24/192 music downloads are silly(xiph.org)
xiph.org
24/192 music downloads are silly
http://xiph.org/~xiphmont/demo/neil-young.html
424 comments
I agree with the silliness of 192kHz, but not 24-bits. Here is why:
In typical PCM recordings, like CDs, mid-range frequencies (e.g. 1kHz to 4kHz) are recorded with lower amplitudes because our ears are more sensitive to them.
Sampling theory is correct and 16-bits can reproduce any waveform with ~100dB of range, however, in a complex waveform consisting of low, mid and high frequencies, the mid- and hi-range frequencies quite simply get shortchanged.
Imagine a recording of a bass sinusoid and a mid-range sinusoid of equal volume. It might use e.g. 10 bits to store the bass and only 6 to store the high frequencies. (2^10sin(200wt)+2^6sin(4000wt)). That means the resolution of the high frequencies is less than the lower frequencies. When the volume of those frequencies changes dynamically, the high frequencies' amplitudes are more quantized. That is quite simply why 16-bits are not enough.
This is similar to the problem with storing waveforms unprecompensated on vinyl. The precompensation makes up for the non-uniformity of the medium. It could be done with 16-bit digital as well. Or alternatively, larger sample sizes like 24 can be used.
I haven't A/B tested this. The A/B test in the article compares CD with SACD. SACD isn't PCM, so its artifacts are going to be totally different from 24-bit PCM.
In typical PCM recordings, like CDs, mid-range frequencies (e.g. 1kHz to 4kHz) are recorded with lower amplitudes because our ears are more sensitive to them.
Sampling theory is correct and 16-bits can reproduce any waveform with ~100dB of range, however, in a complex waveform consisting of low, mid and high frequencies, the mid- and hi-range frequencies quite simply get shortchanged.
Imagine a recording of a bass sinusoid and a mid-range sinusoid of equal volume. It might use e.g. 10 bits to store the bass and only 6 to store the high frequencies. (2^10sin(200wt)+2^6sin(4000wt)). That means the resolution of the high frequencies is less than the lower frequencies. When the volume of those frequencies changes dynamically, the high frequencies' amplitudes are more quantized. That is quite simply why 16-bits are not enough.
This is similar to the problem with storing waveforms unprecompensated on vinyl. The precompensation makes up for the non-uniformity of the medium. It could be done with 16-bit digital as well. Or alternatively, larger sample sizes like 24 can be used.
I haven't A/B tested this. The A/B test in the article compares CD with SACD. SACD isn't PCM, so its artifacts are going to be totally different from 24-bit PCM.
There's a special irony to the fact that this high fidelity audio format is being promoted by Neil Young. Young's a rock musician. He's been around loud noises (e.g. rock concerts) most of his life. He's also 69 years old. Our ability to hear high frequencies decreases dramatically with age and exposure [1]. If anyone were able to discriminate 24/192 from 16/44.1, it sure as heck wouldn't be an elderly rock musician.
[1]: http://www.patient.co.uk/health/presbyacusis-hearing-loss-of...
[1]: http://www.patient.co.uk/health/presbyacusis-hearing-loss-of...
The industry wants to be able to sell you something "better" and 24/192 is clearly bigger and therefore better than 16/48.
This is the same reason I'm convinced we're going to get 8k phone displays someday.
If the recording industry wants to sell me a "platinum" version of recordings, what I'd really like to have is different mastering of an album: at least one for noisy environments like the car, and one for higher-quality environments like my home theater. If you're familiar with "The Loudness Wars", this is a reaction to that. NiN tried to do this with their "audiophile" mix of Hesitation Marks (although a lot of people think they did not succeed, http://www.metal-fi.com/terrible-lie/ )
On the other hand, I don't need to buy any new equipment to support that, so the equipment guys aren't going to be happy. I don't know if there's any silver bullet for them--if there is a hypothetical advancement that would cause me to upgrade my system, I can't envision it.
This is the same reason I'm convinced we're going to get 8k phone displays someday.
If the recording industry wants to sell me a "platinum" version of recordings, what I'd really like to have is different mastering of an album: at least one for noisy environments like the car, and one for higher-quality environments like my home theater. If you're familiar with "The Loudness Wars", this is a reaction to that. NiN tried to do this with their "audiophile" mix of Hesitation Marks (although a lot of people think they did not succeed, http://www.metal-fi.com/terrible-lie/ )
On the other hand, I don't need to buy any new equipment to support that, so the equipment guys aren't going to be happy. I don't know if there's any silver bullet for them--if there is a hypothetical advancement that would cause me to upgrade my system, I can't envision it.
No one can see X-rays (or infrared, or ultraviolet, or microwaves). It doesn't matter how much a person believes he can. Retinas simply don't have the sensory hardware.
The author seems to have stumbled into a poor example, as a recent study shows that humans can indeed see infrared light using an unexpected process. Should we read anything into the audio case from this? Probably not, but it's a sign that even those who are sure they are right because they have science on their side should retain some degree of openmindedness.
The author seems to have stumbled into a poor example, as a recent study shows that humans can indeed see infrared light using an unexpected process. Should we read anything into the audio case from this? Probably not, but it's a sign that even those who are sure they are right because they have science on their side should retain some degree of openmindedness.
Human infrared vision is triggered by two-photon chromophore isomerization
This study resolves a long-standing question about the
ability of humans to perceive near infrared radiation (IR)
and identifies a mechanism driving human IR vision. A few
previous reports and our expanded psychophysical studies
here reveal that humans can detect IR at wavelengths longer
than 1,000 nm and perceive it as visible light, a finding
that has not received a satisfactory physical explanation.
We show that IR light activates photoreceptors through a
nonlinear optical process. IR light also caused
photoisomerization of purified pigments and a model
chromophore compound. These observations are consistent
with our quantum mechanical model for the energetics of
two-photon activation of rhodopsin. Thus, humans can
perceive IR light via two-photon isomerization of visual
pigment chromophores.
Anyone here who has not already seen Xiph's Digital Show and Tell ( http://xiph.org/video/vid2.shtml ) should do themselves a favor and sit down for a watch. It makes sense of a lot of mysteries and misconceptions around digital audio.
You can't simultaneously say that you're dithering to represent low amplitudes while also saying you're keeping enough samples to capture all audible frequencies. Dithering doesn't create resolution out of nowhere, it sacrifices temporal resolution for amplitude resolution. It's also bad for compression (hence why modern video encodes are done at 10-bit even for output to 8-bit devices), and worse if you want to use a source as the basis for further work (i.e. a remix). If you want to store your signal in a simple, convenient way, and not have to carefully tweak the levels for each individual recording, 16 bits isn't quite enough. And as the article admits, extra resolution certainly can't hurt; worst case is the extra bits are thrown away.
Also 44.1KHz is a pain to do in realtime (there's not enough headroom to really filter out the higher frequencies without damaging the 20KHz response), meaning you need a separate mastering step which is inconvenient and frankly unnecessary. 48KHz is a much more sensible standard to work with.
192KHz may be dumb, but 48KHz/24-bit is perfectly sensible. It gives you fewer ways to make mistakes than CD-quality (44.1KHz/16-bit), and at some point the extra space is worth that, particularly since it may well compress better than a dithered 16-bit signal.
Also 44.1KHz is a pain to do in realtime (there's not enough headroom to really filter out the higher frequencies without damaging the 20KHz response), meaning you need a separate mastering step which is inconvenient and frankly unnecessary. 48KHz is a much more sensible standard to work with.
192KHz may be dumb, but 48KHz/24-bit is perfectly sensible. It gives you fewer ways to make mistakes than CD-quality (44.1KHz/16-bit), and at some point the extra space is worth that, particularly since it may well compress better than a dithered 16-bit signal.
What's so funny is to see yet another occurrence of basically "because Nyquist" yet fails to address that Nyquist only holds true over infinite time. Over a window of any finite length perfect reproduction is NOT guaranteed.
This paper, http://www.academia.edu/8412078/Is_The_Nyquist_Rate_Enough, for one, refutes this.
More reading material:
http://www.wescottdesign.com/articles/Sampling/sampling.pdf
This paper, http://www.academia.edu/8412078/Is_The_Nyquist_Rate_Enough, for one, refutes this.
More reading material:
http://www.wescottdesign.com/articles/Sampling/sampling.pdf
The author is largely right. It all comes down to the master. And the lossless format. The numbers here don't matter, what matters is will the masters be better for 24/96..24/192..32/384? If yes, prefer the 24/96. Don't prefer it for numbers' sake.
Personally I can't ABX anything above 192kbps, lame vs flac. Very occasionally a pre-echo reveals itself, but hardly ever. V2 is fine, 16/44.1 is fine, V0 & 320 are indistinguishable from flac, but buying anything but lossless is a crap deal.
The 'volume knob' trick, once you notice it, makes it basically impossible to objectively compare 2 headphones, or two amps. I can't match within a 1/4 dB with my fingers.
Personally I can't ABX anything above 192kbps, lame vs flac. Very occasionally a pre-echo reveals itself, but hardly ever. V2 is fine, 16/44.1 is fine, V0 & 320 are indistinguishable from flac, but buying anything but lossless is a crap deal.
The 'volume knob' trick, once you notice it, makes it basically impossible to objectively compare 2 headphones, or two amps. I can't match within a 1/4 dB with my fingers.
Humans almost universally consider louder audio to sound better,
and .2dB is enough to establish this preference.The argument is fine for listening to a finished product but those absolutes in the post have some hidden assumptions that don't hold in all real-world scenarios.
That probably sounds like BS. Hear me out.
A common practice for DJs is to use special decks to DJ from iPods, or other digital sources (eg. regular turntables with timecode vinyls operating a PC.) DJing involves playback speed adjustment when there is a BPM difference between the two tracks during a transition. 192kHz is overkill for basic DJing and you'd probably be fine in the vast majority of cases with 48kHz, but if the DJ is a turntablist (scratch etc.), you want all you can get when going from zero to target speed—which is happening constantly. It sounds awful when consumer-grade¹ audio is used. As for the ultrasonics, filtering is the answer in this scenario. It may be quite a good thing for these DJs (and their fans) that Apple is doing this. (I'm not under the impression that this is why they're doing it.)
A lot of the music I listen to uses samples that are played at something like half speed, or tuned down at any rate. I tend to tune samples down by about a fifth myself. Point being: a lot of detail that would otherwise be present with higher samplerate source material goes missing. It doesn't help that tuning down like this dumps a good portion of the low end.
There are also neat ways of exploiting nonlinearities from ultrasonic sources, which I use, but that's harder to describe.
¹not meant as a derogatory remark.
That probably sounds like BS. Hear me out.
A common practice for DJs is to use special decks to DJ from iPods, or other digital sources (eg. regular turntables with timecode vinyls operating a PC.) DJing involves playback speed adjustment when there is a BPM difference between the two tracks during a transition. 192kHz is overkill for basic DJing and you'd probably be fine in the vast majority of cases with 48kHz, but if the DJ is a turntablist (scratch etc.), you want all you can get when going from zero to target speed—which is happening constantly. It sounds awful when consumer-grade¹ audio is used. As for the ultrasonics, filtering is the answer in this scenario. It may be quite a good thing for these DJs (and their fans) that Apple is doing this. (I'm not under the impression that this is why they're doing it.)
A lot of the music I listen to uses samples that are played at something like half speed, or tuned down at any rate. I tend to tune samples down by about a fifth myself. Point being: a lot of detail that would otherwise be present with higher samplerate source material goes missing. It doesn't help that tuning down like this dumps a good portion of the low end.
There are also neat ways of exploiting nonlinearities from ultrasonic sources, which I use, but that's harder to describe.
¹not meant as a derogatory remark.
Such "studio" formats don't make sense for end-user listening, but they make sense as inputs to further processing, mixing, etc. Having a pile of headroom in frequency and amplitude means that after further processing you can subsequently output a sensible 16-bit 48kHz file without loss. If you start with a 16-bit 48kHz file and then do a pile of processing, you won't necessarily preserve the same degree of quality.
I can't emphasise enough how good the linked presentation video is...
http://xiph.org/video/vid2.shtml
That is how you do a presentation!
http://xiph.org/video/vid2.shtml
That is how you do a presentation!
The summary here is to just rip everything losslessly and then to go ahead and use 44.1/16 since it's actually better in some ways and not worse in others.
I think most effective way to improve sound quality is to get a good DAC as close as possible to the output. Headphone amps with integrated DACs do wonders for little money. When possible go for XLR on the last mile to the speakers (good neutral studio speakers) to cancel out distortion from external electromagnetic pulses. To me differential, or balanced signalling is still the most clever, yet so simple, analog information preserving method I ever heard of.
http://en.wikipedia.org/wiki/Differential_signaling
Also; of cause 24bit increases the resolution of the signals amplitude, that's exactly what it does by definition, that it sets the noise floor is only a function of that. To stress the visual analogy: it's like looking at 16bit images with banding and then on 24 bit images. Of cause the effect is not very pronounced but it's there. Especially if you have a high dynamic recording of a soundscape but you would like to zoom in into a certain volume range like that of the human voice it's good to have that extra resolution not only in a studio environment. Think about it like developing a picture from RAW into JPEG to stress this analogy again, this doesn't make a difference for compressed pop music but for uncompressed recordings of live performances with analog instruments or natural soundscapes it does. You can then choose at which volume range your most at home for listening and do the compression or not.
I like to think about 24bit audio like having access to the RAW files of images, it doesn't matter in most cases, and most likely you should leave the mixing to professionals artists for the intended effect, but it also enables you to experience the sound in many more different ways.
Also; of cause 24bit increases the resolution of the signals amplitude, that's exactly what it does by definition, that it sets the noise floor is only a function of that. To stress the visual analogy: it's like looking at 16bit images with banding and then on 24 bit images. Of cause the effect is not very pronounced but it's there. Especially if you have a high dynamic recording of a soundscape but you would like to zoom in into a certain volume range like that of the human voice it's good to have that extra resolution not only in a studio environment. Think about it like developing a picture from RAW into JPEG to stress this analogy again, this doesn't make a difference for compressed pop music but for uncompressed recordings of live performances with analog instruments or natural soundscapes it does. You can then choose at which volume range your most at home for listening and do the compression or not.
I like to think about 24bit audio like having access to the RAW files of images, it doesn't matter in most cases, and most likely you should leave the mixing to professionals artists for the intended effect, but it also enables you to experience the sound in many more different ways.
I have worked a lot with audio programming and while most of the article is right in what is known, it could be wrong at what we don't know. Technical myopia(you see too much of what you know but you don't see the big picture).
We are using HDR in pictures even when the eye could not differentiate between HDR colors because it adapts to the general luminosity of the image but CAUTION the general luminosity level affects lots of biological cycles like the circadian rhythm.
In normal pictures we discard this info, but this info is enough for a person to differentiate a picture from a real image in the real world.
Also natural sensors work different than our eyes and ears.
This difference makes HDR a necessary because we can see a huge shadow along with an illuminated area at the same time because our cones in the retina adapt locally, but if you make a single picture we can only choose to picture the bright areas, making shadows too dark, or choose the shadows making the bright areas appear too white.
Artificial sensors linearize over a fixed level. Nature sensors are really continuous exponential, even touch.
The same happens with our ears. So you are doing an spectral analysis using an arithmetic frequency decomposer called the FFT?
Well, sorry to burst your bubble but the cloclea frequency analysis runs circles around anything we have. It does a geometric analysis, and also does it locally. Using just a single tone as an example is a fallacy. Most real sounds are not a single tone but changes in lots of frequencies at the same time.
The law of diminishing returns applies to sound and video, we have a good enough experience for what we want to do, but by no means it is perfect.
Ask John Carmack that is trying to create an immersive experience. Sound is one of the big problems. Yes, you can understand the sound, but you know that it is not the real sound you hear in the real world.
We are using HDR in pictures even when the eye could not differentiate between HDR colors because it adapts to the general luminosity of the image but CAUTION the general luminosity level affects lots of biological cycles like the circadian rhythm.
In normal pictures we discard this info, but this info is enough for a person to differentiate a picture from a real image in the real world.
Also natural sensors work different than our eyes and ears.
This difference makes HDR a necessary because we can see a huge shadow along with an illuminated area at the same time because our cones in the retina adapt locally, but if you make a single picture we can only choose to picture the bright areas, making shadows too dark, or choose the shadows making the bright areas appear too white.
Artificial sensors linearize over a fixed level. Nature sensors are really continuous exponential, even touch.
The same happens with our ears. So you are doing an spectral analysis using an arithmetic frequency decomposer called the FFT?
Well, sorry to burst your bubble but the cloclea frequency analysis runs circles around anything we have. It does a geometric analysis, and also does it locally. Using just a single tone as an example is a fallacy. Most real sounds are not a single tone but changes in lots of frequencies at the same time.
The law of diminishing returns applies to sound and video, we have a good enough experience for what we want to do, but by no means it is perfect.
Ask John Carmack that is trying to create an immersive experience. Sound is one of the big problems. Yes, you can understand the sound, but you know that it is not the real sound you hear in the real world.
The most interesting development in terms of surround audio may be the Oculus Rift. A number of people were working on combining high-quality VR HMD head-tracking with headphones and a spatial audio system http://www.mee.tcd.ie/thrive/ http://www.technologyreview.com/news/527826/microsofts-3-d-a... . Then Oculus licensed a spatial audio system itself http://www.roadtovr.com/oculus-rift-dk2-realsense-3d-audio-p... and announced that the first consumer version of the Rift would have integrated headphones.
Not sure about this one. I'm pretty sure at one point I could hear the difference between 44.1 and 48 khz sampling. I agree 192 is overkill, but 44.1 is just above the Nyquist limit. At that resolution, the top breathy harmonics of a piccolo are only getting 2-3 samples per cycle, which seems to leave room for some possible aliasing if you are not 100% sure about your filters. So why not just go crazy and throw 4X samples at the problem, eliminating any question of proper anti-aliasing?
As to # of bits, the issue there is the wide dynamic range of music, and the fact that our ears can adjust to this wide range. Probably you would get the same effect as 24 or 32 bits with the right dynamic range adjustments, but then we'll have to argue about which algorithm is "right". A surfeit of bits just makes the question go away.
As to # of bits, the issue there is the wide dynamic range of music, and the fact that our ears can adjust to this wide range. Probably you would get the same effect as 24 or 32 bits with the right dynamic range adjustments, but then we'll have to argue about which algorithm is "right". A surfeit of bits just makes the question go away.
So I appreciate everyone's point of view and applaud using an empirical approach, for those of you who share the author's point of view, but I disagree unfortunately. For those of us who have worked in DSP, either using it or implementing new things with it, there's s highly mathematical reason to record the source of you audio with a higher sample rate than what the author suggests is a generous maximum.
It has to do with waveforms and how continuous they are. So, for starters, true, if you have a perfectly continuous wave form, at 22K, then your sample rate must be at least 44K. In fact, with sample rate of 44k you can perfectly discretize a continuous wave form, like a sine wave.
Does you see the problem with this? Sounds are not always continuous! If you look at the waveform of a violin, distorted guitar, cymbal, etc... They're very jagged. To effectively approximate these analog waveforms as a finite set of sums you need a much higher sample rate. It makes s HUGE difference, trust me.
So basically, technically speaking, 44K works just five if you only listen to music made by orjan pipes and penny whistles, but most sounds are very complicated, and to be properly captured you actually need a higher sample rate. It's simple and mathematical. Also, this whole "44.1K is all you need and if you don't agree with me then you're dumb and don't understand math" ra ra ra has been going on all over the Internet for ages, and while u appreciate the motivations that people may have, it gets a little annoying. Basically, instead of immediately jumping to the conclusion that people's ears are wrong, maybe the more patient and mindful approach is to ask oneself, "why does my mathematical knowledge of a subject fall short of explaining what many people seem to experience?".
Note: everything I said was regarding the source of capturing a sound. There's an entire science behind compression and all that sauce.
Also Stanford's DSP lectures (available online) explain this much more indepth, albeit abstractly.
It has to do with waveforms and how continuous they are. So, for starters, true, if you have a perfectly continuous wave form, at 22K, then your sample rate must be at least 44K. In fact, with sample rate of 44k you can perfectly discretize a continuous wave form, like a sine wave.
Does you see the problem with this? Sounds are not always continuous! If you look at the waveform of a violin, distorted guitar, cymbal, etc... They're very jagged. To effectively approximate these analog waveforms as a finite set of sums you need a much higher sample rate. It makes s HUGE difference, trust me.
So basically, technically speaking, 44K works just five if you only listen to music made by orjan pipes and penny whistles, but most sounds are very complicated, and to be properly captured you actually need a higher sample rate. It's simple and mathematical. Also, this whole "44.1K is all you need and if you don't agree with me then you're dumb and don't understand math" ra ra ra has been going on all over the Internet for ages, and while u appreciate the motivations that people may have, it gets a little annoying. Basically, instead of immediately jumping to the conclusion that people's ears are wrong, maybe the more patient and mindful approach is to ask oneself, "why does my mathematical knowledge of a subject fall short of explaining what many people seem to experience?".
Note: everything I said was regarding the source of capturing a sound. There's an entire science behind compression and all that sauce.
Also Stanford's DSP lectures (available online) explain this much more indepth, albeit abstractly.
I remember when CDs first came out and Neil Young was very critical of their sound, and I believe he was entirely justified in his criticism. When they first came out CDs were Record Company's poor stepchildren, and they were treated very poorly. I believe the engineers mastering CDs were given tapes that were several generations away from the original master tapes, and I suspect they may have even been already equalized for vinyl. No wonder audiophiles preferred the sound of vinyl over the sound of early CDs.
Nowadays CDs are made from digital recordings to digital masters to digital discs (DDD). I love Neil Young's music, but his grasp of the details of digital audio recording and reproduction is not particularly strong.
Nowadays CDs are made from digital recordings to digital masters to digital discs (DDD). I love Neil Young's music, but his grasp of the details of digital audio recording and reproduction is not particularly strong.
I wonder how many people see 24/192 and think "24-bit, 192kbps" sound instead of "24-bit, 192khz sound", getting the units wrong.
MP3s are (often/traditionally) 128kbps, so 192kbps would be better, and 24 is more than 16, so it too must be better.
For the people who do get the units, you still have the 'more is better' problem. We've finally gotten past the point of everyone trying to make ultra-compact 30MP cameras because consumers have realized that their current camera is 'good enough' and that more MP doesn't always make the picture sharper.
Could this just be the same thing in the sound world?
MP3s are (often/traditionally) 128kbps, so 192kbps would be better, and 24 is more than 16, so it too must be better.
For the people who do get the units, you still have the 'more is better' problem. We've finally gotten past the point of everyone trying to make ultra-compact 30MP cameras because consumers have realized that their current camera is 'good enough' and that more MP doesn't always make the picture sharper.
Could this just be the same thing in the sound world?
TL;DR, but skimming through an interesting problem comes to mind: The OP "orthogonalizes" the question of the sample accuracy's (24 vs 16 bit) and sample rate's (48 vs 192 kHz) impact on quality, answering one independent of the other. But even with my limited background in mathematics it's quite obvious that that approach is not entirely correct: the Nyqvist theorem only really applies when you have infinite sample accuracy. It would be interesting to see an analysis of how the two interact; i.e. how the discretization error impacts the highest representable frequency.
Fascinating article. Is there any reason to believe that upsampling an audio file should produce a better analog signal than playing it at its native sample rate?
I find that playing high quality mp3's through an upsampling DSP filter and a 24/192 DAC seems to produce a better listening experience. (As the article points out, this could be due to confirmation bias, or the filter making the music a tiny bit louder.) Intuitively it makes sense to me that the DAC sending signals twice as frequently to the headphones would produce a smoother analog signal, but is that actually true?
I find that playing high quality mp3's through an upsampling DSP filter and a 24/192 DAC seems to produce a better listening experience. (As the article points out, this could be due to confirmation bias, or the filter making the music a tiny bit louder.) Intuitively it makes sense to me that the DAC sending signals twice as frequently to the headphones would produce a smoother analog signal, but is that actually true?
Completely unrelated question:
"Can you see the Apple Remote's LED flash when you press a button [4]? No? [Some other remotes] may be just barely visible in complete blackness with dark-adjusted eyes [5]. All would be blindingly, painfully bright if they were well inside the visible spectrum."
Ok, what happens if you point a remote at an infrared camera?
A little research on the FLIR website shows cameras with a range of 7500-13500nm, which would be 22-40THz (?); the remotes are 300-380THz. Sigh.
"Can you see the Apple Remote's LED flash when you press a button [4]? No? [Some other remotes] may be just barely visible in complete blackness with dark-adjusted eyes [5]. All would be blindingly, painfully bright if they were well inside the visible spectrum."
Ok, what happens if you point a remote at an infrared camera?
A little research on the FLIR website shows cameras with a range of 7500-13500nm, which would be 22-40THz (?); the remotes are 300-380THz. Sigh.
This is just so anthropocentric, of cause the true selfless audiophile wants his music to be experienced in it's full natural range by bats and dogs just as well.
Monty's example files made me realize that my computer's sound chip is probably kind of lo-fi.
No, I couldn't hear anything in my headphones when I played the 30Khz/33Khz tones. In comparison, I heard something after it stopped. Somehow it was quieter during the playback.
Does it make any sense that my computer is better at making no audible sound when it's asked to play a loud inaudible sound, than when it's being asked to play nothing?
No, I couldn't hear anything in my headphones when I played the 30Khz/33Khz tones. In comparison, I heard something after it stopped. Somehow it was quieter during the playback.
Does it make any sense that my computer is better at making no audible sound when it's asked to play a loud inaudible sound, than when it's being asked to play nothing?
Some related comments: https://xiphmont.livejournal.com/63490.html
I am sorry, but the authors' examples and parallels doesn't make much sense to me.
First of all, what does sample rate has even remotely to do with hearing range?
If we are comparing this to visual information, perhaps it would be best to draw parallels with video. Namely, sample rate would be equal to frames per second and bit depth to colour depth. In both areas we are still seeing improvements from TV manufacturers (100Hz TVs) and gamers are still racing to get better fps so it looks more realistic.
The question then would be, why 25 fps is the minimum average human can withstand? What are the limits here and can it look more realistic for certain people with more frames per second.
After all, this race for fidelity is just attempt to get more accurate representation of analog sound, which opens different problem - absolute all of todays music is electronic - produced from samples and samples above 44KHz are simply not available. You may find some vinyl rips of more classical music (which does explain the phenomenon why so many rock stars would start listening to classic music - after getting rich and dumping money into expensive sound systems they'd discover elegance beyond distortion).
Saying all this fidelity does not matter is kind of equivalent of throwing away all the vinyls (although transistors play a role here too).
First of all, what does sample rate has even remotely to do with hearing range?
If we are comparing this to visual information, perhaps it would be best to draw parallels with video. Namely, sample rate would be equal to frames per second and bit depth to colour depth. In both areas we are still seeing improvements from TV manufacturers (100Hz TVs) and gamers are still racing to get better fps so it looks more realistic.
The question then would be, why 25 fps is the minimum average human can withstand? What are the limits here and can it look more realistic for certain people with more frames per second.
After all, this race for fidelity is just attempt to get more accurate representation of analog sound, which opens different problem - absolute all of todays music is electronic - produced from samples and samples above 44KHz are simply not available. You may find some vinyl rips of more classical music (which does explain the phenomenon why so many rock stars would start listening to classic music - after getting rich and dumping money into expensive sound systems they'd discover elegance beyond distortion).
Saying all this fidelity does not matter is kind of equivalent of throwing away all the vinyls (although transistors play a role here too).
Should this have [2012] added to it?
One of my favorite is the noise harvester by PS audio.
https://www.youtube.com/watch?v=cJGUSHBuE_0
I could easily imagine the CEO of this company sell eternal life elixir in the 19th century wild wild west.
https://www.youtube.com/watch?v=cJGUSHBuE_0
I could easily imagine the CEO of this company sell eternal life elixir in the 19th century wild wild west.
Is it possible this was just a move to get advertising space in audio files, like with devices that use ultrasonic frequencies to trigger events in devices? (For example, http://lisnr.com/)
I even transcoded some 24/192 FLAC Pink Floyd I had lying around and made him do a double blind test to show him that he'd prefer the slightly louder song every time, even if the louder song was 192kbps vs the FLAC. He did. He still doesn't believe me.
He still thinks he can hear the difference between FLAC and MP3 to this day. He works as a sound engineer now.
I don't think any amount of reasoning will make some people change their minds. Some people buy $500 wooden knobs to make their volume pots sound better. (or was that a hoax? i can't tell anymore)