[Live-devel] Add audio "Mediasession" SDP in to current video Server

[Xingjun Chen]

Hi, Ross,

Thank you for your reply,  I should not express myself well, I think your
method provided is the normal way in live555 which add live aac source
audio in to current Media session, but I want to remain the aac audio RTP
streamed by Gstreamer, and the H264 video is finished by live555, then is
that possible that add the audio which RTP streamed out by Gstreamer to
current live555 video RTSP media session.

Thank you.
Kevin


On Fri, Apr 10, 2015 at 1:43 PM, Ross Finlayson <finlayson at live555.com>
wrote:

> what kind od ServerMediaSubsession objects I should use?
>
>
> As with the video stream, you must write your own subclass of
> “OnDemandServerMediaSubsession”, and implement the
> “createNewStreamSource()” and “createNewRTPSink()” virtual functions.
>
> Like the existing “ADTSAudioFileServerMediaSubsession” class, your
> subclass will implement the “createNewRTPSink()” virtual function by
> calling “MPEG4GenericRTPSink::createNew()” (see
> “liveMedia/include/MPEG4GenericRTPSink.hh”), with the following parameters
> (based on the SDP description that you gave):
> rtpTimestampFrequency: 48000
> sdpMediaTypeString: “audio”
> mpeg4Mode: "AAC-hbr”
> configString: “1190”
> numChannels: 2
>
> You must also implement the “createNewStreamSource()” virtual function to
> deliver one AAC frame at a time from your input source.  It’s important
> that you set “fPresentationTime” properly, if you want the audio to be
> properly synchronized with the video, once you later combine them in a
> single “ServerMediaSession”.
>
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>


-- 
Xingjun Chen
M.S. Degree *Earned*
Electrical and Computer Engineering
University of Arizona, Tucson AZ 85721
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