Muxing two tcpclientsrc sources to rtmp server

soylent soylent.tv at gmail.com
Mon Jul 14 10:33:33 PDT 2014


... also tried with:

gst-launch-1.0 -v audiotestsrc ! audioconvert ! audio/x-raw, format=S16BE,
channels=2, rate=44100 ! queue ! audioconvert ! audioresample ! audiorate !
lamemp3enc bitrate=128000 ! mpegaudioparse ! audio/mpeg, mpegversion=1,
layer=3, channels=2, rate=44100 ! flvmux streamable=true name=mux ! queue !
rtmpsink sync=false
location=rtmp://x.xxxxxxxx.fme.ustream.tv/ustreamVideo/xxxxxxxx/xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx

and works good, but :

gst-launch-1.0 -v tcpclientsrc port=@APORT ! audio/x-raw, format=S16BE,
channels=@CHANNELS, rate=@FREQ ! queue max-size-buffers=0 max-size-time=0 !
audioconvert ! audioresample ! audiorate ! lamemp3enc bitrate=@ABITRATE000 !
mpegaudioparse ! audio/mpeg, mpegversion=1, layer=3, channels=@CHANNELS,
rate=@FREQ ! flvmux ! queue max-size-buffers=0 max-size-time=0 ! rtmpsink
sync=false location=@URL

... gives the same error:

(gst-launch-1.0:7693): GStreamer-CRITICAL **: gst_segment_to_running_time:
assertion 'segment->format == format' failed
(gst-launch-1.0:7693): GStreamer-CRITICAL **: gst_segment_to_running_time:
assertion 'segment->format == format' failed

Thanks.
karl



--
View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Muxing-two-tcpclientsrc-sources-to-rtmp-server-tp4667922p4667925.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.


More information about the gstreamer-devel mailing list