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Wolf480pl

I have one opus file, 2 versions of ffmpeg.

I use each ffmpeg to convert the file to s16le (which I'm assuming is raw PCM audio)

The results differ.

why?

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Q.U.I.N.N.
@wolf480pl floating point decoding isn't guaranteed to be bit identical maybe?
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Wolf480pl

@icedquinn hmm... do you know if opus uses floats?

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Alexander Monakov

@wolf480pl @icedquinn I can confirm that, both the ffmpeg decoder and reference library use floats; the reference library can be configured to avoid floating-point arithmetic, but that's not the default.

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Alexander Monakov

@wolf480pl @icedquinn but even without floating-point, you can get different decodes if your other version of ffmpeg doesn't know about the clever backward-compatible upgrade:

In Opus 1.5, we added a deep learning based redundancy encoder that enhances audio in lossy networks by embedding one second of recovery data in the padding data of each packet. The underlying algorithm behind encoding and decoding the recovery data is called the deep redundancy (DRED) algorithm. By leveraging the padding data within the packet, Opus 1.5 is fully backward compatible with prior revisions of Opus. Please see the README under the "dnn" subdirectory to understand DRED.
Alexander Monakov

@wolf480pl @icedquinn hm, but on second thought, that is specifically for streaming over networks, and wouldn't be used in conventional encoding?

Aug 27, 2024, 09:36 AM·Public·Pinafore
0boosts·1favorite
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Wolf480pl

@amonakov @icedquinn
streaming over networks?
like youtube?

So if you youtube-dl the opus audio from a youtube video without reencoding, it'd have that?

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Alexander Monakov

@wolf480pl @icedquinn possibly? from the sound of it, the encoder would have to opt-in for that, and I barely know what I'm talking about here.

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Alexander Monakov

@wolf480pl @icedquinn again, on second^2 thought, in case of Youtube that wouldn't make sense, because they are streaming over TLS, which is not lossy. It is for use cases where the protocols are specifically built with dropouts in mind, like videoconferencing.

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Q.U.I.N.N.
@amonakov @wolf480pl packet recovery is for live situations, yes. when you are firing packets over UDP and there is no time to otherwise reassemble the data without it being woefully outdated
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Q.U.I.N.N.
@amonakov @wolf480pl the secret to modern "low latency" protocols is just that they carry error recovery data. instead of having to renegotiate data they repair it locally. you can get this on just about anything with the right proxy servers (some chinese people were using that to play games around the firewall.)
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Wolf480pl

@icedquinn
> some chinese people were using that to play games around the firewall

not just chinese people

also, when reading @amonakov 's screenshot for some reason I didn't notice that "lossy" referred to the network, not to the codec.

But then, if the extra data is only some form of error-correcting code, why would it affect the decoded stream when I have the complete encoded stream with no holes in it?

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Q.U.I.N.N.
@wolf480pl @amonakov is this all an academic curiosity?

its a lossy codec so bit identical decoding is a non-goal. you want flac or wavpack for that (lossless but bigger.)
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Q.U.I.N.N.
@wolf480pl @amonakov i remember one version of ffmpeg inserted unwanted resampling steps when making flacs. i'm not sure if its doing this to your opus files. but some versions did things like that. or used different dithering when converting bitrates.

later versions fixed it and stopped resampling to flac but that's one way that non-identical outputs were happening.
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Alexander Monakov

@icedquinn @wolf480pl I don't think it's academic, checking decoded data is how you catch unintended bugs; floating-point decoding can be reproducible if you know what you're doing

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Wolf480pl

@amonakov @icedquinn
I looked at the opus test vectors, and looks like they don't check it bit-by-bit, instead they use opus_compare, which seems to calculate per-band energy in a sliding window manner, and then compare those with some epsilon or sth...

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Q.U.I.N.N.
@wolf480pl @amonakov the lossless codecs are bit-perfect.

the integer decoders *might* be but i haven't played with those.
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Wolf480pl

@icedquinn @amonakov
yeah, I think rounding errors are a sufficient explanation.

I would've assumed that with a lossy codec, all of the lossiness would be in encoding, and the decoder would be deterministic. But I guess nobody cares and that leads to subtle differences between decoders.

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Michal :verified: :btw:

@wolf480pl @amonakov @icedquinn "streaming over networks" is meant more for things like "voice calling over lossy network", the feature is disabled by default: opus-codec.org/demo/opus-1.5/

opus-codec.orgOpus 1.5 Released
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Wolf480pl

@michal @amonakov @icedquinn can ffprobe tell me if this is enabled in a file I have?

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Michal :verified: :btw:

@wolf480pl @amonakov @icedquinn Sorry, I don't know, maybe try `ffprobe -show_frames` with the two ffmpeg versions?

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