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Topic: How does the opus encoder deal with a 24-bit source? (Read 508 times) previous topic - next topic
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How does the opus encoder deal with a 24-bit source?

I know opusenc can take a 24-bit audio file as an input.

But I want to know if the source audio file will be converted to 16-bit first and then be encoded into an opus file.

Re: How does the opus encoder deal with a 24-bit source?

Reply #1
As far as I know everything is converted to 16/48 before it's encoded. Also if you need to keep the 24 bit depth I suggest to use a lossless codec instead of a lossy codec. In practice the bitdepth can get lower than 16 bit theoretically.

Re: How does the opus encoder deal with a 24-bit source?

Reply #2
As far as I know everything is converted to 16/48 before it's encoded. Also if you need to keep the 24 bit depth I suggest to use a lossless codec instead of a lossy codec.

I don't prefer 24-bit over 16-bit (can't hear the difference), I am just curious.

In practice the bitdepth can get lower than 16 bit theoretically.

Could explain what does that mean?

Re: How does the opus encoder deal with a 24-bit source?

Reply #3
Actually all input is converted to 32-bit floating point and processed in that format.

Re: How does the opus encoder deal with a 24-bit source?

Reply #4
Actually all input is converted to 32-bit floating point and processed in that format.

Correct. It still means you won't be keeping the (theoretically) very low noise of your 24-bit recording (for that you need lossless). However, you *will* be keeping all of the dynamic range. Opus is capable of representing the full 24-bit dynamic range, and then some.

Re: How does the opus encoder deal with a 24-bit source?

Reply #5
But I want to know if the source audio file will be converted to 16-bit first and then be encoded into an opus file.

Almost all practical PC lossy encoders use floating point, so everything gets converted to 32 bit. 

Re: How does the opus encoder deal with a 24-bit source?

Reply #6
except fdk-aac, I think

Re: How does the opus encoder deal with a 24-bit source?

Reply #7
fdk-aac must deal with clipped integer samples, and is hard coded to specific integer precision. I even tried decoupling that a bit, to support fixed point that exceeds ±1.0, but the SBC and/or PS code break horribly if the input and output aren't clipped first.

 
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