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Can audio encoders target quality w/o caring about bit rate/file size?, [OP = softrunner / split from “IETF Opus codec now ready for testing”]
softrunner
post Feb 14 2013, 02:33
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QUOTE (Martel @ Jan 1 2013, 14:46) *
QUOTE (softrunner @ Dec 29 2012, 04:50) *
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
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Big_Berny
post Feb 14 2013, 12:11
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QUOTE (softrunner @ Feb 14 2013, 02:33) *
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.
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m0rbidini
post Feb 20 2013, 03:46
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I think encoders in true VBR mode could in theory detect "pure speech" sections (although this may not be very simple to define) and end up outputting lower average bitrates for those sections while maintaining a quality level that results in higher average bitrates for non speech sections, ie with both sections having the same average results in listening tests. That may be a future improvement.

If that's not so simple maybe it's just the case that devs prefer to err on the safe side.

Are there any lossy codec devs or researchers that can share any insight on this?
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