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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: 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 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 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, gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H263-1998" ! rtph263pdepay ! 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 It is important that the depayloader copies the incomming GStreamer timestamp
directly to the depayloaded output buffer. It should never attempt to perform 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, gst-launch-1.0 udpsrc caps="application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H263-1998" ! 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 The latency property on the jitterbuffer controls the amount of delay (in
milliseconds) to apply to the outgoing packets. A higher latency will produce milliseconds) to apply to the outgoing packets. A higher latency will produce
@ -271,7 +271,7 @@ for example).
Some gst-launch-1.0 lines: 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 ... Setting pipeline to PAUSED ...
/pipeline0/videotestsrc0.src: caps = video/x-raw, format=(string)I420, /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 Write down the caps on the udpsink and set them as the caps of the UDP
receiver: 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, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H263-1998,
ssrc=(guint)527842345, clock-base=(guint)1150776941, seqnum-base=(guint)30982" 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 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 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 Stream a quicktime file with mpeg4 video and AAC audio on port 5000 and port
5002. 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 d. ! queue ! rtpmp4gpay ! udpsink port=5002
.... ....
/pipeline0/udpsink0.sink: caps = application/x-rtp, media=(string)video, /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-rate=(int)90000, encoding-name=(string)MP4V-ES, ssrc=(guint)1162703703,
clock-base=(guint)816135835, seqnum-base=(guint)9294, profile-level-id=(string)3, clock-base=(guint)816135835, seqnum-base=(guint)9294, profile-level-id=(string)3,
config=(string)000001b003000001b50900000100000001200086c5d4c307d314043c1463000001b25876694430303334" 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, 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-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, ssrc=(guint)3246149898,
clock-base=(guint)4134514058, seqnum-base=(guint)57633, encoding-params=(string)2, clock-base=(guint)4134514058, seqnum-base=(guint)57633, encoding-params=(string)2,