How does that compare with ffmpeg? The arguments seem about the same.
You are using Opus on most VoIP/call providers.
Meanwhile, AAC, MP3, OGG... are history today.
No. Opus support in the PSTN is still unfortunately almost non-existent. Adoption is growing but the pace is very slow. Source: me, working in the European telecommunications industry for over a decade, specializing in telephony and SMS.
I like opus on VLC for youtube-dl rips but if I’m buying music I want lossless uncompressed.
Way back then (iPod era), I made an exposé for a science exam (small paper), proving that AAC was indeed better than MP3 both subjectively and objectively at most bitrates. This is how I got introduced to Fourier transforms and the likes; it was very interesting to see that you can literally "see" the difference in quality on the encoded waveform output.
But I just used the default encoders on the Mac and I didn't think about going in deep into encoder comparisons at the time. Does it matter that much? From what I know, it just about properly programming a math spec, with some tricks but I wonder if that makes that much of a difference.
In any case, above 256kbps it takes a very skilled listener to correctly identify encoded music. Apple has some useful tools for that, particularly AU Lab that allows you A/B testing of tracks with on-the-fly encoding. https://www.apple.com/apple-music/apple-digital-masters/
You are deluded, then. Or deaf. Most MP3 podcasts at 128 KBPS sound worse than an OPUS file @64.
> Does it matter that much? From what I know, it just about properly programming a math spec, with some tricks but I wonder if that makes that much of a difference.
It does matter a lot and the difference can be staggering. Producing the audio bitstream is a case of both effectively analyzing the original input and programmatically expressing the waveform as correctly and as efficiently possible. Two excellent examples: the old MP3 encoder Xing, which compared to LAME produces very poor material even at higher bitrates, and one of the earliest open-source AAC encoders, FAAC, which also renders a very poor product. A lot of "early adopters" got bitten by FAAC and many of them still stubbornly cling to the misunderstanding that AAC is a worse format than MP3.
I have used Macs since my youth and used iTunes before it was even called that, so encoder quality was never a problem I had. I guess I'll revisit the topic more deeply one of these days, still not that keen on that much math though.
Incorrect. The specification only concerns the bitstream and the decoding. Encoders are free to do whatever they want, as long as they produce a valid bitstream.
Of course, the implementations are going to have varying performance depending on code quality/optimisations but that's a given.
Edit: some other poster gave some cues to my question. I guess you can do some trickery by decomposing the waveform before encoding and recomposing it.
I'll read more on the specs.