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Which, isn't substantially different than evaluating a codec on introduced SNR. These classic signal processing analyses tell you nothing about the correct ope
by saidajigumi 10y ago
Which, isn't substantially different than evaluating a codec on introduced SNR.
These classic signal processing analyses tell you nothing about the correct operation of a psychoacoustic model codec design (i.e. MP3, AAC, Vorbis, etc.) An analysis of A (original signal) vs. A' (signal passed through a compression-decompression cycle) is just extracting the quantization noise introduced by the codec. That provides no information about how effective the codec was in masking that noise with the original signal content.
To illustrate, imagine a perversely designed codec: it runs two models, the first "good model" is a normal psychoacoustic model. The second "bad model" is the one the compressor uses: it applies the total amount of quantization noise allowed by the good model, but applies it in ways that are maximally annoying to human listeners. This isn't just avoiding masking, it's things like using noise correlated to the original signal, which is generally more obtrusive than uncorrelated noise, etc.
A codec using just the "good model" and one using the "good + bad model" would have (by definition) exactly the same introduced noise, but the latter would sound FAR worse to a human listener.
- ArkyBeagle 10y agoI am not familiar with the term "introduced SNR"... Googling. No joy. I can't really follow your thinking above; sorry. In audio, there is distortion, which is correlated with the signal. Noise is uncorrelated. Codec error would seem very much to me to be at least much more correlated than uncorrelated. MP3 artifacts sound at least more "phase-ey" than they sound anything like quantitization noise ( at least to me ). This may be because I have heard badly aligned tape machines make that sort of error happen. "Phase-ey" also triggers my (feeble) mind into thinking about allpass filters as the model for the error. In terms of intelligibility, it's possible to improve intelligibility by adding noise, and by adding clipping - aviation comms does this at times. What destroys intelligibility is phenomes being destroyed by phase changes and bad amplitide errors ( where there are actually good amplitude errors ). I've run an ABACUS voice quality analyzer several times, and I don't think it's purely a distortion analyzer - adding clipping at least can improve MOS/PESQ score surprisingly. Even more surprisingly, there's no mechanism available for calibrating gain staging on one.
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- saidajigumi 10y agoAh, that's a mistake, that should have read "introduced [quantization] noise", which reduces the SNR in A' vs A. If I understand you correctly, your discussion around intelligibility primarily applies to signal processing of voice vs general audio. Codecs such as MP3, AAC, etc. don't have the luxury of making assumptions about the signal content, and so aren't designed along those principles. E.g. speech codecs can generally run well at much lower bitrates than general audio codecs because they operate on a constrained domain of audio (i.e. speech). Regarding distortion management, see Rate-distortion optimization[1] for lossy audio codecs: where the purpose is to manage distortion within the limits of the bit rate supported by a communication channel or storage medium. [1] https://en.wikipedia.org/wiki/Quantization_(signal_processing)#Rate.E2.80.93distortion_optimization https://en.wikipedia.org/wiki/Quantization_(signal_processin...
- ArkyBeagle 10y agoThanks for clarifying.