capacitors (e.g., WIMA). This targets the signal purity before it even reaches the op-amp.
Phase B is certainly more complex, especially the modification for the OPA1612. My primary focus for now is on the high-impact changes in Phase A.

I look forward to hearing your thoughts and any potential pitfalls I may have overlooked.
 
My primary concern started with thermal performance.
This is an unfounded concern on the face of it IMO. I say this out of experience with my two now-elderly A07s which, although notably deficient when it comes to thermal management (undersized heatsink and small, unventilated case), have been operating absolutely trouble-free at 48V pretty much 24/7 for well over four years. Of course there's nothing wrong with operating the TPA3255 at a lower voltage -- in fact my girlfriend's newer A07 runs on a typical sub-20V laptop power brick, which is perfectly adequate given the rather low volumes at which she listens -- but AFAICT there's no real world benefit to that choice, so offhand I'd say "stand down from red alert" on this particular issue.
 
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This is an unfounded concern on the face of it IMO. I say this out of experience with my two now-elderly A07s which, although notably deficient when it comes to thermal management (undersized heatsink and small, unventilated case), have been operating absolutely trouble-free at 48V pretty much 24/7 for well over four years. Of course there's nothing wrong with operating the TPA3255 at a lower voltage -- in fact my girlfriend's newer A07 runs on a typical sub-20V laptop power brick, which is perfectly adequate given the rather low volumes at which she listens -- but AFAICT there's no real world benefit to that choice, so offhand I'd say "stand down from red alert" on this particular issue.
Thank you for sharing your valuable real-world experience with the A07s, Bruce. It's reassuring to know they've been running reliably at 48V for that long. You're absolutely right that the TPA3255 is a robust chip and basic functionality isn't compromised at higher voltages.

My approach, however, stems from a slightly different philosophy, perhaps influenced by the typical Japanese housing situation – our listening rooms can be quite small and poorly ventilated, so any additional heat becomes a noticeable comfort issue quickly! Haha.

But beyond just comfort, my primary concern is about optimizing performance margins and ensuring absolute stability, rather than simply preventing catastrophic failure. Operating at 36V provides a significant thermal margin, which I believe contributes to minimizing any potential for thermally-induced noise or performance drift over long listening sessions. It’s about creating a foundation of unwavering stability.


To clarify my modification strategy further, it's perhaps more important to state what I am deliberately NOT changing, and why:

Components I'm Intentionally Avoiding:
  • Output Filter Components (Inductors & Capacitors): As you know, the PFFB implementation is critically dependent on the precise characteristics of this LC filter. Altering these values without a full re-characterization of the feedback loop could compromise stability, degrade performance, or, in the worst-case scenario, induce oscillation. The stock components are integral to the PFFB design.
  • Small Capacitors in the Signal/Feedback Path: These aren't just simple coupling caps; they are integral parts of the feedback network that define its frequency response and stability margins. Changing them would be akin to performing surgery without understanding the anatomy.
My modifications are therefore surgically targeted only at the "supporting cast" – the input stage op-amp (swapping for lower noise/distortion) and the main power supply capacitors (swapping for lower ESR to maximize transient delivery). The goal is to allow the core TPA3255 and its PFFB architecture to perform at its theoretical best, while preserving the integrity of the feedback design that makes this chip so impressive in the first place.
 
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