19 ms·

Let's go over this one more time: - Q: Can you hear the difference between CD-quality lossless audio and anything higher fidelity? A: No, no one even has the b

• pkulak 3y ago
  I’ve personally ABX tested people who swore up and down that they could spot 128 AAC, even, and they couldn’t. Never found anyone who could. I know they exist, but they are rare, and probably not the folks who say they can.
  • Espressosaurus 3y ago
    There's probably an element of the quality of the DACs and speakers you're using too. If it's a subtle difference it's unlikely we're going to notice it being played through some low-end computer speakers.
    • c0pium 3y ago
      I have very good equipment and lots of people in my circle who are audio enthusiasts. None of them have ever been able to demonstrate in a blind test that they can tell the difference.
  • jerf 3y ago
    Some of it is people ask the wrong questions. On a loudness-war-wrecked pop song I may not be able to tell 128Kbps from the original, but on specific content I have been able to tell. I'm not even claiming golden ears or anything; some specific audio content is the audio equivalent of visual confetti [1], and anyone can hear the difference, because the codec isn't even close. And let me underline, I mean, anyone. No special claims being made here. But all in all, that content is relatively rare, and generally transient even in the music they appear in. [1]: https://www.youtube.com/watch?v=r6Rp-uo6HmI https://www.youtube.com/watch?v=r6Rp-uo6HmI
    • TylerE 3y ago
      The giveaway for low-mid nitrate MP3 is the high hats. The lower the nitrate, the more you get a sort of temporal ghosting that sounds like an almost “crunchy” swishy sizzle sort of sound, a bit like a jazz player using brushes, but more lo fi.
      • tedunangst 3y ago
        Low nitrate MP3 is a fantastic typo.
      • hunter2_ 3y ago
        I agree, and my hypothesis is that it's exacerbated by the combination of three particular things: 1. It's a high frequency complex waveform with a fast envelope, so it demands bitrate. 2. Drum miking often involves multiple mics spaced apart, so more than one typically picks up any given cymbal with a phase offset, and those mics are panned quite differently, leading to a very "wide" result, i.e., left and right output is fairly uncorrelated as seen on a vectorscope [0]. 3. A perceptual codec at a given total bitrate often sounds better when stored as a mid-side transformation (instead of storing a left channel and a right channel, store a L+R "mid" a.k.a. sum channel and a L-R "side" a.k.a. difference channel), also known as "joint stereo" which is a common flag on MP3 encoders, because it allows for assigning more bits to the mid channel (correlated signals) and fewer bits to the side channel (uncorrelated signals). More bits for mono center-panned stuff like vocals is the goal, which is generally for the best, but fewer bits remain available for wide stuff like those cymbals! Contrast with regular stereo mode where half of the total bitrate is assigned to each channel. MP3 below 256kbps typically needs joint stereo mode enabled in order to sound decent. [0] https://en.m.wikipedia.org/wiki/Vectorscope#Audio https://en.m.wikipedia.org/wiki/Vectorscope#Audio
  • mattgreenrocks 3y ago
    128k AAC is quite good, and is roughly akin to 160k MP3. Personally, past 160k on MP3, it gets very hard for me to distinguish bitrates, so I ripped at VBR, averaging at around 200k. 128k MP3s, though, fall apart with more complex instrumentation.
    • pimeys 3y ago
      And 128k opus is perfect to my ears. I store all my music in the best possible quality FLAC files, but stream it to my phone in 128k opus. Such a great format, encodes very fast even with my Intel atom and sounds great.
      • samtheprogram 3y ago
        I’m curious as someone with a local library, how do you stream it in a different format than it’s encoded in, from your phone?
        • lozf 3y ago
          Lossy to lossy transcodes always cause more degradation, so it's better to keep your lossy files in their original format without any re-encoding. Easy transcodes to various lossy formats are the primary reason for a well maintained and curated library of proper FLACs (or ALACs). The main advantage being that your can fit more great sounding music on your space-limited device. Disk space is cheap, but upgrades on your phone are probably not. Even if your Music collection fits easily, why not have more room for other media / apps?
        • doubled112 3y ago
          I used to use Subsonic and one of the many clients. I don't know if Subsonic is still an option, but there are many server side applications, including a few forks, that fill that role. There are others that work with those same Subsonic clients. It will allow you to choose a codec for streaming that is different than storage.
        • pimeys 3y ago
• Syzygies 3y ago
  "You" is excellent wording here. Who's listening? I've heard the resonable assertion that the most gifted audio engineers in the world cannot distinguish 192 kHz sample rates from a raw line feed, but some can distinguish 96 kHz. I certainly can't. I used to build audio equipment. There were legendary "golden ears" that people would drive hours to meet, for design feedback. Whatever they heard was reproducible blind with other "golden ears". How does this square with the logical assertion that there's a sharp cutoff to our biological ability to hear frequencies? Those tests don't account for our ability to sense the presence/absence of overtones. Now, computers may want better resolution, not to "listen" so much as to better transform in novel ways. Just as "DogTV" recolorizes for its audience, a computer could make the inaudible audible in novel ways. Reconstructing better 3D sound stages. Accurately reconstructing a singer's facial expressions via AI video, rather than simply splatting out something plausible. The whole point of computers are to extend our reach, coevolving in every conceivable way.
  • mrob 3y ago
    >Those tests don't account for our ability to sense the presence/absence of overtones. Overtones are frequencies. If you filter the higher frequencies you remove the higher overtones. The only way you're going to "hear" ultrasound is if it's very loud and you hear audio-frequency distortion generated in your own ear, e.g. like when bats are squeaking nearby. But this isn't useful musically, because everybody's ears distort differently.
    • Syzygies 3y ago
      Ok, so despite anecdotal evidence that some individuals can distinguish better-than-CD quality audio, we're questioning the existence of convincing double-blind studies. Yet we accept the varying cutoffs for what frequencies a person can consciously detect in isolation, as proof that we are incapable of perceiving audio information above those frequencies. Are people asserting that an ear removed from a cadaver, hooked up to the best available scientific equipment, measures as a perfect biologically derived low pass filter? Or that we even partially understand how neurons work, when there may be quantum effects to be uncovered a century from now? Intellectual history is a graveyard of models confused with reality.
• miav 3y ago
  Unless I'm reading it wrong, your second source does very much imply some people can tell the difference quite reliably. As expected, regular people can scarcely tell the difference, but musicians are better at it and sound engineers are in fact quite accurate. This matches my own experience well: most of my friends do not care about various levels of compression, nor what headphones they use - that's fine, I'm glad they're enjoying art in their own way - but I, and some others, do in fact stand to benefit from less compressed audio. I've personally done blind tests on myself using a python script that randomly plays compressed and uncompressed snippets of the same track and mp3@320 was not transparent to me (though opus@256 was). Can I tell the difference when casually listening? I don't know, but when the cost of lossless is having my music collection take 60gb instead of 20gb on my 512+gb device, I have no reason not to go for lossless.
  • jrajav 3y ago
    Examine Figure 1 - The key is the 4th and 5th columns there, CD/256 and CD/320. The results show no significant ability to discriminate between them.
  • high_priest 3y ago
    The thing about being or not being able to point out differences in audio quality is that it all boils down to pattern recognition. If you know anything about pattern recognition, you understrnd that you can't have pattern recognition without prior training through provision of tagged samples of such patterns. If you would give high quality audio experience, to a person that has been listening through 80s general store headphones, to low quality radio rips on magnetic tapes, you might be surprised how few people are going to describe one as "better", without prior description of work and technology required to produce each experience. And one would be even more surprised by how many people choose the cassette tapes because of nostalgia and a long time satisfying experience.
    • keeganpoppen 3y ago
      isn't perception itself a matter of mere pattern recognition? hearing? the whole point is that you can hear the difference. whether or not it sounds "worse"... is certainly debatable, but is definitely a value judgment. and the burden of proof is definitely on the "it doesn't matter side" to prove that a lower fidelity version is "better" than one truer to the original master.
• analog31 3y ago
  I created some computed waveforms for audio testing on my PC, and on a whim, stored them as both WAV and MP3. Counterintuitively, the MP3's worked just fine for all of my tests. I didn't dig into the reasons why.
  • kstrauser 3y ago
    At its core, an MP3 says that for the next slice of time, play these frequencies at these volumes. If your waveforms are simple, an MP3 can encode them perfectly.
• bayindirh 3y ago
  The thing is, the “heard” difference between 320kbps MP3 and CD quality & CD quality and higher resolution formats are not “details” per se. The audible difference can be described as sound stage size, instrument separation and atmosphere. The problem is making these details audible needs a good system. “Good” doesn’t mean $10K+ here. Two high quality, large-ish two way bookshelf speakers and a good amp with enough punch (50W+ Yamaha or similar will do) plus a good source in a sizable room is enough. There’ll be people who can’t tell any difference, there’ll be people who can “feel” it, and there’ll be people who can pinpoint differences. This is because the ear training and biological limits of said people. I have a friend who can pinpoint a half note (natural vs. sharp) mistake in a 90+ people symphony from YouTube recordings, incl. the instrument. His natural ear resolution is around 1/9th of the tone. He always tunes his instruments via ear and verifies with a tuner. So, this is not impossible. My ears are not that absolute, but I can divide music to layers and pinpoint details, for example. Lastly, taking a “diff” of CD quality and 320kbps MP3 version of the same track will leave an audible residue. There are other comments I left over the years here. Search them for more info. I’m on mobile. I have no practical way to link all of them.
  • duped 3y ago
    This is the kind of snake oil that double blind studies in controlled reference environments have established is snake oil. Like for example, the fact there is an "audible" diff is meaningless. The threshold of hearing is not linear nor frequency independent. This is called "masking" and it's exploited by lossy codecs to allow for better encoding as well as audio watermarking. You can add a noise that would be perceptible by itself to content that is entirely masked by the content itself. And the reverse is true, you can remove content without it being perceptible.
    • bayindirh 3y ago
      This is the first reaction I always get: "This is snake oil, and you don't know what you're talking about". While there are undisputed snake oils in audio and audiophile market, the difference I'm talking about, is not. The idea of lossy codecs is they filter out the things you theoretically don't hear, yes. However, the presumption that you don't hear these when they are present is not completely true. Because they have a secondary order effects in overall sound. The audible residue you claim that I don't hear when it's in the CD is the part of the sound which adds this instrument separation and soundstage expansion. Same is for higher sampling rates. While you can't pinpoint the difference with words, it shows itself as smoothness and "richer" sound. Saying that you can't hear that difference is akin to saying "Human eye can't see faster than 30/60/X FPS anyways", which is not true. When anyone presented with a lossy-encoded audio file produced with a state of the art encoder and not brick-wall mastered, will be impressed, yes. This includes me, too. However whenever you listen to the same file in lossless or, if present, higher resolution formats, with a sufficiently transparent audio system a couple of times, you start to notice the differences. There are a couple of caveats in all of this audio business. First of all, you need to know how your audio system sounds and behaves to be able to discern differences. This requires time with the same system for a long time, to understand how it responds. In my case, I have the luck of having the same amplifier (An AKAI AM-2850) for ~30 years. I know how that thing responds to any genre of music, and I know how anything should sound at any quality level. Again, as I aforementioned, you need to do these ABX tests a couple of times back to back, esp. if you don't know the track, to be able to decode the details in sufficient manner. Digitalfeed's ABX test (https://abx.digitalfeed.net https://abx.digitalfeed.net) understands this and makes you listen to the same thing 5-10 times according to your available time. See, I'm an ex-orchestra player. I played in concerts, listened master recordings, and YouTube uploads of the concerts I played as well. I have also listened tons of CDs, MP3s of the same albums, etc. Some of the albums I listen have a captivating sound when I listen to them from CDs. MP3 versions of the same albums do not nail me to my chair, yet I can't leave the CD version of the same album to get a cup of tea. Both are ran through a Yamaha CD-S300 CD player with an iPod interface and MP3 playing capability over USB. I can also write how CD tracking quality affects audio clarity, but this comment is long enough. In short, Yamaha's old CD-Recorder, CRW-F1 really improved sound quality by abusing Red Book standard by lengthening the pits of audio CDs. It reduced to capacity to 68 minutes, but it was worth it, esp. on lower end CD players.
• crazygringo 3y ago
  > 44100khz, 16-bit audio can perfectly reproduce audio as far as we can physically tell. I agree on the kHz (as well as on MP3), but I deeply disagree on 16 bits. Because yes, if you keep your headphone volume at a single reference level and never turn it up, then 16 bits is fine. This is very much proven. BUT this ignores the fact that people often turn up the volume a ton to hear the quiet part of the classical music, or on that YouTube video where the volume is inexplicably 5% as loud as it should be. So in practice, 24-bit audio allows you to retain perfect fidelity even when you have to turn the volume up. 16-bit doesn't. I don't understand why nobody ever talks about this. (Or why you have to install special utilities on your Mac to be able to turn up the volume to 200% or 400% in order to listen to those YouTube videos that are maddeningly recorded at 5% volume.)
  • jrajav 3y ago
    You're right, it's true that 24-bit reduces the noise floor and extends the dynamic range available. However, 16-bit audio already has a range of -96db (for reference, a quiet recording studio typically has an ambient noise floor of around -60db). In practice, this is beyond the noise floor of even the very best hi-fi systems. As you turn the volume up, you will start hearing the noise floor of your equipment long before you hear the noise floor of 16-bit audio. Unless you mean that 24-bit allows for representing audio that is stored at an extremely quiet level at the peaks, wasting most of the dynamic range. That would make more sense - but if audio is printed in such a flawed way, I would expect other quality issues to be present as well.
    • amluto 3y ago
      I haven’t done the math, but I wouldn’t be utterly shocked if undithered 16-bit audio, cranked up some silly amount (such that full scale is 130dBA perhaps) has an audible noise floor. This is consistent with my other comment about badly encoded MP3 being far from transparent.
      • SSLy 3y ago
        Yeah, 18-20 bits make sense in loudly tuned cinemas.
• tzs 3y ago
  You aren't considering the listening environment. It is possible for ultrasonic sounds to interact with objects in the environment or with other ultrasonic sounds to produce lower frequency sounds that are in the range of normal human hearing. There was an even a creepy ad campaign several years ago that took advantage of this. They had a billboard in New York for A&Es new show "Paranormal State" with the tagline "It's not your imagination". They used an ultrasonic system on the billboard to make audible sounds appear in a small region on the sidewalk but not anywhere else. When people walking along the sidewalk got to that region they would hear a woman whisper "Who's there? Who's there? It's not your imagination". That system worked by making a single ultrasonic beam that somehow as it dispersed became audible. There are other systems that use multiple ultrasonic beams that produce audible sound via interference where the beams meet. Many acoustic instruments do produce significant amounts of sound above normal human hearing range. Cymbals for example have nearly 70% of the sound power above 20 kHz. Trumpets with a mute have almost 2% above 20 kHz. It seems possible then that if you wanted to produce a recording that reproduces the sound you would get if live acoustic instruments were playing in the same environment you might need to include ultrasonics unless you are making a binaural recording. This does raise the question of what we actually want playback of a recording to achieve. Is a recording of a string quartet when played back in my living room supposed to sound like that string quartet is playing in my living room, or is it supposed to sound like what I'd have heard if I was there when the piece was recorded, or is it supposed to be something else? (For those who haven't heard of binaural recordings, they are stereo recordings made by placing microphones inside the ears of a model human head so they record the sounds that actually ends up in each ear when something is recorded live for a listener at a specific location in an environment. This page of headphone tests [2] includes a binaural test if you'd like to such a recording). [2] https://www.audiocheck.net/soundtests_headphones.php https://www.audiocheck.net/soundtests_headphones.php
  • vel0city 3y ago
    These ultrasonic playback things are depending on specific interference patterns materializing at certain spots. The design of the signals, shape of the space, arrangement and characteristics of the playback devices all need to be seriously tuned for this. You're not experiencing these ultrasonics with headphones. You're not going to faithfully recreate some ultrasonic interference pattern in a random room with a pair of tower speakers placed in any arrangement. If anything, you're more likely to just make extra noise that wouldn't have been there originally from those interference patterns happening haphazardly and chaotically, coupled with the fact the vast majority of audio gear isn't designed to reproduce ultrasoincs (they're often specifically made to try and not generate them!) And even then, by recording in the room with the instrument, you are capturing the resulting ultrasonic interference that you could have experienced if you were there.
• soulofmischief 3y ago
  Your first point stands up to experimental scrutiny, but your second needs qualification: Anyone can be trained to pick up the differences between 320kbps mp3 and lossless formats. Compression kills the high end, and learning to recognize tell-tale compression artifacts will forever ruin your ability to appreciate streamed music, low-bandwidth wireless audio systems, or just 320kbps rips of music, certain genres faring worse than others.
  • c0pium 3y ago
    That is absolutely not true. People can be trained to hear that difference when the testing is not blinded, however in triangle tests that ability vanishes.
    • soulofmischief 3y ago
      Maybe not anyone. I could be biased with sensitive hearing. I know that in grade school, part of the requirements for joining band, due to the size of my school and overwhelming demand, was passing an audiometry test, where we were evaluated on a few different contexts related to ability to discern detail in audio, such as pitch and volume. I remember being pulled away into the principal's office where some of the test administrators were present, and they accused me of cheating and demanded to know how I did it. Apparently, I was the only student in the entire state to get a perfect score on that test, at least for that particular year. Unsure if they were implying I was the first ever, but that seems ridiculous to me because passing the test boiled down to just paying close attention. So I really don't know. Maybe the average person can't hear it, but I know just what to look for in the high-end and usually guess even 320kbps mp3 correctly from my own self-tests wherein I would randomly select between different encodings of a music file. I'm confident I would do well in an administered ABX test if I'm simply being tasked with finding the difference between mp3 lossy encodings and a lossless reference.
      • saltcured 3y ago
        I don't claim to have golden ears but I am a picky listener. I've been pretty satisfied with good MP3 encodings like 320kb/s though I'm not sure I'd agree they're indistinguishable. I had an experience I mentioned elsewhere in this discussion, about hearing a difference between 24bit 96 kHz and CD-quality in a sound studio. I don't know whether the operator screwed up or what. It doesn't mean I don't think CD quality is "good enough". But that's not the same question as "non-discernable". On the other end, I have had a few encounters with satellite radio (SiriusXM) and I cannot understand how that product exists. In each car I've encountered it, I've had a visceral reaction and wanted it turned off after a few minutes of trying to listen to it. It put me somewhere between anhedonia and dysphoria.
• goalieca 3y ago
  > 16-bit audio can perfectly reproduce audio as far as we can physically tell. Imagine encoding a sort of real world dynamic range across 16-bits. This would go from 0db to 100db in volume. This would need more than 16-bits which yields an SNR of about 96db. The dB values are different and not comparable but you can see we don’t capture the full dynamic range of human hearing very well.
  • mrob 3y ago
    Humans don't have 100dB dynamic range across their full hearing spectrum. We're less sensitive to high frequencies, which means you can apply high-frequency dithering to improve the dynamic range without adding audible noise. https://en.wikipedia.org/wiki/Noise_shaping https://en.wikipedia.org/wiki/Noise_shaping
• ptx 3y ago
  What about when the output of the lossy codec is passed through another lossy codec, e.g. MP3 through AAC over Bluetooth? I would expect better results (from the second codec) when starting from a pristine lossless source.
  • dTal 3y ago
    Or just poor listening conditions. Everyone always talks about needing super high end setups to hear the difference, but it's actually the low end setups with wonky frequency responses that totally mess up the assumptions behind psychoacoustic masking.
  • criddell 3y ago
    That would be an interesting experiment. Take a hi-res file and encode it with encoder A then B then A then B then ... How many encodings does it take before a trained listener using good equipment in an ideal setting can tell?
  • CharlesW 3y ago
    > What about when the output of the lossy codec is passed through another lossy codec, e.g. MP3 through AAC over Bluetooth? I would expect better results (from the second codec) when starting from a pristine lossless source. It's true that, technically, you'll get better results from the second codec when starting from the uncompressed source. Generally, it's always better to avoid unnecessary generation loss. That doesn't necessarily mean that you'll hear a difference since that depends on the cumulative output quality.
  • rplst8 3y ago
    This is a great example of why I prefer lossless source material. I can tell an MP3 from a CD nearly every time (on recordings of good original quality) once it's sent through Bluetooth.
• replete 3y ago
  We AB tested 16-44.1 and 24-96 versions of some really good classical recordings recently - you need good listening equipment (ears and electronic) but undoubtedly the dynamic range and top end (particularly) sounded better. It really depends on the listener, the source, and the equipment. A few years ago I did lots of AB testing with some Sony xm1000w3s (Sony LDAC) and Tidal Hifi with some 24bit masters and it was an incredible experience that changed my mind in the whole "640K.. 16bit is enough" argument.
  • nyolfen 3y ago
    "you need more than anecdotal evidence" "have some anecdotal evidence"
    • replete 3y ago
      [flagged]
    • replete 3y ago
      Apologies, that flippant comment was super rude and juvenile and I am taking it as a signal that I really need a break, I don't normally speak in this way. Yikes. Also totally fair, I shouldn't be hastily writing comments but I am interested in audio and wanted to share my experiences, naively. I have some 'hires' recordings that I can't tell the difference between a CD at all and do not care for, and some where I hear more details (on very high end), more separation between instruments - and from what I'm reading it seems more like this is a mastering issue. The difference on some of these recordings enable a kind of subjective 'holographic' spatial effect in me (perhaps the cause of my emphatic response) and it seems I have probably falsely attributed this to the higher resolution as the factor.
      • nyolfen 3y ago
        i appreciate your apology and i didn't mean it as an attack on you personally, it was just meant to highlight the irony. perhaps i could have phrased it more sensitively
  • 3y ago
• asveikau 3y ago
  > No, no one even has the biological ability to. 44100khz, 16-bit audio can perfectly reproduce audio as far as we can physically tell. The only reason to store anything higher is for production or archiving (that is, for computers to listen to). I'm not an expert, but one claim I saw somewhere is that a higher bit width and sample rate is good for people who are mixing and doing audio processing, even where the final result might get downsampled to 44100 hz and 16 bits per sample at the last stage.
  • SSLy 3y ago
    that's what 'production' in the quoted passage means.
  • ok_computer 3y ago
    24 fixed bit and 32 variable/ floating bit rate masters have more head room that may avoid clipping but doesn’t guarantee that. 48 or 96 kHz is useful for time stretching and maintaining fidelity (maybe other post processing without aliasing). That is all intermediate formats and doesn’t really say anything about what is best for consumers like the standard mastered cd quality at 16 bit 44.1 khz. Bandcamp is a cool market because I can download wavs from albums to store on my phone. You can see what people use as masters and its all over the place. There are many 96khz masters around and 24 bit depth is popular. I have a usb audio IO that supports 192KHz across 8xin+out. Those file’s just clog up hard drives so I figure 96 is good enough for bat music.
    • ok_computer 3y ago
      Also, I’ll note that I think the amp and speakers are far more contributing than the master file format. And the quality of the master and mix and tracking even moreso. I’ll run youtube rips of dj sets through some light hardware compressors and preamps and it sounds great. You cannot have specs determine quality.
  • Blackthorn 3y ago
    Yes, that's for antialiasing headroom purposes during the production process.
• Slow_Hand 3y ago
  There is one situation where 44k/24 bit and 88k/24 bit CAN sound appreciably different, and that's when aliasing is introduced into the recording, mixing, or the sample rate conversion. If proper precautions are not taken during the recording/mixing/mastering phases aliasing artifacts can be heard in the recording. This may account for the differences that some people hear when judging whether there are differences between the two. Higher sample rate files are more permissive of aliasing and exhibit less perceptible artifacts. So you're less likely to hear it at a higher sample rate. The artifacts of aliasing manifest as inharmonic distortion that starts at the top octaves and then folds back into lower frequencies as the effect is intensified. This can be easily perceived by most listeners if it is pointed out to them. It is not a pleasant effect like first-order or second-order distortion. It does not compliment the record at all. That said, if proper precautions are taken to mitigate latency artifacts during the record-making process then a listener shouldn't perceive any difference between a 44k and an 88k record. The best case scenario is often a record that's recorded, mixed, and mastered, at high sample rates, even if it's ultimately be down-sampled to CD quality (44 kHz).
  • ReactiveJelly 3y ago
    So if you had an 88k recording, you could run it through a well-known anti-aliasing filter to create a 44k recording that sounded the same? So the only situation where 44k and 88k can sound wrong is if... the 44k file is different and wrong?
    • high_priest 3y ago
      If you think about this "aliasing" as in, what occurs in 3d graphics, then you can understand this. What these 3d fiters do is either remove infirmation with blur (FXAA) or use information that is not available in the image (MSAA and derivatives) In audio recording, sampling at 88k would be like generating MSAA x2 image, so it can be displayed with higer fidelity, despite the outgut resolution being in lower 44k sampling rate.
      • saltcured 3y ago
• beebeepka 3y ago
  "The human eyes can only do 24 fps" and "if I can't do thing, then no one can" all over again.
  • zepolen 3y ago
    No one ever said the human eyes can only do 24fps.
    • TylerE 3y ago
      No one who knows what they’re talking about, but I’ve absolutely seen that argument advanced on multiple occasions, albeit usually with 30 rather than 24fps.
      • beebeepka 3y ago
        Last time I heard this crap in real life was back in 2018. Things have progressed immensely over the last decade, though. Now almost everyone "knows better" due to relentless marketing from big companies, including phone and TV vendors. Before that, 30 was totally fine for the masses. In fact it was preferable. Cinema is the last big holdout and, apparently, it's going to take at least another decade before even mere 48 is standard. As someone who has been riding the 120+ fos for over two decades, going to the movies is awful, especially action scenes and panning.
        • throwaway54_56 3y ago
          24 fps is the standard for cinema because that gives the preferred look for most content, with nice looking motion blur and whatnot. High frame rate may make sense for some movies, but it's not a win for the whole industry to go 48 or 60 or higher. I'm not sure what you're getting at with the 120 fps comment, because that is obviously not the frame rate of the finished product, so it's not the same conversation.
          • beebeepka 3y ago
• daneel_w 3y ago
  You can absolutely hear the difference between 44.1 and 96 kHz sample rate. Even with typical reproduction filters on the output, sampling at 44.1 kHz prevents you from accurately preserving e.g. a sine tone at above ~6 kHz, and that's even without taking into account all the aliasing problems you're facing when the samples don't align with with the peaks of this or that tone. 44.1 kHz is "good enough", but it's not accurate, and you can definitely tell the difference. As for anything beyond 16 bits amplitude on line level, no, you cannot hear a difference. For such a low-voltage signal the resolution at 16 bits is so fine that it already drowns in all the natural noise and THD in the cables, in the amplifier, in your speakers/headphones etc.
  • pja 3y ago
    Please do watch & internally digest the explanatory videos at https://xiph.org/video/ https://xiph.org/video/ It explains why you’re wrong in easily digestible terms & how a 44kHz sample rate will accurately encode signals right up to the Nyquist limit. The second video is an end to end demo showing the process in action.
    • user_7832 3y ago
      > https://xiph.org/video/ https://xiph.org/video/ Thanks a lot for those videos, they were absolutely excellent. For anyone wondering they're presented by "Monty", the guy behind the ogg container and vorbis codec. I probably understood 10% of what he said but that's still a lot.
  • temp0826 3y ago
    It's been a long time since my DSP classes at uni, but I don't think this is true. 44.1kHz sampling is enough to reproduce up to 22.05kHz sound accurately without aliasing. Unless there is another type of distortion you might be picking up. This stuff is pretty far out of my realm these days. https://en.wikipedia.org/wiki/Nyquist_frequency https://en.wikipedia.org/wiki/Nyquist_frequency
    • daneel_w 3y ago
• sircastor 3y ago
  And it’s worth noting that if you’re of a certain age - and generally that age correlates closely with the ability to afford equipment that can reproduce the very high quality you’re demanding - your hearing has likely deteriorated past the ability to discern the difference.
  • mrob 3y ago
    You don't need super high-end equipment to hear subtle details in audio, you just need reasonably good headphones. A few hundred dollars worth of headphones will get you quality that would cost tens of thousands with speakers and the room treatment speakers need to perform at their best (digital room correction can make a good room sound great but it can't fix a bad one).
    • elzbardico 3y ago
      The same can be said about loudspeakers, a good set of loudspeakers is far more important than buying a super-expensive set of DACs, Pre and Power Amps.
• saaaaaam 3y ago
  Some time ago - though not as far back as this article was published - we did an experiment at a conference that we held in the demo facilities of a Very Well Known Audio Company. We played a range of snippets of music - rock, classical, electronic, pop - at various qualities over what was quite possibly the best sound system in the world. The audience was a significant number of record label executives, distribution execs and general audio/music industry experts. We played pairs of the same snippet and asked people to tell us which was higher or lower quality. One person got them all correct. Turned out he’d mastered one of the early tracks we played so had a good reference and then used that as a baseline for the others. Everyone else it was completely scattershot. It wasn’t a controlled experiment but it was definitely interesting.
  • lmm 3y ago
    This is why an ABX test is the way to go. If you don't know which version of a snippet is "right" there's no way to objectively say which one is "better" - maybe you like the distortions (cf the famous vinyl "warm sound").
  • zigzag312 3y ago
    > asked people to tell us which was higher or lower quality This test didn't measure what you probably wanted it to measure.
    • saaaaaam 3y ago
      What did I probably want it to measure?
      • zigzag312 3y ago
        My guess would be, if there was a perceivable difference in quality between sample pairs.
        • saaaaaam 3y ago
          Genuinely interested to know why - without being present - you think this would not have provided some sort of measurement of this. While it wasn’t scientifically devised it also wasn’t just “play a couple of things”.
• nix0n 3y ago
  > The listening test linked in the article leads nowhere, I would have liked to see their methodology. Here you go: https://web.archive.org/web/20080322114622/https://www.stereophile.com/asweseeit/1007awsi/index.html https://web.archive.org/web/20080322114622/https://www.stere...
  • fuzzfactor 3y ago
    Here's the full results from 2013 by the original blogger: https://archimago.blogspot.com/search?updated-max=2013-02-24T21:57:00-08:00&max-results=15&start=51&by-date=false https://archimago.blogspot.com/search?updated-max=2013-02-24... Starts about halfway down the webpage.
• ScoobleDoodle 3y ago
  What quality and power speakers are needed to get good output from the files so it can be heard?
• m463 3y ago
  I wonder how far we still have to go. Computer graphics is pretty good, but how does it compare to walking out into a bright sunny day. Audiowise, I wonder how listening to live music, then listening to something that went through capture and playback end-to-end. I'll bet there are differences and I wonder where the "bottlenecks" are.
  • vel0city 3y ago
    If you're listening to most professional live sets that has amplifiers you're listening to something that went through "capture and playback end to end". Loads of boards are digital these days, and you're probably going to be altering the mix and EQ and going through compressors and all kinds of stuff in between.
• dghughes 3y ago
  > Can you hear the difference Do you think it matters if I play the song on my $10 cheapo earbuds or on $60,000 Sennheiser HE-1 Summit headphones?
  • brewdad 3y ago
    On $10 cheap buds? Yes. On $100 middling buds? Very, very few people will notice a difference.
• joshspankit 3y ago
  I wish years ago we would have switched to A/B testing taking weeks or months for each side.
• dwroberts 3y ago
  Worth noting 48KHz audio is now a commonly encountered standard for video. Not to say it’s necessarily audibly discernible from 44K but it’s obviously not quite as straightforward as 44K being the end of the story.
  • lmm 3y ago
    48 is just about convenience, it's not meant to be "better" than 44. 45, 46, 47 or 49 would be fine too, but 48 is a rounder number.
    • rplst8 3y ago
      AFAIK, 48k was preferred over 44.1k in movie and cinema recordings because it divides evenly by 24. Both 44.1k and 48k are evenly divided by 30 and 60 fps. However, none of that really matters for TV where the actual frame rates are 59.97 and 29.98 to reduce flicker with US household mains frequency. In the early days of digital audio, and before oversampling was possible, the anti aliasing filters were analog circuitry. It's very difficult to cheaply implement a 20kHz brick wall filter with a 2kHz sideband. Doing it in 4kHz yielded better results at the cost of slightly faster ADC designs. I believe this is why 48kHz designs got the foothold in professional audio circles. The analog parts of those designs were WAY better sounding. Once oversampling became common and affordable, the anti aliasing filters where implemented much more easily in the digital domain.
      • adgjlsfhk1 3y ago
        > to reduce flicker with US household mains frequency This isn't the reason for the odd frequencies (and black and white television was exactly 60i). The reason for the odd frame-rate is that you need to do some very fancy math to fit the color data in the same analog signal without breaking backwards compatibility.
        • rplst8 3y ago
          Ah you're right. It still breaks the "even division" argument though.
• rplst8 3y ago
  Agreed on the 44.1k 16-bit is all you need as a delivery medium for a final audio product. However higher bit rates and sample rates are needed for multi track recordings so that during the mixing stage and mastering, the fidelity is preserved when _math_ causes rounding errors and what not. Unless, you are using nondestructive editing. As for listening to the final product, i.e. store bought CDs and their equivalent MP3 and AAC rips... I can often hear the difference in specific recordings, no matter the bit rate, because the perceptual encoding schemes often butcher certain recordings. For example, on RUSH's Red Barchetta from Moving Pictures, there is a synth intro that slowly vamps in volume. Every MP3 encoder (that is normally worth it's salt) I've ever tried encoding that with outputs a garbled, distorted, electronic sounding distortion. It clears up immediately once it reaches full volume, but during the crescendo it falls on it's face.
• jtriangle 3y ago
  If you know what to listen for in the high frequencies, you can pick out mp3's pretty easily, provided said mp3's have that sort of content. The question really isn't 'can you tell', it's 'does it matter', and, well, most of the time, no, it does not, even for lower bitrate mp3's. There are many people, of course, that don't like the idea of lower quality audio, and they can tell at least sometimes, so they 'dislike' mp3 in general. That's all well and good until they start saying silly things like 'mp3's sound bad', which is not true in any sense.
• MuffinFlavored 3y ago
  > Can you hear the difference between 320kbps MP3 or the equivalent, and CD-quality lossless? I find it hard to believe an entire industry exists with high-end audio equipment ($100k+ on speakers/receivers/room treatment) just to play 320kbps MP3s?
  • vel0city 3y ago
    The poster was talking about digital formats and compression. The expense of speakers and receivers is usually higher quality analog reproduction. Two vastly different concepts at play there. Even then going into the $100k+ territory usually yields very little results on top of the many thousands of dollars setup. Its probably better spending at least half of that just making the room and the other sound reflecting/absorbing things better if you're going that high-end.
  • RockRobotRock 3y ago
    It's also hard to believe homeopathic medicine exists.
• midoridensha 3y ago
  [dead]
• ekianjo 3y ago
  > Can you hear the difference between 320kbps MP3 or the equivalent While I agree with the premise this also depends massively on the equipment used to reproduce the sound. If you have a good amp and large speakers you are more likely to notice than your cheap headphones or thru your crappy laptop speakers.
• dclowd9901 3y ago
  I see audio storage like I see cars: there’s no such thing as a “best” version — more like the right version for the right time. On a plane, I’m glad I can pull down a ton of albums on Spotify and listen to them without access to internet connectivity. For Eno, Steely Dan, Roxy Music and Thomas Dolby produced albums, I’m glad for lossless. For sitting at home on a Friday evening with a bourbon in my hand and the lights down low, nothing beats my vinyl player. I could get into the cars side of it too but that’s more a “me” thing than a “relevant to this conversation” thing.
• zigzag312 3y ago
  The MP3 study you referenced uses flawed methodology: > participants were asked to listen to both versions as many times as needed and to choose the version they preferred in a double blind A/B comparison task I have trouble finding high quality studies comparing 16-bit to 24-bit audio. This one one is kind of interesting: https://www.researchgate.net/publication/338989993_Study_on_Comparison_of_24_with_16_Bit_Headphone_High-_Resolution_Music_Box_Audio https://www.researchgate.net/publication/338989993_Study_on_...