>Don't get me wrong, differences between 16-bit and 8-bit are clearly audible, and are demonstrated on my Dynamic Range, Dithering and Noise Shaping page. However, because contemporary popular music has such a limited dynamic range, these differences become subtle, if not inaudible, when tested on it.
From the summary thing above the blind test. They're aware of this, this test was mostly done for irony (as also explained in the blurb.)
Oh boy, this test is going to ruffle some feathers. However, the choice of audio clip ensures that you can't tell the difference easily.
It's known that the only difference between well-prepared files of different bit depths is the noise floor. Basically, in any PCM audio file, you can expect noise of about 1 ULP. So in a 16-bit file you get a noise floor of -90 dB, and in an 8-bit file it's at -42 dB. This is the ONLY DIFFERENCE.
People who are here listening for distortion artifacts aren't going to find any, and they're going to be surprised. That's because you can always choose to introduce noise instead of distortion artifacts when you do the conversion. The noise is actually added on purpose, it's "dithering noise", and it's not there to mask the distortion, but to entirely replace it.
The math works like this. Take a high-resolution input, say, 16-bit. You're converting it to 8-bit. If you just convert, you'll introduce a distortion signal of about 1ULP. However, if you introduce 1ULP of uncorrelated rectangular noise, the distortion signal is now also uncorrelated noise, which is less objectionable than correlated noise. It's also harder to hear.
Then, how do you tell the difference between 16-bit and 8-bit audio files? You can only do it if there's a quiet part somewhere in there. As soon as there's a quiet part, you can hear a "hissing" sound in the 8-bit file at -42 dB. If you're listening to the middle of a rock song like this, the hissing sound is going to be buried in the mix. Heck, a typical guitar amp will already be putting out uncorrelated noise.
The fade out at the end makes it possible to identify it. I couldn’t do it perfectly (also played over my laptop speakers because I’m lazy), but I still got 8/10.
Oh thanks for that. I can now pass the test with 10/10 correctness. There's a noticeably louder noise floor in the 8bit sample that is really clear between the samples.
Note also that many people (me, for example, and most folks over 40 in general) can't hear a -42dB white noise layered on top of an audible signal at all. The pipes just don't pick it up over the general noise in the receiver "circuit".
10/10, but I had to listen closely. Sennheiser 650s & external DAC/amp, quiet room, hearing is still good to ~16khz.
The primary place I picked out to hear this not the quiet parts, but rather a hiss above 10khz, which was audible even at the beginning - and particularly at the beginning. It's almost mistakable for brighter cymbals, if you're not listening closely. It also changes in perceptibility over the duration of the sample.
I tested myself using that same trace: I got 10/10 with the sound of the cymbals, then confirming with the fadeout at the end.
This is with HD-598's with 16 feet of cheapo headphone extension cords, loud old refrigerator running in tiled studio apartment, and motherboard built-in sound output.
10/10 using HD215s plugged into whatever cheapo onboard DAC this laptop has. But like you I did know what to listen for, having done a signals engineering course in the deep dark past.
I scored 10/10 in the test, but only because I experimented with this myself after I read Christopher Montgomery's article in 2012: As explained there, with active dithering, bit-depth and signal-to-noise level are interchangeable -- a signal with a small bit depth has a small signal-to-noise ratio. So one can clearly hear a subtle noise in the 8-bit samples, especially in the fade-out at the end.
In my experiments I found that for almost anything, that is not orchestral music, and that is newer than 1990, 8-bit sample depth are enough. That was really depressing.
By the way, a nice jest is to tell some random bystander that you're going to present them "8-bit" music and then play the 8-bit version of a song that was a victim of the loudness war. Supposedly, the 8-bit times were better than most people think. ;-)
When you do experiments you must understand your experimental design enough to understand exactly what you're testing and what conclusions can be drawn from that. If you do A/B testing of 8-bit/16-bit (which is a good idea) you have to understand you're doing the test through your current audio hardware.
The whole point of the Pono player is to have higher quality hardwhere everywhere, including pre-amps and DACs so you have a chance to hear subtle differences.
If you do this test through your laptop speakers you really might not be able to tell the difference. If you'd like to "debunk" the Pono player then do this test through a Pono player pushing music through quality studio headphones.
Making things worse they're using Neil Young's Rockin In The Free World which contains, get this, large amounts of harmonic distortion to begin with. The pre-existing harmonic distortion will only serve to mask any distortion and fidelity loss from truncating to 8-bits.
If you don't own high-fidelity music equipment then this test is like proving high definition television is impossible because you can't tell the difference between a standard and high definition signals on your standard definition TV.
> The whole point of the Pono player is to have higher quality hardwhere everywhere, including pre-amps and DACs so you have a chance to hear subtle differences.
Yes, hardware quality might make a difference in fidelity. Still, 16 bit CD-quality audio is enough. The marketing of the Pono player with 24bit/196kHz is just nonsense.
And of course any listening test requires quality hardware to make sure it can actually reproduce the test-signal correctly.
> The pre-existing harmonic distortion will only serve to mask any distortion and fidelity loss from truncating to 8bit.
With dithering, smaller sample depths only decrease the signal to noise ratio. There are no distortions. The sample is chosen "badly" (but that was by design) because it contains little to no dynamics.
Dithering basically raises the noise-floor by trading harmonic distortions for random noise at the cost of the last bit of information (for the uninitiated dithering means you randomly flip the last bit to create a soft truncation.) It's "noise" which isn't technically harmonic distortion but that's kind of splitting hairs isn't it?
I have no patience for articles like this because instead of setting the record straight it just adds more pseudoscience to the mix. Their "experiment" doesn't prove what they claim it proves.
Is 24-bit useful for playback? Yes it is. Do you need it for properly mastered audio? Absolutely not. But there are 10,000 home studios across the country who have music files at 24-bits and a large chunk of those engineers don't know how to master audio. Is a raw 24-bit track better than a poorly mastered 16-bit track? It is to me.
Is 192kHz useful for anything? Probably not. Does it hurt anything? Maybe a little. I would expect a quality music player to go up to 96 anyhow so 192 isn't so bad. Similar reasons as before.
Does this reflect badly on the Pono player? Not really. Over-engineering isn't a bad thing in my book.
> It's "noise" which isn't technically harmonic distortion but that's kind of splitting hairs isn't it?
That's not splitting hairs. Harmonic distortion and uncorrelated noise are completely different from a mathematical stand point. They also sound audibly different to my ears.
Merely saying that "24-bit is useful" or "it is better to you" does not mean that you can actually hear the difference, and you include no empirical justifications for why you believe this to be true. Perhaps I could help you conduct a scientific experiment at home?
24 bits has more information. It's useful for mixing and in cases where things haven't been properly gain-staged, an engineer had an off day, a track hasn't been mastered, etc. Seems pretty basic to me.
It probably sounds basic to you because you still aren't doing the math, and you still aren't doing the experiments.
The whole point of 24-bit is to give you flexibility when the singer decides that he's going to whisper one verse from three feet away and then shove the mic down his throat while he screams the chorus. Or you can be sloppy when you set the preamp gain—just give yourself enough headroom, and you'll adjust the levels later. Notice that I said "later", not "never". Even the most amateurish home engineer with a hangover the size of Texas is going to adjust the levels of different tracks in the mix. Once you've done the mixdown, the levels are "reasonable", and you can listen at home on 16-bit system. You won't hear the difference.
No mastering, no compression. 16-bit audio is still enough, once you've mixed a song.
What math exactly would you like me to do? 16 bits gives a theoretical maximum range of 96dB while the human ear can hear over 130dB. Good headphones can handle over 100dB signal to noise.
Maybe you should save your grandiose lectures for somebody who doesn't know enough to see through it. There's no reason to assume levels will be at a "reasonable" level and there's no reason to think a high-quality portable music player should only play back 16 bits. That's absurd. Should consumer level music players stop at 16? Sure. I want better than that but if you want to listen to mp3s on your phone nobody's stopping you.
I'm left wondering how much of this anti-Pono talk is shilling on behalf of smartphone manufacturers. I need 24 bit playback in roughly the same way I need a car that does 0-60 in under 3.5 seconds - I like things that are over-engineered so I can geek out on how awesome they are. But thanks to all the killjoys we can't even get excited about the first high quality portable music player because some amateur sound engineers want to show off they once read a blog post on Shannon-Nyquist theory.
130 dB dynamic range is unreasonable. It's not like having a car that can go 0-60 in under 3.5 seconds, it's more like having a furnace that you can set to keep your house at 50°C. 130 dB SPL is downright bad for you. It will result in pain and permanent hearing loss. 130 dB SPL is very close to OSHA limits for instantaneous (not sustained) volumes, and it is physically painful to experience.
Can you elaborate? What makes you think that people are shilling for smartphone companies?
Everything you say is true and it's clear that you understand these issues a lot better than Neil Young himself appears to.
That's not sarcasm. Superior components such as a better power supply, pre-amp, and DAC can make a big difference. Something like the Audioquest Dragonfly or some of Schiit's low-end gear can make a difference that can be amazing in some instances.
But in every interview I've ever seen, Neil Young focuses on the high resolution audio files which do not make a difference that human beings can hear. He has also, in general, been saying kooky things about digital audio for decades now, ever since CDs became popular.
> The pre-existing harmonic distortion will only serve to mask any distortion and fidelity loss from truncating to 8-bits.
This is blatantly incorrect. The 8-bit conversion is not truncation, it is dithering. Dithering to 8-bit does not introduce any distortion at all, whatsoever, of any kind. If you don't understand the mechanisms and the science of how bit depths work, then you're going to come to false conclusions, like the conclusion that there's any point to the Pono player at all. We're not talking about just "subtle differences" here. In order for it to be even theoretically possible to hear the difference between 16-bit and 24-bit audio, you have to bring your audio system into a quiet room and then crank the volume levels well above into the threshold at which you can damage your ears, and even then, you still won't be able to tell the difference with the most dynamic music.
So, suppose you have a quiet room in your house, with an ambient noise level of about 30 dB. If you raise the 16-bit audio level so the noise floor is above 20 dB, then the peaks are going to be well into the 120 dB range. That's like having a symphony orchestra in the room with you, at the very peak of their performance, with all the instruments playing at once. If you've ever listened to a symphony orchestra, you know that the background noise is NOT 30 dB, but somewhat higher. So even at the peak of a symphony, your CD recording should still be able to reproduce the various unwelcome bits of noise that the musicians produce (stomachs gurgling, breathing, shuffling in their chairs, etc.)
Here is my take on dithering, with online demos : http://www.audiocheck.net/audiotests_dithering.php (still 8-bit though, because at 16-bit, it is barely noticeable, and it won't serve educational purposes very well)
As I ceded in another comment, dithering trades harmonic distortion for noise.
> In order for it to be even theoretically possible to hear the difference between 16-bit and 24-bit audio,
Only for professionally mastered audio which is not a safe assumption in this day and age. If some home engineer recorded a track with too much headroom and you get the 24-bit track you're fine, at 16-bits you have a problem.
I would love a music player where I could play tracks at 88kHz/24-bit because that's what most music is during the mixing process and then an audio engineer can give you the raw version of what they're working with without having to deal with issues of headroom, downsampling and dithering.
24-bit audio doesn't hurt anything other than file size and it has real uses, be it remix culture or just high-quality unmastered music.
You haven't done the math. Even home engineers won't have recordings with noise floors below -90 dBFS. That's just absurd. Double-blind tests have shown that trained listeners in ideal listening environments with high-end, calibrated equipment can't tell the difference between 44.1/16 and 88.2/24.
Audio engineers don't have to "deal" with the issues of downsampling and dithering. The DAW just does it. It's a solved problem. It's not even a button you have to press, it's all automatically set up for you these days. We know what algorithms to use: band-limited interpolation with dithering, possibly combined with noise shaping.
Headroom is a different issue, but even the sloppiest home engineer is going to take a look at the levels at some point. If they don't, they can just check the "normalize" checkbox when they bounce. Or they can just ignore it and leave it checked. They're still not going to give you files with noise floors below -90 dBFS, and therefore, there's still no point in giving you a 24-bit file.
Even for remixes, you're not getting any benefit, since the noise floor is above -90 dBFS anyway.
However... you want to work on a project together? Let's keep things 24-bit until the final mixdown.
I disagree that these things are, in the real world, "solved problems" and I'd rather have an unmastered raw track at 88/24 than a poorly mastered track at 44/16.
I don't disagree that CD quality basically maxes out the ear's natural capabilities, I don't think it's that easy to do.
For a remix I'd rather get a 24-bit stream than a 16-bit stream (that's really 15 bits) and has to be padded up to 24 anyway.
Can you hear the difference between 16-bit audio with dither and 16-bit audio without dither?
Most people can. Now consider - that difference is created by adding a noise signal which is more than 90dB down compared to the maximum possible level.
By all reasonable expectations that difference should be completely inaudible under normal listening conditions.
But the effect it has isn't inaudible at all.
When you understand why, then you'll understand the difference between peer-reviewed and objectively tested psychoacoustic theory, and hand-waving about numbers.
You'll also understand why it's trivially easy to tell amplifiers and converters apart even when they have distortion products well below -90dB.
That aside - you're making the usual mistake of confusing dynamic range with resolution.
What's the effective bit resolution of a -48dB signal on a 16 bit system?
What's the resolution of the same signal on a 24dB system with the same output level?
What's the minimum number of bits needed to make quantisation noise inaudible? (Clue: rather more than 8.)
I'm honestly confused here, because you claim to be disagreeing with me, but when I read the content of your post, it sounds like you actually agree with me?
When I was talking about hearing the difference between 16-bit and 24-bit, I was assuming that we dithered our audio. You can't hear white noise at -90 dBFS in typical listening conditions. You'd have to be in a quiet room with the volume turned way up, and you'd have to have very low-noise equipment.
The effective resolution of a -48dB signal on a 16-bit system will depend on that signal's bandwidth. If you don't understand that part, then you don't understand the math.
It makes no difference if you're sampling a sine wave or broad-band noise - the effective resolution stays the same, because it's solely dependent on quantisation error, not on signal bandwidth.
The latter depends on sample rate not of sample resolution.
If you're making mistakes like that, it's not a brilliant idea to tell people who write DSP code and have designed audio hardware that they don't understand the math.
The other point still stands. If 16-bit resolution is already good enough to represent signals without audible distortion, why does it need dither to sound acceptable, while 24-bit audio doesn't?
I'm still waiting for anyone who believes 16-bit recording is perfect to explain why the industry bothered to invent a clearly audible conditioning process for signals that are supposed to be ideal already.
Let's not resort to comparing credentials here. For the record, I've designed and built audio hardware, and I'm the author of a sample-rate conversion library, which does SIMD band-limited sample rate conversion, and I also wrote the accompanying test suite. I'm not just some dude who read a blog post about audio.
Let's talk about bandwidth. If you have a pure sine wave and want to measure its amplitude, you can do a DFT on your signal and measure the appropriate bin. Let's assume that the sine wave does land in one particular bin. If your data is 16-bit with dithering, the dithering and quantization will add noise to all of the bins, but the noise will be equally divided. As you increase the length of the sample that you're analyzing, the bandwidth of each bin decreases, and the amount of noise in each bin decreases as well. However, the signal will always be concentrated in that one bin.
So, as you decrease the bandwidth, the quantization noise decreases as well. This is equivalent to saying that you have increased resolution.
I know this is counterintuitive. However, this is the foundation of how most modern ADCs work. It's called delta-sigma modulation, and it uses a low-resolution ADC internally to derive a high-resolution digital output. It's also been used in DACs. For an extreme example, look at DSD, which gives high-resolution outputs using a 1-bit signal.
The argument that "if 16 bits is enough, why do we need dithering" is kind of pointless, because we don't use 16-bit audio without dithering. It's like asking, "if this amplifier is good enough, why does it use negative feedback?" The answer of course is that negative feedback increases the linearity and flattens the response of the amplifier, and makes it less sensitive to variations in manufacturing and temperature.
10/10 using the fade-out at the end. Choice of clip is a poor example of the benefits of higher sample depth, because the dynamics are relatively constant and there are a lot of noisy overtones.
10/10 using the fade out, with cheap ass gaming headphones, and not knowing the first goddamn thing about sound, bitrates, what to listen for, or anything in this thread, except seeing your comment first.
The fade outs just seemed fuzzy.
Edit: I also have no idea if this goes for or against audiophiles. Maybe both? I know they're largely ridiculed.
Depends on the kind of audiophile you're talking about I guess. I think the easily identified difference shows that 8bit just isn't enough, and audiophiles that say bit depth matters are validated. But 24bit just isn't necessary for consumer audio. Plenty of audiophiles will insist on it still, though. They're the crazy ones
Using the fade out makes it trivial to detect. Listening closely with headphones it's fairly easy (i.e. get 9+ right) to hear it in the first couple of seconds and throughout the song as well. I guess there's a reason we use 16 bits. Now I want to try it with 24 bits, maybe with a song that has more quiet parts in it.
Edit: The other test using Gangnam Style is much harder to detect (for me), I scored 6/10 on my first attempt (and I won't try again, hearing that riff 10 times in a row is plenty): http://www.audiocheck.net/blindtests_16vs8bit.php
Yeah. The whole thing about these tests is that it's like testing whether you can tell a solid black box in an 8bit or 16bit png. Yeah there cases where you'll see the difference and ones that you won't. Why pick such limited examples, when it effectively undercuts _any_ value the test might have in terms of people's perception of bit depth?
Same here, but as I listened to more samples, I started hearing the noise throughout the sample, a swishing sound mixed in with the music. $25 headphones here (Sennheiser HD202).
This is a terrible source material for this kind of test. The biggest audible difference between 8 bit and 16 bit, assuming the same sample rate, is the noise floor. But, this source material is LOUD throughout, which masks the noise floor. It is in dynamic material that bit depth makes an audible difference. The prior listening test used Gangnam Style, which is also loud and lacking in dynamics that would reveal audible distortion due to low bit depth.
The thing is that I agree with Monty on the (lack of) need for 24/196 audio for consumers. My degree is in audio, and I (mostly) understand the physics of the thing (and have done tons of blind listening experiments in controlled environments using professional equipment, as well as using high quality measurement tools for spectrum analysis, etc.). But, this particular test isn't really a useful proof that we don't need 24 bit and 196k. It only proves that loud material can mask noise, even at 8 bits.
Using a more dynamic Neil Young song would have been a better choice. Maybe Old Man, or similar. Acoustic guitar is good for allowing one to discern audio equipment flaws. It is commonly the source material people use when trying to rate good pieces of equipment that have very small differences across a very broad price spectrum (microphones, preamps, etc.).
Nonetheless, I got 8/10 right using crappy ear buds in a noisy place, which isn't much better than random chance. I'm pretty confident I'd be able to recognize 8 bit vs 16 bit on any reasonably dynamic source played on good equipment in a good listening environment. But, I also know I can't recognize 16 bit vs 24 bit (I've tested), and I know Neil Young can't either. Neil Young was around for the era when ADC/DAC quality was very poor; it actually was true that many early digital recordings were inferior to the very high end analog recordings of the time. And, it's even true that when recording, mixing, and mastering, it is useful to work at higher bit depths (because there's a lot of summing, raising the noise floor each time; though that may just mean you need to process at 24 or 32 bits, rather than actually record at higher depths).
It seems obvious, to me, that the reason most music sounds like shit is because of the loudness wars (dynamics reduction), lossy compression (cramming a lot of music down tiny network pipes), and the low quality of most people's listening equipment and environment. Those factors utterly dwarf the noise floor of 16 bit audio, and completely nullify any frequency advantages of higher bit rates (even if we could hear them).
In short: Pono is elitist bullshit, as most audiophile stuff is.
If your hearing's good north of ~12khz, and you have a system that can reproduce highs well, there's a hiss on the cymbals in this sample which can be used to identify the low bitrate version - despite the high loudness overall.
> Acoustic guitar is good for allowing one to discern audio equipment flaws. It is commonly the source material people use when trying to rate good pieces of equipment.
My go-to track for testing, recommended if rock is your preference, is Muse's "Undisclosed Desires". The beginning of the second verse has a "hidden" bit you will only hear on good headphones. First time I heard that song on some Grado SR80s I literally jumped from being spooked.
I tested myself using my MacBook Pro's speakers. 7/10 correct.
I'm a musician and have a large CD, vinyl record, and digital audio collection. Honestly the knots some people twist themselves into over formats amuse me. I tend to focus on the music and the rhythm rather than the recording quality, at least above a minimum quality level. I prefer vinyl just because I do. :-)
[edit] the test page suggested my results were close to random, and I agree. I couldn't really tell a difference through the laptop speakers. It felt like I was guessing for each sample.
I did 10/10, but that's because I knew exactly what to look for, and how to listen to it. I didn't use my expensive open headphones, but some cheaper closed headphones that passively block external noise, and I used a far louder volume than usual.
Of course the source is specially selected to make the test hard. With music with real dynamics, that hasn't been compressed to death, the test becomes trivial. 8 bit gives you 48dB of SNR, that is lower than vinyl and tape.
However lots of music is meant to play on the radio, and not to be too dynamic since that would just get lost. The clip is likely less compressed than a considerable portion of radio play, which in turn is likely how most recorded music is listened too, even today.
I didn't bother with the test, because the 16 bit version of the audio clip sounds bad. It has some of the qualities of a demo that had been recorded with a casette-based four-tracker.
16/44.1 can reproduce a shimmeringly deep, richly textured audio experience. A valid 16/8 test will take a sample of that sort: a great sounding 16 clip, converted down to 8.
This article is a veiled ad hominem, which says that Neil Young is wrong about audio because he once recorded some music that sounds like shit. The underlying fallacy is the idea that whenever some artist promotes higher sample rates or wider sample sizes, we should arbitrarily criticize that claim based on some poor example of their own music, rather than using science.
The original Soundblaster was a pretty noisy thing and was limited to 22.5 kHz, AFAIK. You could use noise dithering to improve the perceived quality, but it wouldn't be as effective at that sample rate.
10/10 but perhaps my sensing method would be considered cheating : the underlying noise throughout the whole track was noticeably more dithery/glitchy in one set of clips than the others, so I presumed that identified the 8-bit clips.
Speakers are tinny laptop speakers which I think probably overemphasised the effect.
Whether a person prefers 8, 16, or 24 bit is besides the point and has to do with their tastes.
That article clearly debunks any arguments saying that 24 bit is superior in re-creating reality. In fact he makes some points against it.
I love the analogy about a television that can display UV rays.
Why would you need a 4k video when your only display is 1080P?
Same thing goes for audio. Your ears are only capable of hearing a subset of what is reproducible with 16 bit audio, no need for 24 bit.
The only time you'd ever need more than 16 bits is if you're going to do more than just listen to the music.
If you're mixing/mastering that is another story.
I had 6/10 on Neil Young and 5/10 on Gangnam Style.
That said, he claims "less than 9 is not meaningful". But by my calculations an 8/10 is significantly different from chance alone; to some extent, 7/10 is as well.
Suppose random guesses are drawn from a Binomial distribution -- each sample is an urn containing balls black and white (both equally likely) and we count how many white balls we can get.
This is the CDF: people who get 1 answer right score better than only 1% of people.
---
1 0.0107422
2 0.0546875
3 0.171875
4 0.376953
5 0.623047
6 0.828125
7 0.945313
8 0.989258
9 0.999023
--
MP3 files, seriously? A lossy audio encoder will remove much of a static noise floor (and sacrifice some of the signal in the process). That's the point.
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[ 0.12 ms ] story [ 138 ms ] threadBut to be honest I don't think the snippets are good candidates for the comparison
From the summary thing above the blind test. They're aware of this, this test was mostly done for irony (as also explained in the blurb.)
It's known that the only difference between well-prepared files of different bit depths is the noise floor. Basically, in any PCM audio file, you can expect noise of about 1 ULP. So in a 16-bit file you get a noise floor of -90 dB, and in an 8-bit file it's at -42 dB. This is the ONLY DIFFERENCE.
People who are here listening for distortion artifacts aren't going to find any, and they're going to be surprised. That's because you can always choose to introduce noise instead of distortion artifacts when you do the conversion. The noise is actually added on purpose, it's "dithering noise", and it's not there to mask the distortion, but to entirely replace it.
The math works like this. Take a high-resolution input, say, 16-bit. You're converting it to 8-bit. If you just convert, you'll introduce a distortion signal of about 1ULP. However, if you introduce 1ULP of uncorrelated rectangular noise, the distortion signal is now also uncorrelated noise, which is less objectionable than correlated noise. It's also harder to hear.
Then, how do you tell the difference between 16-bit and 8-bit audio files? You can only do it if there's a quiet part somewhere in there. As soon as there's a quiet part, you can hear a "hissing" sound in the 8-bit file at -42 dB. If you're listening to the middle of a rock song like this, the hissing sound is going to be buried in the mix. Heck, a typical guitar amp will already be putting out uncorrelated noise.
Even though, the difference is obviously small.
The primary place I picked out to hear this not the quiet parts, but rather a hiss above 10khz, which was audible even at the beginning - and particularly at the beginning. It's almost mistakable for brighter cymbals, if you're not listening closely. It also changes in perceptibility over the duration of the sample.
This is with HD-598's with 16 feet of cheapo headphone extension cords, loud old refrigerator running in tiled studio apartment, and motherboard built-in sound output.
Someone needs to check their p values.
In my experiments I found that for almost anything, that is not orchestral music, and that is newer than 1990, 8-bit sample depth are enough. That was really depressing.
By the way, a nice jest is to tell some random bystander that you're going to present them "8-bit" music and then play the 8-bit version of a song that was a victim of the loudness war. Supposedly, the 8-bit times were better than most people think. ;-)
When you do experiments you must understand your experimental design enough to understand exactly what you're testing and what conclusions can be drawn from that. If you do A/B testing of 8-bit/16-bit (which is a good idea) you have to understand you're doing the test through your current audio hardware.
The whole point of the Pono player is to have higher quality hardwhere everywhere, including pre-amps and DACs so you have a chance to hear subtle differences.
If you do this test through your laptop speakers you really might not be able to tell the difference. If you'd like to "debunk" the Pono player then do this test through a Pono player pushing music through quality studio headphones.
Making things worse they're using Neil Young's Rockin In The Free World which contains, get this, large amounts of harmonic distortion to begin with. The pre-existing harmonic distortion will only serve to mask any distortion and fidelity loss from truncating to 8-bits.
If you don't own high-fidelity music equipment then this test is like proving high definition television is impossible because you can't tell the difference between a standard and high definition signals on your standard definition TV.
Yes, hardware quality might make a difference in fidelity. Still, 16 bit CD-quality audio is enough. The marketing of the Pono player with 24bit/196kHz is just nonsense.
And of course any listening test requires quality hardware to make sure it can actually reproduce the test-signal correctly.
> The pre-existing harmonic distortion will only serve to mask any distortion and fidelity loss from truncating to 8bit.
With dithering, smaller sample depths only decrease the signal to noise ratio. There are no distortions. The sample is chosen "badly" (but that was by design) because it contains little to no dynamics.
I have no patience for articles like this because instead of setting the record straight it just adds more pseudoscience to the mix. Their "experiment" doesn't prove what they claim it proves.
Is 24-bit useful for playback? Yes it is. Do you need it for properly mastered audio? Absolutely not. But there are 10,000 home studios across the country who have music files at 24-bits and a large chunk of those engineers don't know how to master audio. Is a raw 24-bit track better than a poorly mastered 16-bit track? It is to me.
Is 192kHz useful for anything? Probably not. Does it hurt anything? Maybe a little. I would expect a quality music player to go up to 96 anyhow so 192 isn't so bad. Similar reasons as before.
Does this reflect badly on the Pono player? Not really. Over-engineering isn't a bad thing in my book.
That's not splitting hairs. Harmonic distortion and uncorrelated noise are completely different from a mathematical stand point. They also sound audibly different to my ears.
Merely saying that "24-bit is useful" or "it is better to you" does not mean that you can actually hear the difference, and you include no empirical justifications for why you believe this to be true. Perhaps I could help you conduct a scientific experiment at home?
The whole point of 24-bit is to give you flexibility when the singer decides that he's going to whisper one verse from three feet away and then shove the mic down his throat while he screams the chorus. Or you can be sloppy when you set the preamp gain—just give yourself enough headroom, and you'll adjust the levels later. Notice that I said "later", not "never". Even the most amateurish home engineer with a hangover the size of Texas is going to adjust the levels of different tracks in the mix. Once you've done the mixdown, the levels are "reasonable", and you can listen at home on 16-bit system. You won't hear the difference.
No mastering, no compression. 16-bit audio is still enough, once you've mixed a song.
Maybe you should save your grandiose lectures for somebody who doesn't know enough to see through it. There's no reason to assume levels will be at a "reasonable" level and there's no reason to think a high-quality portable music player should only play back 16 bits. That's absurd. Should consumer level music players stop at 16? Sure. I want better than that but if you want to listen to mp3s on your phone nobody's stopping you.
I'm left wondering how much of this anti-Pono talk is shilling on behalf of smartphone manufacturers. I need 24 bit playback in roughly the same way I need a car that does 0-60 in under 3.5 seconds - I like things that are over-engineered so I can geek out on how awesome they are. But thanks to all the killjoys we can't even get excited about the first high quality portable music player because some amateur sound engineers want to show off they once read a blog post on Shannon-Nyquist theory.
Can you elaborate? What makes you think that people are shilling for smartphone companies?
That's not sarcasm. Superior components such as a better power supply, pre-amp, and DAC can make a big difference. Something like the Audioquest Dragonfly or some of Schiit's low-end gear can make a difference that can be amazing in some instances.
But in every interview I've ever seen, Neil Young focuses on the high resolution audio files which do not make a difference that human beings can hear. He has also, in general, been saying kooky things about digital audio for decades now, ever since CDs became popular.
This is blatantly incorrect. The 8-bit conversion is not truncation, it is dithering. Dithering to 8-bit does not introduce any distortion at all, whatsoever, of any kind. If you don't understand the mechanisms and the science of how bit depths work, then you're going to come to false conclusions, like the conclusion that there's any point to the Pono player at all. We're not talking about just "subtle differences" here. In order for it to be even theoretically possible to hear the difference between 16-bit and 24-bit audio, you have to bring your audio system into a quiet room and then crank the volume levels well above into the threshold at which you can damage your ears, and even then, you still won't be able to tell the difference with the most dynamic music.
So, suppose you have a quiet room in your house, with an ambient noise level of about 30 dB. If you raise the 16-bit audio level so the noise floor is above 20 dB, then the peaks are going to be well into the 120 dB range. That's like having a symphony orchestra in the room with you, at the very peak of their performance, with all the instruments playing at once. If you've ever listened to a symphony orchestra, you know that the background noise is NOT 30 dB, but somewhat higher. So even at the peak of a symphony, your CD recording should still be able to reproduce the various unwelcome bits of noise that the musicians produce (stomachs gurgling, breathing, shuffling in their chairs, etc.)
> In order for it to be even theoretically possible to hear the difference between 16-bit and 24-bit audio,
Only for professionally mastered audio which is not a safe assumption in this day and age. If some home engineer recorded a track with too much headroom and you get the 24-bit track you're fine, at 16-bits you have a problem.
I would love a music player where I could play tracks at 88kHz/24-bit because that's what most music is during the mixing process and then an audio engineer can give you the raw version of what they're working with without having to deal with issues of headroom, downsampling and dithering.
24-bit audio doesn't hurt anything other than file size and it has real uses, be it remix culture or just high-quality unmastered music.
Audio engineers don't have to "deal" with the issues of downsampling and dithering. The DAW just does it. It's a solved problem. It's not even a button you have to press, it's all automatically set up for you these days. We know what algorithms to use: band-limited interpolation with dithering, possibly combined with noise shaping.
Headroom is a different issue, but even the sloppiest home engineer is going to take a look at the levels at some point. If they don't, they can just check the "normalize" checkbox when they bounce. Or they can just ignore it and leave it checked. They're still not going to give you files with noise floors below -90 dBFS, and therefore, there's still no point in giving you a 24-bit file.
Even for remixes, you're not getting any benefit, since the noise floor is above -90 dBFS anyway.
However... you want to work on a project together? Let's keep things 24-bit until the final mixdown.
I don't disagree that CD quality basically maxes out the ear's natural capabilities, I don't think it's that easy to do.
For a remix I'd rather get a 24-bit stream than a 16-bit stream (that's really 15 bits) and has to be padded up to 24 anyway.
Can you hear the difference between 16-bit audio with dither and 16-bit audio without dither?
Most people can. Now consider - that difference is created by adding a noise signal which is more than 90dB down compared to the maximum possible level.
By all reasonable expectations that difference should be completely inaudible under normal listening conditions.
But the effect it has isn't inaudible at all.
When you understand why, then you'll understand the difference between peer-reviewed and objectively tested psychoacoustic theory, and hand-waving about numbers.
You'll also understand why it's trivially easy to tell amplifiers and converters apart even when they have distortion products well below -90dB.
That aside - you're making the usual mistake of confusing dynamic range with resolution.
What's the effective bit resolution of a -48dB signal on a 16 bit system?
What's the resolution of the same signal on a 24dB system with the same output level?
What's the minimum number of bits needed to make quantisation noise inaudible? (Clue: rather more than 8.)
I'm honestly confused here, because you claim to be disagreeing with me, but when I read the content of your post, it sounds like you actually agree with me?
When I was talking about hearing the difference between 16-bit and 24-bit, I was assuming that we dithered our audio. You can't hear white noise at -90 dBFS in typical listening conditions. You'd have to be in a quiet room with the volume turned way up, and you'd have to have very low-noise equipment.
The effective resolution of a -48dB signal on a 16-bit system will depend on that signal's bandwidth. If you don't understand that part, then you don't understand the math.
The latter depends on sample rate not of sample resolution.
If you're making mistakes like that, it's not a brilliant idea to tell people who write DSP code and have designed audio hardware that they don't understand the math.
The other point still stands. If 16-bit resolution is already good enough to represent signals without audible distortion, why does it need dither to sound acceptable, while 24-bit audio doesn't?
I'm still waiting for anyone who believes 16-bit recording is perfect to explain why the industry bothered to invent a clearly audible conditioning process for signals that are supposed to be ideal already.
Let's talk about bandwidth. If you have a pure sine wave and want to measure its amplitude, you can do a DFT on your signal and measure the appropriate bin. Let's assume that the sine wave does land in one particular bin. If your data is 16-bit with dithering, the dithering and quantization will add noise to all of the bins, but the noise will be equally divided. As you increase the length of the sample that you're analyzing, the bandwidth of each bin decreases, and the amount of noise in each bin decreases as well. However, the signal will always be concentrated in that one bin.
So, as you decrease the bandwidth, the quantization noise decreases as well. This is equivalent to saying that you have increased resolution.
I know this is counterintuitive. However, this is the foundation of how most modern ADCs work. It's called delta-sigma modulation, and it uses a low-resolution ADC internally to derive a high-resolution digital output. It's also been used in DACs. For an extreme example, look at DSD, which gives high-resolution outputs using a 1-bit signal.
The argument that "if 16 bits is enough, why do we need dithering" is kind of pointless, because we don't use 16-bit audio without dithering. It's like asking, "if this amplifier is good enough, why does it use negative feedback?" The answer of course is that negative feedback increases the linearity and flattens the response of the amplifier, and makes it less sensitive to variations in manufacturing and temperature.
Having $700 headphones helps, too.
The fade outs just seemed fuzzy.
Edit: I also have no idea if this goes for or against audiophiles. Maybe both? I know they're largely ridiculed.
Edit: The other test using Gangnam Style is much harder to detect (for me), I scored 6/10 on my first attempt (and I won't try again, hearing that riff 10 times in a row is plenty): http://www.audiocheck.net/blindtests_16vs8bit.php
The thing is that I agree with Monty on the (lack of) need for 24/196 audio for consumers. My degree is in audio, and I (mostly) understand the physics of the thing (and have done tons of blind listening experiments in controlled environments using professional equipment, as well as using high quality measurement tools for spectrum analysis, etc.). But, this particular test isn't really a useful proof that we don't need 24 bit and 196k. It only proves that loud material can mask noise, even at 8 bits.
Using a more dynamic Neil Young song would have been a better choice. Maybe Old Man, or similar. Acoustic guitar is good for allowing one to discern audio equipment flaws. It is commonly the source material people use when trying to rate good pieces of equipment that have very small differences across a very broad price spectrum (microphones, preamps, etc.).
Nonetheless, I got 8/10 right using crappy ear buds in a noisy place, which isn't much better than random chance. I'm pretty confident I'd be able to recognize 8 bit vs 16 bit on any reasonably dynamic source played on good equipment in a good listening environment. But, I also know I can't recognize 16 bit vs 24 bit (I've tested), and I know Neil Young can't either. Neil Young was around for the era when ADC/DAC quality was very poor; it actually was true that many early digital recordings were inferior to the very high end analog recordings of the time. And, it's even true that when recording, mixing, and mastering, it is useful to work at higher bit depths (because there's a lot of summing, raising the noise floor each time; though that may just mean you need to process at 24 or 32 bits, rather than actually record at higher depths).
It seems obvious, to me, that the reason most music sounds like shit is because of the loudness wars (dynamics reduction), lossy compression (cramming a lot of music down tiny network pipes), and the low quality of most people's listening equipment and environment. Those factors utterly dwarf the noise floor of 16 bit audio, and completely nullify any frequency advantages of higher bit rates (even if we could hear them).
In short: Pono is elitist bullshit, as most audiophile stuff is.
My go-to track for testing, recommended if rock is your preference, is Muse's "Undisclosed Desires". The beginning of the second verse has a "hidden" bit you will only hear on good headphones. First time I heard that song on some Grado SR80s I literally jumped from being spooked.
I'm a musician and have a large CD, vinyl record, and digital audio collection. Honestly the knots some people twist themselves into over formats amuse me. I tend to focus on the music and the rhythm rather than the recording quality, at least above a minimum quality level. I prefer vinyl just because I do. :-)
[edit] the test page suggested my results were close to random, and I agree. I couldn't really tell a difference through the laptop speakers. It felt like I was guessing for each sample.
Of course the source is specially selected to make the test hard. With music with real dynamics, that hasn't been compressed to death, the test becomes trivial. 8 bit gives you 48dB of SNR, that is lower than vinyl and tape.
16/44.1 can reproduce a shimmeringly deep, richly textured audio experience. A valid 16/8 test will take a sample of that sort: a great sounding 16 clip, converted down to 8.
This article is a veiled ad hominem, which says that Neil Young is wrong about audio because he once recorded some music that sounds like shit. The underlying fallacy is the idea that whenever some artist promotes higher sample rates or wider sample sizes, we should arbitrarily criticize that claim based on some poor example of their own music, rather than using science.
... not sure if trolling, or "audiophile".
Then of course it became obsolete for 99.9% of use cases
Subwoofer Harmonic Distortion test: http://www.audiocheck.net/testtones_subwooferharmonicdistort...
Polarity test: http://www.audiocheck.net/audiotests_polaritycheck.php
Practical effects of bit depth: http://www.audiocheck.net/audiotests_dithering.php
Headphone Ultimate Test (the binaural audio one is pretty awesome): http://www.audiocheck.net/soundtests_headphones.php
> https://news.ycombinator.com/item?id=10522850
Whether a person prefers 8, 16, or 24 bit is besides the point and has to do with their tastes. That article clearly debunks any arguments saying that 24 bit is superior in re-creating reality. In fact he makes some points against it.
I love the analogy about a television that can display UV rays.
It is all well explained in this article: https://xiph.org/~xiphmont/demo/neil-young.html
The only time you'd ever need more than 16 bits is if you're going to do more than just listen to the music. If you're mixing/mastering that is another story.
That said, he claims "less than 9 is not meaningful". But by my calculations an 8/10 is significantly different from chance alone; to some extent, 7/10 is as well.
Suppose random guesses are drawn from a Binomial distribution -- each sample is an urn containing balls black and white (both equally likely) and we count how many white balls we can get.
This is the CDF: people who get 1 answer right score better than only 1% of people. --- 1 0.0107422 2 0.0546875 3 0.171875 4 0.376953 5 0.623047 6 0.828125 7 0.945313 8 0.989258 9 0.999023 --
A "7" is almost at 95% significance.
Even after reading all the hints about what I should listen to in this thread.
http://www.klippel.de/listeningtest/lt/
A couple of years ago, we asked our colleagues to do the tests and the results were quite interesting.
(The tests require Microsoft Silverlight, which might be a little hard to setup these days.)