6 ms·

There's a very straightforward practical reason why you might want to keep downloading 24/192. The people who put those together are on average "audiophile" sno

by lz400 9y ago

There's a very straightforward practical reason why you might want to keep downloading 24/192. The people who put those together are on average "audiophile" snobs so the rips tend to be perfect, often from high quality sources like SACDs, hdtracks, etc. The source of the recording is usually the best master known for that record (special remastered editions, etc.). If you download mp3/spotify chances are the copy you download is from a worse source. Sure, you don't need 24/192, a well ripped 320kbps mp3 would be the same in practice, but looking for 24/192 in the internet is an easy way of getting better quality music on average in practice.

derefr 9y ago
Similarly, seeing 24/192 on a rip from an analogue source (e.g. vinyl) usually implies that a high-quality player+amp+ADC was used to digitize the album.
kazinator 9y ago
So, in other words: bite the bandwidth bullet and get the 192 material, but for Pete's sake, do yourself a favor and downsample to 44.1.
- lz400 9y ago
  If you have the patience, sure. In my desktop I keep everything on the big file source and listen to it that way. I also have dedicated DACs/AMPs that can decode all the way up to DSD. I know it probably doesn't make a difference but whatever, storage is cheap. For my phone I listen to spotify.
mbell 9y ago
> The source of the recording is usually the best master known for that record (special remastered editions, etc.). Usually what is happening here is you're getting a master that hasn't been compressed to death (see loudness wars). Vinyl is a shit source for 'quality', but most records aren't compressed to death so they can still sound better on a good set of speakers due to the dynamic range.
gruez 9y ago
so like, signaling?[1] [1] https://en.wikipedia.org/wiki/Signalling_(economics) https://en.wikipedia.org/wiki/Signalling_(economics)
em3rgent0rdr 9y ago
> "If you download mp3/spotify" You are misconstruing Monty's argument here. He is very much against mp3...in fact he says he could tell the difference between high bitrate mp3 and 16 bit 44k wav. The real point of the video is that 16 bit 44k wav is beyond sufficient...don't need to go beyond that to 24-bit 192kHz.
- lz400 9y ago
  As far as I know, people consistently fail to tell the difference in blind A/B tests in listening between (any decent) FLAC and a well ripped mp3 from the same source. I know I can't. I can't link to proper double blind studies but that's the general consensus in the non-BS-audiophile community as far as I know.
  - Tepix 9y ago
    c't magazine did a test all the way back in 2000 (see https://www.heise.de/ct/artikel/Kreuzverhoertest-287592.html https://www.heise.de/ct/artikel/Kreuzverhoertest-287592.html) and even the best test listeners were unable to tell 256kbit mp3 and the original apart.
    - Applejinx 9y ago
      Was this the Arny Krueger stuff? I did that. I ABXed 320kbit mp3 from an uncompressed original, I think 9/10 IIRC. It was a recording of castanets, and listening for frequency response differences was useless so I keyed off of 'personality' differences in the sounds and did it that way. I was also just as horrible at detecting wow and flutter, as I was good at detecting lossy compression 'changes of sonic personality'. Comes from my experience being with analog audio, which is less true of people these days. The idea that 'the best' listeners cannot tell 256k from even something as limited as 16/44.1 is ridiculous. Any mp3 format… any lossy format… is seriously compromised. Talk to techno/house DJs and producers about how useful it is to perform off mp3s, this is not a hypothetical argument.
  - ycombinete 9y ago
    My anecdote is that I've done a number of these tests (Foobar has a plugin that allows you to do them on yourself), and I can reliably tell the difference between FLAC and 128 MP3, but can't tell the difference on 256 MP3 and up.
shams93 9y ago
The 192 is really for low latency not for anything you can actually hear. You can't hear the difference past 48khz. An audio interface that can do 192 gets you down to 5ms latency where you can finger drum without being thrown off. But it has nothing to do with audio quality. Like if you use guitar rig you will have the lowest possible latency for realtime computer generated guitar effects at 192.
- Buge 9y ago
  Where do you get the 5ms from? Why is 192 more likely to have lower latency? I would think the higher bandwidth required by 192 has the possibility to slow your computer down and cause more latency.
  - toothbrush 9y ago
    I believe the parent might be referring to the myth about attack of signal falling "between the cracks" of digital sampling. This is debunked in the video linked elsewhere, in the chapter titled "bandlimitation and timing". I might be wrong about the parent poster's meaning, though.
    - rocqua 9y ago
      See the excelent answer by zaxomi for another more favorable interpretation of the parent. https://news.ycombinator.com/item?id=15131280 https://news.ycombinator.com/item?id=15131280
  - zaxomi 9y ago
    I believe the parent is referring to the latency of real-time effects used during performance: * With analogue equipment the latency between the input and output of the effect is often below 1 ms (unless the effect is supposed to have a delay). * Standalone digital effect equipment that processes the samples one at a time can also have a latency below 1 ms. (48kHz ≈ 0.02 ms) * If you use a computer for real-time effects, the samples are not transferred sample by sample from the audio interface, but in a block of many samples. The number of buffers and the size of them can usually be changed by the user. With a buffer of 1024 samples, the oldest sample is already about 21 ms old before the computer can process it. After the buffer is processed it has to be transferred to the audio interface, and that will add another 21 ms. So the minimum latency for any real-time effect in the computer is about 42 ms at 48 kHz if the size of the buffers is 1024 samples. Often it is much worse because the operating system adds more latency. If the equipment can handle a sample rate of 192 kHz, the same latency is about 10 ms. If the computer can handle smaller buffers, the latency can be lowered. With 256 samples per buffer the minimum latency will be about 11 ms at 48 kHz and 3 ms at 192 kHz.
tempestn 9y ago
It seems to me what we really need then is some other kind of litmus test that indicates a quality recording, without having to waste a ton of bandwidth and storage to get it. Are there other things to look for that also tend to correlate with quality? Newer formats maybe?

by lz400 9y ago

derefr 9y ago

Similarly, seeing 24/192 on a rip from an analogue source (e.g. vinyl) usually implies that a high-quality player+amp+ADC was used to digitize the album.

kazinator 9y ago

So, in other words: bite the bandwidth bullet and get the 192 material, but for Pete's sake, do yourself a favor and downsample to 44.1.

lz400 9y ago
If you have the patience, sure. In my desktop I keep everything on the big file source and listen to it that way. I also have dedicated DACs/AMPs that can decode all the way up to DSD. I know it probably doesn't make a difference but whatever, storage is cheap. For my phone I listen to spotify.

mbell 9y ago

> The source of the recording is usually the best master known for that record (special remastered editions, etc.). Usually what is happening here is you're getting a master that hasn't been compressed to death (see loudness wars). Vinyl is a shit source for 'quality', but most records aren't compressed to death so they can still sound better on a good set of speakers due to the dynamic range.

gruez 9y ago

so like, signaling?[1] [1] https://en.wikipedia.org/wiki/Signalling_(economics) https://en.wikipedia.org/wiki/Signalling_(economics)

em3rgent0rdr 9y ago

> "If you download mp3/spotify" You are misconstruing Monty's argument here. He is very much against mp3...in fact he says he could tell the difference between high bitrate mp3 and 16 bit 44k wav. The real point of the video is that 16 bit 44k wav is beyond sufficient...don't need to go beyond that to 24-bit 192kHz.

lz400 9y ago
As far as I know, people consistently fail to tell the difference in blind A/B tests in listening between (any decent) FLAC and a well ripped mp3 from the same source. I know I can't. I can't link to proper double blind studies but that's the general consensus in the non-BS-audiophile community as far as I know.
- Tepix 9y ago
  c't magazine did a test all the way back in 2000 (see https://www.heise.de/ct/artikel/Kreuzverhoertest-287592.html https://www.heise.de/ct/artikel/Kreuzverhoertest-287592.html) and even the best test listeners were unable to tell 256kbit mp3 and the original apart.
  - Applejinx 9y ago
    Was this the Arny Krueger stuff? I did that. I ABXed 320kbit mp3 from an uncompressed original, I think 9/10 IIRC. It was a recording of castanets, and listening for frequency response differences was useless so I keyed off of 'personality' differences in the sounds and did it that way. I was also just as horrible at detecting wow and flutter, as I was good at detecting lossy compression 'changes of sonic personality'. Comes from my experience being with analog audio, which is less true of people these days. The idea that 'the best' listeners cannot tell 256k from even something as limited as 16/44.1 is ridiculous. Any mp3 format… any lossy format… is seriously compromised. Talk to techno/house DJs and producers about how useful it is to perform off mp3s, this is not a hypothetical argument.
- ycombinete 9y ago
  My anecdote is that I've done a number of these tests (Foobar has a plugin that allows you to do them on yourself), and I can reliably tell the difference between FLAC and 128 MP3, but can't tell the difference on 256 MP3 and up.

shams93 9y ago

The 192 is really for low latency not for anything you can actually hear. You can't hear the difference past 48khz. An audio interface that can do 192 gets you down to 5ms latency where you can finger drum without being thrown off. But it has nothing to do with audio quality. Like if you use guitar rig you will have the lowest possible latency for realtime computer generated guitar effects at 192.

Buge 9y ago
Where do you get the 5ms from? Why is 192 more likely to have lower latency? I would think the higher bandwidth required by 192 has the possibility to slow your computer down and cause more latency.
- toothbrush 9y ago
  I believe the parent might be referring to the myth about attack of signal falling "between the cracks" of digital sampling. This is debunked in the video linked elsewhere, in the chapter titled "bandlimitation and timing". I might be wrong about the parent poster's meaning, though.
  - rocqua 9y ago
    See the excelent answer by zaxomi for another more favorable interpretation of the parent. https://news.ycombinator.com/item?id=15131280 https://news.ycombinator.com/item?id=15131280
- zaxomi 9y ago
  I believe the parent is referring to the latency of real-time effects used during performance: * With analogue equipment the latency between the input and output of the effect is often below 1 ms (unless the effect is supposed to have a delay). * Standalone digital effect equipment that processes the samples one at a time can also have a latency below 1 ms. (48kHz ≈ 0.02 ms) * If you use a computer for real-time effects, the samples are not transferred sample by sample from the audio interface, but in a block of many samples. The number of buffers and the size of them can usually be changed by the user. With a buffer of 1024 samples, the oldest sample is already about 21 ms old before the computer can process it. After the buffer is processed it has to be transferred to the audio interface, and that will add another 21 ms. So the minimum latency for any real-time effect in the computer is about 42 ms at 48 kHz if the size of the buffers is 1024 samples. Often it is much worse because the operating system adds more latency. If the equipment can handle a sample rate of 192 kHz, the same latency is about 10 ms. If the computer can handle smaller buffers, the latency can be lowered. With 256 samples per buffer the minimum latency will be about 11 ms at 48 kHz and 3 ms at 192 kHz.

tempestn 9y ago

It seems to me what we really need then is some other kind of litmus test that indicates a quality recording, without having to waste a ton of bandwidth and storage to get it. Are there other things to look for that also tend to correlate with quality? Newer formats maybe?