Can audio encoders target quality w/o caring about bit rate/file size?, [OP = softrunner / split from “IETF Opus codec now ready for testing”] Options

[softrunner]

I don't know weather it is possible for encoder to do such an analysis of a source audio, but it would be great it yes.

It's only a matter of finding the right formula/algorithm.

x264 video encoder has encoding mode called Constant Rate Factor. In this mode number (16, 17, etc) is used to define desired quality (lower - better quality and higher bitrate), and encoder does not care about bitrate, only about keeping rate factor constant. It is a question, why nobody has invented something similar for audio encoding (except lossyWAV, which needs too much bitrate for acceptable quality)?
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Opus 1.1 Alpha has some bugs, which can be found using samples from thread High Frequency Listening Test Samples. For example, at 16-24 kbps Opus gives this:

and for 32-40 kbps it gives this:

For samples 1_12kHz, 1_20kHz, 2_8kHz, 2_12kHz and 2_20kHz Opus sounds wrongly even at 512 kbps.
Full set of files is here (problematic sampes are marked with exclamation mark). Hope, developers will use this samples in their work.

This post has been edited by softrunner: Feb 14 2013, 02:34

[Big_Berny]

x264 video encoder has encoding mode called Constant Rate Factor. In this mode number (16, 17, etc) is used to define desired quality (lower - better quality and higher bitrate), and encoder does not care about bitrate, only about keeping rate factor constant. It is a question, why nobody has invented something similar for audio encoding (except lossyWAV, which needs too much bitrate for acceptable quality)?

I think every encoder with real vbr (not abr) does that? Lame has V(0-9), QT AAC has --tvbr (0-127), Vorbis has -q((-2)-10). The bitrate may vary a lot with these settings between different songs/genres.

[IgorC]

Speech isn't that easy to code. http://research.nokia.com/files/public/%5B..._Opus_Codec.pdf

Opus uses hybrid mode only at very low bitrates. Speech requires comparable bitrate as for music for (near) transparent or high quality . There is no such thing as smart encoder that does"64 kbps for speech and 128 kbps for music".
That's enough to say that Opus 1.1 alpha (--bitrate 64) produces bitrates considerably >64 kbps on speech. It doesn't go anyhow lower.

[db1989]

Speech isn't that easy to code. http://research.nokia.com/files/public/%5B..._Opus_Codec.pdf […] Speech requires comparable bitrate as for music for (near) transparent or high quality .

Thanks for this! It supports earlier suppositions that bitrates for speech that are similar to music point to speech being more complex than we estimate, not to any failing in VBR modes.

I guess we’re conditioned to think of speech as requiring low bitrates, when in fact it’s often just a case of people forcing low bitrates due to constraints upon bandwidth or capacity, or even just habit. I can appreciate that actually encoding speech at a level that matches music may be more of a challenge than is assumed. That was the case for me, anyway.