The VideoLAN Forums

Hi and happy holidays,

I have two question:

I have a server that broadcasts RTP over UDP on the localhost (without any RTCP packets)
It is only Audio so no need for synchronization really, here is the SDP file I use:

v=0
o= - 1 1 IN IP4 127.0.0.1
s=Audio
i=Sample Audio
t=0 0
a=recvonly
a=ptime:20
a=type:broadcast
c=IN IP4 127.0.0.1/127
m=audio 5006 RTP/AVP 96
a=rtpmap:96 L16/48000/2

Looking at the messages window (debug-level 2) I can see VLC gets the packets fine but I get the following messages (and audio is not played back): 1- I was wondering if this could be because of the lack of RTCP packets or maybe RTP packets are corrupted, if VLC/LIVE555 requires RTCP for all kind of multicast then I need to fix the server.
Reading through RFCs RTP broadcast do not require RTCP I guess (?)

Second question:
2- Are non-standard sampling rates supported by VLC?
I have tried different options, if I change my sampling rate (both server and SDP of course) to something non-standard (L16/30000/2).
I see the ugly-resmapler is doing something and does not give me errors, but the warning changes: I appreciate any hints/articles/advice/rants

dashesy

1/ This is a live555 question and we cannot help you. RTCP is mandatory by the specifications, though I agree with you that it brings no practical benefit in your particular case.

2/ Generally speaking yes, but this is codec-dependent.

Hi Remi,

Thanks for the information!
I actually found the problem was that my packets were corrupted, when fixed VLC was able to play them fine!
However I will add RTCP, because not every media player is as robust as VLC is, plus it might be useful even in my scenario:

I have now two more questions, I appreciate if you can help (please let me know if I should ask live555):

Although the nominal sample-rate is fixed, my audio source can slow-down or generate faster samples (via GUI interaction).
In one way this is to some extent similar to the problem of having different clock-rates at source and destination machines,
1- Does VLC use NTP timestamp in RTCP to synchronize to the audio source? if yes what is the tolerance of clock drift VLC can handle?
2- What is a fast codec with variable sampling rate that VLC also supports? I want to use that in an almost realtime application?

Thanks
dashesy