>> I just have another question. Is there any code snippet in the test
>> programs that shows how to safely destroy and restart an RTSPClient
>> and/or RTSPServer without stopping the event loop? Whenever I do
>> something like Medium::close(rtspServer), the event loop gets a "bad
>> file descripto
> OK, but what if the queue *is* empty? In this case, you’ll need to (somehow)
> arrange for the code to get called again in the future, when an OPUS packet
> becomes available.
>
Ha, yes, you are completely right ;-)
I took care of this already but did not paste that part of the code.
> One
> You’re correct that “SimpleRTPSink” is the correct ‘sink’ class to
> use. (You can do this because the RTP payload format for OPUS audio
> - defined in RFC 7587 - is relatively straightforward.)
> Note that - from RFC 7587, section 4.2 - a RTP packet contains
> exactly one ‘OPUS packet’, which
Hi,
I'm trying to stream an OPUS encoded audio signal, but I could not find
out yet how to implement this.
I'm using a H264 encoder and a class derived from FramedSource - that
one works perfectly and I can see the stream in VLC. So I wrote my own
OPUS encoder along the same lines with "frames" o
Hi,
I just found a single post from quite some time ago discussing this
issue without real outcome. My goal is to compress live camera input
from a windows mobile or android phone, probably with XVID, and stream
it over Live555 and WLAN to a single client. From the sources I have
seen that there a
