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rtp/README: update pipelines to work with 1.0

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- Use gst-libav encoders/decoders instead of gst-ffmpeg
- gstrtpjitterbuffer -> rtpjitterbuffer
- gst-launch-0.10 -> gst-launch-1.0
- Add 'videoconvert' element
- xvimagesink -> autovideosink

https://bugzilla.gnome.org/show_bug.cgi?id=729247
This commit is contained in:
Guillaume Desmottes 2014-04-30 11:13:12 +02:00 committed by Olivier Crête
parent ec38c62563
commit d089f99a39
1 changed files with 8 additions and 8 deletions
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16
gst/rtp/README
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@ -181,7 +181,7 @@ of the sender.
Some pipelines to illustrate the process:
gst-launch-1.0 v4l2src ! ffenc_h263p ! rtph263ppay ! udpsink
gst-launch-1.0 v4l2src ! videoconvert ! avenc_h263p ! rtph263ppay ! udpsink
v4l2src puts a GStreamer timestamp on the video frames base on the current
running_time. The encoder encodes and passed the timestamp on. The payloader
@ -206,7 +206,7 @@ following pipeline:
gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H263-1998" ! rtph263pdepay !
ffdec_h263 ! xvimagesink
avdec_h263 ! autovideosink
It is important that the depayloader copies the incomming GStreamer timestamp
directly to the depayloaded output buffer. It should never attempt to perform
@ -239,7 +239,7 @@ The following pipeline illustrates a receiver with a jitterbuffer.
gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H263-1998" !
gstrtpjitterbuffer latency=100 ! rtph263pdepay ! ffdec_h263 ! xvimagesink
rtpjitterbuffer latency=100 ! rtph263pdepay ! avdec_h263 ! autovideosink
The latency property on the jitterbuffer controls the amount of delay (in
milliseconds) to apply to the outgoing packets. A higher latency will produce
@ -271,7 +271,7 @@ for example).
Some gst-launch-1.0 lines:
gst-launch-0.10 -v videotestsrc ! ffenc_h263p ! rtph263ppay ! udpsink
gst-launch-1.0 -v videotestsrc ! videoconvert ! avenc_h263p ! rtph263ppay ! udpsink
Setting pipeline to PAUSED ...
/pipeline0/videotestsrc0.src: caps = video/x-raw, format=(string)I420,
@ -289,10 +289,10 @@ Some gst-launch-1.0 lines:
Write down the caps on the udpsink and set them as the caps of the UDP
receiver:
gst-launch-0.10 -v udpsrc caps="application/x-rtp, media=(string)video,
gst-launch-1.0 -v udpsrc caps="application/x-rtp, media=(string)video,
payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H263-1998,
ssrc=(guint)527842345, clock-base=(guint)1150776941, seqnum-base=(guint)30982"
! rtph263pdepay ! ffdec_h263 ! xvimagesink
! rtph263pdepay ! avdec_h263 ! autovideosink
The receiver now displays an h263 image. Since there is no jitterbuffer in the
pipeline, frames will be displayed at the time when they are received. This can
@ -302,7 +302,7 @@ Some gst-launch-1.0 lines:
Stream a quicktime file with mpeg4 video and AAC audio on port 5000 and port
5002.
gst-launch-0.10 -v filesrc location=~/data/sincity.mp4 ! qtdemux name=d ! queue ! rtpmp4vpay ! udpsink port=5000
gst-launch-1.0 -v filesrc location=~/data/sincity.mp4 ! qtdemux name=d ! queue ! rtpmp4vpay ! udpsink port=5000
d. ! queue ! rtpmp4gpay ! udpsink port=5002
....
/pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video,
@ -324,7 +324,7 @@ Some gst-launch-1.0 lines:
clock-rate=(int)90000, encoding-name=(string)MP4V-ES, ssrc=(guint)1162703703,
clock-base=(guint)816135835, seqnum-base=(guint)9294, profile-level-id=(string)3,
config=(string)000001b003000001b50900000100000001200086c5d4c307d314043c1463000001b25876694430303334"
! rtpmp4vdepay ! ffdec_mpeg4 ! xvimagesink sync=false
! rtpmp4vdepay ! ffdec_mpeg4 ! autovideosink sync=false
udpsrc port=5002 caps="application/x-rtp, media=(string)audio, payload=(int)96,
clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, ssrc=(guint)3246149898,
clock-base=(guint)4134514058, seqnum-base=(guint)57633, encoding-params=(string)2,