I cannot hear the difference between 16/44.1 (and by extension, 16/48) and High-Res Content generally, be they HDCD, SACD, or just straight-up Masters from Qobuz. This is on multiple sets of equipment, ranging from El Cheapo earbuds all the way to HD800 cans and full-fledged tower speakers being bi-amped.
That’s not why I go for High-Res stuff, though.
It’s all about archival, at least for me. With a 24/192 Master in FLAC or ALAC, I can downsample to whatever the destination form factor is. I can transcode to a 320kbps MP3, or a 16/48 WAV stream for a smart speaker, or a 24/96 stream for the theater. The point isn’t that I can hear the difference, it’s the fear that I might lose something irrecoverable by sticking with lower-quality files for bulk storage. Once data has been discarded, it cannot be retrieved, and that influences my preference for storage (and is also why my BD/UHD rips are into MKVs, no re-encoding).
Now that being said, I will absolutely hem and haw and ABX different releases to determine if I opt for the 16/44.1 CD rip of an album from the 80s or the new 202X remaster in 24/192 (spoiler: almost always the former), and I absolutely prefer anything with classic instruments (Jazz, Classical) in higher-quality formats because of a subjective perception of a wider, clearer sound stage, though this is almost certainly a psychological effect from performing in concert bands and orchestras rather than physical or objective in nature.
Like I tell newcommers: if it sounds better enough to you to warrant the purchase price, then that’s all that really matters. Enjoy the hobby.
The article says "I've run across a few articles and blog posts that declare the virtues of 24 bit or 96/192kHz by comparing a CD to an audio DVD (or SACD) of the 'same' recording. This comparison is invalid; the masters are usually different."
It may be simultaneously true that:
A) Humans cannot tell the difference between 44.1kHz/16-bit audio and any higher resolution, and
B) For a particular song, the best commercially available 44.1kHz/16-bit version may not be the best commercially available version
"The quality of the particular mastering can still make a noticeable difference regardless of the ability for the digital sampling rates to perfectly represent it perceptually"
Just to be clear that the statement applies to any such releases, not just 44.1 kHz @ 16-bit ones.
That’s true, but I consider myself a collector. Think of how a comic book collector operates.
If I have an option to get a 16bit version of a recording or a high-res version, I choose the highest quality version very time
Same with a physical copy. A limited edition, better quality vinyl LP is more attractive if you are going through the trouble of curating a collection.
I’ve been curating a music library of digital files since before the iPod was released and I will always go for the highest quality version out of principle. I can always downsample it to any thing that makes sense.
This really is driving a muscle/super car, or drinking expensive wine. At the end none of specs or tests matter. It is a form of art. If it makes the listener feel better (even if its just psychological) then its probably worth it.
To expand on this a bit, I appreciate some audio overkill because, if I do hear sizzle or distortion, it eliminates one possible reason and helps me figure out what’s actually happening.
It’s like having gigabit internet to my house: I don’t actually need it, but when a website is slow, I know the problem isn’t in my internet connection.
Correct. I've paid for Tidal for a decade because I just like the peace of mind that it's closer to the original recording. I'm sure it's mostly placebo, but I like it.
I'd distinguish between differences that anyone can detect but some may not care about, and differences that may not be objectively detectable at all. Muscle cars, at least, are different in a way that anyone can see. Push that pedal to the floor and it feels different from a Honda Civic or whatever. Whether that difference is actually interesting or good is, of course, a matter of taste. Whereas audiophile nonsense is often indistinguishable even to the connoisseur and depends entirely on some form of self-deception. Still could be worth it, depending on what one considers worthy.
That’s actually a really good comparison, especially because - yes I can hear the difference between an excruciatingly lossless digitization of a piece of music that I’m intimately familiar with, played back on expertly configured hardware… but the difference is so little, that most of the time, I’m find just listening to it at medium high quality streaming on a pair of <$50 headphones.
I’ve played with the nice toys, and they are nice, but for 100x the price, they barely deliver 1.5x the experience.
My good enough amplifier and DAC combo claims up to 24bit/192kHz, I use a cheap optical interface from my computer that claims up to 32bit/192kHz, and the streaming service I use serves most albums at 24bit/44.1kHz.
It would have cost the same for the entire stack to be 16bit/44.1kHz at every step, but with excessive resolution I can control the volume anywhere. The bits right before the analog conversion at the end are essentially the same whether I turn down the volume in the software player, the operating system, or the DAC/amplifier.
Oh great. And here I thought that fantasy literature where forest elves could hear the screams of the plants they stepped on when they walked was just that -- fantasy.
The point of this article and video is there is no problem with 16-bit 44-kHZ PCM. It thoroughly covers the audible range and is there is absolutely no need for more when distributing music for humans to listen to.
The problem is the people spreading myths and disinformation out of ignorance or to promote their enterprise.
The weak links are producers/mastering-engineers, speakers/headphones and the room when using speakers.
Nobody downloads music these days and everybody just streams. Audio at 24 bit still takes a small fraction of the bandwidth that 1080p video takes, so I don’t understand the hate for it.
I use a DAC by focusrite which can do 24-bit, and if I want to listen to higher fidelity audio on my planer headphones then I should be able to. Why should I limit myself to 16-bit
Counterpoint: bandcamp is doing well. Vinyl sales are doing well.
If I like an artist that I find on streaming, I buy an LP and get a lossless download for free. I still have a music library and I will never rent my favorite music.
Artists prefer to connect directly with their fans and BC is probably the best platform for people who care to pay and support acts directly. They have high res downloads and I import them.
Just get one of those "hi fi" valve amplifiers from Amazon you see under $100. The valve already distorts the sound, so you don't need to bother paying more for low distortion anywhere else in the audio chain. Saved you thousands of dollars, done!
Foobar2000 has an extension that allows you to blindly test whether you can tell the difference between two tracks.[1] The prime use is to compare different encodings of the same song from the same lossless master.
It kind of changed me a bit when I ran through 20 lossless tracks I had re-encoded to various mp3 bitrates and realized that even on a fancy system, it can be really hard if not impossible to discern even moderate lossy from lossless.
If you are an audiophile geek, really think about if you want to try this, the reality check might crack your foundations.
I'm curious if the audio was being sent bit-perfect to the DAC for all of these tests (ALSA direct), or if it was being run through the audio mixer and being resampled
I can always tell if my 44.1 songs are being resampled to 48 because they're being run through the OS mixer
@xiphmont also made an amazing video response to the many responses he received to this article. Using analog equipment he busts a bunch of myths and demonstrates what really happens with digital audio.
At a minimum, anything above 16/44.1 requires far more than just files: monitors, a treated room, listening position, DAC, etc... but most importantly - a trained ear. That last one is the most uncomfortable truth.
If you want to hear the difference between an audio file recorded at 44.1 and 88.2kHZ, then you need slow the audio playback down. Otherwise, a trained ear cannot physically hear the difference.
Pretty good analogy. Thing is though, the person who receives the 16-bit, 44.1khz music file can always upsample it to 192khz and not lose anything in the process (heck, lots of audio stuff oversamples internally to this level or beyond, for extra aliasing headroom!). I'm not sure about expansion from 16bit to 24bit though, downward expansion isn't necessarily perfect.
The whole audiophile industry is built on stuff which doesn't make any sense
My favourite: "audiophile-grade" audio players which allocate a single continuous buffer of RAM into which they load/decode the whole .WAV/.FLAC file, because supposedly the CPU "jumping" between "fragmented audio" causes audible "jitter".
Of course, they don't know that what looks like continuous memory to user-code is probably discontinuous in kernel/physical RAM.
Didn't check in many years, I wonder if they created kernel level players to account for that, to have "true continuous memory"
> My favourite: "audiophile-grade" audio players which allocate a single contignuous buffer of RAM into which they load/decode the whole .WAV/.FLAC file, because supposedly the CPU "jumping" between "fragmented memory" causes audible "jitter".
Thanks for the laugh... this is absolutely bonkers. In case anyone is wondering, before sound hits our ears it has to go through a digital to analog conversion, which takes place on hardware independent of the CPU, operating with its own clock and buffers etc.
If you try to use empiricism when it comes to certain groups audiophiles, you are going to be sorely reminded that it's basically the equivalent of healing crystals for a different type of person. 24/192 is useful for mixing/mastering, but completely unnecessary for the end product to distribute for listening.
24/192 is also great for digital synthesizers--if you're generating a waveform like a sawtooth that has theoretically instantaneous transitions, they can eat as much frequency as you can give them. Running at 44khz loses noticeable high-end content.
Most modern digital synths have already caught onto this and run internally at much higher sampling rates even if their output gets downsampled, but sometimes you run across a vintage plugin that runs at the host audio rate and working in a higher sampling rate is audible.
32-bits are great for recording too because they do an incredible job of capturing the dynamic range without having to be precise on the preamp settings. It removes an entire job from the recording workflow.
192 for mixing and mastering can be useful especially if you're doing a lot of effects, especially anything that pitch shifts. But I've seen low quality phone-microphone recordings make it to the master; if you capture lightning in a bottle, it hardly matters what the settings were, what the microphone was, or anything else.
I completely accept that human audition has limits that are easy to determine by playing a pure sound. But is it the same with music, where multiple frequencies are played and interfere with each other? Aren't some harmonics or effects created by these "inaudible" frequencies?
To try to imagine something similar: the human eye is unable to see UV light, yet fluorescent paint has a visible quality of its own compared to "normal" pigments.
32-bit float has become popular in filmmaking/field recording equipment lately because, with a microphone preamp that supports it, you can capture the entire dynamic range of the microphone--there's no accidental clipping if you drive the gain stage too hard.
It's a bit redundant for a skilled technician, they're already used to setting the gain staging, inbound compression, and feathering the mics to avoid this in 24-bit, but if you're handing a boom mic to a novice and have a scene where e.g. someone's whispering and another person's screaming, it can be nice to not have to worry about it.
sheeesh , measly 24-bit/192kHz
of course it makes no sense, unless it is downloaded through low oxyegen wire, which somehow and unfathomably, must have been omited or forgotten.
For typical listening (though humans can perceive bone-conducted vibrations up to 100 kHz or even 120 kHz) 16-bit-fixed/44.1kHz is a high-fidelity transport format. As a DSP researcher, I prefer 32-bit-float/44.1kHz as a transport format. I often upsample to 32-bit-float/188.2kHz or even 32-bit-float/192kHz for signal processing applications such as high-fidelity reverberation via direct and FFT convolution. While the author advocates for the transport to ear use case, I would argue that 24-bit/192kHz provides greater fidelity and resolution for sound processing. I found the pedantic arrogance of the author to be annoying. But yes, the sampling theory is an important consideration -- but so is the quality of the actual digital filters used in the DAC->ADC pipeline. They are much more forgiving and less lossy at 192kHz.
I cannot hear the difference between 16/44.1 (and by extension, 16/48) and High-Res Content generally, be they HDCD, SACD, or just straight-up Masters from Qobuz. This is on multiple sets of equipment, ranging from El Cheapo earbuds all the way to HD800 cans and full-fledged tower speakers being bi-amped.
That’s not why I go for High-Res stuff, though.
It’s all about archival, at least for me. With a 24/192 Master in FLAC or ALAC, I can downsample to whatever the destination form factor is. I can transcode to a 320kbps MP3, or a 16/48 WAV stream for a smart speaker, or a 24/96 stream for the theater. The point isn’t that I can hear the difference, it’s the fear that I might lose something irrecoverable by sticking with lower-quality files for bulk storage. Once data has been discarded, it cannot be retrieved, and that influences my preference for storage (and is also why my BD/UHD rips are into MKVs, no re-encoding).
Now that being said, I will absolutely hem and haw and ABX different releases to determine if I opt for the 16/44.1 CD rip of an album from the 80s or the new 202X remaster in 24/192 (spoiler: almost always the former), and I absolutely prefer anything with classic instruments (Jazz, Classical) in higher-quality formats because of a subjective perception of a wider, clearer sound stage, though this is almost certainly a psychological effect from performing in concert bands and orchestras rather than physical or objective in nature.
Like I tell newcommers: if it sounds better enough to you to warrant the purchase price, then that’s all that really matters. Enjoy the hobby.
The article says "I've run across a few articles and blog posts that declare the virtues of 24 bit or 96/192kHz by comparing a CD to an audio DVD (or SACD) of the 'same' recording. This comparison is invalid; the masters are usually different."
It may be simultaneously true that:
A) Humans cannot tell the difference between 44.1kHz/16-bit audio and any higher resolution, and
B) For a particular song, the best commercially available 44.1kHz/16-bit version may not be the best commercially available version
While 100% true, I'd phrase B as:
"The quality of the particular mastering can still make a noticeable difference regardless of the ability for the digital sampling rates to perfectly represent it perceptually"
Just to be clear that the statement applies to any such releases, not just 44.1 kHz @ 16-bit ones.
As they say, most people listen to their music with equipment. Audiophiles listen to their equipment with music.
That’s true, but I consider myself a collector. Think of how a comic book collector operates.
If I have an option to get a 16bit version of a recording or a high-res version, I choose the highest quality version very time
Same with a physical copy. A limited edition, better quality vinyl LP is more attractive if you are going through the trouble of curating a collection.
I’ve been curating a music library of digital files since before the iPod was released and I will always go for the highest quality version out of principle. I can always downsample it to any thing that makes sense.
This is perfect, thank you this goes straight into my long-term memory bank.
On a tangent, whenever someone mentions LP sounding warmer or whatever I like to point out that I prefer wax cylinders (a.k.a. phonograph cylinders).
This really is driving a muscle/super car, or drinking expensive wine. At the end none of specs or tests matter. It is a form of art. If it makes the listener feel better (even if its just psychological) then its probably worth it.
To expand on this a bit, I appreciate some audio overkill because, if I do hear sizzle or distortion, it eliminates one possible reason and helps me figure out what’s actually happening.
It’s like having gigabit internet to my house: I don’t actually need it, but when a website is slow, I know the problem isn’t in my internet connection.
Well, at least there are objective performance benchmarks on cars, and some of them are okay proxies of performance in motorsports.
https://www.carwow.co.uk/blog/carwow-quarter-mile-400-metre-...
https://en.wikipedia.org/wiki/List_of_N%C3%BCrburgring_Nords...
Correct. I've paid for Tidal for a decade because I just like the peace of mind that it's closer to the original recording. I'm sure it's mostly placebo, but I like it.
I'd distinguish between differences that anyone can detect but some may not care about, and differences that may not be objectively detectable at all. Muscle cars, at least, are different in a way that anyone can see. Push that pedal to the floor and it feels different from a Honda Civic or whatever. Whether that difference is actually interesting or good is, of course, a matter of taste. Whereas audiophile nonsense is often indistinguishable even to the connoisseur and depends entirely on some form of self-deception. Still could be worth it, depending on what one considers worthy.
That’s actually a really good comparison, especially because - yes I can hear the difference between an excruciatingly lossless digitization of a piece of music that I’m intimately familiar with, played back on expertly configured hardware… but the difference is so little, that most of the time, I’m find just listening to it at medium high quality streaming on a pair of <$50 headphones.
I’ve played with the nice toys, and they are nice, but for 100x the price, they barely deliver 1.5x the experience.
My good enough amplifier and DAC combo claims up to 24bit/192kHz, I use a cheap optical interface from my computer that claims up to 32bit/192kHz, and the streaming service I use serves most albums at 24bit/44.1kHz.
It would have cost the same for the entire stack to be 16bit/44.1kHz at every step, but with excessive resolution I can control the volume anywhere. The bits right before the analog conversion at the end are essentially the same whether I turn down the volume in the software player, the operating system, or the DAC/amplifier.
If you can't hear the squeals of the plants [1] in the studio's reception area, are you really getting the full experience of a piece of music?
[1]: https://www.cnn.com/2023/03/30/world/plants-make-sounds-scn
Oh great. And here I thought that fantasy literature where forest elves could hear the screams of the plants they stepped on when they walked was just that -- fantasy.
What a human centric view. I like my music to scare neighbor's pets.
Counter: An ultra high bit rate solves the problem and you can stop worrying if it's the weakest link.
You can the focus on other things.
Example: I Bought the best skis possible. Now I know I need to just focus on my skills and not blame the equipment.
The point of this article and video is there is no problem with 16-bit 44-kHZ PCM. It thoroughly covers the audible range and is there is absolutely no need for more when distributing music for humans to listen to.
The problem is the people spreading myths and disinformation out of ignorance or to promote their enterprise.
The weak links are producers/mastering-engineers, speakers/headphones and the room when using speakers.
(2012) https://news.ycombinator.com/item?id=3668310 316 comments
(2014) https://news.ycombinator.com/item?id=8689231 424 comments
(2015) https://news.ycombinator.com/item?id=10520639 228 comments
(2017) https://news.ycombinator.com/item?id=15127633 428 comments
(2019) https://news.ycombinator.com/item?id=19318898 314 comments
Nobody downloads music these days and everybody just streams. Audio at 24 bit still takes a small fraction of the bandwidth that 1080p video takes, so I don’t understand the hate for it.
I use a DAC by focusrite which can do 24-bit, and if I want to listen to higher fidelity audio on my planer headphones then I should be able to. Why should I limit myself to 16-bit
Counterpoint: bandcamp is doing well. Vinyl sales are doing well.
If I like an artist that I find on streaming, I buy an LP and get a lossless download for free. I still have a music library and I will never rent my favorite music.
Artists prefer to connect directly with their fans and BC is probably the best platform for people who care to pay and support acts directly. They have high res downloads and I import them.
Just get one of those "hi fi" valve amplifiers from Amazon you see under $100. The valve already distorts the sound, so you don't need to bother paying more for low distortion anywhere else in the audio chain. Saved you thousands of dollars, done!
Foobar2000 has an extension that allows you to blindly test whether you can tell the difference between two tracks.[1] The prime use is to compare different encodings of the same song from the same lossless master.
It kind of changed me a bit when I ran through 20 lossless tracks I had re-encoded to various mp3 bitrates and realized that even on a fancy system, it can be really hard if not impossible to discern even moderate lossy from lossless.
If you are an audiophile geek, really think about if you want to try this, the reality check might crack your foundations.
[1]https://www.foobar2000.org/components/view/foo_abx
I'm curious if the audio was being sent bit-perfect to the DAC for all of these tests (ALSA direct), or if it was being run through the audio mixer and being resampled
I can always tell if my 44.1 songs are being resampled to 48 because they're being run through the OS mixer
Proper audio resampling should not be identifiable. Of course, the OS mixer probably doesn't do proper (CPU expensive) resampling.
But a quality audio player should account for this and do it's own.
@xiphmont also made an amazing video response to the many responses he received to this article. Using analog equipment he busts a bunch of myths and demonstrates what really happens with digital audio.
https://video.xiph.org/vid2.shtml
or on YT if you can't play it https://www.youtube.com/watch?v=cIQ9IXSUzuM
At a minimum, anything above 16/44.1 requires far more than just files: monitors, a treated room, listening position, DAC, etc... but most importantly - a trained ear. That last one is the most uncomfortable truth.
Are you, per chance, a dog posting on the internet? Since 44.1khz sample rate is already past the range of the human ear, regardless of training.
If you want to hear the difference between an audio file recorded at 44.1 and 88.2kHZ, then you need slow the audio playback down. Otherwise, a trained ear cannot physically hear the difference.
A treated room would be the most impactful, DACs the least.
huh...
So I guess the programmer equivalent is distributing .pdb's (or, symbols)
Pretty good analogy. Thing is though, the person who receives the 16-bit, 44.1khz music file can always upsample it to 192khz and not lose anything in the process (heck, lots of audio stuff oversamples internally to this level or beyond, for extra aliasing headroom!). I'm not sure about expansion from 16bit to 24bit though, downward expansion isn't necessarily perfect.
The whole audiophile industry is built on stuff which doesn't make any sense
My favourite: "audiophile-grade" audio players which allocate a single continuous buffer of RAM into which they load/decode the whole .WAV/.FLAC file, because supposedly the CPU "jumping" between "fragmented audio" causes audible "jitter".
Of course, they don't know that what looks like continuous memory to user-code is probably discontinuous in kernel/physical RAM.
Didn't check in many years, I wonder if they created kernel level players to account for that, to have "true continuous memory"
Don't forget: "most players use malloc to get memory while new is the c++ method and sounds better."[1]
[1] https://www.audioasylum.com/messages/pcaudio/119979/
> My favourite: "audiophile-grade" audio players which allocate a single contignuous buffer of RAM into which they load/decode the whole .WAV/.FLAC file, because supposedly the CPU "jumping" between "fragmented memory" causes audible "jitter".
Thanks for the laugh... this is absolutely bonkers. In case anyone is wondering, before sound hits our ears it has to go through a digital to analog conversion, which takes place on hardware independent of the CPU, operating with its own clock and buffers etc.
I can tell when my CPU usage spikes because it causes a hum through my speakers, so this does not seem that far-fetched.
The latter is probably true, but the former does actually happen, and it's easy to accidentally do--lossless or not.
If you try to use empiricism when it comes to certain groups audiophiles, you are going to be sorely reminded that it's basically the equivalent of healing crystals for a different type of person. 24/192 is useful for mixing/mastering, but completely unnecessary for the end product to distribute for listening.
24/192 is also great for digital synthesizers--if you're generating a waveform like a sawtooth that has theoretically instantaneous transitions, they can eat as much frequency as you can give them. Running at 44khz loses noticeable high-end content.
Most modern digital synths have already caught onto this and run internally at much higher sampling rates even if their output gets downsampled, but sometimes you run across a vintage plugin that runs at the host audio rate and working in a higher sampling rate is audible.
32-bits are great for recording too because they do an incredible job of capturing the dynamic range without having to be precise on the preamp settings. It removes an entire job from the recording workflow.
192 for mixing and mastering can be useful especially if you're doing a lot of effects, especially anything that pitch shifts. But I've seen low quality phone-microphone recordings make it to the master; if you capture lightning in a bottle, it hardly matters what the settings were, what the microphone was, or anything else.
They literally sell actual crystals that you’re supposed to place on top of speakers and amplifiers to make them sound better.
Even with mixing/mastering 96khz is enough for persisting to files. But as another commenter said, 192 is useful, if you bend and stretch samples!
(2012)
I wonder how many people think that 24 bit audio encodes 50% “more”
It is 50% more headroom above the noise floor in logarithmic decibels.
I completely accept that human audition has limits that are easy to determine by playing a pure sound. But is it the same with music, where multiple frequencies are played and interfere with each other? Aren't some harmonics or effects created by these "inaudible" frequencies?
To try to imagine something similar: the human eye is unable to see UV light, yet fluorescent paint has a visible quality of its own compared to "normal" pigments.
(2012)
Some previous discussions:
2023 https://news.ycombinator.com/item?id=34698427
2022 https://news.ycombinator.com/item?id=30138561
2019 https://news.ycombinator.com/item?id=19318898
2017 https://news.ycombinator.com/item?id=15127633
2015 https://news.ycombinator.com/item?id=10520639
2014 https://news.ycombinator.com/item?id=8689231
2012 https://news.ycombinator.com/item?id=3668310
Obligatory mention of https://xiph.org/video/ which clears up a lot of misconceptions.
24 bits is now ubiquitous and 32 bit is becoming the norm in recording studios.
32-bit float has become popular in filmmaking/field recording equipment lately because, with a microphone preamp that supports it, you can capture the entire dynamic range of the microphone--there's no accidental clipping if you drive the gain stage too hard.
It's a bit redundant for a skilled technician, they're already used to setting the gain staging, inbound compression, and feathering the mics to avoid this in 24-bit, but if you're handing a boom mic to a novice and have a scene where e.g. someone's whispering and another person's screaming, it can be nice to not have to worry about it.
That use case is literally addressed in the first sentence.
sheeesh , measly 24-bit/192kHz of course it makes no sense, unless it is downloaded through low oxyegen wire, which somehow and unfathomably, must have been omited or forgotten.
If it has been transmitted via hollow-core fibres it will obviously sound hollow.
For typical listening (though humans can perceive bone-conducted vibrations up to 100 kHz or even 120 kHz) 16-bit-fixed/44.1kHz is a high-fidelity transport format. As a DSP researcher, I prefer 32-bit-float/44.1kHz as a transport format. I often upsample to 32-bit-float/188.2kHz or even 32-bit-float/192kHz for signal processing applications such as high-fidelity reverberation via direct and FFT convolution. While the author advocates for the transport to ear use case, I would argue that 24-bit/192kHz provides greater fidelity and resolution for sound processing. I found the pedantic arrogance of the author to be annoying. But yes, the sampling theory is an important consideration -- but so is the quality of the actual digital filters used in the DAC->ADC pipeline. They are much more forgiving and less lossy at 192kHz.
The more the bits the better the music, easy as one two three
Don't forget to buy the new low oxygen platinum plated HDMI cables for the better experience!
/s