troubles with audio branch via rtp

[Giacomo D]

Hi everyone!

I have two sources (my webcam and an .AVI file) and I want to send them to
my pc via RTP so I made two client scritps which stream audio and video
parts on two different ports on my PC. Videos work perfectly but I am
experiencing problems with the audio.

For videos on the receiver side            

for d in
(self.udpsrc,self.camerafilter,self.rtpjitterbuffer,self.rtph264depay,self.avdec_h264,self.tee,self.queue):
                self.pipeline.add(d)

I add and link the same elements to capture the videos on two different
ports that I link on a videomixer and an xvimagesink, and one of them on
another xvimagesink

Now, I want to send the two audios too, for the audio camera and the audio
of .AVI file.
On the sender side and  client 1 I have

       self.alsasrc.link(self.audioconvert)
       self.audioconvert.link(self.audioresample)
       self.audioresample.link(self.camerafilter_A)
       self.camerafilter_A.link(self.tee_A)
       self.tee_A.link(self.queue_A)
       self.queue_A.link(self.autoaudiosink)

       self.tee_A.link(self.queue_A_udp)
       self.queue_A_udp.link(self.speexenc)
       self.speexenc.link(self.rtpspeexpay)
       self.rtpspeexpay.link(self.udpsink_A)

On the sender side and  client 2 I have

(first I have a decodebin )
       self.audioconvert.link(self.audioresample)
       self.audioresample.link(self.camerafilter_A)
       self.camerafilter_A.link(self.tee_A)
       self.tee_A.link(self.queue_A)
       self.queue_A.link(self.autoaudiosink)

       self.tee_A.link(self.queue_A_udp)
       self.queue_A_udp.link(self.mulawenc)
       self.mulawenc.link(self.rtppcmupay)
       self.rtppcmupay.link(self.udpsink_A)


 I am using other ports. On my receiver side I capture them and I link like
this

            for d in (self.udpsrc,self.tee,self.queue,self.fakesink):
                self.pipeline.add(d)

I want to hear, for example, the .AVI audio so I built another audio branch
like this

        self.tee_A_ON_AIR.link(self.queue_A_ON_AIR)
        self.queue_A_ON_AIR.link(self.rtpjitterbuffer_A_ON_AIR)
        self.rtpjitterbuffer_A_ON_AIR.link(self.rtppcmudepay_A_ON_AIR)
        self.rtppcmudepay_A_ON_AIR.link(self.mulawdec_A_ON_AIR)
        self.mulawdec_A_ON_AIR.link(self.audioconvert_A_ON_AIR)
        self.audioconvert_A_ON_AIR.link(self.camerafilter_A_ON_AIR)
        self.camerafilter_A_ON_AIR.link(self.audioresample_A_ON_AIR)
        self.audioresample_A_ON_AIR.link(self.autoaudiosink_A_ON_AIR)

When I add this audio branch I get a freezed video output without no error.
Is there something wrong in this branch?
But, first of all, I want to know If there is a way to decode-encode an
audio stream with the same elements like for the videos?

Thanks in advance
Giacomo








--
View this message in context: http://gstreamer-devel.966125.n4.nabble.com/troubles-with-audio-branch-via-rtp-tp4674039.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.