[SimonPac]

Am googling for "implement transient voltage protection"? Is that the correct term?

That should do it. But if the voltage remains, for example as DC because your amp output stage goes bad and gives a fixed output close to supply rail, the zeners or similar might not be able to handle the energy and would fry. If so, they might fail short and still protect the headphones. This is another situation where a series R in the amp output can help matters.

 

[SimonPac]

Ok. Do you use cross-feed?

Never tried it, is that your schematic on your avatar?

 

[SimonPac]

71dB I'm assuming you need a higher total series R to bring the '650 to critical damping compared to the '600. This because the 650 is better damped mechanically. Therefore extra 355 ohms due to the amp? That's a lot.

 

[pinnahertz]

Zeners or avalanche protectors back to back is a good idea with thin wire phones or windings not that well secured to the former.

They might not be zeners, though...standard silicon diodes have a knee voltage similar to a zener, but less sharp, at .6V to .7V, which is about right for something like the HD650. They are MUCH cheaper too!

 

[SimonPac]

They might not be zeners, though...standard silicon diodes have a knee voltage similar to a zener, but less sharp, at .6V to .7V, which is about right for something like the HD650. They are MUCH cheaper too!

0.6 volts peak to peak is too low a voltage for high impedance headphones.

 

[pinnahertz]

0.6 volts peak to peak is too low a voltage for high impedance headphones.

On the edge...103dB SPL is a tad low.

 

[71 dB]

Interesting. Thank you for the information!

What I am not understanding is how a diode in parallel with the driver will protect? How does it work with AC? Do you have a link for me that is worth reading?

(edit: Wait - 2 diodes per driver?)
(edit: What fault must occur that the diodes "work"? )

I'm not an expert of the model. I have never listened to any Beyerdynamics headphone. When you mentioned that the resistor is in fact a diode, it occurred to me it's voltage protection. Googled it and found an old thread on this forum about it with a picture. The diodes (yes, two per driver) are in series (in opposite polarities to work with AC). When the voltage exceeds a certain level, the diodes will limit the voltage. I don't know the details, but diodes "leak" when the voltage goes over 0.6-0.7 volts in forward direction and for example at 10 volts in reverse direction. Together it means limiting to 10.6 volts or about 0.2 W at 600 Ω, but I don't know the exact type of the diodes. Check out wikipedia , section "Waveform clipper".

 

[Strangelove424]

The 880 600 Ohm has a diode at the driver, not a resistor. Don't ask me why though. I do wonder why?

Beyer states that the high impedance cans have "lighter wire" and can theoretically react "faster".
Not sure how that works, because using lighter wire means one needs more?

Interesting, I wonder if that means a pair of DT880s are immune to being overdriven.

I am curious how they managed to artificially raise the electrical resistance so high with just thin wire. Even at 40 gauge, you'd need 570 feet to get 600ohms worth of resistance.

 

[71 dB]

71dB Thanks for the work put into iteration of parameters.

You are welcome!

I believe the Q and therefore damping is a function of the L, C and R of the parallel circuit.

Correct. Damping ratio = 0.5 * sqrt(L/C) / R (without electric damping, i.e. Rout = infinite)

The LC alone would be totally undamped.

In theory yes, but of course real world L and C contain some resistance so there is some damping even without R. Totally undamped resonator would "explode".

The total series R (coil plus z out) is effectively an extra parallel resistance component in the LCR circuit that brings the overall damping up. The series L is ineffective at frequencies at or near primary resonance. Is you 355 Ohms with or without the coil R?, i.e. do we need 45 Ohms or 355 Ohms amplifier output R for critical damping?

Yes, but damping goes down when Rc + Rout goes up, because the effective resistance R* increases.

Damping ratio = 0.5 * sqrt(L/C) / R*, where R* = (R*(Rout+Rc)) / (R+Rout+Rc)

The Rout = 355 Ω means output impedance alone. You have coil resistance Rc too. So, 355 Ω is what you need to add, well 330 Ω is close enough. However, remember such a large output impedance means bass boost of about 1.6 dB.

Your second '650 model seems better and the initial plot you worked from questionable. I have no reason to doubt your results, I just wondered how you did it. I was going to send you my own 650s to measure and resolve into lumped parameters

If you could sent me the impedance values of step 2, it is quite fast to calculate these parameters.

I would have thought that electromotive force on the voice coil was current-governed if the magnetic assembly stays the same. Therefore the acceleration of the coil assembly relative to the magnet would be greater for a given current with lighter wire if everything else stays the same. Lower moving mass. But then the resonant frequency would go up all other things being equal.

Yes, but faster movement means more SPL. The sound doesn't become "faster" when you turn up volume. It becomes louder. that's why the term "fast" is a bit silly in audio. Lighter wire means increased sensitivity. But it means also resonances at higher frequency which means the response doesn't go as low. So, you may want to compensate it and you are back where you started. Making moving parts light isn't a magic trick in audio. You can't break free of the laws of physics. Audio engineering is about optimazing a lot of different (partly contradictory!) parameters to avoid compromizes as much as possible.

Never tried it, is that your schematic on your avatar?

Yes, my avatar is the schematic of a basic Linkwitz-Cmoy cross-feeder. I discovered cross-feed in 2012 and it revolutionazed my headphone listening. A loudspeaker guy became a headphone guy. I should have discovered it 15 years earlier because of my education in acoustics, but I was so ignorant about headphones until one day trying to get sleep it hit me: Headphone listening without cross-feed is fundamentally wrong. If you are curious, you can read my messages at this thread:

https://www.head-fi.org/threads/to-...t-is-the-question.518925/page-3#post-13729820

71dB I'm assuming you need a higher total series R to bring the '650 to critical damping compared to the '600. This because the 650 is better damped mechanically. Therefore extra 355 ohms due to the amp? That's a lot.

151 Ω for HD 600 and 355 Ω for HD 650 so yes. However, in my opinion finetuning output impedance to achieve critical damping is pointless and I don't think overdamping is bad in audio. Underdamping is the real problem.

 

[SimonPac]

I'm not an expert of the model. I have never listened to any Beyerdynamics headphone. When you mentioned that the resistor is in fact a diode, it occurred to me it's voltage protection. Googled it and found an old thread on this forum about it with a picture. The diodes (yes, two per driver) are in series (in opposite polarities to work with AC). When the voltage exceeds a certain level, the diodes will limit the voltage. I don't know the details, but diodes "leak" when the voltage goes over 0.6-0.7 volts in forward direction and for example at 10 volts in reverse direction. Together it means limiting to 10.6 volts or about 0.2 W at 600 Ω, but I don't know the exact type of the diodes. Check out wikipedia , section "Waveform clipper".

As I'm sure 71dB knows, two back-to-back zeners conduct in both directions when you exceed a certain voltage. A small zener such as a BZY88 acts like a normal diode, i.e. it conducts in the forward direction with an exponential curve having a 'knee point' of around 600mV. Below that it passes very little current. In the reverse direction the zener is much like a normal diode, tiny leakage current only, until it reaches its breakdown voltage, where it suddenly conducts strongly. The breakdown voltage is the prime factor in the zener specification and can be from about 3V up to 100V plus. When you have two back-to-back, as in the Beyer photo, you get very little conduction until you hit about 600mV+zener breakdown voltage. Same in both directions so works as a symmetric clipper for AC.

Yes, but damping goes down when Rc + Rout goes up, because the effective resistance R* increases.

Agreed. I agree with you on everything you have said about electrical equivalent damping and LCR elements. I also trust you have made good agreement in deducing equivalent lumped elements though I haven't tried to regenerate the curves from your parameters

Yes, but faster movement means more SPL. The sound doesn't become "faster" when you turn up volume. It becomes louder. that's why the term "fast" is a bit silly in audio. Lighter wire means increased sensitivity. But it means also resonances at higher frequency which means the response doesn't go as low. So, you may want to compensate it and you are back where you started. Making moving parts light isn't a magic trick in audio. You can't break free of the laws of physics. Audio engineering is about optimazing a lot of different (partly contradictory!) parameters to avoid compromizes as much as possible.

I agree that higher maximum rate of change of driver displacement at a given frequency means louder sound on a sine wave. However I'm not convinced that a maximum d/dt on displacement has no effect on music just because a 20KHz sine wave can be generated at adequate level. In a similar way I'm not fully convinced by 22KHz brickwall filters for audio either, having heard analogue master tapes fed through various digital codecs. I recently heard the LCD4's, and they have amazing delicacy and transparency on acoustic material, though I don't know for sure why. But light moving assembly was a prime design goal. Listening it is easy to hear the progression in resolution from 650 to LCD3 to LCD4. I still enjoy my 650's and 518's though.

 

[castleofargh]

ok so your problem is with audio and what is considered audible, not with how the headphone works. following that logic, I'm guessing you disagree with recordings done using mics that roll off soon in the upper freqs?

 

[SimonPac]

ok so your problem is with audio and what is considered audible, not with how the headphone works. following that logic, I'm guessing you disagree with recordings done using mics that roll off soon in the upper freqs?

I agree with the technical analysis performed by 71dB concerning how headphones work, at least on the specifics discussed so far. I assume he is a good engineer or physicist. I am thinking about the implications of the LF over-damping issue. The LCD4s I mentioned I assume are mechanically highly over-damped and have a very light membrane and strong magnetic field. For me, if I'm right and judging by the end result, that is a good way to go. All these things are a compromise, as 71dB says. Some work better than others. Having said that, in the case of the HD650s and indeed '518s , I prefer them with some series R. I'm not totally sure why, which is why I was interested in the damping issues. I had not done the work there that 71dB has. incidentally he prefers low z out and overdamping on dynamic phones. On that, with my phones, recordings and system, I disagree, at least subjectively.

At the risk of drifting off thread, digital audio is a complex business largely because brickwall filters do not perform well in the time domain, as has been realized. An acquaintance of mine in professional recording told me 25 years ago that the first DAT machines were not transparent at all in his opinion. Opinions vary, but I agree with him. Even 192/24 is not transparent on a setup I heard. The corollary of brickwall in the frequency domain is temporal smear, although this should be less of a problem with higher sample rates.. Meridian are attempting to rectify this with MQA. There's also an interesting graph of impulse response on this page: https://www.plusmusic-us.com/index.php/technology/sla/dynamic-vs-sampling

I think the total system impulse response is of very high relevance to sound quality. The ADC/DAC system including the filtering and any rate change interpolation is a big part of that, as obviously is the headphone or speaker.

I think quality sound is less about the >20KHz bandwidth and more about abrupt frequency domain behaviour with its time domain implications.

 

[gregorio]

[1] I think the total system impulse response is of very high relevance to sound quality. [2] The ADC/DAC system including the filtering and any rate change interpolation is a big part of that, as obviously is the headphone or speaker.

1. Correct, assuming of course you only listen to recordings of impulse responses.
2. You're vastly confusing scale. A football and the sun can both be described as enormously big compared to an atom but of course, just because they are both "enormously big" doesn't mean that a football and the sun are roughly the same size! This is effectively what you've done with your statement, the ADC/DAC, filtering and SRC will all be completely audibly transparent, in fact way below any audibility threshold. So it's effectively an extremely tiny "part of that" compared to the headphone or speaker.

I suspect your confusion might be due to the article to which you linked, which is a very good article from a marketing point of view but nonsense from a factual and practical point of view!

G

 

[pinnahertz]

At the risk of drifting off thread, digital audio is a complex business largely because brickwall filters do not perform well in the time domain, as has been realized. An acquaintance of mine in professional recording told me 25 years ago that the first DAT machines were not transparent at all in his opinion. Opinions vary, but I agree with him.

As compaired to what? An analog tape?

If you want to compare anecdotes, I worked at a professional studio back when digital audio was very, very young. Once, by accident, our monitor source was the return from a digital audio recorder instead of the 2-mix bus of the console. It stayed there for a very long time and nobody, none of us, including our young hot-shot engineer with golden ears, knew the difference! We were listening, in effect to a complete record/play cycle through ADC and immedately through DAC. And that was in the days before oversampling filters, and 44.1kHz sampling frequency.

Funny thing....analog tape never fooled anyone like that.

How's that?

Even 192/24 is not transparent on a setup I heard.

Again, as compared to what? Did you have a live analog source to compare to?

The corollary of brickwall in the frequency domain is temporal smear, although this should be less of a problem with higher sample rates.. Meridian are attempting to rectify this with MQA. There's also an interesting graph of impulse response on this page: https://www.plusmusic-us.com/index.php/technology/sla/dynamic-vs-sampling

....And the marketing hype again raises it's ugly head. Are you aware that Meridian coined the term "temporal smear", and that there's not a single shred of scientific proof that it's audible?

I think the total system impulse response is of very high relevance to sound quality.

Sure...but correlation of impulse response to audibility is a total mess.

The ADC/DAC system including the filtering and any rate change interpolation is a big part of that, as obviously is the headphone or speaker.

One of these things does nothing to the reproduced sound, the other changes it radically. Care to guess which is which?

I think quality sound is less about the >20KHz bandwidth and more about abrupt frequency domain behaviour with its time domain implications.

Ever seen an impulse response, or any time-domain data on analog tape recorders? No? I have. They're a hot mess, just a different one. They are NOT free of time domain issues at high frequencies by any means. In fact, don't bother with impulse response, just try to reproduce a square wave...any frequency you like, using analog tape. Take a look at it on a scope, you'll see a very interesting wave form...but not a square wave! It'll tilt, ring and overshoot, and worse, be unstable over time.

Research into the audibility of nonlinear phase shift has shown it has to be massive...many times that of any antialiasing filter, or reconstruction filter...to be audible.

It's all what you compare to. And yes, this should pretty much drift off thread...

 

[SimonPac]

2. You're vastly confusing scale. A football and the sun can both be described as enormously big compared to an atom but of course, just because they are both "enormously big" doesn't mean that a football and the sun are roughly the same size! This is effectively what you've done with your statement, the ADC/DAC, filtering and SRC will all be completely audibly transparent, in fact way below any audibility threshold. So it's effectively an extremely tiny "part of that" compared to the headphone or speaker.

I suspect your confusion might be due to the article to which you linked, which is a very good article from a marketing point of view but nonsense from a factual and practical point of view!

I don't think I'm confused, subjectively or technically. I used to design parts of PCM based systems for telecoms. Yours is a common opinion but I think manifestly false, and widely (though not uniformly) held as such within the industry. ADC-DAC systems measure very well on some common and traditional parameters such as THD, noise and audio band response, but they don't always sound too good. The brain-ear works differently to most test gear in my opinion.

A true brickwall filter gives you sinc (x) output in the time domain with pre and post ringing of quite high amplitude at around the cutoff frequency. This is a hard fact. Research FFT windows and conjugate variables, or even QM conjugate variables for an roughly analogous situation.

Certainly on THD and FR any CD are higher res digital system will outperform any headphone or speaker.