[opus] Antw: Re: Antw: Re: Possible bug in Opus 1.3

Emily Bowman silverbacknet at gmail.com
Mon Nov 5 19:46:00 UTC 2018

On Mon, Nov 5, 2018 at 11:01 AM Jan Stary <hans at stare.cz> wrote:

Attached I send the spectrogram (vic SoX) of the first 20 seconds
> for the wav file and the opus file. Indeed, there is extra noise
> for the low frequencies, but somewhere around -100 dB.
>         Jan

That might be entirely due to SoX treating it as a 16-bit file, which it is
not; -100dB is almost exactly the limitation of 16-bit. All Opus files are
infinite-precision, and they'll encode the input at whatever precision is
fed to them, but they do have a silence-detection mechanism which defaults
to 16-bit in opusenc. SoX is either reading that value and erroneously
assuming that's that internal precision, or just falling back on a 16-bit
default because it has no information, but it would be more proper to
_always_ decode to 24-bit to eliminate that broadband noise low bit depths
create (except in the case of hardware limitations).

As you've found, Opus is always 48kHz, never more, never less. Its
resampler is very accurate, and should never introduce noise.

My speakers and headphones definitely have issues with the sweep, so it's
hard to isolate any differences. (I do love it though, it really gets my
floor shaking; I'll keep it around for testing purposes.) The only
remaining issue is the way that Opus doesn't even come close to respecting
the requested bitrate for this sample. For instance, encoding it at 130
gives me a file of 210kbps. Over a wide corpus of music, I've noticed
libopus almost invariably overshoots the bitrate, rather than averaging out
close to it; only speech is consistently at or below it.

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