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For audio at least, things are looking up. Opus, a new royalty-free codec backed by both Mozilla Foundation and Skype (Microsoft), has been standardized as an R
by graue 14y ago
For audio at least, things are looking up. Opus, a new royalty-free codec backed by both Mozilla Foundation and Skype (Microsoft), has been standardized as an RFC[1], and will likely be mandatory in WebRTC[2]. So we should be seeing that across all major browsers in a year or two.
Opus beats almost all other codecs (MP3, AAC and HE-AAC, Vorbis) in subjective quality[3], so it's a good standard to have.
[1] https://tools.ietf.org/html/rfc6716 https://tools.ietf.org/html/rfc6716
[2] http://jmspeex.livejournal.com/11042.html http://jmspeex.livejournal.com/11042.html
[3] http://www.wired.co.uk/news/archive/2012-09/15/opus-codec http://www.wired.co.uk/news/archive/2012-09/15/opus-codec, also http://people.xiph.org/~greg/opus/ha2011/ http://people.xiph.org/~greg/opus/ha2011/
- voltagex_ 14y agoIsn't Opus for speech?
- VMG 14y agoApparently not http://www.opus-codec.org/ http://www.opus-codec.org/ > Opus is a totally open, royalty-free, highly versatile audio codec. Opus is unmatched for interactive speech and music transmission over the Internet, but also intended for storage and streaming applications
- graue 14y agoNot solely. Opus is a hybrid of the SILK codec, which is more for speech, and CELT, which is more aimed at music. It can seamlessly switch between the two methods and use them simultaneously: https://wiki.xiph.org/OpusFAQ#Why_not_keep_the_SILK_and_CELT_codecs_separate.3F https://wiki.xiph.org/OpusFAQ#Why_not_keep_the_SILK_and_CELT... The main goal was streaming (of both music and speech), and hence, low latency. Matching or bettering high-latency codecs (like Vorbis) on quality was just a bonus, and I believe somewhat of a surprise to the developers when listening test results came out.
- vr000m 14y agoThe IETF is forming a new working group for an open video codec[1]. It is still a BoF and will be chartered in 3-6 months, which is when the real work will begin. More details on the charter can be found in this email [2]. As for the audio, WebRTC chose G.711 and Opus as mandatory to implement (MTI)[3]. The reason for G.711 is so that WebRTC can interoperate with legacy devices. [1] http://www.ietf.org/mail-archive/web/video-codec/current/maillist.html http://www.ietf.org/mail-archive/web/video-codec/current/mai... [2] http://www.ietf.org/mail-archive/web/video-codec/current/msg00013.html http://www.ietf.org/mail-archive/web/video-codec/current/msg... [3] http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267...