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Post by Deleted on Feb 16, 2015 20:28:21 GMT
In DACs NOS = Non Over Sampling Sent from my GT-I9100 using proboards Ah! I was nowhere near it. That was the early ones then. I once owned a Technics CD player that made a big thing of being Bitstream (I think). I don't really understand digital to be honest. You can probably tell.
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solderdude
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Post by solderdude on Feb 16, 2015 20:29:24 GMT
Ian is 100% correct.
The filter goes between the source and the amp and it creates the 'opposite' frequency curve as the headphone has. So if the headphone has a +3dB hump of a certain width than the filter has a frequency response that shows a -3dB dip with the same width as the hump. Add both the +3dB of the headphone and -3dB of the filter and you end up with 0dB so compensated.
As ALL headphones wobble around and have peaks and/or dips of various dB's all headphones thus need their own 'opposite' curve to become 'flat'. This is what the modules do.. they define the opposite curve.
Of course you can come close by using EQ and pull up and down various sliders but it will never be as exact as the 'exact' curve. A filter can come closer with good headphones.
The Ember and O2 (as well as all other amps out there) all have equally flat frequency responses. What Ian says is correct... you often read amp A has boosted bass or suppressed mids or whatever. It is complete bollocks they are all equally flat (if well designed)
They can still sound different for 2 reasons. 1: output resistance interacting with headphone impedance creating a frequency dependent voltage divider (forget about the damping factor nonsense) 2: Added harmonics.
See it like this... A pure sinewave of 70Hz has an amplitude of lets say 0dB and lets assume that is 1V A pure sine wave has no harmonics. Now we have a tube amp that puts out the same 70Hz tone also at 1V It ADDS harmonics so the first harmonic = 140Hz and is at -40dB which is a factor 100 smaller in amplitude so 0.01V The total output voltage is thus 1.01V and consists of 70Hz + a small signal of 140Hz When you will be looking at the FR plot of the non distorting amp (lets call this the O2) than it will show 0dB at 70Hz (1V = 0dB). When you will be looking at the FR plot of the distorting amp (lets call this the Ember) than it will show 0dB at 70Hz (1.01V = +0.08dB) BUT a plot is created referenced to 1kHz and 0dB.
at 1kHz the amps do the same and also at the other frequencies so from 20Hz to 20kHz both amps will show the same. with the same input signal (a pure sinewave) the O2 will always give 0dB and the Ember always +0.08dB so both are equally 'flat' When we make a plot of the Ember and want 1kHz as 0dB than the input signal is lowered by -0.08dB so at 1kHz it shows 0dB and also at all the other frequencies.
BUT... here is the kicker. The O2 will only let you hear the 70Hz and the Ember will let you hear the 70Hz + a MUCH softer 140Hz but you'll hear it non the less. Just not as 2 separate tones but as one tone with a more natural 'warmth' to it. When the 70Hz and 140Hz were not related and would be 'free running' you would hear 2 tones.
So both amps are equally flat but because the added harmonic signal is VERY small in amplitude it won't SHOW on the plot (well as a marginal raise in amplitude) but your ears will hear it and measurement equipment that shows THD will show the signal and an FFT plot will also show it.
Now when you have 'corrected' the anomalies of the FR of the headphone it will be tonally correct but the Ember will still add the harmonics (tube goodness).
Of course it should be noted that when a 70Hz (or another frequency) pure tone is played through a headphone that headphone itself will also add quite a few harmonics. The amount of harmonics is frequency dependent as well so will differ from headphone to headpone.
The HD650 has exceptionally low distortion b.t.w. and once FR corrected it will thus sound 'better' than some others. The 'added tube goodness' will be faithfully reproduced by the headphone and is thus audible.
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solderdude
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Post by solderdude on Feb 16, 2015 20:31:22 GMT
The Technics has a MASH converter. Basically a 4 bit converter with lots of oversampling. Some of the current Delta Sigma DACs do a similar thing. NOS DACs WITHOUT a proper reconstruction filter roll-off the treble in a specific way. It is not easy to explain in just a few words. basically below a drawing of what happens in NOS DAC's that do not have a proper reconstruction filter. the thin sinewave line is what the original waveform looked like AND would look like again when a proper reconstruction filter (which does have to 'ring' in order to do so). Without the reconstruction filter a high frequency tone would look like the thick (and not so neatly) drawn line. You can see that at some point the amplitude is correct but at other points in time the amplitude is low. It becomes 'modulated' by the sample frequency. the average amplitude of the sinewave is thus lower and becomes 'rolled-off'. Lower frequencies do not have this problem because more samples describe the original waveform so will always be reproduced closer (not exactly like) the original waveform. The reconstruction filter is thus needed, not only for the higher but also the lower frequencies as they also become more accurate. No idea why many insist a non post filtered NOS DAC is better.. it may sound better to them because of the roll-off though. Of course the 'low BW' setting cannot replicate the exact same effect as described above but the 'average' roll-off one gets using a NOS DAC is approximated so it sounds similarly rolled-off. The same 'effect' can be created with an Ember while using ECC83 type tubes. These roll-off similarly as well but are also much noisier and increase the gain significantly.
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Rabbit
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Post by Rabbit on Feb 16, 2015 20:39:40 GMT
You explain things SO clearly Frans.
Your description kind of reminds me of additive synthesis as in the old Yamaha DX synths, except the frequencies are added above the fundamental which creates a very harmonic laden and possibly 'jangly' sound that is so prevalent in the 80s.
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Javier
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Post by Javier on Feb 16, 2015 20:40:06 GMT
Plus they have tons of alliasing (NOS DACs I mean)
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Post by Deleted on Feb 16, 2015 20:43:15 GMT
Wow!
I do actually understand what you're saying there. I sometimes feel that a lot of so-called experts barely know any more than I do myself. Not so with you Frans. It's awesome that you're prepared to share that kind of knowledge with folk like me. Many wouldn't take the time. Kudos to you my friend.
And yes, MASH. That was it. I think it might have been Philips who used the bitstream tag. I was the first to get a CD player among my friends. I got it on my 21st birthday, lol. A Denon DCD-1100 I think. It was awesome at the time. My local Woolworths had 3 CD's. Brothers in Arms, by Dire Straits, Zoolook by Jean-michel Jarre & a rendition of Tchaikovsky's 1812 overture and Marché Slàve with REAL CANNONS!
I bought all three.
Gordon.
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Post by Deleted on Feb 16, 2015 20:44:06 GMT
You explain things SO clearly Frans. Your description kind of reminds me of additive synthesis as in the old Yamaha DX synths, except the frequencies are added above the fundamental which creates a very harmonic laden and possibly 'jangly' sound that is so prevalent in the 80s. Lol, I'm listening to Hipsway right now.
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Post by Deleted on Feb 16, 2015 20:44:41 GMT
Plus they have tons of alliasing (NOS DACs I mean) As do I (on Facebook I mean)
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Rabbit
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Post by Rabbit on Feb 16, 2015 20:50:32 GMT
Wow! I do actually understand what you're saying there. I sometimes feel that a lot of so-called experts barely know any more than I do myself. Not so with you Frans. It's awesome that you're prepared to share that kind of knowledge with folk like me. Many wouldn't take the time. Kudos to you my friend. Gordon. He's a friggin' expert who can explain in plain English in spite of speaking Dutch as a native language!!!! He never ceases to amaze me with his knowledge and use of English. We take his language skills for granted, but when you think, it's not his first language, he's pretty amazing.
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Post by Deleted on Feb 16, 2015 20:56:53 GMT
Yeah mate, it kinda puts us islanders to shame.
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Post by Deleted on Feb 17, 2015 12:15:28 GMT
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Jakkal
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Post by Jakkal on Feb 17, 2015 14:41:15 GMT
.... The Ember and O2 (as well as all other amps out there) all have equally flat frequency responses. What Ian says is correct... you often read amp A has boosted bass or suppressed mids or whatever. It is complete bollocks they are all equally flat (if well designed) They can still sound different for 2 reasons. 1: output resistance interacting with headphone impedance creating a frequency dependent voltage divider (forget about the damping factor nonsense) 2: Added harmonics. ................ So there should be no audible difference between solid state amps if they are design correctly and have low output impedance? What about the speed and the other amp characteristics besides FR, should they affect what we are hearing?
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solderdude
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Post by solderdude on Feb 17, 2015 16:40:29 GMT
That's correct ... When we isolate the 'speed' aspect of signals.
The 'speed' of the opamp would actually depend on the maximum output voltage and frequency range. Some research has been done that could indicate rise times can be perceived >> than our hearing limit would suggest. The question is if this is STILL the case for complex signals like music where the brain has a LOT to process.
So lets assume we can detect up to 100kHz which could be almost be achieved with 192kHz digital signals. (most adults won't perceive > 16khz) That would be a sinewave and NOT squarewave. Squarewaves (as used in tests) are not present in nature and music signals. Even synthesizer generated squarewaves woun't be recorded as such due to antialiasing filters in the recording chain.
To accurately reproduce a 100kHz sinewave the amp output voltage needs to rise at least faster than the 'speed' it reaches around the 0V mark, the zero crossing if you will. Now lets assume we have an output voltage swing of +/-12V (Solstice, Sunrise, Starlight)
Normally a risetime is given between 10% and 90% of a signal but for sinewaves the maximum speed is only present near the 0 crossing so we can't use 'normal' rise time numbers here. So I took an scope and used a 24Vpp 100kHz sinewave and zoomed in at the 0 crossing. This gave me a 'slope' of 7.5V per microsecond (0.75V in 100ns measured) thus 7.5V/us.
Thus for Solstice, Sunrise, Starlight and all other amps that run on say +/-12V to 15V (most opamps) 7.5V/us is more than enough.
For 48V amps like Ember/Polaris the output voltage is twice as high and to reach the same 'speed' round the 0 crossing 15V/us is needed... In theory.
FYI, as this is a Polaris thread the Polaris itself reaches 16V/us for positive flanks and 20V/us for negative flanks (OPA551 limits) which exceeds the theoretical signal above already.
Now this is to reproduce a full swing 100kHz signal. In audio there is NO such a signal. When the music would play at its loudest (ear deafening) the bass notes would be loudest and 100kHz 'artefacts' would easily be 20dB below that (in practice it will be much lower even). 20dB is a factor 10 so the 100kHz in music would not reach 48Vpp but 4.8Vpp. This comes down to 3V/us that isn't that fast at all for music.
Suppose we need to reach 20kHz at full amplitude. You could not hear it and would probably get a severe headache from the pressure.
at 20kHz FULL SCALE we would need 0.75V/us for opamp signals and 1.5V/us for amps like Ember and Polaris.
So it's safe to say that 2V/us is fast enough for all amps. I believe NwAvGuy deteremined that 1V/us would be enough for his opamp based (+/-9V) amplifier which is already overdoing it. So technically most opamps should have no problem following the fastest signals.
Those that have a computer + audio software and can zoom in to sample level may have 'examined' fast transients in music signals by zooming in. If you haven't I would advise you to do so. Once you do that you will realise that music that sounds very 'fast' doesn't have ANY fast rise times at all and those signals span quite a few samples and rise only fractionally compared to the FSD over that period.
This means that even if you have a really loud and fast transient in music the actual signal is SLOOOOWWWW compared to say a FSD 10kHz sinewave (which doesn't exist in music). So the question is ... how much speed do we actually need to hear 'fast' transients in music.
Well in practice even speeds belonging to a 10kHz sinewave at max output will NOT be there in any music signal when we look at speed as in V/us. That's the theory at least.
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Rabbit
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Post by Rabbit on Feb 17, 2015 16:41:11 GMT
The FR of most amps nowadays is pretty flat. I find it very difficult to differentiate between amps and when I read descriptions of how an amp change the sound drastically, I always suspect something is amiss. Generally, I find many overstate the differences massively. Solid state amps are amazingly close to each other. Most react very fast as well. You'd need a quick ear to hear any differences. We just don't hear as fast as we think. Op amps are way faster than us. Generally, in an amp, I listen out for noise issues, volume pot travel and how it relates to volume, connections being good, and most important .... Enough power to supply proper transients and dynamics on recordings that actually contain that kind of information. Unfortunately, too many recordings have been processed to death so that the dynamic range is squashed to pulp. On some reviews, the reviewers write what they have listened to. If you check the recordings, you often find that they have been compressed terribly, so when the reviewer goes on about massive dynamics, then you know that something is really amiss. I'd say that most modern ss amps are pretty similar and the last fraction of quality comes with huge expense. Getting distortion lowered, noise level or noise floor minimal, but more important, supplying a decent amount of power comes at quite a cost and unfortunately, many people might well not be able to tell the difference in spite of what they write. In fact, I doubt whether they really do hear those differences. As Frans eloquently put it....... Complete bollocks!!!!
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Jakkal
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Post by Jakkal on Feb 17, 2015 18:42:14 GMT
Thank you Frans for all the information, it really helps.
@rabbit Ye, the music compression is the worst thing happening in the music industry now. I check all of my music with Audacity and most of the new music is brickwalled. Otherwise great albums sound awful because of that, which is really sad.
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