What's the meaning of "sample rate" in MP3/Vorbis/AAC?, Considering the audio was transformed Options

[sheh]

What's the meaning of "sample rate" in MP3/Vorbis/AAC? Wouldn't it be meaningless after transform, same as bit depth?

[jensend]

I'm not intimately acquainted with the details of any of these formats, but from what little I do know, I think many of the answers thus far are misleading.

If you transform a periodic signal with the DCT, and thus really were only storing information in the frequency domain, you could decode that at any sample rate. Yes, like Saratoga said, the original sample rate would normally be used to define a standard list of frequencies so you can just store coefficients rather than both frequencies and coefficients, but you could still natively decode it at any frequency.

But since we're looking at the MDCT, a lapped/windowed transform, the width of the window and the overlap are time-domain information, and combining the windows, and the filtering that allows you to do that without introducing artifacts, surely involves some sample rate inflexibilities. For instance, at most sampling rates the window won't be an integer number of samples.

But it still may be possible to decode at a number of different rates which are integer multiples of each other. In particular, though [JAZ] said "an Opus stream is a 48kHz stream," Opus was designed so it can be decoded at any of 8, 16, 24, or 48 kHz. The native-lower-sampling-rate decoding isn't normally used, since they've only targeted their code for that at highly resource-constrained environments and thus that decoder gives lower quality, so even if you tell opusdec on the PC to give you 16 kHz output it will use its high-quality 48kHz decoding and then downsample, but it is there (and somebody could implement a higher-quality native-24-kHz decoder if they really felt like it).