OpusTodo: Difference between revisions

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== Code ==
== For 1.2 ==
* Low bitrate quality improvements
* AVX optimizations
* Fix compilation as a single module for gecko


=== For IETF draft ===
== Spec ==
* Code cleanup
* Matroska mapping. See: [[MatroskaOpus]] And firefox/ffmpeg implementation
** Remove SKP_ prefixes
* RTP payload format. Mono/stereo mapping is complete [[https://tools.ietf.org/html/rfc7587 RFC 7587]], no multichannel mapping yet.
** remove silk_ prefixes in file names
* mp4 mapping. See [[https://opus-codec.org/docs/opus_in_isobmff.html ISO Base Media File Format draft]]
* Multi-channel signalling
 
== Website ==
* De-uglify webpage - some suggestions:
** write about codecs obsoleted by OPUS (Speex, CELT, Vorbis(?) and the proprietary ones)
** write about implementations (libopus encoder/decoder, libavcodec decoder, any others?)
** [https://en.wikipedia.org/wiki/Comparison_of_audio_coding_formats audio codec comparison table] (Opus, Vorbis, Speex, ..., MP5) of features (channels, freq, bits per sample, license, language (C89), integer impl. (Vorbis decoder only, Opus YES, ...)
** future use in video files (Theora? Dirac? WebM? other future codecs...)
** audio files for storage (like Vorbis, no raw Opus defined, only inside OGG), ...
* Promotional material (some nice free/public-domain sounds/radio stations in Opus format)
 
== Other ==
 
* Oggz-validate (should also validate opus toc)
 
== Opus-tools ==
* Port opusdec to libopusfile/libopusurl.
* A simple real time streaming example tool
** Start with opusrtp.c in [https://git.xiph.org/?p=opus-tools.git opus-tools]
** Make <code>opusrtp rtp://example.com:5431/</code> listen to that host and port and mux packets from there. Generalize the cpac bases --sniff implementation
** Make sending similarly generic. Maybe just <code>opusrtp source.opus -o rtp://example.com:5431/</code> to send source.opus out to the destination?
** Make --sniff save one file per
** Implement DTLS-SRTP. See webrtc.
** audio capture/encode, decode/playback?
** Parse and act on sdp for convenience and testing.
 
* EBU R128/Replaygain (half done— needs a gain tool)


== SILK issues ==
== Surround work ==
* PLC buffer not fully initialized (fix needs verifying)
* Mid and side using different sampling rates (fix needs verifying)
* <s>LLBR stereo issue (has a proposed fix)</s>
** <s?Introduces prefill bug (fixed on greg's tree)</s>
* Artefacts on SILK mono<->stereo switching
* Artefacts on Hybrid/CELT switching


=== Later ===
* Apply spreading to energy masking
* <strike>Exposed CELT constrained VBR</strike>
* More conservative energy masking (not just mean difference) and dynalloc
* <strike>Fixed-point build</strike>
* Allow SILK/hybrid on center channel for voice?
* <strike>Fix build system (right now it fails to build shared libraries, drops .o files all over)</strike>
 
* <strike>Floating point API</strike>
== Psychoacoustic stuff ==
* Usable command-line tools (opus is a major regression from libcelt right now)
 
* Smart automatic mode decision
* Adaptive width narrowing and forced intensity stereo bands
* psymodel based VBR
* Remove copy in inverse MDCT


== Spec ==
== Optimisations ==


* Finish codec draft
* Vectorising comb_filter()
* Ogg mapping (including multi-channel). See: [[OggOpus]]
* Use 16-bit mul plus shift in denormalise_bands()
* RTP payload format
* Optimise MDCT somehow


== Other ==
== Third-Party tool enhancements ==
* mutagen: [https://bitbucket.org/lazka/mutagen/issue/202/oggopus-support-in-place-rewrites-for support padding in comments header], [https://bitbucket.org/lazka/mutagen/issue/203/oggopus-allow-updating-the-output_gain allow updating output gain in ID header]


* Logo
== Future work ==
* Test vectors
* psymodel based VBR
* Listening tests
* Remove copy in inverse MDCT
* Documentation (at a minimum every exported symbol should have complete and accurate documentation)
* Save some float<->int conversions
* Add content to opus-codec.org
* Improvements to LP mode CBR (greg has some code)
** The above documentation
* Unconstrained SILK VBR
** Presentations
* Better handling for the case where FEC has a different bandwidth than the current mode
** Examples and test results  (hyperlink to Monty's demo, gmaxwell's HA results page, etc)
* PLC transitions on unprotected SILK-SILK bandwidth changes?
* Useful example software (e.g. streaming software that works correctly) (opus-tools in xiph git)
* Figure out how to use speech/music detection optimally
** <s>Support for resampling in tools</s>
** find optimal switching time (low energy/tonality)
* Improve variable frame size


== Third party software ==
[[Category:Opus]]
* Support in ekiga
* Support in mumble
* Support in asterisk
* Support in firefox (rtcweb and in ogg)
* Support in VLC
* Support in ogg123
* Support in ffmpeg
* Support in rockbox
* Support in foobar2000
* Support in gstreamer
* Support in mplayer
* Support in xmms
* Support in oggdsf
* Support in xiphqt

Latest revision as of 18:59, 19 July 2016

For 1.2

  • Low bitrate quality improvements
  • AVX optimizations
  • Fix compilation as a single module for gecko

Spec

Website

  • De-uglify webpage - some suggestions:
    • write about codecs obsoleted by OPUS (Speex, CELT, Vorbis(?) and the proprietary ones)
    • write about implementations (libopus encoder/decoder, libavcodec decoder, any others?)
    • audio codec comparison table (Opus, Vorbis, Speex, ..., MP5) of features (channels, freq, bits per sample, license, language (C89), integer impl. (Vorbis decoder only, Opus YES, ...)
    • future use in video files (Theora? Dirac? WebM? other future codecs...)
    • audio files for storage (like Vorbis, no raw Opus defined, only inside OGG), ...
  • Promotional material (some nice free/public-domain sounds/radio stations in Opus format)

Other

  • Oggz-validate (should also validate opus toc)

Opus-tools

  • Port opusdec to libopusfile/libopusurl.
  • A simple real time streaming example tool
    • Start with opusrtp.c in opus-tools
    • Make opusrtp rtp://example.com:5431/ listen to that host and port and mux packets from there. Generalize the cpac bases --sniff implementation
    • Make sending similarly generic. Maybe just opusrtp source.opus -o rtp://example.com:5431/ to send source.opus out to the destination?
    • Make --sniff save one file per
    • Implement DTLS-SRTP. See webrtc.
    • audio capture/encode, decode/playback?
    • Parse and act on sdp for convenience and testing.
  • EBU R128/Replaygain (half done— needs a gain tool)

Surround work

  • Apply spreading to energy masking
  • More conservative energy masking (not just mean difference) and dynalloc
  • Allow SILK/hybrid on center channel for voice?

Psychoacoustic stuff

  • Adaptive width narrowing and forced intensity stereo bands

Optimisations

  • Vectorising comb_filter()
  • Use 16-bit mul plus shift in denormalise_bands()
  • Optimise MDCT somehow

Third-Party tool enhancements

Future work

  • psymodel based VBR
  • Remove copy in inverse MDCT
  • Save some float<->int conversions
  • Improvements to LP mode CBR (greg has some code)
  • Unconstrained SILK VBR
  • Better handling for the case where FEC has a different bandwidth than the current mode
  • PLC transitions on unprotected SILK-SILK bandwidth changes?
  • Figure out how to use speech/music detection optimally
    • find optimal switching time (low energy/tonality)
  • Improve variable frame size
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