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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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db1989
post Feb 15 2013, 00:26
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QUOTE (jensend @ Feb 14 2013, 23:15) *
If you have a mixed-content file, then for an encoder to do a good job of targeting a bitrate for the whole file while providing "constant quality" it would have to do a two-pass type deal. It has no other way of knowing, when you ask for 64kbps, if e.g. half is effectively-mono speech (both channels identical) and half is stereo music and thus it can target 32kbps for the speech, 96kbps for the music, and give you basically the bitrate you asked for.

The "VBR bitrate setting=quality level" idea we've heard so many times says an ideal VBR encoder is supposed to encode things at a constant quality level which averages out to the target bitrate across some generic ideal reference collection

At least for me, I was specifically not referring to any quality setting that targets a bitrate: I was thinking about settings that target a given quality (level of noise, whatever) without any obligation to meet a particular bitrate and therefore might be predicted to allocate different bitrates based upon material.

If I fed separate speech and loud music files to a single encoder using ‘pure’ VBR and no target bitrate, I would expect the music to come out at a higher bitrate. Extending that reasoning, I would assume to expect the same thing if the two were placed adjacently in one file. I don’t see how two passes would be necessary if no bitrate is being targeted.

Sure, it’s nice to have some vague idea of what sort of bitrate to expect from a giving setting, but IMHO, a proper VBR setting should just work with psychoacoustics and not worry about bitrate.

I think this may be exactly the point you’re making, so I don’t mean to sound like I’m repeating or contradicting you; this is just to clarify what I was trying to convey in my posts.
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