Opus Codec 1.1.1 Brings Encoder/Decoder Optimizations

[Dick Palmer]

...mostly matters for phones...

Exactly. The NEON optimisations will certainly be handy - quality SIP calls with better battery life.

[bug77]

Ok. I just thought that you wouldn't appreciate optimizations.

I'm a developer with a thing for sneaking in code cleanups whenever I can.
I have even worked on a SIP project for a while, so I know how important a good codec is. So you see, this article struck a chord with me.

[tegs]

That's nice and everything, but what does it really mean? Can we get better sound now?
Because I doubt many end users will care that someone shove 10ms off their encoding process or that the decoding uses 1 less microwatt.

Let me rephrase the initial question: are these just routine improvements or do they enable something bigger?

Its more than that. It is a libre and patent free audio codec that is of high quality and is already the basis of WebM version 2 (VP9 and Opus). It is a dual codec that can handle voice at lower rates (like Speex) and uses CELT (better than AAC, MP3, Vorbis) at higher rates. I have been following this codec for a while and it is really awesome.

[mmstick]

I'm a developer with a thing for sneaking in code cleanups whenever I can.
I have even worked on a SIP project for a while, so I know how important a good codec is. So you see, this article struck a chord with me.

Why would it strike a chord? Is it not a good thing that someone decided to improve Opus with SSE optimizations? People work best when they work on what they want to work on.

[bug77]

Why would it strike a chord?

Memories...

[mr_tawan]

That's nice and everything, but what does it really mean? Can we get better sound now?
Because I doubt many end users will care that someone shove 10ms off their encoding process or that the decoding uses 1 less microwatt.

Let me rephrase the initial question: are these just routine improvements or do they enable something bigger?

Well Opus is meant to used in 'real-time' application. Reduce 10ms out of its latency really means something. Imagine having things like an online music rehersal service or even music game. This kind of stuff really need super low latency. At 30ms can already drive someone crazy.

[Prescience500]

Very unlikely, that there will be any significant improvements in the future. Maybe some metrics get added for a better vbr management, but even that is doubtful considering how close together all results are in this listening test: http://listening-test.coresv.net/results.htm 1.1 was done because of bad quality in harpsichord encoding.

They are actively developing quality improvements right now. It's true that people who use it for music won't likely find many improvements in the near future, but they want to get really good quality at absurdly low bitrates for telephony. As I understand it...some day, PSTN will be totally phased out in favor of the converged Internet and when that happens, networking firms like Cisco are betting on Opus to rule telephony.