Hi,
Here i am using to stream an axis camera from WAN . I just tried use rtsp first failed !! . Then i tried "rtspt" then the problem arises is the synchronisation of audio and video. I created a pipeline in which the acdio and video decoded , encoded and pass throug a avimux. If the audio is connected to the fakesink then the video is recorded and and played normally. When the audio is connected to the avimux the recording is done ,but when playing the video rate is too high
Here is the pipeline i created( audio encoder connected to the avimux)
gst-launh -e rtspsrc location="rtspt://
spartan.dyndns.biz:554/axis-media/media.amp?videocodec=mpeg4&resolution=320x240" name=rtsp ! queue ! rtpmp4vdepay ! video/mpeg, width=320, height=240, framerate=30/1 ! ffdec_mpeg4 ! videorate ! ffenc_mpeg4 ! avimux name=avi ! filesink location=/root/s.avi sync=true rtsp. ! queue ! rtppcmudepay ! mulawdec ! audioconvert ! audioresample ! audiorate ! faac ! avi.
Then the Ouput...!!
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
WARNING: from element /GstPipeline:pipeline0/GstFaac:faac0: Could not get/set settings from/on resource.
Additional debug info:
gstfaac.c(484): gst_faac_configure_source_pad (): /GstPipeline:pipeline0/GstFaac:faac0:
bitrate 128000 exceeds maximum allowed bitrate of 48000 for samplerate 8000. Setting bitrate to 48000
WARNING: from element /GstPipeline:pipeline0: Internal GStreamer error: clock problem. Please file a bug at
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer.
Additional debug info:
gstbin.c(2308): gst_bin_do_latency_func (): /GstPipeline:pipeline0:
Failed to configure latency of 0:00:03.033333333
Some problem with the bitrate problem in faac ;
Then i tried to set up the bitrate in faac by insrting the ! faac bitrate=48000 ! ( failed).
Plz help on this...
The entire log is
http://pastebin.com/m58034093
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