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You've completely misunderstood the article. It's discussing differences in the the PCM sampling rate, not the MP3 encoding bitrate. Or in otherwords, all these

You've completely misunderstood the article. It's discussing differences in the the PCM sampling rate, not the MP3 encoding bitrate. Or in otherwords, all these samples they compare are FLAC. 192 KHz has nothing to do with 192 kbps. I'm one of those audio engineers who mistakenly thinks I can tell between an MP3 and a FLAC. Yet somehow I understand the difference between sample rate and encoding bitrate, and you do not. All of this is besides the point. I'd much prefer a 24/44 sample than a 16/96 or 16/192. Bit depth has a much larger impact on the sound than sample rate.

ChrisGranger 12y ago
Bit depth affects dynamic range, and that's it. The only thing a 24-bit sample can do better than a 16-bit sample is accurately reproduce the difference between very loud sounds and very quiet ones. That's all. For the vast, vast majority of music listeners, the difference will be insignificant, as they don't listen in an environment where a dynamic range of 144 dB can be used to anywhere near its full effect.
- retrogradeorbit 12y ago
  Yes. And 16 bit cannot by itself represent without distortion the full dynamic range of a lot of music. Most samples, most of the time, do not use a full 16 bits. This is why during CD mastering dithering is used. Take it from me, when you master 24 bit stereo tracks, and you don't dither, huge amounts of low level detail disappear. The detail in the quiets is there in 24 bit, and lost when its truncated to 16 bits. Add the dithering, and you get increased noise, but the detail comes back. One could suggest that with dithering 16 bits can represent it. But that's with a whole bunch of noise added to the signal. You can argue that noise is not audible, but it is _just_, and when mastering you can audition the different dither spectrums to find which dither least impacts the music. http://www.digido.com/articles-and-demos12/13-bob-katz/16-dither.html http://www.digido.com/articles-and-demos12/13-bob-katz/16-di...
  - ChrisGranger 12y ago
    I certainly won't argue that 16-bit is just as good as 24-bit from an objective standpoint, as 24-bit is obviously superior, full stop. I'm just saying that for most listeners (everyone except those who listen at high levels in dedicated, treated listening rooms in very quiet environments) the difference will be inaudible almost all the time. Extremely low level detail doesn't really matter if it's lost in the >20 dB of natural noise in your room. At that point the issue may become moot as other problems like standing waves, harmonic distortion, inaccurate speaker frequency response and so on creep in and affect music playback to a subjectively larger degree than '16-bit versus 24-bit does', IMO. All that said, 24-bit is definitely the way to go since we might as well do it right even if only x percent of listeners will notice. As an aside, thank you for being one of the conscientious 'good guys' in the studio. I collect music and wish I had a nickel for every sloppy recording I've heard.
    - retrogradeorbit 12y ago
      Yes. I completely agree. From my perspective, even if one person in a hundred can hear a difference, then I'm going to pay attention. I don't want to boil everything down to meet the average. I think it's fine to release 16/44. I think well produced music sounds excellent in that format. It's one of the reasons the CD has done so well. It's just hifi enough to capture everything. And it's amazing to think this technology had its début in 1982! But for so long, for those of us who want a higher quality (hearing it exactly as they would have heard in the studio during the production) there was nothing we could do. Willing to pay more for it, doesn't matter. You just can't get it. It's still that way. What gripes me is the attitude of many, including this xiph article, that hires versions "make no sense", that "there is no point" and thus everyone should just be happy with what they've got and that anyone protesting is an "audiophool" or believes in magic fairies or something. We all get lumped in with those people buying $3000 IEC power cables. For many people it's all black and white, there is no room for grey. You either think that 128kbps mp3s sound identical to the analogue master tape, or you are a fool spending $20,000 on magical stickers to increase the light speed in your CD player. All I want is to be able to buy the mix and hear it as the engineer heard it in the studio. That would be nice. I know it's not for everyone, but it doesn't make me crazy. As food for thought, have a read of what Rupert Neve said about Geoff Emerick's hearing ability (being able to discern a 3dB rise at 54kHz) here: http://poonshead.com/Reading/Articles.aspx http://poonshead.com/Reading/Articles.aspx "The danger here is that the more qualified you are, the more you 'know' that something can't be true, so you don't believe it. Or you 'know' a design can't be done, so you don't try it."
- ecocentrik 12y ago
  This is slightly incorrect. The combination of bit length and sampling rate determine both dynamic range and frequency fidelity. Although it's common to hear the two values used to represent these two separate physical measurements, it's just another case of explaining new tech (80s CDs) to old technical consumers (70s hi-fi types). You can measure a reduction in dynamic range by reducing either sampling rate or bit length. You'll frequently find 1-bit A/Ds and D/A at > 5Mhz on high fidelity systems. That 1-bit signal is converted to/from a higher bitrate, lower sampling rate signal without loss of fidelity. If you're interested in looking at alternate bitrate encodings you should just look at the Super Audio CD format https://en.wikipedia.org/wiki/Super_Audio_CD https://en.wikipedia.org/wiki/Super_Audio_CD On your second point, I agree, we live in a noisy world and hearing 144dB of dynamic range would require serious isolation. What most people should be able to hear with 90Dbs of dynamic range, are the harmonics created by undersampling a high frequency signal. To quickly explain I'll use a 1-bit lower frequency scenario. Lets say we have a 2Khz sine wave and a 1-bit 5Khz sampling rate. The 2Khz signal is going to be represented by a different 2 samples every cycle. The result will be a signal that is no longer a sine wave and closer in frequency to 800Hz (wild approximation) than 2Khz. Low pass filters are used to to keep those harmonics from being too pervasive but they still sneak into the signal near the high frequency range. Transpose this example to our current audio standard and you might realize that in order to accurately represent the high end we need a little more than 16bit 48Khz.
  - badsock 12y ago
    In your last example about the harmonics, I can see that being so in the most simple case, but surely that's a limitation of the playback system, not the storage medium. And isn't that addressed by the oversampling that is used almost universally in DACs now?
    - ecocentrik 12y ago
      It's a limitation of the recording/storage medium and is addressed by oversampling and low pass filters in ADCs in acoustical recording. Most audio recording, processing and mixing is performed in 24 or 32 bit. Once the data is down sampled for distribution in 16bit 44.1Khz you run into the limitation again where you have fewer samples to represent higher frequencies. The only remedy is to attenuate those frequencies before downsampling. I'm unsure of the role oversampling plays in DACs so I can't speak to that.
ChrisGranger 12y ago
I think you're misreading what the OP said. He wasn't making the mistake of associating 192 kbps with 192 kHz. It was just a coincidence that he used 192 kbps MP3 transcodes. He could just as easily have used 224 kbps MP3s versus 24/192 FLACs in the test he described. The point was that his friend couldn't tell the difference between MP3s and FLACs.
- retrogradeorbit 12y ago
  Fair enough. Then the OP comment is completely off topic. What has encoding bitrates got to do with the original article?

ChrisGranger 12y ago

Bit depth affects dynamic range, and that's it. The only thing a 24-bit sample can do better than a 16-bit sample is accurately reproduce the difference between very loud sounds and very quiet ones. That's all. For the vast, vast majority of music listeners, the difference will be insignificant, as they don't listen in an environment where a dynamic range of 144 dB can be used to anywhere near its full effect.

retrogradeorbit 12y ago
Yes. And 16 bit cannot by itself represent without distortion the full dynamic range of a lot of music. Most samples, most of the time, do not use a full 16 bits. This is why during CD mastering dithering is used. Take it from me, when you master 24 bit stereo tracks, and you don't dither, huge amounts of low level detail disappear. The detail in the quiets is there in 24 bit, and lost when its truncated to 16 bits. Add the dithering, and you get increased noise, but the detail comes back. One could suggest that with dithering 16 bits can represent it. But that's with a whole bunch of noise added to the signal. You can argue that noise is not audible, but it is _just_, and when mastering you can audition the different dither spectrums to find which dither least impacts the music. http://www.digido.com/articles-and-demos12/13-bob-katz/16-dither.html http://www.digido.com/articles-and-demos12/13-bob-katz/16-di...
- ChrisGranger 12y ago
  I certainly won't argue that 16-bit is just as good as 24-bit from an objective standpoint, as 24-bit is obviously superior, full stop. I'm just saying that for most listeners (everyone except those who listen at high levels in dedicated, treated listening rooms in very quiet environments) the difference will be inaudible almost all the time. Extremely low level detail doesn't really matter if it's lost in the >20 dB of natural noise in your room. At that point the issue may become moot as other problems like standing waves, harmonic distortion, inaccurate speaker frequency response and so on creep in and affect music playback to a subjectively larger degree than '16-bit versus 24-bit does', IMO. All that said, 24-bit is definitely the way to go since we might as well do it right even if only x percent of listeners will notice. As an aside, thank you for being one of the conscientious 'good guys' in the studio. I collect music and wish I had a nickel for every sloppy recording I've heard.
  - retrogradeorbit 12y ago
    Yes. I completely agree. From my perspective, even if one person in a hundred can hear a difference, then I'm going to pay attention. I don't want to boil everything down to meet the average. I think it's fine to release 16/44. I think well produced music sounds excellent in that format. It's one of the reasons the CD has done so well. It's just hifi enough to capture everything. And it's amazing to think this technology had its début in 1982! But for so long, for those of us who want a higher quality (hearing it exactly as they would have heard in the studio during the production) there was nothing we could do. Willing to pay more for it, doesn't matter. You just can't get it. It's still that way. What gripes me is the attitude of many, including this xiph article, that hires versions "make no sense", that "there is no point" and thus everyone should just be happy with what they've got and that anyone protesting is an "audiophool" or believes in magic fairies or something. We all get lumped in with those people buying $3000 IEC power cables. For many people it's all black and white, there is no room for grey. You either think that 128kbps mp3s sound identical to the analogue master tape, or you are a fool spending $20,000 on magical stickers to increase the light speed in your CD player. All I want is to be able to buy the mix and hear it as the engineer heard it in the studio. That would be nice. I know it's not for everyone, but it doesn't make me crazy. As food for thought, have a read of what Rupert Neve said about Geoff Emerick's hearing ability (being able to discern a 3dB rise at 54kHz) here: http://poonshead.com/Reading/Articles.aspx http://poonshead.com/Reading/Articles.aspx "The danger here is that the more qualified you are, the more you 'know' that something can't be true, so you don't believe it. Or you 'know' a design can't be done, so you don't try it."
ecocentrik 12y ago
This is slightly incorrect. The combination of bit length and sampling rate determine both dynamic range and frequency fidelity. Although it's common to hear the two values used to represent these two separate physical measurements, it's just another case of explaining new tech (80s CDs) to old technical consumers (70s hi-fi types). You can measure a reduction in dynamic range by reducing either sampling rate or bit length. You'll frequently find 1-bit A/Ds and D/A at > 5Mhz on high fidelity systems. That 1-bit signal is converted to/from a higher bitrate, lower sampling rate signal without loss of fidelity. If you're interested in looking at alternate bitrate encodings you should just look at the Super Audio CD format https://en.wikipedia.org/wiki/Super_Audio_CD https://en.wikipedia.org/wiki/Super_Audio_CD On your second point, I agree, we live in a noisy world and hearing 144dB of dynamic range would require serious isolation. What most people should be able to hear with 90Dbs of dynamic range, are the harmonics created by undersampling a high frequency signal. To quickly explain I'll use a 1-bit lower frequency scenario. Lets say we have a 2Khz sine wave and a 1-bit 5Khz sampling rate. The 2Khz signal is going to be represented by a different 2 samples every cycle. The result will be a signal that is no longer a sine wave and closer in frequency to 800Hz (wild approximation) than 2Khz. Low pass filters are used to to keep those harmonics from being too pervasive but they still sneak into the signal near the high frequency range. Transpose this example to our current audio standard and you might realize that in order to accurately represent the high end we need a little more than 16bit 48Khz.
- badsock 12y ago
  In your last example about the harmonics, I can see that being so in the most simple case, but surely that's a limitation of the playback system, not the storage medium. And isn't that addressed by the oversampling that is used almost universally in DACs now?
  - ecocentrik 12y ago
    It's a limitation of the recording/storage medium and is addressed by oversampling and low pass filters in ADCs in acoustical recording. Most audio recording, processing and mixing is performed in 24 or 32 bit. Once the data is down sampled for distribution in 16bit 44.1Khz you run into the limitation again where you have fewer samples to represent higher frequencies. The only remedy is to attenuate those frequencies before downsampling. I'm unsure of the role oversampling plays in DACs so I can't speak to that.

ChrisGranger 12y ago

I think you're misreading what the OP said. He wasn't making the mistake of associating 192 kbps with 192 kHz. It was just a coincidence that he used 192 kbps MP3 transcodes. He could just as easily have used 224 kbps MP3s versus 24/192 FLACs in the test he described. The point was that his friend couldn't tell the difference between MP3s and FLACs.

retrogradeorbit 12y ago
Fair enough. Then the OP comment is completely off topic. What has encoding bitrates got to do with the original article?