Re: [codec] #16: Multicast?

["Raymond (Juin-Hwey) Chen" <[e-mail]>]

Hi Koen,

In-line below...

You wrote:
> The essence, if I understand you correctly, is that there still exist  
> low-end platforms with barely enough processing power to run a VoIP  
> call.  If such platforms use a naive FIFO scheduler, they'll create up  
> to one frame of processing delay for encoder and decoder each, on top  
> of the frame of buffering delay.

[Raymond]: It doesn't have to be low-end platforms.  I wouldn't consider 
high-density VoIP gateways "low-end".  What matters is whether the 
processor is heavily loaded (i.e. busy at a high percentage of time) 
with real-time tasks (and thus is just fast enough). I think this is 
true for typical implementations of IP phones and VoIP gateways.

I also wouldn't use the term "a naïve FIFO scheduler" to describe the 
"run to completion" real-time scheduler that I talked about in my last 
email, because that term seems to imply that it is a very simple-minded 
and inferior approach used by an inexperienced person who doesn't know 
anything better.  My understanding from talking to the three senior 
technical leads of Broadcom is that the reality is when you have many 
real-time tasks that you need to handle concurrently, using a 
prioritized interrupt-driven scheduler is just way too complex and 
messy, and it doesn't even guarantee that you will get a lower delay if 
you do go through the trouble.  In contrast, the kind of "run to 
completion" real-time scheduler that I talked about is a more elegant 
solution as it simplifies the scheduling problem substantially and also 
allows you to have more efficient utilization of the processor. 

Other than these two points, your understanding of my main point is 
correct.

> The good news is that Moore's law will continue to drive down the  
> fraction of platforms with such processing delay problems.

[Raymond]: This may be true for PC but probably not true in general. 
PC is a general-purpose computing device that has to handle numerous 
possible tasks, and a voice phone call takes only a very small fraction 
of the worst-case computational power requirement of a PC.  In contrast, 
for special-purpose dedicated hardware devices such as IP phones or 
VoIP gateways, it would make no sense to use a processor that is many 
times faster than the worst-case computational power requirement.  For 
the sake of cost and power efficiency, the designers of such special-
purpose devices will want to use a processor that's just slightly faster
than required, because then they can use the cheapest and/or lowest 
power-consuming processor that's fast enough to get the job done.  
If they choose to use a processor much faster than is required, then 
competitors using processors just fast enough can have lower costs 
and power consumption and can take market share away from them.

A case in point: after its first appearance several decades ago, 8-bit 
microprocessors are still widely used in many devices today despite the 
several orders of magnitude of speed improvement provided by Moore's 
Law, because those devices just don't need anything faster, so using 
anything faster would be a waste of money and power consumption.

My point is that we should not expect that future IP phones or gateways 
will operate at a very low percentage point of the processor load just 
because Moore's Law can improve processor speed over time. Therefore, 
don't expect the 3X multiplier for codec frame size to go down much 
below where they are now.

In fact, if in addition to a VoIP call, a PC is heavily loaded with a 
lot of other concurrent tasks, many of which may be real-time tasks 
(e.g. video, playing/burning CD/DVD, networking, etc.), then it will be 
difficult for the PC to have small encoding and decoding RTS delays (d2 
and d5 in my delay analysis).  In this case, the codec frame size 
multiplier will be closer to 3X than to 1X, unless you are willing to 
let the voice stream occasionally run out of real time and produce an 
audible glitch (which is not acceptable from the voice quality 
perspective).  If you agree with this and agree that a PC sometimes 
does get very heavily loaded, then if you don't want the voice stream
to run out of real time, the worst-case codec-dependent delay for 
PC can still be around 3X the codec frame size.

> I'm a bit surprised by your analysis of "packet transmission delay",  
> as it has little bearing on our multiplier (ie the change in delay as  
> a function of frame size). See old posts.

[Raymond]: I am not sure I understand what you are saying.  You probably 
misunderstood the goal of my analysis. I mentioned in my last email that 
my delay analysis aimed to derive the lower and upper bounds of the 
codec-dependent one-way delay as functions of both the codec frame size 
AND the packet size.  That "packet transmission delay" does depend on 
the packet size, so it should be included.  Also, including it doesn't 
increase the lower bound of the delay (and the codec frame size 
multiplier there); it only affects the upper bound.

Or, are you saying the "packet transmission delay" depends on the packet 
size, not the codec frame size, and therefore is not codec-dependent? 
Well, we know the packet size should be a positive integer multiple of 
the codec frame size.  Once the codec frame size is determined, there 
are only limited choices of packet sizes you can use, so in this sense 
the packet size does depend on the codec frame size.  Therefore, the 
"packet transmission delay" indirectly depends on the choice of the 
codec.

Best Regards,

Raymond