[Speex-dev] New jitter.c, bug in speex_jitter_get?

[SteveK]

On May 3, 2006, at 9:54 PM, Jean-Marc Valin wrote:

>> Perhaps, but then you need to assume that the jitterbuffer can just
>> throw away the data, and that limits how you can use it.  In object-
>> oriented terms, you might want to pass objects to the JB, and then
>> call a destructor on them.  In C terms, you may want to allocate
>> frames via malloc(), and then call free() on them later.  You might
>> want to pass in reference-counted objects of some sort, etc.
>
> Are we talking about the same thing here? I'm talking about IP  
> packets,
> more specifically datagrams. These contain a bunch of bytes and a
> length. With RTP you get a timestamp as well. That's all. No object
> oriented stuff until you decode them (which is done after it leaves  
> the
> jitter buffer).

It depends on the protocol.  RTP packets, though, don't just have the  
audio payload and a length;  they also have the timestamp, the  
payload type, etc.  Some RTP packets may be audio data, some may be  
video, some may be DTMF digits, etc.

It's not brain surgery, but you'd generally parse these into some  
kind of structure, even if the structure is just a mapping onto a  
buffer.  Anyway, the point is that if you want to an abstraction  
above the jitterbuffer, it makes sense that the jitterbuffer would  
treat it's payload as opaque.  That would mean it can't just throw it  
away.  Then if it wants to destroy it, you could either allow the  
jitterbuffer to callback to an application-passed destroy function,  
or have it return it with a flag of some kind.

>
>> Mainly they're different because you don't ever want the jitterbuffer
>> to throw them away -- you always want to deliver them.  They probably
>> have zero duration (are impulses), and will overlap in timestamps
>> with the audio frames.  You may not want to consider them in your
>> jitter calculations.
>
> Depending on how the control stuff works, you probably don't *need* a
> jitter buffer in the first place. At best you'll want to reorder the
> data, no?

Hmm, I just had this discussion here:  http://bugs.digium.com/ 
view.php?id=6011  Here's the case where you would want it:

The theory behind doing that is that in general, you want control to be
synchronized with voice.  Imagine a situation where you are leaving a
voicemail, and you have a big (say 1500ms) jitterbuffer;  The VM system
lets you press "#" to end the recording.   So you say "I'll pay you A
million dollars; just kidding!", and press "#";  If the system acted on
the DTMF immediately, your message might be seriously misinterpreted.

The same concept holds true for processing HANGUP frames, etc.

I don't know what is done by other applications, and buffering  
control frames, etc certainly leads to complication.

>
>>> Also, why would you want to give it structs? AFAIK, IP packets
>>> can only contain bytes anyway.
>>
>> Of course.  But, in the way I've used the JB, and I would imagine in
>> most cases, the application which uses it is going to be parsing the
>> network stuff before putting it into a JB, and would put it into a
>> structure or object.  Clearly, everything is just bytes, and you
>> could do something similar with your JB api by passing in pointers
>> and len==4, _if_ your jitterbuffer didn't have the ability to just
>> drop frames internally.
>
> Why would you parse and do work *before* putting it in the jitter
> buffer, especially not even knowing whether you'll actually use it.

In the case of IAX2, there's lots of reasons:

1) You need to determine which call it belongs to.
2) If it's a reliably-sent frame, you acknowledge it immediately.
3) Some frame types are not buffered.


>
>> The time in my implementation doesn't need to be wall time, nor do
>> timestamps;  They're all relative to each other, and the beginning of
>> the "session".   I think everything would work OK +- some constants
>> if the scale were different.
>
> But why do you need that time in the first place?

Time is just what drives the output of the jitterbuffer.  Time could  
be determined by the number of samples needed to keep a soundcard  
full, or it could be time according to the system clock, if you're  
not (or not able to) synchronize to a soundcard.

I don't think it's very different from your concept of ticks.  I use  
actual measures of time, because the implementation of my  
jitterbuffer sometimes makes decisions where knowing what time means  
helps.


>
>>> Why no overlap? What if you want to include a bit of redundancy
>>> (doesn't
>>> have to be 100% either) to make your app more robust to packet
>>> loss? You
>>> could want to send a packet that covers 0-60 ms, followed by 40-100
>>> ms,
>>> followed by 80-140 ms, ...
>>
>> I see now.  I hadn't considered this, but it could also be expressed
>> as a sequence of 20ms frames, some of which are dups, and some which
>> have identical arrival times.  I'm not sure how my implementation
>> would handle this, but I don't think it breaks the API.
>
> What do you mean expressed as a sequence. You mean you'd break the
> frames down before sending them? Sounds complicated and even  
> technically
> impossible for the general case (what if the frames *can't* be broken
> down for a particular codec?).

Again, I don't think the API prohibits the use that you've  
described.  You can call jb_put() with that overlapping data.  My  
_implementation_ may not like it, but the API can support it.

>
>>> Well, that API clearly has limitations that mean I can't use them
>>> to do
>>> what I need. Unless you're willing to change that (and even then
>>> I'm not
>>> sure), there's no way we can use the same API. I still suspect it
>>> may be
>>> possible to wrap by current API in that API. Of course, some  
>>> features
>>> would just not be available.
>>
>> I think it would, except that your API lets the jb destroy data on
>> it's own, which would be bad, for example, if the data was a control
>> frame, or in every case, because frames are usually malloced.
>
> What's the problem about the jitter buffer destroying control  
> frames. If
> you need to send them reliably, don't use UDP in the first place and
> don't use a jitter buffer.

See above.  I'm not married to the idea of buffering control frames,  
but there's a
valid argument to be made for it.

>
>> Yours may indeed be better than mine, but before you say it won't get
>> confused, let's see what happens if it gets into asterisk and a lot
>> of real-world broken streams get thrown at it :)
>
> Of course, I'm always interested in more testing. However, I've  
> already
> (voluntarily and especially involuntarily) abused it nonsensical data
> and I have yet to see it fail (i.e. go into an irrecoverable state).
>
>> What really would help in the long run is if we had some kind of test
>> harness to run these things in, and good test data culled from real-
>> world situations.   I had some hacky tools like this I used when I
>> built my implementation, but nothing really good.
>
> Sure. I guess it comes down to collecting (recording timestamps and  
> all)
> data from real applications in real scenarios. Then you figure out the
> "right" behaviour and compare with what you obtain.

There may not always be a single "right" behavior, because the system  
is always trying to predict future events.  But there's definitely  
"wrong" behaviors that can easily be tested objectively.

-SteveK