Den 07. april 2016 23:17, skrev Thomas Davies (thdavies):
> I don’t think a time delay is a requirement that a codec can fulfil all
> on its own, so this requirement could not be directly applied as it
> stands. What is important is that if you have a system with a specific
> latency requirement, down to 100ms or below, is that when you’ve worked
> out what contribution to the latency you can allow a codec to have, you
> can readily design an encoder to meet that. For low delay that means no
> reordering of pictures, but that is trivial for a conventional hybrid
> codec. More importantly I think it means supporting bitstream features
> that allow for fine-grained rate control so that you can meet really
> tight buffer constraints with single-pass encoding.

I think this hits the nail on the head.
"No forward references" = "zero structural delay", but in addition the
*implementation* of the codec has to be capable of meeting a budget
constraint.

One limit is that the encoder has to have enough CPU or hardware to
encode one frame in less than one frame interval under ~all
circumstances. Another one is that it's able to clock it out on the wire
in less than one frame interval.

> 
>  
> 
> In practice I think that will mean supporting some kind of sub-frame
> quantisation method so you can hit a bit rate target for a frame or even
> parts of a frame. We probably want such mechanisms to allow for visual
> optimization and region-of-interest coding too.

I tried implementing a measurement tool for delay imposed by bufffering
into a limited-bandwidth pipe (accounting for the delay imposed by, for
instance, over-large i-frames) as part of compare-codecs.appspot.com. It
showed that for commercially available codecs, this wasn't a big issue -
it's a good tool to have in the toolbox, though.


> 
>  
> 
> Best regards
> 
>  
> 
> Thomas
> 
>  
> 
>  
> 
> *From:*video-codec [mailto:[email protected]] *On Behalf Of
> *Martin Thomson
> *Sent:* Thursday, April 07, 2016 7:57 PM
> *To:* Adam Roach
> *Cc:* Filippov Alexey; [email protected]; Maire Reavy; Timothy B.
> Terriberry; Randell Jesup; Jose Alvarez
> *Subject:* Re: [video-codec] NETVC encoding/decoding delay
> 
>  
> 
> As a user who frequently enjoys a network delay in the order of 320ms, I
> can confirm that this is highly unpleasant.
> 
>  
> 
> On 7 April 2016 at 15:27, Adam Roach <[email protected]
> <mailto:[email protected]>> wrote:
> 
> To be clear, the 320 figure from G.1091 is consistent with the data from
> G.114: 320ms is approximately where the the R-value drops below "users
> satisfied." The issue at play here is that both documents (G.1092 and
> G.114) deal with total system end-to-end delay. But this is a total
> number that represents a budget, and the budget must be shared by
> capture, encoding, packetization, network transit, dejittering,
> decoding, and playout. Grabbing the entire budget amount for encoding
> and decoding requires an assumption that the remaining steps will take 0 ms.
> 
> /a
> 
> 
> 
> On 4/7/16 15:07, Jose Alvarez wrote:
> 
>     Dear Mo,
> 
>      
> 
>     The end-to-end delay of 320 ms value came from section 10.1 of
>     reference [2] <https://www.itu.int/rec/T-REC-G.1091-201410-I/en>
>     Recommendation ITU-T G.1091: Quality of Experience requirements for
>     telepresence services, 2014. Certainly your data shows that this
>     value is not adequate and we should consider adjusting it.
> 
>      
> 
>     Best Regards,
> 
>      
> 
>     *José Roberto Alvarez*
> 
>     *Futurewei Technologies, Inc. *
> 
>     2330 Central Expressway, Santa Clara, CA 95050
> 
>     *Email: [email protected]
>     <mailto:[email protected]>*
> 
>     *Tel: 408-330-5443 | Mbl: 408-829-5219*
> 
>     cid:[email protected]
> 
>      
> 
>     
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>      
> 
>     *From:*Adam Roach [mailto:[email protected]]
>     *Sent:* Thursday, April 07, 2016 10:48 AM
>     *To:* [email protected] <mailto:[email protected]>
>     *Cc:* Randell Jesup; Maire Reavy; Timothy B. Terriberry; Jose
>     Alvarez; Filippov Alexey
>     *Subject:* NETVC encoding/decoding delay
> 
>      
> 
>     [as participant]
> 
>     In today's meeting, Jesup brought up an issue that Maire had raised
>     at our previous session: the section on Video Conferencing (section
>     2.2 in the current version) calls out a "preferable" end-to-end
>     delay value of 100ms. I think the discussion at today's meeting may
>     have introduced some confusion on this point.
> 
>     To be clear, this section of the document talks about "two-way video
>     and audio transmission for communication in real-time." For this use
>     case, there is a dramatic drop-off in system usability that starts
>     around 200ms. Keeping in mind that this data also must traverse a
>     network that frequently will add on the order of 100ms or
>     thereabouts, it's pretty clear that 100ms encoding/decoding delay
>     per direction is well outside of ideal.
> 
>     I'll also note that specifying 320ms as a "maximum acceptable" would
>     result in an experience that is only barely passable even without
>     network delay, and likely to become actually unusable for most users
>     once you send it across a network:
> 
> 
>     So, for conversational cases, I would argue that the "maximum
>     acceptable" value may be in the 100ms range. The preferable delay is
>     clearly much smaller than that.
> 
>     /a
> 
>  
> 
> 
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>  
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