Perhaps "pinnacle" is a bit of an over statement, but the point is sound.
EVS is quite capable, but note that that entire presentation makes no mention of Opus at all. One arises from the 3GPP the other from the IETF. Fundamentally different groups, with very different perspectives. Like AMR before it, or MPEG, EVS is basically a revenue engine for the various patent holders. Much of what EVS can do follows Opus, after-the-fact. The real strength of EVS are the compatibility modes with legacy telecom codes (AMR, AMR-WB, AMR-WB+) which virtually ensured adoption in mobile telecom. Opus is open source and free to use by anyone. It also accommodates an arbitrary number of channels, supporting various surround schemes. There are those who, fearing the appearance of some patent holder making a claim against Opus, will prefer to pay for a license to use something else. Michael Graves mgra...@mstvp.com http://www.mgraves.org o(713) 861-4005 c(713) 201-1262 sip:mgra...@mjg.onsip.com skype mjgraves -----Original Message----- From: Sursound <sursound-boun...@music.vt.edu> On Behalf Of Stefan Schreiber Sent: Friday, May 31, 2019 10:01 AM To: Surround Sound discussion group <sursound@music.vt.edu> Subject: Re: [Sursound] wifi audio (was Re: Deconstructing soundbar marketing B.S.) (Opus) > It's basically the pinnacle of audio encoding at this point, having > merged the best ideas from CELT, Silk and a few entirely new ones. > It would be hard to see how any proprietary codec vendor could compete > except where addressing a very narrow niche. - - Low delay AAC, in various versions? What about EVS? https://www.researchgate.net/profile/Anssi_Raemoe/publication/282605143/figure/fig6/AS:281480141651970@1444121503098/Combined-results-with-all-72-listeners-and-all-signal-types-with-increasing-bitrate-in.png Opus is really good. But the “pinnacle”? http://www.aes.org/technical/documentDownloads.cfm?docID=548 “A narrow niche? “ 😉🍷 I would see EVS (more or less) as the low-delay version of USAC. Best, Stefan Schreiber - - - Citando mgraves mstvp.com <mgra...@mstvp.com>: > Chris, > > Actually, I too come from a broadcast background, having installed > graphics systems into production and master controls for over 25 > years. I completely appreciate the demand for hard real-time and zero > latency. > > I've tracked Opus since its earliest days in the IETF CODEC working > group. The standard has many operative modes. It's absolutely capable > of full-bandwidth, in both lossy and lossless modes. > > You will find it both in the production/contribution side of the house > (remote codecs, STL, etc.) and distribution. It also dominates video > conference space. > > It's basically the pinnacle of audio encoding at this point, having > merged the best ideas from CELT, Silk and a few entirely new ones. > It would be hard to see how any proprietary codec vendor could compete > except where addressing a very narrow niche. > > Michael Graves > > mgra...@mstvp.com > > http://www.mgraves.org > > o(713) 861-4005 > > c(713) 201-1262 > > sip:mgra...@mjg.onsip.com > > skype mjgraves > > -----Original Message----- > > From: Sursound <sursound-boun...@music.vt.edu> On Behalf Of Chris > Woolf > > Sent: Friday, May 31, 2019 5:45 AM > > To: sursound@music.vt.edu > > Subject: Re: [Sursound] wifi audio (was Re: Deconstructing soundbar > marketing B.S.) > > On 30/05/2019 17:51, mgraves mstvp.com wrote: > >> The RF issue of range, carrier frequency, channel width is quite >> separate from the deliverable audio path. >> >> The Opus audio codec has revolutionized audio coding. It's able to >> deliver full-bandwidth audio at bitrates not much more than what was >> once typical of a telephone call. This means that the RF band need >> not be large to deliver high quality audio over a digital link. > > This answer is quite revealing of the different approaches and > requirements within our audio field. My background is broadcast audio, > so for origination purposes any digital coding has to be lossless, and > latency has to be ~very~ low. Lossy coding is fine as a delivery > format (and so would be OK for speaker feeds) but if the sound has to > be processed en route the psychoacoustic stuff doesn't stand up. > Likewise latency of 5-10ms can begin to alter performance, depending > upon how the foldback is returned to an artist. > > I don't know Opus but having read up its spec (on Wikipedia) it is > lossy and so can only be used as a delivery format. I had to smile at > 30ms latency being reported as adequate for musicians to feel > "in-time" - not for the ones I've ever worked with. Likewise the > suggestion that 45-100ms is acceptable for lipsync is laughable - > that's up to 5 TV frames adrift. Maybe audiences have become inured to > low quality standards. Latency for "live interaction" at each end of a > phone line, and face-to-face a few feet apart in a room require very > different standards - Opus's suggestion of 150ms for VOIP might just > be acceptable for the first, but it would destroy the second > application. > > I don't doubt that it is a clever and well-designed codec, and that it > is extremely useful, but one must keep in mind what it ~actually~ is > rather than what it sounds like. Opus doesn't deliver full bandwidth > audio, any more than other digitally compressed systems do. It > delivers something that convinces most ears that it is a full > bandwidth, full dynamic range signal, but it must always be remembered > what is missing. > > If you used such a system to deliver sound to speakers (assuming there > is a technique for maintaining multichannel phase coherence) it should > work perfectly well. If you used it for passing the output channels of > a microphone I doubt you would not remain happy for long. > > Which also means that the statement "the RF issue of range, carrier > frequency, channel width is quite separate from the deliverable audio > path" must be very carefully qualified - it is only correct in very > specific circumstances. > > Chris Woolf -------------- next part -------------- An HTML attachment was scrubbed... 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