While I despise the loudness war and the loss of dynamic range, it's worth recognizing that where and how people listen to music has changed drastically in the last 20 years.
Moreso than ever in the past, people listen to music in public or otherwise noisy environments. (worse yet) They listen with generally low quality headphones/earbuds.
Basically, if the music isn't compressed, they won't hear most of it.
Of course, this plus the crappy earbuds and the noisy surroundings means they listen at higher volumes and suffer more hearing damage, which leads to less human sensitivity to sound levels (and more desire for compressed, high volume music).
It's very rare for most of us to get to sit in a good room with a good system and listen without interference... but it's really, really nice to do when you can.
I don't see why this needs to be done at the track mastering level though. If audio device manufacturers know their users are primarily going to be listening in noisy environments with cheap headphones they can compress the track at playtime.
As a person studying music production, this can't be done easily without having to mix and then master again. Certainly there could be varying edits of a track, though this is already common practice. The music you hear o the radio, or even Spotify, will usually be mastered differently than what you'll hear on a CD or DVD
You can get away with just mastering the same mix again, most likely. This would be especially true if you employ EBU R128 [0] and other such modern audio engineering practices. I've personally employed automated R128 dynamics matching for the past ten years, in order to output from one stereo PCM source file, a separate dynamically-matched master for each digital distributor's particular LUFS standard. i.e., YouTube's -13 dB LUFS, Tidal & Spotify's -14 dB LUFS, and iTunes' -16dB LUFS.
It'd be cool if there was something akin to Dolby Atmos where you don't deliver a master, but rather individual tracks and associated metadata, such that it can then be mastered on the fly depending on the output equipment.
Many games let you specify the audio output, and then adjusts the mixing accordingly. This would be something similar I guess, but the mixing is probably best done at the streaming provider, to save bandwidth and other resources. (For static content like music it only really has to be done once per output type.)
I'd like something like this, I have a vastly different sound setup at home than on the go, but listen to mostly the same tracks. Most sound pretty good on my home setup, but on the go I often find myself adjusting volume up and down between tracks, despite things like normalization being on.
I think this is one of the things driving the vinyl resurgence. Albums that release on vinyl tend to less alteration to the dynamic range... Ymmv I've not listened to everything out there :-]
The real reason for that is because of the physical limitations of the medium. You have to be very careful when mastering for vinyl that you aren't knocking the stylus out of the groove by making things too loud (especially sudden changes are dangerous). Even for less proactively mastered stuff, if a master is received too loud, the factory will reduce the volume before pressing so that it is playable.
This could still be solved. Streaming services and digital music stores like Spotify and Apple Music could simply allow labels/artists to supply multiple masterings of the same tracks. One would be the default offering; the other would be a 24 bit file mastered with all loudness-optimising compression disabled. Popular music also routinely uses track level compression for artistic reasons and to balance the mix; it would be up to the artist whether or not to pare back any of that. Obviously the artist’s intent should take precedence here.
Then, as an end user, we could choose which we preferred. By default there would be no change in behaviour. Where software updates are available, an the option could be provided to swap between versions at will. For older devices, the streaming service could let the end user choose the high dynamic range version as an account-level default.
Personally I’d want ready access to both versions. When listening to music is the singular activity in a quiet environment, full dynamic range is great. But as soon as I’m not solely focused on music, multitasking or in a noisy environment, I’d actually prefer the compressed version.
24-bit for playback is a waste of space and bandwidth, 16-bit audio has plenty of dynamic range for any kind of music enjoyed by human beings.
In the old days, the recommendation for digital audio was to master for -20 dBFS average, use no compression (or very little) and let any peaks fall where they may in the 20 dB headroom. And nobody complained about the noise floor at ~76 dB below the average level.
I wish everyone would go back to mastering to that spec, instead of slamming everything to 0 dBFS with loads of compression and often clipping on top. Obviously still allowing use of compression for artistic reasons.
24 bits gives you more headroom to work with, making mastering easier. It also quells any concerns about 16 bits being insufficient. As for "waste of space", the simple fact is that we don't have a shortage of space when you're talking about audio files. Hard drives are routinely over 1TB now. Remember, the person you responded to proposed offering the 24-bit version as an optional "audiophile" version of a track, not the default offering. It's not like everyone will want to stream the top-quality version for listening on earbuds.
You’re correct that the file size of audio stopped being relevant many years ago. In a world of Netflix 4K streaming, even the most inefficient audio formats don’t move the needle.
(If it did matter, which it doesn’t, it occurs to me that you could even create a 24 bit file that only contains 18 significant bits [dithered to 18 bits then padded to 24] in such a way that the space was reclaimed by ALAC or FLAC encoding.)
I agree with all of that; I only suggest 24 bits in order to shut up the people who can’t get over the idea of using 16 bits for wide dynamic range content. I just want the better mastering. I don’t care whether it takes extra placebo to make it happen.
The bandwidth consequence would be trivial bordering on nil given that only a small number of albums would ever receive the treatment, and only a small number of end users would choose the option.
If you take a dynamic piece of music and aggressively compress it will create a weird pulse like effect. If you add compression/limiting to each individual track tastefully you can increase the overall volume without it being noticable.
It's not possible to separately compress constituent stems in a mix. Once they're mixed, the impact that you can have on "loudness" is restricted to the whole, frequency-separated bands (multi-band compressor), or extracted parts. From an information-theory perspective, the available leeway for control over presentation is restricted once the audio is mixed.
However, I'd rather have a simple compressor running on playback than have a whole industry ruin their content.
That's way worse than loud mastering. There is no reasonable general method of compressing the mixbus at playtime, it depends on the source material and musical considerations. Its not even possible to find the right compression level without using your ears.
To be fair, I believe mastering is one of those domains that may soon be made obsolete with machine learning. But we're not quite there yet, and the existing methods require a lot of DSP power.
We also fail to protect ourselves from non-music hearing damage. Vacuums, lawnmowers, leaf blowers, traffic, etc will all degrade your hearing over time. I always wear ear protection whenever the sound level is even slightly uncomfortable. I'd rather spend my hearing on music than on noise.
I disagree with this completely. Many midrange to high-end car sound systems are the perfect listening environment for many types of music, and come standard with excellent sound systems.
Not all cars. The optional B&O high-end systems found in higher-end Audi, Mercedes and BMW are quite amazing, especially if fed a FLAC, Apt-X HD or other quality signal. These systems are expensive options, ranging from $3500 to $7000.
The quality of the sound system in the car is possibly secondary to the amount/quality of sound deadening in the car. Road noise is a nasty thing unless there is a lot of well engineered sound deadening material and acoustic design.
Once that's taken care of, then the audio can be appreciated. When I last owned a car, it was an enormous Lexus. The audio experience was sublime just with the premium factory system. But with the audio off, the cocoon was still so much better than what I've experienced in "regular" cars.
Unless you're in a Rolls Royce or Maybach while driving, I seriously doubt you'll tell the difference between FLAC and MP3. There's just too much external interference to compete with the subtleties that a high end audio signal can offer compared to a more typical system.
Regardless of what car you’re in, you’re not going to be able to reliably tell the difference between a flac file and an mp3 with a reasonable bit rate (256+).
I disagree. I have a stock sound system that I can easily tell the difference between a 320kbps MP3 and a CD flac in a blind test. I don't believe I have any special hearing as pretty much anyone I'm with can tell the difference.
-It does, of course, also depend on which MP3 encoder and settings you use; I did a most interesting (to me, anyway!) series of ABX trials on my own ears a few years ago.
From a 24/96 PCM master I prepared lots of transcodings - 16/44.1, 12/44.1, 8/44.1, 16/22, 12/22, 8/22, 320, 256, 192, 160, 128kbps mp3 with various options for encoding, ditto AAC - etc, etc.
I then used the (brilliant!) ABX plugin for Foobar and an excellent set of headphones, and after a few days I had a fair idea of how good my ears really were. Or, as it turns out, weren't. In most cases, 256kbps mp3 and red book were impossible to tell apart; 320kbps being indistinguishable in all but one case, if memory serves. However, at both 256 and 320kbps I had to listen hard for encoding artifacts rather than just enjoying the music.
Most tracks were eminently listenable at 192kbps mp3, but I could then reliably tell whether I listened to an mp3 or PCM without much effort.
Hard disk cost being what it is, I still ripped all my CDs lossless, though.
128kbps? I shudder at the thought of my 20-year-old-self telling myself and others that it was just about as good as uncompressed audio...
Oh, and you'd be surprised at how little dynamic range is actually utilized; on most tracks, telling 16/44.1 from 8/44.1 was next to impossible.
Yeah I've messed with dynamic range a bit, even got caught by the 24/96 hype for a while. But after running my own blind tests, I couldn't tell any difference with it.
On some songs with some setups I do believe you can still hear a difference but it's unlikely you'll have those songs and/or those setups. The vast majority of the time, unless the encoder for the MP3 is garbage, you won't be able to tell them apart.
Lower-fidelity playback systems and environments can actually reveal artifacts in lossily compressed audio. The psychoacoustic encoder makes assumptions about what noises mask other noises - if noises it expects to be present are absent (for instance, if your playback system has a poor frequency response), then low-bitrate noises can be unmasked.
I would not be surprised if additional noise, such as road noise, can also subvert masking phenomena in some way, but I don't know of any studies that tested this.
Its not just the speakers, its the loud rumble from the car moving. Maybe some cars have perfect sound insulation but I would say thats not very safe to be driving if you can't hear the outside.
nitpick: apt-x hd is a lossy audio codec and is almost never the source encoding of the music. most people are streaming aac, vorbis, or mp3, which is transcoded on the fly to apt-x hd. this is a lossy-to-lossy transcode which is basically never going to be transparent. apt-x hd is not a "quality signal" unless you happen to have a bunch of flacs on your phone.
I understand. what I'm saying is, nine times out of ten, the audio files themselves are already going to be in a lossy format. it's not really practical to have lossless content on your phone or stream it over a mobile connection. the lossy-to-lossy transcode is going to sound bad enough that most people would notice in a double blind test. this isn't some mp3 v0 vs flac audiophile nonsense; lossy-to-lossy transcodes sound noticeably bad.
from your other comment, I see that you are using a pretty good DAC in your car. kudos to you, sir.
Hey, that's what I drive as well. It's surely not "audiophile quality" but it's got just enough range that most music I listen to sounds surprisingly good for a stock sound system (as long as the source file is high quality).
I used an app called Boom for years to boost the volume, when I only had my laptop around and want to boost the volume. I haven't used it in a while however as my recent Macbook has much better speakers so I'm not sure if the v2 update is as good as v1.
Moreso than ever in the past, people listen to music in public or otherwise noisy environments. (worse yet) They listen with generally low quality headphones/earbuds.
Basically, if the music isn't compressed, they won't hear most of it.
Of course, this plus the crappy earbuds and the noisy surroundings means they listen at higher volumes and suffer more hearing damage, which leads to less human sensitivity to sound levels (and more desire for compressed, high volume music).
It's very rare for most of us to get to sit in a good room with a good system and listen without interference... but it's really, really nice to do when you can.