According to their documentation, this is "Upper bound of 256kbps AAC & OPUS": https://support.google.com/youtubemusic/answer/9076559?hl=en...
EDIT: FWIW, I'm a Premium member. I'm not sure if this is a standard feature.
Due to what I assume are music licensing oddities, one song in my Spotify now has an entirely different singer. It was taken down for a while, then later reuploaded as a new recording with new vocals. I’ve also seen copies of my songs changing to remix/cover versions, presumably due to metadata adjustments by the artists. Although usually I can re-search and find the original, adding it back to my library.
A few songs just have new outros/intros. One song in my library now has an additional several seconds of silence on the end. One has a new longer intro that I think is really from the music video version of the song.
As convenient as these music services are, I hate that my library changes beneath me without my control. There are dozens of songs just straight up missing from my Spotify library now. And a small handful that have changed audio. These are almost always in indie songs, and Spotify just hides deleted songs by default in the UI so most users don’t notice.
-F (uppercase!) lists all available audio and video formats to download.
Why doesn't this huge AV platform use a better audio time stretch algorithm?
Besides, their Android and iOS apps do slow music as bad if not worse than on web.
[1] https://bungee.parabolaresearch.com/compare-audio-stretch-te...
I would take a guess that a higher bitrate = longer loading times, and viewers care far more about an extra few second of buffering than they care about audio quality, especially when they don't have the original to compare to.
With newer codecs, doing 4k with 2Mb/s isn't unheard of.
For audio, on the other hand, 32 or 64kbps per channel isn't unheard of.
To support seeking you could encode a low bitrate stream, and a high quality stream, and then a number of ramps between these. So when you seek you start with the low bitrate stream and then after a few time units go on the ramp to the high quality stream.
… while yeah… a lower bitrate upfront might lower the required bandwidth and thus, latency, to get enough of a buffer to start playback … all the bloat on the page would be a better first port of call.
Server CPUs can encode audio at hundreds of times faster than realtime so there’s no need for hardware acceleration. Back in the iPod era DSPs were used to decode MP3/AAC but now only the most CPU or battery constrained devices like AirPods need hardware acceleration.
I got better audio quality ripping songs from limewire or Napster in the 2000s.
Why do we settle with this substandard quality? Oh wait, YT barely has any competition and subsidized by G. No need for competition. Just shove ads down users throats and sell of their usage data.
Nah, you didn't, at least not reliably. Half of that were recodes and upcodes that used Blade, FhG or Xing.
Only with torrent technology and community-driven trackers we got reliable distribution that surpassed the official non-physical channels.
"I decided that analysis should focus on the higher, more conventional rates – 48k and 44k1" - opus is always 48khz, so that doesn't mean much.
1. Poor sensitivity in bass and treble. See:
https://en.wikipedia.org/wiki/Equal-loudness_contour
2. Limited ability to hear multiple sounds simultaneously, or almost simultaneously. See:
https://en.wikipedia.org/wiki/Auditory_masking
Bernhard Seeber has some videos on Youtube with demonstrations of auditory masking:
https://www.youtube.com/watch?v=R9UZnMsm9o8
https://www.youtube.com/watch?v=bU0_Kaj7cPk
The only fair way to evaluate lossy codecs is with double blind listening tests.
And who are therefore forced to hear terrible audio because the compression method only considers the majority.
It's wild how many details sound people have to keep track of. I know when I upload to Youtube things get smoothed noticeably compared to say Soundcloud. Probably because I've mastered over their -14 LUFS requirement.
I wonder how much of the modern 'everyone needs to use closed captioning watching TV now' comes from the streaming services Codecs and other decisions and not just the A/V sound peoples' decisions. Do movies sound people now need to listen through on something like Streamliner above for every decision?
That said, technical aspects matter, recording practices, mastering for loudness with little dynamic range (so that compressed voice blends with foley and the rest), and indeed encoding can definitely affect the ease of understanding human speech. Speaking of encoding… I couldn’t find it on the site, is Streamliner a one-time purchase?
[0] Well, not every film has to (Upstream Color comes to mind, as an example of a film with relatively little plot-advancing dialogue), but it seems that the majority of productions rely on dialogue and it isn’t going away.
https://www.slashfilm.com/673162/heres-why-movie-dialogue-ha...
Upload 320kbit encoded MP3? Sounds great.
Upload a high-khz WAV? It gets butchered, the top-end turns to glittery noise.
Maybe others have different experiences, but honestly it felt like I'd been duped when paying for the subscription but still got trash quality audio, only to have to pay more.
While this does retain the majority of useful information, it explains why the youtube version of your song feels just a little more 'lifeless' than the high quality version you have elsewhere.
The original recording contains high frequency detail that got lost. Your human body uses that high frequency detail to orient itself in space with respect to sound sources (like reverb, reflections, or ambient sounds).
It is interesting from a data storage point of view because this could result in massive savings. Consider audio is recorded at 44.1khz or 48kHz but is actually stored at 32kHz. They have effectively saved 25% in audio file storage at marginal customer experience.
Having hearing sensitivity over 16 kHz is unusual. If you're under 15 years old and kept your ears pristine by not listening to loud noises, you might be able to hear it. Older people are out of luck.
Moreover, even if you can hear above 16 kHz in loud pure tones, there is so little content in real audio/music above 16 kHz that it makes no practical difference.
> massive savings ... effectively saved 25%
Not really. Going from a 48 kHz sampling rate to 32 kHz is indeed 2/3× the size for uncompressed PCM audio. But for lossily compressed audio? Not remotely the same. Even in old codecs like MP3, high frequency bands have heavy quantizers applied and use far fewer bits per hertz than low frequency bands. Analogously, look at how JPEG and MPEG have huge quantizers for high spatial frequencies (i.e. small details) and small quantizers for broad, large visual features.
Good point about the savings. I was using uncompressed format as the reference, but it is indeed unlikely that YouTube serves out lossless audio.
I also should have used the word "delivery" instead of data storage. Those are two separate problems: where the original asset is stored (and how, if they don't store raw originals), and also how the asset is delivered over the web.
Maybe with a browser that doesn't support Opus and gets AAC instead (Safari?). With Firefox or Chromium on Linux I get up to 20 kHz, which by design is the upper limit in Opus codec.
In all seriousness, every aspect of this comparison is somewhere between deeply flawed and invalid. No point dwelling on just one part.
It’s a noticeable problem in audio production if e.g. a filtered kick drum goes out of phase and sucks amplitude when mixed with the original.
Description:
High-Quality Audio
Available until February 22
With high-quality audio, you can listen to music on YouTube in the best audio quality.
How it works: Watch an eligible music video on YouTube and enjoy the benefits of higher-quality audio.
Only available on iOS and Android.
Oversampling can be a useful internal detail for ADCs and DACs. For example, if the digital audio stream is mathematically converted to 96 kHz with a high-quality FIR low-pass filter and then fed to a DAC, then the analog low-pass filter can have a much shallower roll-off and be more easily designed. Same goes for ADCs, where the analog filter can be simple and gentle, then digitized at 96 kHz, then downsampled digital to 48 kHz with high-quality but more computationally intensive filters. ( https://en.wikipedia.org/wiki/Oversampling , https://en.wikipedia.org/wiki/Delta-sigma_modulation , etc.)
But yes, listening to or distributing audio at anything over 48 kHz is a complete waste of resources. Monty@Xiph.Org explained very well in: https://people.xiph.org/~xiphmont/demo/neil-young.html
Now I just check on YouTube again and they are now back to 128 / 130 Kbps for AAC-LC.
Take unprocessed audio and process it. Then take both the processed and unprocessed audio and add them to your audio software (e.g. Audacity). Now flip the polarity of one of the audio signals.
This allows you to listen to the difference between the two signals and if there is nothing there, guess what, they are the same or the differences are so small that theg are inaudible.
This is a great way to anger people with expensive hifi gold cables, because what is true in the digital also works in the analog.
You only need to make sure both ajdio signals are at the same level (by minimizing the level of the difference).
[reads]
...Jesus H. F. Christ....
Every generation thinks they discover sex and audio analysis for the first time.
[And don't call me Shirley]
Good. YouTube audio quality is crap. Plain and simple. 320 bps MP3 sounds better than anything Youtube offers. And 320 bps MP3 is not even "quality".
