Hi, I am trying to create a pipeline to stream AAC using rtp,
then using Darwin Streaming Server to relay it to a client’s quicktime
player. I am able to do this successfully with video (by itself) with the
following pipeline gst-launch -v v4l2src always-copy=false
do-timestamp=true \
num-buffers=$(( 30 * 60 * 5 )) !
\
video/x-raw-yuv,
format=\(fourcc\)UYVY,
\
width=720, height=480, framerate=\(fraction\)30/1 ! \ TIVidenc1
codecName=h264enc engineName=codecServer \
rateControlPreset=2 bitRate=512000 iColorSpace=UYVY \
resolution=720x480 frameRate=30 !
\
rtph264pay
scan-mode=bytestream !
\
udpsink host=225.1.1.2
auto-multicast=true port=9010 Audio however seems to be a different beast. I’ve
found even when piping AAC to a filesink, in order for quicktime to play the
audio, I have to use qtmux or mp4mux after the encoding phase. When I
attempt to mimic my video pipeline for audio like the following: gst-launch -v alsasrc device=hw:0,0 num-buffers=$((30*60*5))
! \ audio/x-raw-int,
endianness=1234 ! \ queue ! \ TIAudenc1
engineName=codecServer codecName=aacheenc ! \ qtmux ! \ rtpmp4vpay pt=96
! \ udpsink
host=225.1.1.2 port=9010 auto-multicast=true The pipeline doesn’t play – I get errors linking
mp4mux to rtpmp4apay (caps don’t match). Using gst-inspect I see that
qtmux has only 1 src - video/quicktime, and when looking at the rtpmp4 plugins
none of them take that as a sink. So I’m unclear how I can stream the
audio with rtp. Is there a different mux I can use that doesn’t put out
video/quicktime, but still works with quicktime player? Or is there some other
plugins to use between qtmux and one of the rtpmp4[a,v,g]pay plugins that will
convert the caps to video/mpeg? Thanks eric _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Hi,
On Fri, May 20, 2011 at 10:33 PM, Eric Shoquist <[hidden email]> wrote:
..snip..
it's understandable as long as it's about playing local files.. streaming is a different beast and usually the muxer role is played from the payloader (see below).
you should use NO muxers if yoiu're already payloading, that is, try connecting the encoder directly to the payloader. It should just work (r). Besides, you should make sure that in the rtpsp negotiation (I didn't understand how it's done prior using these pipelines, but I'll assume some magic happens under the hood) the audio format is specified as something like MP4A-LATM (as for rfc3016). Indeed, RealMedia has its own payloading format and it might happen that the negotiation assumes that one to be used. Regards _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Free forum by Nabble | Edit this page |