- As saratoga says, 24 bits has nothing to do with frequency. It is about the maximum difference between "lowest possible positive volume" and "loudest possible volume". 24 bits is more than your ears can handle. 16 bits is enough to cover what anyone needs to listen to in an ordinary room.
- Frequency is directly related to sampling frequency. A 96 kHz file usually includes more inaudible noise. Usually only a waste of space.
- Why they sell 24-bit files even though you can't tell the difference? Because someone is willing to pay for it (and, maybe they save customer support expenses by less people nagging them about it).
- If you get two files from the same place at the same time, they are usually from the same master. If you get two files of the same recording but one was made, say, some years ago, it could be a different mastering. They could sound different because they could be, in reality, different mixes although it just says "remaster". (Mastering is more than just transferring from file to CD.)
- When comparing, you must match volume. Same file at different volumes, sound different.
Oh, it's interesting you say those. I thought framesize 60 improves compression a bit (while reducing transient representation as a tradeoff). Other codecs like HE-AAC and USAC seems to make a benefit from 100ms+ delay in those extreme low bitrates. I'm surprised it's not the case with Opus.
CVBR was used because I'm testing for mobile broadcasting so bitrates shouldn't fluctuate too much. But hard-cbr seemed too aggressive.
Dude, why? The beauty of the podcasts is that they are talk audios and they can be as low as 16/24kbps.
I wonder which codec is being used with --bitrate 32 --framesize 60 --cvbr --downmix-mono setting right now in 1.2-alpha. Seems like a nice spot for very low bitrate generic music. It's still noticable not transparent, but artifacts are not utterly annoying (at least compared to HE-AACv1 mono at this bitrate). For me it's even listenable with headphones on. But it sounds a bit different, like it was using SILK instead of CELT.I could be using either SILK or CELT, depending on whether it thinks it's speech or music. Also, don't use --framesize 60 and --cvbr. They make things worse.
One of those days, eh? But no harm done; after all, it took me three days to notice that the last version had been uploaded. In comparison, my update today felt like getting it instantaneously.
I wonder which codec is being used with --bitrate 32 --framesize 60 --cvbr --downmix-mono setting right now in 1.2-alpha. Seems like a nice spot for very low bitrate generic music. It's still noticable not transparent, but artifacts are not utterly annoying (at least compared to HE-AACv1 mono at this bitrate). For me it's even listenable with headphones on. But it sounds a bit different, like it was using SILK instead of CELT.
Not closely related, but is there any algorithm which can more gracefully downmix stereo to mono in order to avoid phase cancelation as much as possible? Does Opus involves anything like that with --downmix-mono, or it's just using a simple downmix algorithm? Imo below 48kbps Opus is not capable enought to represent stereo without nasty artifacts at the moment (especially bad when played back on single speaker devices, like mobile phones), so I'm preferring the mono downmix. But that's just my 2 cents. Maybe reducing the stereo width and phase significantly before encoding could help, but then again, you can just downmix to mono then.
If you'd visited the components site, you'd see I forgot to upload it. But it's there now!
edit: just had to re-upload the files because of a typo in changelog.
Last post by saratoga -
It should be zero padded to the least significant bits. If that isnt what you want just lower the gain 48 dB.
No one here is going to be able to tell you that.
24 bit has nothing to do with frequency.