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

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.

Low nitrate MP3 is a fantastic typo.

I agree, and my hypothesis is that it's exacerbated by the combination of three particular things:

1. It's a high frequency and non-repetitive waveform, so it demands bitrate.

2. Drum miking often involves multiple mics spaced apart, so more than one typically picks up any given cymbol with a phase offset, and those mics are panned quite differently, leading to a very "wide" result, i.e., left and right output is somewhat uncorrelated as seen on a vectorscope [0].

3. Perceptual codecs often sound 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 then instead of giving equal bitrate to each channel, you give more bits to the mid (correlated signals) and fewer bits to the side (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 cymbols! MP3 bitrates below 256kbps typically need joint stereo mode enabled in order to sound decent.

[0] https://en.m.wikipedia.org/wiki/Vectorscope#Audio

A realistic take on the issues invovled. I never knew what “joint stereo” meant. Great explanation.

If anyone has a good cymbal crash sample at 24/96 or better that they can provide, it seems like it would be a great example for intentional differentiation of various compressed versions.

Vinyl actually encodes stereo much the same way.

FM radio as well: originally one channel, then a second was added, and the second is used for the difference signal (the first already being the sum signal, as with any conversion to mono). Overhauling the whole thing to broadcast left and right discretely would destroy backwards compatibility, and the newly-added subcarrier had worse SNR (thus receivers ignore it until reception is sufficiently strong) so it only made sense to use it for the difference.

I was going to say this - cymbals are often very noticeably bad on MP3 recordings.

Well, most classical songs are very well compressible, because not much is going on. Punk Rock or any other music were a lot is happening, at the same time, can suffer very audible from 128 kbit lossy compression. So you can hear lossy compression better in a loud pop song than other music.

I don't agree with this from my own experience. To me, classical music at high compression suffers far worse than modern bands.

My unscientific guess would be that classical music might have wider dynamic range than “normal” music. So the same compression amount affects the one with more range first (classical).

Higher dynamic range and typically also more 'pure'. The introduced compression artifacts stand out more in simpler waveforms than in wavevorms that are an addition of many more layers of sound.

Your guess would be correct [1]. Some samples [2].

[1] https://www.mdpi.com/2624-599X/4/3/42

[2] http://www.harmjschoonhoven.com/mp3-quality.html

A typical classical symphony requires 50-80 instruments, several dozen of which might play at the same time, while a typical punk song has maybe four.

Compression algorithms don't care about instrument counts, they care about the complexity of the signal. So there being a total of maybe hundreds of instruments doesn't matter, it's only about how many are playing at any given time. There might be a few dozen instruments playing at any given moment in a classical recording, but they're all highly tuned instruments with several of each one playing just about the exact same note.

Grungy rock music might only have a few instruments, but they're often purposefully highly distorted and have people pretty much screaming and shouting, leading to the actual sinal being closer to literally noise.

So the closer you are to literally noise, the less compressible your signal is.

Imagine a an image with a dozen sharp, clear, colorful squares. Now imagine a similar resolution image with only 5 colors, but they're different shapes and they're kind of fuzzy and they're really more like gradients instead of a pure color. Which is going to compress easier?

This is an interesting argument, but not convincing. You are essentially claiming that the information content of a classical recording is low, and could be replicated with fewer, simpler instruments. However, I suspect that many of the interesting subtleties that make a recording sound “beautiful” would be lost that way. Surely such nuances are why classical musicians are so passionate about their instruments.

(FWIW, I’m way more of a punk fan myself, and usually find most classical music pretty boring.)

> You are essentially claiming that the information content of a classical recording is low, and could be replicated with fewer, simpler instruments.

No, I do agree having multiple instruments does lead to a wider sound than just a single one. Plus multiple instruments will probably help balance out a single one being out of tune or not quite hitting the note right. But still, these instruments usually are way more tuned to produce closer to pure tones and their harmonic overtones than a guitar going through half a dozen different distortion and effect pedals then through a compressor along with a guy screaming all over the place into a microphone.

Just look at these two spectrograms. Look at the rock song where there's a lot of distorted guitar, bass, drums, and singing going on and compare that to an active part of the classical recording. See how the classical recording has a lot more clean, straight lines while the rock song is a lot more fuzzy? Imagine if these were images, which would be more complicated to accurately compress? That's not really a great analogy, but it is touching on the same concept.

Each dot is essentially another part of the signal. Its another wave of the sum output wave that needs to be encoded. Compression algorithms (on a basic level) group together these signals and throw away areas that aren't used.

Rock song: https://youtu.be/BVsp23B8dWo?t=62

Classical song: https://youtu.be/Txp-pHU2K6w?t=210

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.

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.

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.

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.

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?

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.

Plexamp does this automatically if requested.

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?