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Martin Taylor via outlook.com
12:06 PM (1 hour ago)
It amazes me that the forum which claims honesty, truth, accuracy etc., appears to make so little attempt at finding the truth and repeats poor or incorrect assumptions, including an engineer who rates himself above all others (Cawley) and a Turing machine with crayons who is allowed to spout rudeness against others without contributing a single thing of any interest himself (DQ). Then there’s all the hangers-on who worship everything you say, unintentionally absorbing the same incorrect advice.
So much for your laughable stance. Now for the proof. You are all wrong, despite my several attempts to explain how the P10 works. It doesn’t help that PS Audio call it a regenerator but I can understand their desire not to go into great explanations and put potential customers off.
The P10 is not a ‘big amplifier’, it does not convert all incoming power into DC and then create the AC waveform as in older models. It’s a tracking amplifier, or as I call it a waveform repairer. Even reviewers of the P10 get it wrong and PS Audio make no effort to correct them. You have to go back to the previous model, the PPP, in order to get the explanation. Here it is...
“Enter Bob Stadtherr, PS Audio's out-of-the-box thinker. He invented a way of converting the incoming AC signal into a 5-volt square wave by using a synchronization circuit. This square wave is perfectly in phase with the AC signal so it follows it accurately. Now a processor, a DSP chip running at 6.4kHz using Direct Digital Synthesis techniques, assembles a sine wave from the self-generated square wave. Next, the sine wave is fed to a digital-to-analog converter or DAC, the same kind as one would find in a CDP. The DAC here samples the sine wave and puts out an analog 2.5-volt signal again phase-locked with the incoming AC signal. Now we have a completely regenerated but downscaled likeness of the incoming signal which merely requires an amplifier to boost it to the necessary voltage.
Of course there is also a filter that cleans the low voltage analog signal before it is sent off to the power amplifier. Filtering of low voltages is far easier to achieve than on high voltages. As mentioned, the PPP uses 2.5 volts for the analog signal and from this signal, all remnants of DC coming from the DAC plus any remaining sampling noise of the DDS circuit are purged. The basic idea hence is to create an exact miniature replica of the incoming AC signal, clean it up, amplify it to the desired and now precisely stabilized level to pass it on to whatever device wants it. And because it can be done without bulky transformers -- the load is driven directly -- the source impedance can be very low.
Still, this technology is not yet the precise reason why the PPP is so much more efficient than its predecessor (which, incidentally, used the same minimize/filter/amplify technology). The real secret behind the efficiency resides in the power supply for the amplifier. In the P1000, PS Audio used a 200-volts DC power supply. This voltage was fed to a class A/B amplifier and the result was a 50 or 60Hz output. With the 200 volts, the amplifier would waste 50% of the used energy on a good day, more on a bad one. In the PPP, only 70 volts are used to feed the amplifier. That naturally lowers both heat and output voltage substantially. In order to elevate the output voltage, the PPP design now couples the 70-volt DC power supply and amplifier and AC-couples them hot to ride up and down with the incoming sine wave. That's a really bright idea. The resultant output is more than sufficient to produce a powerful and clean sine wave without the need for a high power supply voltage.”
Honestly, I don’t know why I bother but I’m not going to join your forum in order to post the correction. It should go without saying that any DC is going to upset the sensitive circuits generating the low voltage reference waveform. It also explains why you cannot change the output frequency as with older models, why you can switch it off and keep listening and why setting the output voltage to as close as possible the input voltage gives the lowest distortion output. The big transformer is there because you don’t want to try to repair the waveform for the entire local community.
But hey, who am I to correct such auspicious people as some of your members? I’m just an electronics engineer with a degree and a pair of ears.
But I thank you for your input, I made a presumption from what I knew, so the new circuit does things differently, so does Power Inspire for less than 1/10 the cost, and a NVA BMU does similar but better, again at about a 1/10 of the cost. You have got yourself in the pocket of a rip off merchant retailer, you want to waste your money, fine it is your money to waste, but you feed that slurpers pocket by recommending him, so he can feed off the less rich and snobbish members.
BTW I don't "rate myself above all others" but I was lecturing at AES in NYC a few months ago.
Generally doesn’t make any difference to the points being made, the ability to remove mains problems is now limited by the frequency response of the correcting amplifier, which is the point I think Geoff is referring to. But there is still a large transformer that passes all the power that the load requires, and it will still saturate if there is an asymmetric supply.
Be mad if personal insults got to me , please call me what you like.
Martint ,I call you, because your opinions on HiFi are shite and dangerous
But will that affect the output waveform in any way ?
I will refer you to my earlier "don't know" answer, but it will reduce the power handling capacity of that transformer, and likely increase the distortion on its secondary, so will consume more of the available error correction the tracking amplifier has available.