I believe I may have found what I've been looking for. The StreamReplicator class may just solve my problem. Please advise.
Togba Liberty C++ Developer | ipConfigure, Inc. www.ipconfigure.com [Description: cid:7D82F492-5DFF-427B-9295-2AD8F6DD9DA3] From: live-devel-boun...@ns.live555.com [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Togba Liberty Sent: Tuesday, January 29, 2013 3:06 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] About AAC Streaming Hi Ross, Thanks for the quick replies. We are now receiving JPEG, H.264, and Audio at the client end of our pipeline; however, the audio stream has a lot of noise. We suspect that this might be down to audio synchronization. After going through Live555 library, we would like to know if there is a way that Live555 hardcopies one FramedSource into another. The idea is to be able to directly transfer the audio and video RTP packets from the camera, through the transcoding server, to our clients without having to remove the payload of the JPEG and audio streams. Note that the client and server ends of our transcoding server have two different TaskSchedulers. Thanks. Togba Liberty C++ Developer | ipConfigure, Inc. www.ipconfigure.com<http://www.ipconfigure.com> From: live-devel-boun...@ns.live555.com<mailto:live-devel-boun...@ns.live555.com> [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: Friday, January 25, 2013 5:56 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] About AAC Streaming The issue we have now is that in retransmitting the audio stream we don't understand if Live555 adds an extra header to the AAC frames. As I said before - it doesn't. The frames that come from the "MPEG4GenericRTPSource" are simply AAC frames, with no extra header added. We have the jpeg and h.264 pipeline going, we just need to know what headers the MPEG4GenericRTPSource is adding to the aac frames to be able to retransmit it so VLC can play it. VLC should be able to play the stream, provided that you set up the "MPEG4GenericRTPSink" correctly (as I noted in my previous email), and also provided that you are setting "fPresentationTime" correctly on each outgoing frame. The best way to test this is *not* to use VLC at first, but instead to use our "testRTSPClient" application. You should make sure that the "a=config" value in the SDP description is the same as it was in the original ('back-end') stream. Ross Finlayson Live Networks, Inc. http://www.live555.com/
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