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24/192 music downloads make no sense

• iamleppert 11y ago
  This is totally offtopic but I can't stand the "XXX considered harmful" stuff. I had to rage-quit the article.
• Retra 11y ago
  They are useful if you're resampling them or editing them, but I doubt that's something consumer music services are overly concerned with.
  • saidajigumi 11y ago
    I'll note that's the entire point of Monty's (great) article, which has this near the top: Unfortunately, there is no point to distributing music in 24-bit/192kHz format. Its playback fidelity is slightly inferior to 16/44.1 or 16/48, and it takes up 6 times the space. This has all been known to anyone with actual signal processing and/or audio engineering knowledge for a long time now. As in, common knowledge to the kinds of folks attending the AES conference at least back to ~2001 or so. The high sample rate/bit depth stuff is useful for production process, but irrelevant for final distribution.
    • thwest 11y ago
      There's a reasonable argument that fits within DSP theory that frequencies sampled above audible range could have harmonics down in the audible range.
      • JonnieCache 11y ago
        This is addressed in the article. While theoretically relevant to some recording applications, (overdubbing a string section one violin at a time, why would you want to do that?) this kind of intermodulation distortion can only harm the reproduction of mixed material.
  • deleted 11y ago
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  • kuschku 11y ago
    Or if you apply an equilizer, like lots of people do in consumer applications.
• andy_ppp 11y ago
  Why do high quality DACs clearly sound better then? And they sound better with better files. Maybe it really is all in my head but I mean listening to a £20000 hifi the other day (vinyl) really just shocked me. I was listening to Marvin Gaye on my friends system and I could hear that there were several different backing singers all moving and at different distances from the microphone. Are there any double blind trials anywhere of Vinyl/CD/24-192khz with super high end hifi systems? Mostly I see people suggesting that these tests are performed from the phono output of a mac with a pair of average ear buds...
  • JonnieCache 11y ago
    >Why do high quality DACs clearly sound better then? And they sound better with better files. Maybe it really is all on my head but I mean listening to a £20000 hifi the other day... You were listening to £20,000 worth of amps and speakers, and you were most likely in an acoustically treated room. Also, novelty is almost always euphonic when it isn't overtly bad. This fact is often neglected. You hear something you didn't hear before and your brain immediately tells you that it sounds better, even if it doesn't actually represent higher fidelity. Actually making an objective judgement requires a careersworth of experience, or a test lab and the skills to use it. For example: you were listening to vinyl, which is covered in delicious noise and warm harmonic distortion, and is mastered differently. Highly euphonic, very novel if youve only ever heard the CD version before, but definitely not higher fidelity. BTW higher end DACs do sound better, but the rest of your signal chain needs to be really good for you to notice it. It's often to do with better phase accuracy between the left and right channels, which affects the soundstage, or stereo image. If your speakers/amp have loose timing however, you'll never be able to tell.
    • sanoli 11y ago
      > higher end DACs do sound better This hasn't passed the blind tests either. A good, 100 dollar dac (a schiit or an odac) will sound just as good as a 1000 dollar dac.
      • JonnieCache 11y ago
• splitdisk 11y ago
  From my experience, what matters more than sample rate is 24 bit vs. 16 bit sampling in the recording/production process. Using heavy compression and EQ can mean that very quiet sounds can become louder, in this case 24 bit recording is ideal. Sample rate wise, anything above 40khz is fine for most ears (I've probably lost a few khz in the upper range anyways) Another note is that most converters operate at a multiple of 48K, so it makes sense to use 48/96khz if you are recording. It all comes down to how much disk space you have, and want to use up.
  • klodolph 11y ago
    I can still hear around the 20k range, so let's not exclude a few listeners just because we wear hearing protection when it matters. In practice, 20k audio content means 44k or higher sample rates, due to the fact that actual filters have finite transition bands. There's an unfortunate history of engineers with poor hearing who inflict pain on others, such as the horizontal retrace on NTSC TVs which still annoys me when I encounter one. 24bit also means we don't have to record at 0dBFS, which saves a lot of time.
    • splitdisk 11y ago
      I'm very jealous of your healthy hearing range... I now wear hearing protection, but the damage has been done. When it comes to file storage, I will take a 44.1/48K 24 bit FLAC happily, since it usually comes out to the same size as a CD-Q WAV file anyways. I see no reason why everything shouldn't be in this format, but CD's have made a serious dent in formatting standards/habits.
      • klodolph 11y ago
        24bit has been shown to be excessive in actual tests, and it doesn't really pass the "bullshit test". 24 bit allows you to switch between a whisper in a library and a jet engine or shotgun blast in a single recording—yes, it would instantly damage your hearing. By comparison, 16 bit audio can "only" record a whisper in a library and a motorcycle or jackhammer. Double blind tests show that 8 bits are not enough, but 14 bits are.
• TerryADavis 11y ago
  The nigger is a sad stubborn stupid fuck. The CIA has no engineers, just psychologists.
• hatsunearu 11y ago
  The single worst thing that harms audio quality is excessive compression. It's ruining everything, I'd say. Heck, I'm sure everyone who knows their worth would agree that compression harms audio quality. Recommended reading: https://en.wikipedia.org/wiki/Loudness_war https://en.wikipedia.org/wiki/Loudness_war edit: compression as in dynamic range compression, not data compression like mp3 in audio
  • muraiki 11y ago
    Reading "compression" as both a programmer and audio-minded person made that first sentence difficult to parse at first. :) But yeah, it's obvious when I have my car stereo nearly on max to listen to classical or jazz, and then if I turn the radio and get a pop music station my ears are about to explode.
    • JonnieCache 11y ago
      Don't worry, the loudness war is basically over. itunes and youtube normalize everything to -15dbRMS and -13dbRMS respectively. Spotify uses the ReplayGain algorithm. Broadcasters worldwide now use the EBU R128 standard, which is a more effective form of loudness normalization, hopefully to be adopted by online services too at some point. All that's required is for personal media players to widely implement loudness normalization (by default) for preexisting records in peoples collections. http://productionadvice.co.uk/lufs-dbfs-rms/ http://productionadvice.co.uk/lufs-dbfs-rms/
      • muraiki 11y ago
        Yeah, I should have read the wikipedia article in full before commenting. I'm glad the war is over!
      • hammock 11y ago
        The normalization is song-by-song to my knowledge. It doesn't normalize the audio within the song itself. (Over)compression is a key part of the aesthetic of a lot of pop hits today, and while in the face of normalization you won't be able to make your song overall louder than the next, it won't necessarily stop you from creating a "wall of sound" waveform if that's the sound you're looking for.
• sliverstorm 11y ago
  To what can I attribute the consistently horrible quality of 64kHz streams ten or fifteen years ago? Would that fall under the "bad encoder" bucket? Edit: christ, I mixed up bitrates (e.g. 192kbps) with sampling frequency (e.g. 192kHz) again. I was referring to 64kbps streams.
  • CamperBob2 11y ago
    64 kHz isn't a standard sample rate -- you're probably thinking of the bit rate of an MP3 or AAC file. A 64-kbit MP3 does sound pretty awful.
    • sliverstorm 11y ago
      Yup. Further confusing me was the fact that (if memory serves) Apple did offer MP3's at 192kbps for a while, before upping to 320kbps. Edit: apparently my memory is worse than I thought.
      • epmatsw 11y ago
        iTunes has offered 128kps AAC and 256kbps AAC files. Now it's only the 256kbps versions.
      • Xixi 11y ago
        Hasn't Apple always been offering AAC? At first at 128 kbps and then 256 kbps.
      • comex 11y ago
        Apple doesn't sell 320kbps anything, but 256kbps AAC, which is probably better than 320kbps MP3.
  • joosters 11y ago
    mp3 encoders have gotten better over time. As well as general improvements in fidelity, older encoders had bugs that would cause occasional terrible encoding for fragments of a sample.
• guelo 11y ago
  (2012) Previous discussion https://news.ycombinator.com/item?id=3668310 https://news.ycombinator.com/item?id=3668310
• fla 11y ago
  With nowdays bandwidths, why do we keep using destructive compression for songs?
  • icegreentea 11y ago
    Lossy vs non-lossy compression is orthogonal to the sampling rate and bit-depth (which is what this article is about). While the MP3 standard effectively means sampling more than 48kHz is useless, there's no reason you can't have a lossy comprssion scheme that attempts to capture higher frequencies.
    • joosters 11y ago
      But the article makes a compelling case for why > 48kHz is completely pointless.
  • Avenger42 11y ago
    A lot of listening today occurs through streaming services, and huge uncompressed songs will eat into data plans quickly.
• earlz 11y ago
  I can't hear a difference between 96khz/44khz in it's raw form. However, I can tell the difference from effects in audio mixing. The extra detail can really make a difference in how well an audio effect VST works. I have a 96khz/24bit interface that I use and ATH-M30X headphones, and I can tell a difference between at least some 24bit FLAC files and 16bit highest-quality-possible MP3s. I was mixing my own music and the difference was quite obvious to me. The notable thing was that drum cymbals seemed to have a bit less sizzle and such. Now that being said, if I hadn't heard the song a million times in it's lossless form from trying to mix it, I probably wouldn't have noticed, and even then it didn't actually affect my "experience". I'm one of those guys that downloads vinyl rips as well, but I do that mostly just to experience the alternative mastering, not that I think it's higher quality or anything. (though I have heard a terrible loudness-war CD master that sounded great on vinyl with a different master)
  • pizza234 11y ago
    Compressed formats are not really relevant to considerations about the bit depth itself. Besides, mp3 [audio] compressions have difficulty in handling specific samples, or type of samples (eg. sharp attacks), and they may manifest artifacts independently of the bitrate; MP3, AFAIK, also has a ceiling of 320 kbps within the standard specification, which certainly doesn't help. Secondly, I'm not sure if you process further the MP3s (when you refer to mixing), but if you do, you're definitely going to make noticeable, artifacts which weren't so in the unprocessed MP3 form.
    • earlz 11y ago
      I mean, I'm just some hobbyist, but my understanding is everyone renders to lossless and then converts that to the various MP3/AAC formats, never changing anything solely because of the final compressed format.
    • hatsunearu 11y ago
      Old thread, but I thought I made it clear that I wasn't talking about data compression, but rather dynamic range compression.
  • 11y ago
• Johnny555 11y ago
  Wouldn't this question be answered with a large-scale double blind trial? If more people prefer the sound at the higher bitrate and sampling rate, then that's the better format, even if there's no technical reason why that format is superior. Much like how some people prefer the "warm" sound of tube amps, even if that means more distortion.
  • upofadown 11y ago
    From the article: >Empirical evidence from listening tests backs up the assertion that 44.1kHz/16 bit provides highest-possible fidelity playback. You can read the article if you want to find the actual references. No one is arguing that higher rates/bits produces any sort of distortion that anyone would prefer.
  • TheCoelacanth 11y ago
    > Much like how some people prefer the "warm" sound of tube amps, even if that means more distortion. The difference from my perspective is that an amp is a tool for sound production while a digital music format is a tool for sound reproduction. When producing sound, choosing more distortion over less distortion is a valid choice. When reproducing sound, the goal should be accurate reproduction of the original.
• lmm 11y ago
  Dithering is a horrible thing to be doing, and 44.1 is an awkward rate. So while I agree that 192khz is dumb, 24/48 is a better standard than CD.
  • ska 11y ago
    No, dithering (properly) is usually what you should be doing when you quantize. See Vanderkooy and Lipshitz 1987 for why.
    • lmm 11y ago
      Paper seems to be paywalled. I can't imagine any possible purpose for dithering before encoding that wouldn't be better served by dithering on playback.
      • deleted 11y ago
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      • ska 11y ago
        You dither before quantization in order to decorrelate the quantization error. If you don't do this, you risk artifacts in any digital filtering (or for that matter, playback) done afterwards. This includes any requantization. In audio if I recall correctly it is also important to avoid obvious noise modulation.
      • nullc 11y ago
        At 16 bits dithering is probably pointless for listening purposes. What dithering does is it decorrelates the quantization noise with the signal. Absent it, quantization generates harmonic spurs. In theory, on a very clean and noiseless signal these harmonic spurs might be more audible than you'd expect from the overall quantization level. In practice, 16 bits is enough precision that these harmonics are inaudible even in fairly pathlogical cases. But quantization eliminates the potential problem by replacing the harmonic content with white noise. Adding noise on playback just adds noise, it would not remove the harmonic generation. The _best_ kind of dithering scheme is a subtractive dither, where noise is added before quanitization and then the _same_ noise is subtracted from the dequantized signal on playback. This is best in the sense that it's the scheme that completely eliminates the distortion with the least amount of additional noise power. But it's not ever used for audio applications due to the additional complexity of managing the synchronized noise on each side.
• derefr 11y ago
  So, no point in 24/192 because it makes no difference in playback... but having lossless downloads is important in part for enabling remix culture? There's a bit of a double-standard here. Maybe I can't hear 24/192 audio, but isn't it better input for sampling?
  • some-guy 11y ago
    The article is specifying 24/192 as useless for playback quality only. Halfway down he addresses the benefits of 24/192 for the sake of mixing and mastering different digital audio signals, but a final mix offers no benefit to the human listener when choosing between 16bit/44Khz and 24bit/192Khz.
    • derefr 11y ago
      What I was trying to get across is: every file has two potential purposes—listening and serving as input for sampling. So, if we care about enabling "remix culture", wouldn't make sense to offer a "24/192 FLAC" option for download, push DVDA over CD, etc., anyway? I've never seen the hype from artists about 24/192 as being about better listening experience. It's about handing their consumers a better master so as to encourage and enable more of them to be remixers.
      • some-guy 11y ago
        Yeah I think that's not obvious from the title of the article: 24/192 are useful downloads for the sake of editing.
        • deleted 11y ago
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          • derefr 11y ago
            The number after the slash in this conversation is 192kHz (sample rate), not 192kbps (compressed bitrate). The "44100Hz PCM audio" you see on your CDA/WAV/FLAC source, before lossy encoding, would be "192000Hz PCM audio" instead. Unlike bitrate, this number does not affect the "quality" of the sound (unless it's really low), but rather acts as a lowpass filter, dropping out the frequencies above 0.5x of it.
• yzhou 11y ago
  There's a big difference in impulse response with different sample rates, any one can see it on a oscilloscope, I bet some one can hear the difference. Those who don't have a oscilloscope can see the picture here: http://i.imgur.com/wY0wzcW.png http://i.imgur.com/wY0wzcW.png
  • yzhou 11y ago
    The brick wall filters used on low sample rate sound cause ringing in the time domain, which can "blur" the neighboring impulse.
  • yzhou 11y ago
    In the real world, it is almost impossible to make a voltage divider with 24bit resolution. So all the DAC makers have to convert 24bit audio into lower bits(6bits to 1bits), this step requires oversampling the original audio. It is a lot easier to oversample a 192KHz/24bit audio than a 44.1K/24bit audio, and the ringing is much less after oversampling the 192KHz/24bit audio.
  • fancyketchup 11y ago
    The two pictures don't have the same vertical scaling, and it's clear that the probe is ahead of the LPF in the signal chain.
    • yzhou 11y ago
      the probe is placed on the headphone jack. The difference in the vertical scaling is that 0dB DSD signal is 6dB below a 0dB PCM.
  • nullc 11y ago
    What you are showing is _precisely_ the effect of low-passing, nothing more, nothing less. See the digital media primer 2 for more information on that: https://wiki.xiph.org/Videos/Digital_Show_and_Tell https://wiki.xiph.org/Videos/Digital_Show_and_Tell If humans were able to hear audio above 22kHz (or what not) in any meaningful way, we'd expect to be be able to demonstrate that effect in carefully controlled studied and then that lack of low-passing may matter; but that isn't what the best evidence so far shows.
• ChuckMcM 11y ago
  I have a pair of Roger Sound Labs studio monitors for my speakers at home. I got to look at their insides when a technician was replacing a blown midrange speaker (they have a "lifetime" warranty, however that warranty expired when RSL did). Looking at the cross over filter network I could see a network selecting for frequencies > 20khz and it was shunted to a resistor. I asked about it, and the reponse was exactly like the authors, by filtering out signals higher than the tweeter could reproduce, they improved the listening experience. It made sense to me, and I love how the speakers sound. Understanding is not inserting distortion makes even more sense.
• pgrote 11y ago
  So, what are the better settings for ripping songs?
  • aidenn0 11y ago
    when ripping songs you are probably starting out with either 44.1 (CD) or 48kHz (DVD) sound. Just keep whatever the native sampling rate is.
  • Synaesthesia 11y ago
    Compression wise I go for 256kbps AAC. It's quite superior to MP3 as a codec.
  • LeoPanthera 11y ago
    I rip to FLAC, not because I think it sounds better, but simply because if some newer better codec comes along in the future that will let me compress my songs on my smartphone even smaller (Opus?) I don't want to have to get my CDs out again. I can just transcode from the FLAC files.
• tonecluster 11y ago
  Some [consumer] digital low-pass filter can benefit from higher sampling rates, leading to an overall better representation of the analog signal up to 20kHz. But there are diminishing returns as the filter "folds" the octaves above 22kHz; A rate of 96k for certain lowpass filters is better than 48k, but at some point there's little (if any) benefit by going to 192k or 384k. For recording studios, go as high as you can in both bit-rate and bit-depth. Especially when you're processing the signal "in the box". Give the software as much data as possible to operate without introducing errors and artifacts. There are diminishing returns there as well, but RTFM for (for example) UA gear and software and you're good to go.
  • aidenn0 11y ago
    TFA mentions that for recording and mastering there is a use. Furthermore, the headline implies it see the term "downloads" in the title.
    • tonecluster 11y ago
      Yep, I read that too. Even so, there are low-pass filters in some consumer gear that benefit from, say, 96k sampling rates and result in better quality sound. This does imply that at 44.1 or 48 they don't represent up to 20kHz properly, of course.
      • aidenn0 11y ago
        Are you talking about digital low-pass filters? In the event that they are using one that benefits from a higher sampling rate, they could upsample before applying. Or just use a better low-pass filter implementation.
      • fancyketchup 11y ago
        Lossless upsampling the 44.1 kHz recording to, say 192 kHz is trivial for the reproduction equipment. That the LPF on the reproduction end wants the DAC to run at greater than 44.1 kHz has no bearing on the sampling rate of the distribution format.
• idlewords 11y ago
  I love the idea the author mentions in passing of a dedicated speaker assembly for ultrasonics. This seems like something that could be a huge margin business, and the parts costs would be as low as you wanted.
  • xiphmont 11y ago
    You're late to the game. Such high margin products have existed for some time. There are even published papers about them()! The published papers tend to be by the same people making the supertweeters
• some-guy 11y ago
  This article hits close to home: before I became a programmer I worked as an audio engineer at a fledgling studio in my hometown. The amount of misinformation / junk-science in the audio world is preposterous. There's a religious-cult of an industry that feeds off the ignorance and placebos of its participants. I have many friends who swear by their What.cd 24/192 FLAC vinyl rips and spend hundreds of dollars on audiophile AC wall outlets. Not to say that there are no differences in high-end audio equipment, but so much of what's "good" is subjective.
  • SSLy 11y ago
    this is the first mention of what.cd outside of the tracker scene i ever saw. funny when you think about it.
  • lfam 11y ago
    In the case of sites like what.cd, I think that FLAC 16/44 rips of CDs and vinyl are useful for creating distributed backups of our cultural corpus. But I agree that 24/192 FLACs of vinyl are ridiculous.
    • some-guy 11y ago
      I agree, in fact I very much like the sound of vinyl, but to say it's more "accurate" or of higher fidelity and dynamic range than 16/44 is completely false.
• rplst8 11y ago
  First, let me state that I believe that CD audio, played through a modern DAC and quality stereo equipment is pretty much the pinnacle of home audio listening. That is to say, I think 44.1kHz 16-bit PCM audio is plenty good and I'm in no rush to replace my CD collection, nor do I think significant investment in higher bandwidth audio (for playback, mixing and mastering are another story) buys you much. That said, there's one thing the article does not address and that is "beating", or really inter-modulation distortion from instrumental overtones. Instruments are not limited to 20-20kHz. They can have overtones well above this range. Additionally, note that short pulse-width signals, i.e. transients, like drum strikes, especially involving wooden percussion, can have infinite bandwidth. (Not really infinite, but pulse-width is inversely proportional to bandwidth. In a real listening environment (i.e. live performance) these overtones have a chance to interact with one another in the air. It is possible that these overtones may beat with one another and cause inter-modulation products in the audible range. For an example of this, play a 1000 Hz tone through your left speaker, and a 1001 Hz through your right speaker. You will hear a distinct 1 Hz "beat". The audibility of these are largely dependent on listening position and amplitude, but it is possible to occur with instruments. Since most recordings are done using a "close mic" technique (placing the microphone very close to the source) the interactions such as this are never recorded. However, if full bandwidth of the producing instruments is preserved, these interactions of the overtones can be reproduced in a playback environment given equipment having a wide enough bandwidth and degree of quality.
  • cynicalkane 11y ago
    Nope. Intermodulation distortion for out-of-range frequencies is inaudible. The 1hz beat you are hearing is not a 1hz sound wave, it's a 1000.5hz sound wave becoming louder and softer once per second. The comparison of a 1hz beat to a 1hz sound should be absurd on its face: you need about 20-30hz to become audible, and it's a low rumble more felt than heard. Very low frequencies sound absolutely nothing like intermodulation beats.
    • tonecluster 11y ago
      "...and cause inter-modulation products in the audible range." AFAIK, this is true in acoustic environments under conditions as described in the original post.
• weinzierl 11y ago
  Because digital filters have few of the practical limitations of an analog filter, we can complete the anti-aliasing process with greater efficiency and precision digitally. The very high rate raw digital signal passes through a digital anti-aliasing filter, which has no trouble fitting a transition band into a tight space. I always thought digital anti-aliasing filters were creatures from a fairy-tale world. Much talked about but no one has ever seen one. My understanding: If you have a an analog filter of a given steepness the only way to further reduce aliasing effects digitally is oversampling. Or less steep (cheaper) analog filter plus oversampling is the same as steeper (more) expensive analog filter. People tend to say digital anti-aliasing filters when they really mean oversampling. "24/192 music downloads make no sense" seems to be a thoroughly researched and carefully written article. It explains oversampling very well, possible confusion with digital filtering (anti-aliasing or not) is out of question. But then it goes on to talk about digital anti-aliasing filters, which makes me afraid I could be wrong. Do digital anti-aliasing filters exist?
  • hamiltonkibbe 11y ago
    Yes, but obviously they don't work if your signal is already aliased. The most common example would be when decimating a signal, e.g using a digital filter with a cutoff at 22khz before down sampling from 192kHz to 44.1kHz. This is often realized in a single step... Check out polyphase interpolation/decimation if you're interested in learning more
  • deleted 11y ago
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  • squeaky-clean 11y ago
    The digital anti-aliasing filter can only ever work on a digital -> digital signal, but they're still useful in the analog->digital process. > My understanding: If you have a an analog filter of a given steepness the only way to further reduce aliasing effects digitally is oversampling. Or less steep (cheaper) analog filter plus oversampling is the same as steeper (more) expensive analog filter. People tend to say digital anti-aliasing filters when they really mean oversampling You're right, and it's actually both. The ADC can run at a much higher sample rate with a cheaper analog filter, and then that digital signal is again passed through a digital filter and downsampled.
• PaulHoule 11y ago
  It is not just headphones that are the problem, it is the speakers. People today are often amazed when they listen to CD or turntable content through 70's era crossover speakers. Back in the 70's you'd have a stereo with 2 "speakers" that each had 3 subspeakers for a total of six speakers. The fad today is to have 5.1 sound with a single driver in each satellite, also a total of six speakers. The spatial resolution increase is good for movies, games and TV but surround sound in music is marginal. An amazing number of old "classic rock" recordings were done in quad and anything by Donald Fagan will sound pretty good w/ Dolby Pro Logic, there are some more recent Bjork recordings, but almost everything is mixed for stereo and what you loose in frequency response is not compensated by anything, except perhaps the ability to produce more volume with more speakers.
• z3t4 11y ago
  I can hear insects and buzzing electronic devices, and my partner thinks I'm crazy some times. Thinking I might have golden ears I tested * my range and I could hear up to 18kHz. * http://onlinetonegenerator.com/hearingtest.html http://onlinetonegenerator.com/hearingtest.html
  • rubberbandage 11y ago
    Honestly, depending on your age, that still could be “golden” — I’m 31, I’ve taken very good care of my hearing, I’m very acutely aware of audio subtleties, and my hearing range tops out around 16.5KHz. The so-called standard upper limit of 20KHz really only applies to young children, which is why CD audio being able to reproduce frequencies of 22.05KHz is already beyond ideal, and calls for 48! no, 96!, no, 192! (or higher) is literally insane for playback.
    • lstamour 11y ago
      Using a tone generator on my computer and a pair of headphones, I found that I couldn't easily hear past 15-ish myself, then I started turning up the volume, or playing with turning the volume all the way up, then all the way down. Using that technique, I was able to distinguish noise and high pitches up to 20.2khz or so. So I think from now on, if I hear some whine, I'm going to trust that it's there and not my imagination. Of course, it's also the definition of going deaf, I suppose, that I have to turn up the pitches to such a loud volume to hear them in the first place...
• aidenn0 11y ago
  If you want to know more, Monty made one of the best intros to digital sampling I've ever seen: https://www.xiph.org/video/vid2.shtml https://www.xiph.org/video/vid2.shtml
• multinglets 11y ago
  This is clearly overkill, if you ask me, but let's not get too carried away with "44kHz 320kbps mp3 are always good enough and if you say otherwise you're a snakeoil-buying fool." For starters, there are people who can hear well into the 22kHz range. Mp3 is also based on a psychoacoustic model, and that model can be (and absolutely is) more accurate for some people than it is for others, and it is also subject to fundamental flaws (there are sounds that will always produce a compression artifact for even the latest encoders). It drives me absolutely crazy when people start circlejerking over "those dumb ol' flac people." I don't care how much theory you think you have; you don't know what you're talking about.
• gwbas1c 11y ago
  This article really misses the facts of the Nyquest-Shannon theory. In order to decimate a signal to 44.1 or 48khz, and preserve high-frequency content, high frequencies need to be phase-shifted. This phase-shift is similar to how lossy codecs work. For what it's worth: I'm a big fan of music in surround, and most of it comes in high sampling rates. When I investigated ripping my DVD-As and Blurays, I found that they never have music over 20khz. It's all filtered out. However, downsampling to 44.1 or 48khz isn't "lossless" because of the phase shift needed due to the Nyquist-Shannon theory. I still rip my DVD-As at 48khz, though. There isn't a good lossless codec that can preserve phase at high frequencies, yet approach the bitrate of 12/48 flac.
  • Hello71 11y ago
    no, it addresses it fairly clearly (albeit briefly): > So the math is ideal, but what of real world complications? The most notorious is the band-limiting requirement. Signals with content over the Nyquist frequency must be lowpassed before sampling to avoid aliasing distortion; this analog lowpass is the infamous antialiasing filter. Antialiasing can't be ideal in practice, but modern techniques bring it very close. ...and with that we come to oversampling. if you accept that the limit of hearing is around 20 kHz, then you must also accept that frequencies above that can freely be removed without loss of fidelity to the human ear. the article notes that higher frequencies can be heard, but only in the form of ultrasonic intermodulation distortion. (i.e. not in fact the higher frequencies at all)
  • itp 11y ago
    When you say > In order to decimate a signal to 44.1 or 48khz, and preserve high-frequency content, high frequencies need to be phase-shifted. What do you mean by high frequency? If you mean frequencies below but near the Nyquist frequency then no, there is no phase shift. If you mean at or above... I'm struggling to avoid a blatant appeal to authority here, but your position is that the author of the Ogg Vorbis coded doesn't understand digital sampling, which seems challenging to believe.
  • nullc 11y ago
    > In order to decimate a signal to 44.1 or 48khz, and preserve high-frequency content, high frequencies need to be phase-shifted. Your understanding of sampling theorem is incorrect. Sampling alone (not quantization, of course) is completely lossless under the critical frequency. We demonstrated this in a very clear way near the end, at about 21 minutes in, on the primer two video: http://www.xiph.org/video/vid2.shtml http://www.xiph.org/video/vid2.shtml where we show a square wave being phase shifted tiny fractions of the intersample length.
• gwbas1c 11y ago
  This article really misses the facts of the Nyquest-Shannon theory. In order to decimate a signal to 44.1 or 48khz, and preserve high-frequency content, high frequencies need to be phase-shifted. This phase-shift is similar to how lossy codecs work. For what it's worth: I'm a big fan of music in surround, and most of it comes in high sampling rates. When I investigated ripping my DVD-As and Blurays, I found that they never have music over 20khz. It's all filtered out. However, downsampling to 44.1 or 48khz isn't "lossless" because of the phase shift needed due to the Nyquist-Shannon theory. I still rip my DVD-As at 48khz, though. There isn't a good lossless codec that can preserve phase at high frequencies, yet approach the bitrate of 12/48 flac.
• S_A_P 11y ago
  I just recently purchased Izotope Ozone 7 advanced. One feature it has is "codec preview" which lets you "solo" the codec artifacts for MP3 and AAC format. Even at high but rates it's amazing how swishy bit reduction sounds. It also made me realize what I was hearing with mp3s was artifacts from compression. That said, it's not unlike tape hiss or vinyl noise. In fact I think it can have its own charm and in some cases make the music sound more full. It's also probably why 24/192 digital audio can sound so "cold" or lifeless.
  • beat 11y ago
    From mastering records at home, I've found that in all but the most golden-ears focused listening, I can't hear the difference between 192 bit mp3 and 44.1/16 cd quality. But 128 bit mp3 is audibly degraded and irritating. That's a pretty cool feature for Ozone 7, for sure! I'm still using Ozone 5 and don't feel a need to upgrade, but that might make it...
• StavrosK 11y ago
  Everything else is fine and good in the article, but I can see the infrared in the Apple remote (and all the other IR remotes I've tried). It's faint, but plainly visible. Am I the only one?
  • glitch 11y ago
    Went to a dark room with an Apple Remote; let my eyes adjust for a little while. Pressed it many times; I couldn't see the infrared coming from the LED with my naked eye. (But the camera on my iPhone imaged the infrared from the remote's LED.) I envy your biological wavelength detection.
    • StavrosK 11y ago
      Hmm, that's odd. I've noticed this with lots of remotes, I usually just look at them to tell if the batteries are dead. I wonder why I can see it.
• rphlx 11y ago
  24/192 lossless is a digital Veblen good; some people will pay more for it (and/or the HW to play it & store it), and almost all of them will enjoy it more, if only because it costs more. Whether it actually sounds better is somewhat tangential.
• acd 11y ago
  Try what age is your ears https://www.youtube.com/watch?v=VxcbppCX6Rk https://www.youtube.com/watch?v=VxcbppCX6Rk Or generate a tone sweep in audacity. Generate->chirp http://www.audacityteam.org/ http://www.audacityteam.org/ You loose the ability to hear high frequency sounds as you age. Personally I can hear up to about 14kHZ
  • lstamour 11y ago
    Huh. Downloaded a tone generator, and found that while in the video I heard a series of clicks at 16khz and beyond, I could in fact hear 16khz if I raised and lowered the volume from nothing to loudest in the app. It sounded like a whine and much harder to hear than I expected, distinguished most easily by when it quickly went from present to not present and back. In fact, I kept going up the scale doing that, and raising the volume, and found that I was able to hear even 19 to low 20khz as a high pitched noise, very quiet even at -6db. So ... yeah, probably does me no good considering that the loudness of other pitches makes it near impossible for me to hear anything practical in those frequencies. Of course, then I go to listen to music and wow, I can hear all this detail. I think I trained my ears for it, or I'm losing it. ;-)
• lips 11y ago
  My third time reading this, and a new question popped into my head: Are there any volume adjustments (on software or hardware) that take into account the pain threshold curves? That is, volume adjustments that aren't flat, but that will attenuate the frequencies that will cause discomfort at the lowest volumes?
• JadeNB 11y ago
  It may be worth noting (though it doesn't change any of the science) that this is from 2012.