> I am currently using a file, but eventually it will be live source.
> I got the audio streaming, which is good, but it stutters. I looked at the
> packet format with wireshark, but found no issues.
> However I found out that apparently RTP packets are sent too quickly with my
> class. But if yo
Dear Ross,
Because your codec is MP3, you definitely *do not* need to reimplement the
"getAuxSDPLine()" virtual function. I.e., if you have such a reimplementation,
then you should remove it.
Many thanks for these details, that's enlightening!
Our RTP output code - in this case, the "MPEG1o
> I have another question about FramedSource::doGetNextFrame(). Does this
> method requires the actual MP3 frame to be returned, or the MP3 frame
> encapsulated as required by the chosen RTP profile (14 in my case)?
Our RTP output code - in this case, the "MPEG1or2AudioRTPSink" class -
automati
Dear Ross,
I have another question about FramedSource::doGetNextFrame(). Does this
method requires the actual MP3 frame to be returned, or the MP3 frame
encapsulated as required by the chosen RTP profile (14 in my case)?
Thanks a lot for your help!
Best regards,
Fabrice
_
Dear Ross,
We implemented a class based on OnDemandServerMediaSubsession, and it
uses an apparently widely used trick in live555 to get the "SDP lines".
It plays the stream into a "dummy" RTP sink to get those lines and then
stops the stream. That's where things were going wrong, because the R
Dear Ross,
Before you do this, you should first just add a
"MP3AudioFileServerMediaSubsession" with the MP3 file - just to make sure that
this (an audio subsession with data coming from a file) works OK for you.
Thanks for the tip!
I just tried it instead of our custom subsession class, and it
> I am trying to modify an existing RTSP server based on live555. It streams
> live video without problems, and I have to add live audio sub-streams for
> each video streams.
>
> As a first step, I wanted to stream an MP3 file, so I created an audio source
> class based on the "DeviceSource" te
Hello everyone,
I hope somebody could point me to some directions where to investigate.
I am trying to modify an existing RTSP server based on live555. It
streams live video without problems, and I have to add live audio
sub-streams for each video streams.
As a first step, I wanted to stream
