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It defines how some channel can be standalone channels (mono) and some other coupled (stereo), and how one can for example reconstruct the usual surround audio 5.1 by having a set of stereos and audio channels together. The original codec specification details all of the 256 channels opus can handle. It even has an RTP Payload specification, so it can be used directly in webrtc. It has better quality than other default codecs for almost any given usage (see above).

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ItвЂ™s free as in beer (royalty free) and free as freedom (open source implementation and standard from 2012, updated in 2017). Eventually, the standard committee decided to for the more open Opus, whose stereo form (the only one standardised for RTP media transport) would be mandatory to implement.

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(it s slightly more complicated, MP4 would require high profile, which openH264 does to provide but itвЂ™s enough for the sake of this discussion).Д®xtremely quickly the licensor of AAC removed the corresponding option de facto preventing AAC from getting the same treatment.

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Using the fact that the H264 licensing cost is capped, and that Cisco was already paying the maximum, Cisco de facto made H264 software implementation free (Hardware implementation license cost is shouldered by the hardware manufacturer), for anybody who would use Cisco provided pre-compiled binaries. The adoption of H264 in the webrtc standard (I should say rtcweb) was facilitated by the release of openH264 by Cisco in 2014. So why is it not here already? Even RTMP supported AAC, not surprisingly since AAC was released in 1997. If you were to add AAC support, you would have all you need to support the entire MPEG4 format, which has an existing RTP payload format, and a separate backward compatible extortion for Surround Audio. When it comes to Webrtc, H.264 made it in the list of the mandatory to implement codecs. That explains the delay in implementation and adoption of codecs between the media stack and the webrtc stack (question like, does Chrome support codec N are thus difficult to answer with a simple yes or no). You also need to have an RTP payload spec for the codecs you want to use in webrtc, where the definition of the codec itself is enough for streaming through MSE. It is much easier to make a real-time decoder, which everyone want to use, than it is to make a real-time encoder, which only webrtc will use. For AV1, the support was added to browser more than a year ago, and to webrtc only right now.

If we take the example of AAC, the support is here for the Video tag and the Media Source Extension, but not in Webrtc. вЂ“ Media Source Extension (still linked to the Video Tag)(decoder only) вЂ“ What the Video tag can support directly (decoder only) However, if you want interoperability with the web, you are limited to 3 options:

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Replace Apple Tv by Roku, AndroidTV, etc and the sentence is still true. If you use a native iOS app, to send to an AppleTV for example, the HTML5 specifications do not apply to you, you are only limited by what the AppleTV let you do. SURROUND SOUND OVER THE INTERNETДЇirst, if you use native apps, you can do whatever you want. Wait, did I say netflix? ItвЂ™s that VOD thingy over the internet, right? So it is possible to get 5.1 over the internet then? Well, yes, but they are nuances. If you want a crash course on the past 40 years of audio (and feel old like I did), this site does a good job at showing the different technologies in an historical way, and simple words.

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We are in 2020, and #webrtc supports stereo opus at best? LetвЂ™s not even compare to current audio standards AAC 5.1 you can get by default on AppleTV or Netflix, or Dolby Atmos in movie theatre. The first surround sound format (quadraphonic sound) came out in 1969.

While all of those previous claims are false, there is one that was correct: the default audio codecs in webrtc implementations, and especially in browsers, are not as good when it comes to spatial info than their streaming or gaming counterpartsвЂ¦ untill INTRODUCTION TO SURROUND SOUND Webrtc cannot do, itвЂ™s just good enough for webcams, it cannot use more than 2.5 Mbps, вЂ¦.

While scalability is often the first excuse some use not to use WebRTC, a persistent claim even though that myth has been debunked long ago and many time over since, вЂњqualityвЂќ is certainly the second one.