10 ms·

Edit: they cheated, this isn't ANC and will never work as ANC. See below. This sounds pretty fundamentally different from normal ANC. A common misconception is

Edit: they cheated, this isn't ANC and will never work as ANC. See below. This sounds pretty fundamentally different from normal ANC. A common misconception is that ANC headphones "predict" the incoming noise and cancel it (and thus "they're good with constant noise but not sudden changes"). In reality they're just hybrid open-loop/closed-loop feedback systems. Specifically, the mic inside the cavity works with the speaker as a closed loop system to equalize the pressure inside in real time. This works for wavelengths smaller than the cavity size, give or take. So there is no prediction, and they don't care whether the noise is constant or not. They just have a given frequency response, and above a certain frequency all you have is passive muffling (no active cancellation). But this can't do that, because the speakers are much further away from the user. So it had to predict to get any kind of decent frequency response. I wonder how they're doing that. A lot of noise sources are uncorrelated and not predictable... Edit: They cheated. They're taking direct feeds from the source loudspeakers, and then all they have to do is track the transfer function to cancel that out from another speaker. This will never work in the real world, where sound sources are not coincident with your microphones (or lack thereof as in this case). Cool research, zero real world application as a true ANC system. I have no idea why they're selling this as ANC. This looks more like a system for computing real time room/response calibration as the user moves around their head. That's valuable, in a completely different use case.

Youden 6y ago
The actual study is here [0] and the abstract and introduction do a much better job of describing what the study is actually about. I encourage you to read at least those before you dismiss the research and its real-world applications. TL;DR: The purpose of the research is to try out a new way of figuring out which sound is going into the user's ears, so that it can be cancelled out. Current methods work for lower frequencies but not for higher frequencies. They "cheated" because they're testing one component of the system and the "cheating" component is immaterial to the research they're doing. [0]: https://www.nature.com/articles/s41598-020-77614-w https://www.nature.com/articles/s41598-020-77614-w
- marcan_42 6y ago
  They "cheated" because they did the easy part, and the "other component" has no known solution. What they're handwaving away is the real problem. The reason why current methods work for lower frequencies and not for higher frequencies is that your microphone, emitter, and ear all need to be enclosed within a volume << the minimum wavelength you can handle, to eliminate sound field and propagation time effects and let you treat the noise cancellation problem as a point problem in space. Once you put the ears in open air and the emitters away from them, the problem isn't figuring out the sound that's going into the users' ears, it's figuring it out in advance. The ANC speakers have to send the complement signal before the original signal arrives at the ears, so they both arrive coincidentally. In other words, the "missing part" to make this research work is time travel (if you use their sensing technology as the only input). Or maybe full-volume sound field sampling and characterization technology (if you don't). Neither of which exist, and which are orders of magnitude more difficult than this research. The handwave part is "The constraint of the possible locations [of microphones] would affect the control performance and this remains a topic to be further studied in the ANC community." Yes, it would completely destroy high frequency performance, which is what they were trying to achieve, in anything but lab scenarios (such as the sound sources still being loudspeakers and the microphones being directly in line between them and the user). Real world noise doesn't come from loudspeakers, it comes from all around you. Good luck using distant microphones to compute the expected sound at a user's ear in advance, with any kind of accuracy in the high frequencies. At best I imagine they could achieve adaptive cancellation for a set of slowly moving pointlike sources in an otherwise simple room, with a number of microphones greater than the number of sources. It's like RF MIMO systems. But again, this is orders of magnitude more work to implement, and real world constraints are going to kill your high frequency response. And for larger sources - forget it. You just can't characterize the transform for that. Not enough dimensions in your input data to solve for it. So anything mechanical, stuff where the noise isn't coming from a literal speaker with a 1-dimensional input signal - nope. As the uncertainty and source size grows, your high frequency response goes down the drain.
  - wegs 6y ago
    No, the missing part is not time travel. Sound comes from the skin and engines of the plane, where microphones could be located, and could potentially predict sound throughout the plane well in advance with an appropriate microphone array. You'd need a fair number of microphones, but I suspect the noise profile isn't too chaotic, which helps. You're talking about things like jet engines (which are repetitive), turbulence (which seems chaotic, but in practice, you shed vertices at a known rate), and a known cabin configuration (modulo people moving about). I suspect while the general solutions suffers from the curse of dimensionality, the specific one probably doesn't. A reasonable number of microphones, combined with decent models, could characterize the whole sound field. Speakers, located in the seats, if they knew where ears were, could play sound to cancel that. Decent microphones are $2, and what was expensive were things like microphone preamps, ADC, and the whole data pipeline. From there, we need a shit-ton of computation which really wasn't practical until... big-ass computation systems came out for the rise of ML. The problem is hard today, but definitely not impossible. I don't think I would have given that same answer a decade ago. An NVidia Titan V brings over 100 teraflops. That's a lot of cycles one can throw at trying to predict sound by my ear from sound at the skin of the airplane in real-time.
74d-fe6-2c6 6y ago
> This works for wavelengths smaller than the cavity size (A) "This" works for a wavelength < L. > and above a certain frequency all you have is passive muffling (B) "This" does not work for a frequency > F. <=> (B') "This" works for a frequency < F. --- Don't A and B' (or B) contradict each other? Because an upper limit for the wavelength equates to a lower limit for the frequency - not an upper limit. What is your background that justifies your very confident demeanour?
- ohazi 6y ago
  It's clearly a typo. Should be "This works for wavelengths larger than the cavity size." A long wavelength implies that the phase of the reference input and feedback input are similar, which is what allows you to just subtract them. Your processing delay, the distance between the speaker and the microphone, and the speed of sound are all fixed, so as the wavelength gets shorter, the relative phase difference between the two signals you're trying to subtract increases. At a certain point, you run out of phase margin and can no longer just subtract them anymore. If you were designing an ANR system, you'd probably want to put a low pass filter somewhere around this point to prevent your feedback system from oscillating.
  - 74d-fe6-2c6 6y ago
    Maybe that's why the speakers of Sony's xm series are designed to lie flat on (my) ears while Bose's leave more space but are considered slightly less efficient at noise cancelling compared to the xm's. For me ANC is by far the most important feature but having my ears pressed for longer than one hour is painful. And I tend to wear my Boses (QC35 and HP700) easily 8+ hours.
- marcan_42 6y ago
  Sorry, that was an edit typo. I originally had "frequency" but I figured that would be confusing when juxtaposed to "cavity size", took the inverse and forgot to flip the sign. My background is audio processing is one of my hobbies, I have torn down and repaired noise canceling headphones and understand how they work (and have previously talked about this on Reddit and been corroborated by an actual engineer at a company designing such headphones), I have implemented laser galvanometer closed loop feedback systems and worked with them (which is both a similar control system problem, and a different part of OP's research paper on the light path), I have attempted room and speaker response cancelation (and wrote my own DSP firmware for implementing biquad filters, which I use for my DIY living room rig), making an open source speaker calibration system is in my future project list and I've recently done a lot of thinking about this... and, most importantly, since this is neither my professional nor academic field I have no reason to overstate my claims or mislead about the importance of my work, unlike way too many academic researchers (see also: that guy who I recently called out for being a paper mill about side channel leakage methods after he successfully spun a false media story about generating Wi-Fi signals from RAM using a misleading paper title; in that case infosec is in fact one of my professional fields). I don't like to brag or anything, I'm just a curious guy who likes tinkering with (lots of) things... but if you insist on asking "what are your credentials", well, there you go.
wegs 6y ago
This comment is technically incorrect. Noise-cancelling headphones work the way OP described, and are limited to working in a cavity which is smaller than the wavelength. There are predictive noise cancelling systems too. These are commonly built into industrial equipment. You might, for example, have a fan or motor which spins at 5000 RPMs, and what do you know, it has a consistent noise profile on each of those rounds. If you cancel that noise profile, you do pretty well. These systems both reduce noise and extend equipment life by reducing vibration. They also work to higher frequencies (in part because speed-of-sound in metals is > 10x that in air, and what matters is wavelength more than frequency). There are ones which are open-loop used in HVAC systems. If I have a pipe, I can record sound on one end, and play a cancelling waveform on the other end, with an appropriate delay. I haven't followed the field, and when I looked, these were mostly expensive prototype systems (I'm not sure if they ever made it into mainstream use), but they did work. They needed to be calibrated for each HVAC, which made them impractical for most real-world applications. There are all sorts of ways to do active noise cancellation. Many of these could be implemented on airplanes. Most of these techniques were implemented decades ago, when computation was a lot more expensive. A simple feedback loop is a few opamps, capacitors, and inductors, so pretty cheap to toss into headphones with 1990's-era technology. One could do a lot more with teraflops....
- marcan_42 6y ago
  Yes, as I said, this doesn't work in the general case when your noise source is uncorrelated and not predictable. If you have a predictable noise source then you've solved the time travel problem and you can cancel it any way you want as long as you can obtain feedback from the location of the user's ears. We're talking headrest ANC here (for use cases like consumer ANC headphones), not built-in noise reduction for industrial systems.
  - wegs 6y ago
    An airplane is an industrial system.... And in either case, the design of algorithms depends on much more on context than on whether it's "consumer" or "industrial." In this case, the context looks a lot more like industrial machinery.
Solocomplex 6y ago
Some cars such as new Genesis models attempt this.

Youden 6y ago

The actual study is here [0] and the abstract and introduction do a much better job of describing what the study is actually about. I encourage you to read at least those before you dismiss the research and its real-world applications. TL;DR: The purpose of the research is to try out a new way of figuring out which sound is going into the user's ears, so that it can be cancelled out. Current methods work for lower frequencies but not for higher frequencies. They "cheated" because they're testing one component of the system and the "cheating" component is immaterial to the research they're doing. [0]: https://www.nature.com/articles/s41598-020-77614-w https://www.nature.com/articles/s41598-020-77614-w

marcan_42 6y ago
They "cheated" because they did the easy part, and the "other component" has no known solution. What they're handwaving away is the real problem. The reason why current methods work for lower frequencies and not for higher frequencies is that your microphone, emitter, and ear all need to be enclosed within a volume << the minimum wavelength you can handle, to eliminate sound field and propagation time effects and let you treat the noise cancellation problem as a point problem in space. Once you put the ears in open air and the emitters away from them, the problem isn't figuring out the sound that's going into the users' ears, it's figuring it out in advance. The ANC speakers have to send the complement signal before the original signal arrives at the ears, so they both arrive coincidentally. In other words, the "missing part" to make this research work is time travel (if you use their sensing technology as the only input). Or maybe full-volume sound field sampling and characterization technology (if you don't). Neither of which exist, and which are orders of magnitude more difficult than this research. The handwave part is "The constraint of the possible locations [of microphones] would affect the control performance and this remains a topic to be further studied in the ANC community." Yes, it would completely destroy high frequency performance, which is what they were trying to achieve, in anything but lab scenarios (such as the sound sources still being loudspeakers and the microphones being directly in line between them and the user). Real world noise doesn't come from loudspeakers, it comes from all around you. Good luck using distant microphones to compute the expected sound at a user's ear in advance, with any kind of accuracy in the high frequencies. At best I imagine they could achieve adaptive cancellation for a set of slowly moving pointlike sources in an otherwise simple room, with a number of microphones greater than the number of sources. It's like RF MIMO systems. But again, this is orders of magnitude more work to implement, and real world constraints are going to kill your high frequency response. And for larger sources - forget it. You just can't characterize the transform for that. Not enough dimensions in your input data to solve for it. So anything mechanical, stuff where the noise isn't coming from a literal speaker with a 1-dimensional input signal - nope. As the uncertainty and source size grows, your high frequency response goes down the drain.
- wegs 6y ago
  No, the missing part is not time travel. Sound comes from the skin and engines of the plane, where microphones could be located, and could potentially predict sound throughout the plane well in advance with an appropriate microphone array. You'd need a fair number of microphones, but I suspect the noise profile isn't too chaotic, which helps. You're talking about things like jet engines (which are repetitive), turbulence (which seems chaotic, but in practice, you shed vertices at a known rate), and a known cabin configuration (modulo people moving about). I suspect while the general solutions suffers from the curse of dimensionality, the specific one probably doesn't. A reasonable number of microphones, combined with decent models, could characterize the whole sound field. Speakers, located in the seats, if they knew where ears were, could play sound to cancel that. Decent microphones are $2, and what was expensive were things like microphone preamps, ADC, and the whole data pipeline. From there, we need a shit-ton of computation which really wasn't practical until... big-ass computation systems came out for the rise of ML. The problem is hard today, but definitely not impossible. I don't think I would have given that same answer a decade ago. An NVidia Titan V brings over 100 teraflops. That's a lot of cycles one can throw at trying to predict sound by my ear from sound at the skin of the airplane in real-time.

74d-fe6-2c6 6y ago

> This works for wavelengths smaller than the cavity size (A) "This" works for a wavelength < L. > and above a certain frequency all you have is passive muffling (B) "This" does not work for a frequency > F. <=> (B') "This" works for a frequency < F. --- Don't A and B' (or B) contradict each other? Because an upper limit for the wavelength equates to a lower limit for the frequency - not an upper limit. What is your background that justifies your very confident demeanour?

ohazi 6y ago
It's clearly a typo. Should be "This works for wavelengths larger than the cavity size." A long wavelength implies that the phase of the reference input and feedback input are similar, which is what allows you to just subtract them. Your processing delay, the distance between the speaker and the microphone, and the speed of sound are all fixed, so as the wavelength gets shorter, the relative phase difference between the two signals you're trying to subtract increases. At a certain point, you run out of phase margin and can no longer just subtract them anymore. If you were designing an ANR system, you'd probably want to put a low pass filter somewhere around this point to prevent your feedback system from oscillating.
- 74d-fe6-2c6 6y ago
  Maybe that's why the speakers of Sony's xm series are designed to lie flat on (my) ears while Bose's leave more space but are considered slightly less efficient at noise cancelling compared to the xm's. For me ANC is by far the most important feature but having my ears pressed for longer than one hour is painful. And I tend to wear my Boses (QC35 and HP700) easily 8+ hours.
marcan_42 6y ago
Sorry, that was an edit typo. I originally had "frequency" but I figured that would be confusing when juxtaposed to "cavity size", took the inverse and forgot to flip the sign. My background is audio processing is one of my hobbies, I have torn down and repaired noise canceling headphones and understand how they work (and have previously talked about this on Reddit and been corroborated by an actual engineer at a company designing such headphones), I have implemented laser galvanometer closed loop feedback systems and worked with them (which is both a similar control system problem, and a different part of OP's research paper on the light path), I have attempted room and speaker response cancelation (and wrote my own DSP firmware for implementing biquad filters, which I use for my DIY living room rig), making an open source speaker calibration system is in my future project list and I've recently done a lot of thinking about this... and, most importantly, since this is neither my professional nor academic field I have no reason to overstate my claims or mislead about the importance of my work, unlike way too many academic researchers (see also: that guy who I recently called out for being a paper mill about side channel leakage methods after he successfully spun a false media story about generating Wi-Fi signals from RAM using a misleading paper title; in that case infosec is in fact one of my professional fields). I don't like to brag or anything, I'm just a curious guy who likes tinkering with (lots of) things... but if you insist on asking "what are your credentials", well, there you go.

wegs 6y ago

This comment is technically incorrect. Noise-cancelling headphones work the way OP described, and are limited to working in a cavity which is smaller than the wavelength. There are predictive noise cancelling systems too. These are commonly built into industrial equipment. You might, for example, have a fan or motor which spins at 5000 RPMs, and what do you know, it has a consistent noise profile on each of those rounds. If you cancel that noise profile, you do pretty well. These systems both reduce noise and extend equipment life by reducing vibration. They also work to higher frequencies (in part because speed-of-sound in metals is > 10x that in air, and what matters is wavelength more than frequency). There are ones which are open-loop used in HVAC systems. If I have a pipe, I can record sound on one end, and play a cancelling waveform on the other end, with an appropriate delay. I haven't followed the field, and when I looked, these were mostly expensive prototype systems (I'm not sure if they ever made it into mainstream use), but they did work. They needed to be calibrated for each HVAC, which made them impractical for most real-world applications. There are all sorts of ways to do active noise cancellation. Many of these could be implemented on airplanes. Most of these techniques were implemented decades ago, when computation was a lot more expensive. A simple feedback loop is a few opamps, capacitors, and inductors, so pretty cheap to toss into headphones with 1990's-era technology. One could do a lot more with teraflops....

marcan_42 6y ago
Yes, as I said, this doesn't work in the general case when your noise source is uncorrelated and not predictable. If you have a predictable noise source then you've solved the time travel problem and you can cancel it any way you want as long as you can obtain feedback from the location of the user's ears. We're talking headrest ANC here (for use cases like consumer ANC headphones), not built-in noise reduction for industrial systems.
- wegs 6y ago
  An airplane is an industrial system.... And in either case, the design of algorithms depends on much more on context than on whether it's "consumer" or "industrial." In this case, the context looks a lot more like industrial machinery.

Solocomplex 6y ago

Some cars such as new Genesis models attempt this.