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24/192 music downloads make no sense

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Re: 24/192 music downloads make no sense

#151

Post by aidenn0 »

tonecluster wrote: post #77
aidenn0 wrote: post #76

Earlier quoted context omitted.

TFA mentions that for recording and mastering there is a use. Furthermore, the headline implies it see the term "downloads" in the title.

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.

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.

Re: 24/192 music downloads make no sense

#152

Post by shams93 »

earlz wrote: post #26

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 t…

That's your processing overhead, you can mess with the sound a lot more at 96k before you hear audio issues.

The difference you hear is the difference between flac's lossless format and mp3's lossy format it has nothing to do with 16 bit versus 24 bit.

Re: 24/192 music downloads make no sense

#153

Post by alkonaut »

detaro wrote: post #147
alkonaut wrote: post #132

Earlier quoted context omitted.

Doesn't that depend on what type of playback you are doing? More and more playback these days is done via digital transfer with the volume set in software at the sending end, to amplifiers at fixed volume, such as many multi-room systems. If I airplay a song from my iPhone and have the volume at 50% set in software, then a few extra bits can help. Not sure if it makes a noticable difference, but it's a digital mixing…

Why woulda few extra bit in the source data help there? A DAC with more bits would, but that can easily be scaled from the existing data.

If you attenuate digitally you bit shift things out. If you have data in a 16bit (-32k..32k) stream and set a low volume in software before you send it, then it will scale the samples to (say) -8k..8k which is basically now a 14bit stream.

With a 24bit stream you can easily give up a few bits without losing dynamic range.

Re: 24/192 music downloads make no sense

#154

Post by subnaught »

TheOtherHobbes wrote: post #134
ska wrote: post #55

Earlier quoted context omitted.

The article is pretty clear about this too - higher bitdepths and sampling rates can be quite useful in mixing and recording situations. They're pointless for playback.

No they're not. And no matter how many times this gets linked to on the Internet, it's still wrong. The basic problem: the quieter a sound or detail gets, the fewer bits of resolution are used to represent it. In 16-bit recording, there simply aren't enough bits to represent very low level details without distorting them with a subtle but audible crunchy digital halo of quantisation noise. In a 24-bit recording, ther…

"This is why using more bits does not mean you can "reproduce music with a bigger dynamic range" - not without turning the volume up, anyway. What actually happens is that the maximum possible volume of a playback system stays the same, but quieter sounds are reproduced with more or less accuracy."

You are slightly confused. (It may help to remember that a decibel always refers to a ratio, so the setting of your volume knob is not important.) Greater bit depth does allow for greater dynamic range, this stems directly from the definition of dynamic range. 16-bit audio has a theoretical dynamic range of:

  10 * log10 (2^16)^2 ~ 96dB
24-bit audio has a theoretical dynamic range of:

  10 * log10 (2^24)^2 ~ 144dB
For reference, a quiet recording room has a noise floor of ~40dB SPL and the loudest amplified music rarely exceeds 115dB SPL. This gives a dynamic range of 75db, which indicates that a well-recorded 16-bit master should be more than adequate.

The idea that having bits in excess of this amount will somehow result in the perception of a smoother or more accurate sound is fallacious. Even at maximum playback volume, this information will exist well below the noise floor and will simply not be perceived. In fact, this information will likely exist well below the noise floor of the recording studio and thus, in some sense, will not even be recorded.

Re: 24/192 music downloads make no sense

#155

Post by detaro »

alkonaut wrote: post #153
detaro wrote: post #147

Earlier quoted context omitted.

Why woulda few extra bit in the source data help there? A DAC with more bits would, but that can easily be scaled from the existing data.

If you attenuate digitally you bit shift things out. If you have data in a 16bit (-32k..32k) stream and set a low volume in software before you send it, then it will scale the samples to (say) -8k..8k which is basically now a 14bit stream. With a 24bit stream you can easily give up a few bits without losing dynamic range.

Sure, but that only to the stream you actually send to the DAC, not the source material? (as in, you can take a 16 bit stream, scale it to 24 bits and then lower the volume) Am I missing something?

Re: 24/192 music downloads make no sense

#156

Post by hamiltonkibbe »

thescriptkiddie wrote: post #117
cynicalkane wrote: post #84

Earlier quoted context omitted.

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 intermod…

Is there a difference? Audio is one-dimensional, frequency is just the derivative of amplitude. An arbitrarily high frequency sound wave becoming louder and softer 440 times per second is just as much an A as a 440 Hz sound wave at constant volume. A lot of cheap audio gear even uses a "1-bit DAC" that is just very high frequency PWM.

The signal you describe in your comment is very much NOT an A. look at the fourier transform of a 1kHz signal modulated by a 440Hz signal, and you won't see any frequency component at 440Hz, nor any integer multiple 440Hz!

You can look at your "A440 at a constant volume" example as a 0Hz(DC) signal getting louder and softer 440 times a second, but this is the only case in which your example holds. Amplitude modulation creates sum and difference frequencies, so the A that you hear is 440Hz + 0Hz. if you change that 0Hz to 1kHz, you get a signal thats the sum of a sinewave at 560Hz and a sinewave at 1.44kHz, neither of which are an A.

The distinction is that the 1Hz signal is modulating the audible signal, not adding to it, if you look at the spectrum of the sum of those frequencies there is no 1Hz component, whereas if you added a 1Hz signal you'd get something completely different. And in this case the amplitude of the signal is always changing faster than 1Hz.

Edit: another way of looking at it: You wouldn't say you can "hear DC" because you can hear an A440 played at a perfectly constant volume.

Re: 24/192 music downloads make no sense

#157

Post by chimeracoder »

kinghajj wrote: post #88
hammock wrote: post #29

Earlier quoted context omitted.

Dynamic range compression is an important part of the aesthetic of pop music today, like it or not. Pop hits don't sound the same without it. For this reason (unaffected by normalization on itunes/youtube) it will be slow to go away.

I recall listening to a vinyl rip of Lady Gaga's first album, and found that "Pokerface" was much more enjoyable with the additional dynamic range. So I'm not sure if overcompression is necessary for newer songs, but rather that it's just what the mass market has become accustomed to because of the loudness war.

> I recall listening to a vinyl rip of Lady Gaga's first album, and found that "Pokerface" was much more enjoyable with the additional dynamic range.

I was skeptical of this, so I sought out a copy on Youtube[0]. "Better" is subjective, and this is definitely lower quality overall (given the Youtube compression). But it is undeniably a very different song. I've listened to the CD version of this song countless times, and I don't think I ever remember hearing the male voice in the intro so prominently.

For someone hearing the song for the first time (or even the first few), it may not be as obvious, but as someone who's used to hearing that song on a regular basis, it definitely jumps out as having a crisper sound.

[0] https://www.youtube.com/watch?v=98AgdAkjOjo

Re: 24/192 music downloads make no sense

#158

Post by gwbas1c »

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.

Re: 24/192 music downloads make no sense

#159

Post by gwbas1c »

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.

Re: 24/192 music downloads make no sense

#160

Post by Hello71 »

alkonaut wrote: post #132
ska wrote: post #55

Earlier quoted context omitted.

The article is pretty clear about this too - higher bitdepths and sampling rates can be quite useful in mixing and recording situations. They're pointless for playback.

Doesn't that depend on what type of playback you are doing? More and more playback these days is done via digital transfer with the volume set in software at the sending end, to amplifiers at fixed volume, such as many multi-room systems. If I airplay a song from my iPhone and have the volume at 50% set in software, then a few extra bits can help. Not sure if it makes a noticable difference, but it's a digital mixing…

no, because if you play at lower volume, that simply means that the quietest parts are closer to (or underneath) the threshold of hearing.

I see it this way: let's say 16 bits is needed to represent the entire discernable dynamic range between the threshold of hearing and the threshold of pain. if you turn the volume to 50%, then you throw out 8 bits, but you also only need to represent 8 bits worth of hearing range.

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