I agree that your high-pass filters are an application of masking principles. It would be challenging, especially for an untrained person but even for someone who is trained, to perceive enabling/disabling of those filters in the context of the full mix. Instead we solo to isolate the track being adjusted so the change is easily audible -- i.e. no longer masked by the other tracks -- and adjust the filtering anticipating the counter-intuitive but artistically desirable effect when the filtered track is blended back into the mix.
Despite that, your hostility to the skeptics illustrates that you and I still aren't really on the same page (though I don't doubt you produce compelling art, don't question your techniques, and agree 100% about ensuring that aesthetic gestures survive the transition to imperfect end-user environments).
To me, "objective accuracy" is a worthy and desirable intermediate goal for sound artists, a tool which can be leveraged for creative ends the same as a baker understanding food chemistry or an animator understanding Newtonian physics or a modern origami folder understanding programming. However, chasing "objective accuracy" means understanding the wildly non-linear, even bizarre human perceptual mechanism -- which is more of a pattern recognition machine and definitely not a measurement device -- in addition to all the strangeness of acoustics.
The problem with audiophiles is that their approach to "accuracy" is unscientific and doomed from the start. The skeptics on the other hand, have the right idea -- but those who have not done deep study often fail to appreciate just how difficult the problem is and how far off their intuitions are.
I've had some wonderful conversations with individuals whose perspectives are similar to mine, but it still seems impossible to find a hospitable public forum.
When I say "objective", it definitely needs air quotes, because the well-meaning skeptics love to toss it around for some decidedly non-objective opinions. My pet example is quoting "THD" as if it were a useful figure. It's not. The advantage of THD isn't that it is meaningful, but rather that it's easy to measure. Run a 1khz sine wave into the front of the amp, tap a resistive load on the back of the amp, measure the difference. Manufacturers like it because it makes for a competitive-sounding spec. But it has approximately zero to do with how an amplifier behaves in the face of complex musical signals, driving complex speaker loads.
When it comes to recording, my "objective" reality is the sound I hear in the room before it ever hits a microphone. But what comes out of the other end of the microphone is already deeply subjective. The objective ability of hi-fi systems to reproduce that subjective, colored microphone signal (plus whatever processing is involved downstream) is about as objective as they get. But the recording itself? Outside the world of purely electronic music, it's a subjective experience.
I play acoustic guitar every day, listen to unamplified singers/musicians (other than me) at least weekly, play electric guitar at least weekly, gig regularly. These direct, objective experiences of the natural sounds of instruments color my interpretation of both magical-thinking audiophiles and pseudo-scientific skeptics. I know intimately what the natural sound is, I know intimately what the recorded/reproduced sound is, and I know how the entire recording process works, from miking to mixing to mastering. Laws and sausages. The audiophiles and skeptics are both missing that perspective.
So, as a recording/mixing engineer, as a producer, I'm looking not to deliver audiophile accuracy, but rather to deliver the intended intellectual and emotional experience of the music. What feeling is the artist trying to convey? How do I manipulate the sonics to emphasize the musician's intent, as expressed through the lyrics, the composition/arrangement, and the natural tones of their instruments? That's the fun part, for me.
(As an aside, I'm currently working on a particular song of my own, destined for my long-delayed solo album. I wrote the song based on a nightmare I had in great distress. A few years back, I suffered an illness that left me nearly unable to speak, much less sing, and my voice is permanently damaged (I have surgery a few times a year on my vocal cords to keep speaking and not die of suffocation). I recorded the song right after writing it, just a couple of weeks before my first surgery, with a near-useless voice. "My broken voice won't fill the air / I know that you don't really care / You may not need my song but I need to sing it anyway". Fitting with the broken voice is the melody repeated on a piano that hasn't been tuned in over 50 years and has been flooded multiple times, so it's, um, colorful. To both the audiophile and objective mindset, neither the voice nor the piano were worth recording at all. But as an artist trying to communicate an emotional experience, they're vital.)
When a transistor is off, it’s off. Then it takes a certain input voltage to get it going. Then there’s a varying curve to full output.
In contrast, fully switching on or off “gets rid of the curve” and tranforms the problem into one of timing, and one of a modulation scheme for a series of on or off states. In that respect I contend that it is an useful thought to consider transistors’ behavior (in an amp) as more linear and controlled if used in only the on or the off state. The output quality of Class-D amps in relation to price, size, complexity, and power, and cost suggests to me that the gains are not only in power efficiency.
Class D amps do leave the operation of transistors in their linear domain on the table, and do gain efficiency by that. Overall linearity as a control mechanism is something I stand by saying is increased. You may present an alternate take of course.
It is :) But it’s also only a contradiction on the surface of the words. The amplifier is quite linear even though the transistors aren’t used in what’s called their “linear region”, but I’d say they’re still used in a linear fashion.
I think you are confused.
To get a linear amplifier, you operate your output transistors in a state of Saturation. This gives a linear transfer function to the drain based on the Gate to source voltage and you use a method of choice to hold the source constant. This is what a class A amplifier does. An AB amplifier has 2 transistors to extended the range of inputs on the gate for which the output is linear at the cost some distortion when switching between the A and B (hi and low side) transistors.
A class D amplifier uses a smaller amplifier internally (which can be a single transistor or more complicated) that is driven open loop so that it is either hard off or hard on but very fast and efficient. The hard on state is known as full compression because the output power will not change with input power. This is purely non-linear behavior and can be very easily verified by measuring the high IMD components it generates. Class D amplifiers employ significant filtering to suppress these non-linear terms and allow you to recover the signal. Mathematically, they violate the linearity condition of F(ax) = aF(x)
Also you should reread the article you learned from. It directly contradicts you.
It’s not about me; The “you”-ing is incredibly offputting.
We are simply and obviously applying the word “linear” a little differently. I am aware that it is a specific term for certain behavior / a certain operating mode of transistors. I’m referring to it in another wider context on purpose. As another example “linear” in “linear algebra” isn’t about saturation or gates. I’ve gone through this before in this thread and believe my perspective is now pretty complete and clear here and won’t engage further.
I don't think that your first paragraph is an honest assessment of the issue. People who buy tube gear are absolutely not looking for "flat, uncolored sound".
It does, but plenty of albums have been recorded on 4 track machines to 1/4" tape.
The real difference in quality is probably mostly in having a machine that's properly setup and calibrated, with a good transport to minimize flutter, etc.
We had some older B&Ms at our house back in the day (also an audiophile middle class household). As a musician myself, I'd say they're worth the money if you can afford it. (I think they were around $8000 each or something).
there seems to be too many different thinghs mixed up, "professional amplifier" "high end audio" and "class D" are three different worlds. It's like a Humvee a Maybach and a Prius in the same vehicle.
Thanks, those are the kind of chips I was looking for. And S/PDIF to I2S transceivers are also available, so they could be used with common consumer audio equipment.
I'm assuming you mean instruments and microphones excepted, because many high-end recording microphones cost $1000 and up. Yes, there are legends of a few albums being recorded with only a few SM57 (a $100 mic), but they're the exception - and most such albums aren't generally regarded as audiophile-quality.
But regardless, even just the input transformer on most mixing consoles costs at least $100. And many extremely high-end consoles use discrete opamps which can get really expensive. There's also a lot of tube-based outboard gear in studios.
Sure, you could build a mixing desk with only $2 opamps and no transformers, no tubes, etc. But that would sound bad. Unlike the audiophile world, recording isn't about doing everything possible to prevent distortion - if it was, all recording would be done with calibrated measurement microphones and other various lab equipment.
So how can distortion be good in recording if it's bad in reproduction? A few reasons. IM distortion, for starters - distorting every channel individually is going to sound much different (and in most cases, better) than distorting the mix of all the channels.
But also, in recording, the distortion is finely tuned to the exact song, by someone who gets paid hundreds of dollars an hour because they are very good at it (and yes, good at getting it to sound even better on a variety of different systems). A recording engineer can use exactly whatever mic preamp they think will sound best for the song. Yes, audiophiles do tend to adjust tiny settings for each song, but not to the degree that recording engineers do - no audiophile is going to own 20 different amps and switch them out for different parts of different songs (or at least not on a regular basis).
Recording engineers can also do so on an individual channel basis - use one preamp for the vocals, a different one for the large-diaphragm condenser over the piano, another different one for a small-diaphragm condenser pointed right at the piano hammers, etc. Unless they're using Dolby Atmos or some other multichannel format, an audiophile doesn't even have that option, no matter how much they enjoy tweaking minor settings.
So no, the signal path is certainly not way less than $100 worth of components - and it wouldn't sound nearly as good if it was.
Yes I exclude instruments and transducers. Those are where the investment should be if anything.
You have no idea what you are talking about regarding signal chain. Discrete opamps are inferior on every possible measurement. This is total nonsense. Utter rubbish.
I'll never forget my wakeup call on the differences of sound between amplifiers. I had been listening to headphones on my Techniqs receiver, a well regarded unit with a THD of 0.04%. For some reason I plugged the headphones into my Nakamichi cassette recorder instead, and was blown away by how much better they sounded. Until then I had thought that amplifiers were a solved problem and the inaccuracies of transducers like headphones would mask the tiny differences between them. I never expected to be wrong.
I needed a good set of headphones a few weeks ago, as my work environment has changed and I have more background noise than before. Instead of trying to find something audiophile (I have Grados already) I went to Guitar Center. They had a huge selection and in 30 minutes of trying out different models I had something I was happy with and could leave in an unlocked drawer.
https://www.audiosciencereview.com has measurements of various DACs. Tl;dr: the topping D50 ($250 MSRP) measures really well, and for a combo amp&dac unit the topping dx7s (currently on massdrop for $360) measures really well. Just a heads up, that site is biased towards objective measurements, the Toppings measure really well, so the site is biased towards them. A lot.
According to that site, Schiit stuff measures really bad.
> Unless you are using really shitty speakers, room acoustics trumps everything else pretty quickly.
I agree completely, but emphasising that point tends to lead to flame-wars in most places on the internet. And it's such an imprecise subject to debate because everyone's room is different. Some rooms can be atrocious, while others can be unintentionally excellent. Some rooms can be easy to improve, some require knowledge and experience in order to improve. Some only require repositioning, some require treatment products that can cost a lot more than many enthusiasts realise.
My favourite video that demonstrates the effectiveness of acoustic treatment:
TPA3116 can do about double the power of the TA2024 in the TP31 at about the same distortion.
The TPA3250 from the article, and the TPA3251/TPA3255 that have higher power from the same range are in a different class to those IMO, with much less distortion, and are just ridiculous value for money.
That's interesting, thank you. I got the TP31 as it's got both headphone and speaker out plus it's own DAC. I've had it a few years so will have a look at the newer ones. Thanks.
I have a Corda HA2 mk3 and it still gives everything I need: enough punch to drive 250ohm headphones and a crossfeed to prevent fatigue when using them hours every day.
Maybe in 50 years when it breaks I'll get a new one :)
Yes, you definitely don't want a bluetooth combo. Also, you don't want to buy the 100W version that combines two TPA3116 chips. These models are using a special sync pin/line, but when I bought this model, the manufacturers screwed up the design, it doesn't sync properly, and it's super noisy.
41 comments
[ 3.1 ms ] story [ 101 ms ] threadDespite that, your hostility to the skeptics illustrates that you and I still aren't really on the same page (though I don't doubt you produce compelling art, don't question your techniques, and agree 100% about ensuring that aesthetic gestures survive the transition to imperfect end-user environments).
To me, "objective accuracy" is a worthy and desirable intermediate goal for sound artists, a tool which can be leveraged for creative ends the same as a baker understanding food chemistry or an animator understanding Newtonian physics or a modern origami folder understanding programming. However, chasing "objective accuracy" means understanding the wildly non-linear, even bizarre human perceptual mechanism -- which is more of a pattern recognition machine and definitely not a measurement device -- in addition to all the strangeness of acoustics.
The problem with audiophiles is that their approach to "accuracy" is unscientific and doomed from the start. The skeptics on the other hand, have the right idea -- but those who have not done deep study often fail to appreciate just how difficult the problem is and how far off their intuitions are.
I've had some wonderful conversations with individuals whose perspectives are similar to mine, but it still seems impossible to find a hospitable public forum.
When it comes to recording, my "objective" reality is the sound I hear in the room before it ever hits a microphone. But what comes out of the other end of the microphone is already deeply subjective. The objective ability of hi-fi systems to reproduce that subjective, colored microphone signal (plus whatever processing is involved downstream) is about as objective as they get. But the recording itself? Outside the world of purely electronic music, it's a subjective experience.
I play acoustic guitar every day, listen to unamplified singers/musicians (other than me) at least weekly, play electric guitar at least weekly, gig regularly. These direct, objective experiences of the natural sounds of instruments color my interpretation of both magical-thinking audiophiles and pseudo-scientific skeptics. I know intimately what the natural sound is, I know intimately what the recorded/reproduced sound is, and I know how the entire recording process works, from miking to mixing to mastering. Laws and sausages. The audiophiles and skeptics are both missing that perspective.
So, as a recording/mixing engineer, as a producer, I'm looking not to deliver audiophile accuracy, but rather to deliver the intended intellectual and emotional experience of the music. What feeling is the artist trying to convey? How do I manipulate the sonics to emphasize the musician's intent, as expressed through the lyrics, the composition/arrangement, and the natural tones of their instruments? That's the fun part, for me.
(As an aside, I'm currently working on a particular song of my own, destined for my long-delayed solo album. I wrote the song based on a nightmare I had in great distress. A few years back, I suffered an illness that left me nearly unable to speak, much less sing, and my voice is permanently damaged (I have surgery a few times a year on my vocal cords to keep speaking and not die of suffocation). I recorded the song right after writing it, just a couple of weeks before my first surgery, with a near-useless voice. "My broken voice won't fill the air / I know that you don't really care / You may not need my song but I need to sing it anyway". Fitting with the broken voice is the melody repeated on a piano that hasn't been tuned in over 50 years and has been flooded multiple times, so it's, um, colorful. To both the audiophile and objective mindset, neither the voice nor the piano were worth recording at all. But as an artist trying to communicate an emotional experience, they're vital.)
When a transistor is off, it’s off. Then it takes a certain input voltage to get it going. Then there’s a varying curve to full output.
In contrast, fully switching on or off “gets rid of the curve” and tranforms the problem into one of timing, and one of a modulation scheme for a series of on or off states. In that respect I contend that it is an useful thought to consider transistors’ behavior (in an amp) as more linear and controlled if used in only the on or the off state. The output quality of Class-D amps in relation to price, size, complexity, and power, and cost suggests to me that the gains are not only in power efficiency.
Class D amps do leave the operation of transistors in their linear domain on the table, and do gain efficiency by that. Overall linearity as a control mechanism is something I stand by saying is increased. You may present an alternate take of course.
A class D amplifier uses a smaller amplifier internally (which can be a single transistor or more complicated) that is driven open loop so that it is either hard off or hard on but very fast and efficient. The hard on state is known as full compression because the output power will not change with input power. This is purely non-linear behavior and can be very easily verified by measuring the high IMD components it generates. Class D amplifiers employ significant filtering to suppress these non-linear terms and allow you to recover the signal. Mathematically, they violate the linearity condition of F(ax) = aF(x)
Also you should reread the article you learned from. It directly contradicts you.
We are simply and obviously applying the word “linear” a little differently. I am aware that it is a specific term for certain behavior / a certain operating mode of transistors. I’m referring to it in another wider context on purpose. As another example “linear” in “linear algebra” isn’t about saturation or gates. I’ve gone through this before in this thread and believe my perspective is now pretty complete and clear here and won’t engage further.
It's important for the room to be quiet, as the sound is very faint and easily masked, but it's there.
It's possible that Bluetooth headphones could lose it because of compression, or cheap DAC in the sound card simply ignores least-significant bits.
It looks like this particular board has tiny SMT components, which make this sort of thing a pain in the but.
The real difference in quality is probably mostly in having a machine that's properly setup and calibrated, with a good transport to minimize flutter, etc.
Thank you :)
But regardless, even just the input transformer on most mixing consoles costs at least $100. And many extremely high-end consoles use discrete opamps which can get really expensive. There's also a lot of tube-based outboard gear in studios.
Sure, you could build a mixing desk with only $2 opamps and no transformers, no tubes, etc. But that would sound bad. Unlike the audiophile world, recording isn't about doing everything possible to prevent distortion - if it was, all recording would be done with calibrated measurement microphones and other various lab equipment.
So how can distortion be good in recording if it's bad in reproduction? A few reasons. IM distortion, for starters - distorting every channel individually is going to sound much different (and in most cases, better) than distorting the mix of all the channels.
But also, in recording, the distortion is finely tuned to the exact song, by someone who gets paid hundreds of dollars an hour because they are very good at it (and yes, good at getting it to sound even better on a variety of different systems). A recording engineer can use exactly whatever mic preamp they think will sound best for the song. Yes, audiophiles do tend to adjust tiny settings for each song, but not to the degree that recording engineers do - no audiophile is going to own 20 different amps and switch them out for different parts of different songs (or at least not on a regular basis).
Recording engineers can also do so on an individual channel basis - use one preamp for the vocals, a different one for the large-diaphragm condenser over the piano, another different one for a small-diaphragm condenser pointed right at the piano hammers, etc. Unless they're using Dolby Atmos or some other multichannel format, an audiophile doesn't even have that option, no matter how much they enjoy tweaking minor settings.
So no, the signal path is certainly not way less than $100 worth of components - and it wouldn't sound nearly as good if it was.
You have no idea what you are talking about regarding signal chain. Discrete opamps are inferior on every possible measurement. This is total nonsense. Utter rubbish.
Do you know what noise figure is?
Do you know what CMRR is?
Do you know what PSRR is?
According to that site, Schiit stuff measures really bad.
List of all reviews: https://www.audiosciencereview.com/forum/index.php?threads/m...
I agree completely, but emphasising that point tends to lead to flame-wars in most places on the internet. And it's such an imprecise subject to debate because everyone's room is different. Some rooms can be atrocious, while others can be unintentionally excellent. Some rooms can be easy to improve, some require knowledge and experience in order to improve. Some only require repositioning, some require treatment products that can cost a lot more than many enthusiasts realise.
My favourite video that demonstrates the effectiveness of acoustic treatment:
https://www.youtube.com/watch?v=cp56A6TcL1E
The TPA3250 from the article, and the TPA3251/TPA3255 that have higher power from the same range are in a different class to those IMO, with much less distortion, and are just ridiculous value for money.
Maybe in 50 years when it breaks I'll get a new one :)