What features would you like to see in opus-tools? Options

[NullC]

I'm looking for suggested features to get into opus-tools prior to the libopus 1.1 release.

Before I get 1001 suggestions for it: One frequently requested feature which was recently added is flac input in opusenc. I'm contemplating changing opusdec to use the new opusfile library— which would give it seeking and integrated http(s) streaming support. Also already on my todo list are default comment packet padding so updating metadata doesn't require rewriting the files and adding a replaygain tool.

Some people would really like it if opusenc/opusdec supported taking multiple input files e.g. opusenc *.flac but the implicit output file naming is pretty ununixy, and would break the interface and I got flamed all to heck last time I changed the opusenc interface.... so I'm not sure if/how I want to accommodate that usage.

What other things have you found missing?

[Dynamic]

I think there was a time when VBR quality scales tended to be tuned for transparency only at one specific, 'standard' setting and often didn't have much of a variable quality scale above and below that, although they often offered extreme or insane settings too.

I guess historically, that was LAME --alt-preset standard (now -V2) and Musepack standard (aka q 5) where the respective psymodels were extensively tuned.

There's maybe some idea that Vorbis -q5 is sort of equivalent and that neroaacenc at 0.5 might be in the same ballpark, but both formats have toolkits that greatly help in obscuring artifacts without the sudden degradation in quality at lower bitrates. Even LAME now does very nicely at -V4 (formerly medium) and -V5 despite having fewer tools to hide artifacts of encode with great-efficiency than more modern codecs.

In many ways the typical generalised graphs that show a sort of rising quality reaching some sort of 'knee' before rolling over to a fairly flat region of diminishing gains with bitrate are a reasonable picture, and I suspect that the special low-bitrate tricks like Contrained Energy per critical band (very well implemented in Opus/CELT, fairly well in Vorbis) and PNS and SBR have reduced the rate of deterioration 'below the knee' in the most modern music codecs so that artifacts and stereo changes can be discerned but are rarely annoying (though for AAC+v2, I find 32kbps stereo high end pretty annoying and much prefer Opus at 32)

SPEECH/MUSIC mixes
I would say that a difference arises in Opus (and to some extent other codecs) depending on whether one is encoding speech or music. Speech seems to be remarkably good at much lower quality settings and much lower audio bandwidths than music, which offer more codec-killing sounds, more stereo separation and more high frequency content than speech. I'd guess that certain VBR codecs would dip their bitrate during speech faster than opus does, albeit from a higher starting point - often LAME -V5 would dip a lot just thanks to the monophonic content of most speech, as it does with a lot of vintage material that's mono or hard-panned stereo. I'd guess that Opus's efficiency in low-rate stereo encoding wouldn't allow so much of a dip because it's better in the first place.

In Opus I can imagine that if I were after highly acceptable speech coupled with very good sounding stereo music for, say, a podcast, my optimum might be around 24 to 32 kbps for the speech segments and around 64 kbps for the music, or 80 to 96 for really rather good music reproduction for music enthusiasts. Then again, for normal podcasts with only incidental music, 32kbps produces nice sounding music (quite like cassette tape without the hiss) and great speech, and from about 48 kbps things really start to sound fairly good with music - superficially very nice, having full bandwidth and stereo and no egregious artifacts like warbling or splashy applause that codecs like RealAudio used to produce.

I could see some advantage in a future switch for podcasts and audiobooks (or a mode built into a podcast/radio-show creation program) that automatically switches between quite a high bitrate target for stingers and music sections (32, 48, 64, 80, 96, 128 say) and a much lower bitrate target for speech sections (16, 20, 24, 32, say). Essentially, that mode is telling Opus that quality preservation isn't the main aim so long as speech is fairly good and music pretty nice, and that bitrate economy is more important. The easiest solution (if we wish to optimize more than using 32 kbps) is to put it in a podcast editor, of course, which can be told which clips are music or can have these clips ready-encoded to Opus at the optimum bitrate target (e.g. theme music and stingers) and can simply take advantage of the seamless switching of any Opus decoder.

Many talk podcasts I listen to now, would be awesome at a steady 32kbps target like now. Some could save costly hosting bandwidth by using 16/32 mix (and being talk would barely average over 16) or plain 16 or 24 target with theme music carefully selected so as not to sound awful (try transcoding the Windows 7 sample tune "C:\Users\Public\Music\Sample Music\Sleep Away.mp3" to 32 kbps or less to hear awful, with the brused snares especially in SILK mode, whereas "Ninja Tuna.mp3" and "Maid With The Flaxen Hair.mp3" in the same folder transcode to sound OK at 32kbps - all with libopus 1.0.2, that is).