<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Let’s write one pair of A/V to disk to see if A/V sync problem occurs.<div class=""><br class=""></div><div class=""><div class="">gat-launch-1.0 uridecodebin name=dec uri=<a href="udp://239.50.50.50:1234" class="">udp://239.50.50.50:1234</a> \</div><div class="">dec. ! videoconvert ! deinterlace ! videoscale method=0 ! tee name=vt \</div><div class="">dec. ! audioconvert ! audioresample ! audio/x-raw,rate=48000,channels=1 ! audiorate ! queue ! voaacenc bitrate=32000 ! tee name=at \</div><div class="">vt. ! queue ! videorate ! video/x-raw,width=640,height=512,framerate=25/1 ! x264enc ! h264parse ! mpegtsmux name=muxer alignment=7 ! filesink location=test.ts \</div><div class="">at. ! queue ! aacparse ! muxer. \</div></div><div class=""><br class=""></div><div class="">=====</div><div class="">Then check UDP output of one pair of A/V to see if A/V sync problem occurs.</div><div class=""><br class=""></div><div class=""><div class="">gat-launch-1.0 uridecodebin name=dec uri=<a href="udp://239.50.50.50:1234" class="">udp://239.50.50.50:1234</a> \</div><div class="">dec. ! videoconvert ! deinterlace ! videoscale method=0 ! tee name=vt \</div><div class="">dec. ! audioconvert ! audioresample ! audio/x-raw,rate=48000,channels=1 ! audiorate ! queue ! voaacenc bitrate=32000 ! tee name=at \</div><div class="">vt. ! queue ! videorate ! video/x-raw,width=640,height=512,framerate=25/1 ! x264enc ! h264parse ! mpegtsmux name=muxer alignment=7 ! rtpmp2tpay ! udpsink host=127.0.0.1 port=1111 \</div><div class="">at. ! queue ! aacparse ! muxer.</div></div><div class=""><br class=""></div><div class=""><br class=""><div class="">
<div style="color: rgb(0, 0, 0); letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><div style="color: rgb(0, 0, 0); letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><div class="">--</div><div class="">Yi-Lung (Bruce) Tsai</div><div class=""><a href="mailto:wagamama.tsai@gmail.com" class="">wagamama.tsai@gmail.com</a></div><div class=""><br class=""></div></div><br class="Apple-interchange-newline"></div><br class="Apple-interchange-newline"><br class="Apple-interchange-newline">
</div>
<br class=""><div><blockquote type="cite" class=""><div class="">On Apr 21, 2017, at 6:01 PM, Dmitriy Novash <<a href="mailto:programer@wnet.ua" class="">programer@wnet.ua</a>> wrote:</div><br class="Apple-interchange-newline"><div class="">
<meta content="text/html; charset=utf-8" http-equiv="Content-Type" class="">
<div bgcolor="#FFFFFF" text="#000000" class=""><p class="">Hi.<br class="">
First, problem with synchronization video and audio.<br class="">
Second, with videoscale i get:<br class="">
gstbin.c(2586): gst_bin_do_latency_func ():
/GstPipeline:pipeline0:<br class="">
Impossible to configure latency: max 0:00:02.700000000 < min
0:00:02.820000000. Add queues or other buffering elements.<br class="">
<br class="">
Without videoscale 1 pair video+audio working, if I add second
pair and adding videoscale after vt no errors, such a video and
audio and if I remove videoscale and left other:<br class="">
$ gst-launch-1.0 uridecodebin name=dec uri=<a href="udp://239.50.50.50:1234" class="">udp://239.50.50.50:1234</a>
\<br class="">
dec. ! videoconvert ! deinterlace ! tee name=vt \<br class="">
dec. ! audioconvert ! audioresample !
audio/x-raw,rate=48000,channels=1 ! audiorate ! queue ! voaacenc
bitrate=32000 ! tee name=at \<br class="">
vt. ! queue ! videorate ! x264enc ! h264parse ! mpegtsmux
name=muxer alignment=7 ! rtpmp2tpay ! udpsink host=127.0.0.1
port=1111 \<br class="">
at. ! queue ! aacparse ! muxer. \<br class="">
vt. ! queue ! videoscale method=0 ! videorate !
video/x-raw,width=360,height=288,framerate=25/1 ! queue ! x264enc
! h264parse ! mpegtsmux name=muxer1 alignment=7 ! rtpmp2tpay !
udpsink host=127.0.0.1 port=1112 \<br class="">
at. ! queue ! aacparse ! muxer1.<br class="">
I get 2 working copies of video+audio witout scale and with
audio+video synchronization problem.<br class="">
</p>
<br class="">
<div class="moz-cite-prefix">21.04.2017 10:35, Bruce Tsai пишет:<br class="">
</div>
<blockquote cite="mid:A36ECB36-A73B-45D4-A3E6-E5FC81992138@gmail.com" type="cite" class="">
<meta http-equiv="Content-Type" content="text/html; charset=utf-8" class="">
I made a mistake when connecting “mpegtsmux” to “rtpmp2tpay”.
<div class=""><br class="">
</div>
<div class="">Here is modified version:</div>
<div class=""><br class="">
</div>
<div class="">
<div class="">gat-launch-1.0 uridecodebin name=dec uri=<a moz-do-not-send="true" href="udp://239.50.50.50:1234" class="">udp://239.50.50.50:1234</a> \</div>
<div class="">dec. ! videoconvert ! deinterlace ! videoscale
method=0 ! tee name=vt \</div>
<div class="">dec. ! audioconvert ! audioresample !
audio/x-raw,rate=48000,channels=1 ! audiorate ! queue !
voaacenc bitrate=32000 ! tee name=at \</div>
<div class="">vt. ! queue ! videorate !
video/x-raw,width=640,height=512,framerate=25/1 ! x264enc !
h264parse ! mpegtsmux name=muxer alignment=7 ! rtpmp2tpay !
udpsink host=127.0.0.1 port=1111 \</div>
<div class="">at. ! queue ! aacparse ! muxer. \</div>
<div class="">
<div class="">vt. ! queue ! videorate !
video/x-raw,width=320,height=256,framerate=25/1 ! x264enc !
h264parse ! mpegtsmux name=muxer1 alignment=7 ! rtpmp2tpay !
udpsink host=127.0.0.1 port=1112 \</div>
<div class="">at. ! queue ! aacparse ! muxer1. \</div>
</div>
<div class="">
<div class="">vt. ! queue ! videorate !
video/x-raw,width=160,height=128,framerate=25/1 ! x264enc !
h264parse ! mpegtsmux name=muxer2 alignment=7 ! rtpmp2tpay !
udpsink host=127.0.0.1 port=1113 \</div>
<div class="">at. ! queue ! aacparse ! muxer2.</div>
<div class=""><br class="">
</div>
</div>
<div class="">
<div style="letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">
<div style="letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">
<div class="">--</div>
<div class="">Yi-Lung (Bruce) Tsai</div>
<div class=""><a moz-do-not-send="true" href="mailto:wagamama.tsai@gmail.com" class="">wagamama.tsai@gmail.com</a></div>
<div class=""><br class="">
</div>
</div>
<br class="Apple-interchange-newline">
</div>
<br class="Apple-interchange-newline">
<br class="Apple-interchange-newline">
</div>
<br class="">
<div class="">
<blockquote type="cite" class="">
<div class="">On Apr 21, 2017, at 2:32 PM, Dmitriy Novash
<<a moz-do-not-send="true" href="mailto:programer@wnet.ua" class="">programer@wnet.ua</a>>
wrote:</div>
<br class="Apple-interchange-newline">
<div class="">
<meta content="text/html; charset=utf-8" http-equiv="Content-Type" class="">
<div bgcolor="#FFFFFF" text="#000000" class=""><p class="">Hi, Bruce.<br class="">
<br class="">
Problem with combine video and audio.<br class="">
$ GST_DEBUG=1 gst-launch-1.0 uridecodebin name=dec
uri=<a moz-do-not-send="true" href="udp://239.50.50.50:1234" class="">udp://239.50.50.50:1234</a>
dec. ! videoconvert ! deinterlace ! tee name=vt dec. !
audioconvert ! audioresample !
audio/x-raw,rate=48000,channels=1 ! audiorate ! queue
! voaacenc bitrate=32000 ! tee name=at vt. !
videoscale method=0 ! videorate !
video/x-raw,width=640,height=512,framerate=25/1 !
x264enc ! h264parse ! mpegtsmux name=muxer alignment=7
at. ! queue ! aacparse ! muxer. ! rtpmp2tpay ! udpsink
host=127.0.0.1 port=1111<br class="">
0:00:00.015467297 3955 0x1f2a980 ERROR
GST_PIPELINE grammar.y:959:priv_gst_parse_yyparse:
syntax error<br class="">
0:00:00.015490467 3955 0x1f2a980 ERROR
GST_PIPELINE grammar.y:959:priv_gst_parse_yyparse:
syntax error<br class="">
<br class="">
Simplify code:<br class="">
$ GST_DEBUG=1 gst-launch-1.0 uridecodebin name=dec
uri=<a moz-do-not-send="true" href="udp://239.50.50.50:1234" class="">udp://239.50.50.50:1234</a>
dec. ! videoconvert ! deinterlace ! tee name=vt dec. !
audioconvert ! audioresample !
audio/x-raw,rate=48000,channels=1 ! audiorate ! queue
! voaacenc bitrate=32000 ! tee name=at vt. !
videoscale method=0 ! videorate !
video/x-raw,width=640,height=512,framerate=25/1 !
x264enc ! h264parse ! mpegtsmux name=muxer alignment=7
at. ! fakesink<br class="">
Установка конвейера в состояние PAUSED…<br class="">
Конвейер работает и не требует состояния PREROLL…<br class="">
Установка конвейера в состояние PLAYING…<br class="">
New clock: GstSystemClock<br class="">
</p>
<br class="">
<div class="moz-cite-prefix">20.04.2017 18:58, Bruce
Tsai пишет:<br class="">
</div>
<blockquote cite="mid:C305F643-B41E-4661-8CC9-D87F7D5CFA72@gmail.com" type="cite" class="">
<meta http-equiv="Content-Type" content="text/html;
charset=utf-8" class="">
<div class="">First, there is no need to mix-use
“decodebin” and “tsdemux”.</div>
<div class="">“decodebin” already has demux
capability.</div>
<div class="">Second, there are too many “queue” in
pipeline.</div>
<div class="">Each “queue” introduces a new thread.</div>
<div class="">Too many unnecessary threads is not a
good idea.</div>
<div class="">Third, “tee” for audio stream is also
necessary.</div>
<div class="">Each “mux” needs a copy of video and a
copy of audio.</div>
<div class="">Use “tee” to get three copies of video
and audio respectively.</div>
<div class="">Then combine each pair of video and
audio to a “mpegtsmux”.</div>
<div class=""><br class="">
</div>
<div class="">I tried to rewrite your pipeline as
follows:</div>
<div class=""><br class="">
</div>
<div class="">gat-launch-1.0 uridecodebin name=dec
uri=<a moz-do-not-send="true" href="udp://239.50.50.50:1234" class="">udp://239.50.50.50:1234</a>
\</div>
<div class="">dec. ! videoconvert ! deinterlace !
videoscale method=0 ! tee name=vt \</div>
<div class="">dec. ! audioconvert ! audioresample !
audio/x-raw,rate=48000,channels=1 ! audiorate !
queue ! voaacenc bitrate=32000 ! tee name=at \</div>
<div class="">vt. ! queue ! videorate !
video/x-raw,width=640,height=512,framerate=25/1 !
x264enc ! h264parse ! mpegtsmux name=muxer
alignment=7 \</div>
<div class="">at. ! queue ! aacparse ! muxer. !
rtpmp2tpay ! udpsink host=127.0.0.1 port=1111 \</div>
<div class="">
<div class="">vt. ! queue ! videorate !
video/x-raw,width=320,height=256,framerate=25/1 !
x264enc ! h264parse ! mpegtsmux name=muxer1
alignment=7 \</div>
<div class="">at. ! queue ! aacparse ! muxer1. !
rtpmp2tpay ! udpsink host=127.0.0.1 port=1112 \</div>
</div>
<div class="">
<div class="">vt. ! queue ! videorate !
video/x-raw,width=160,height=128,framerate=25/1 !
x264enc ! h264parse ! mpegtsmux name=muxer2
alignment=7 \</div>
<div class="">at. ! queue ! aacparse ! muxer2. !
rtpmp2tpay ! udpsink host=127.0.0.1 port=1113</div>
</div>
<div class=""><br class="">
</div>
<div class="">I didn’t verify above pipeline on my
machine.</div>
<div class="">You could try it with first pair of
output only.</div>
<div class="">Simplicity is good for debug.</div>
<div class="">Then you could add second pair and then
third pair.</div>
<div class=""><br class="">
</div>
<br class="">
<div class="">
<div style="letter-spacing: normal; orphans: auto;
text-align: start; text-indent: 0px;
text-transform: none; white-space: normal; widows:
auto; word-spacing: 0px;
-webkit-text-stroke-width: 0px; word-wrap:
break-word; -webkit-nbsp-mode: space;
-webkit-line-break: after-white-space;" class="">
<div style="letter-spacing: normal; orphans: auto;
text-align: start; text-indent: 0px;
text-transform: none; white-space: normal;
widows: auto; word-spacing: 0px;
-webkit-text-stroke-width: 0px; word-wrap:
break-word; -webkit-nbsp-mode: space;
-webkit-line-break: after-white-space;" class="">
<div class="">--</div>
<div class="">Yi-Lung (Bruce) Tsai</div>
<div class=""><a moz-do-not-send="true" href="mailto:wagamama.tsai@gmail.com" class="">wagamama.tsai@gmail.com</a></div>
<div class=""><br class="">
</div>
</div>
<br class="Apple-interchange-newline">
</div>
<br class="Apple-interchange-newline">
<br class="Apple-interchange-newline">
</div>
<br class="">
<div class="">
<blockquote type="cite" class="">
<div class="">On Apr 20, 2017, at 9:49 PM, Dmitriy
Novash <<a moz-do-not-send="true" href="mailto:programer@wnet.ua" class="">programer@wnet.ua</a>>
wrote:</div>
<br class="Apple-interchange-newline">
<div class="">
<meta http-equiv="content-type" content="text/html; charset=utf-8" class="">
<div bgcolor="#FFFFFF" text="#000000" class="">
Hi all.<br class="">
<br class="">
I have many TV streams (Mpeg2) in udp
multicast.<br class="">
I want to have stream in H264+AAC in udp
multicast with 3 resolutions: 1:1
(original),1:2,1:4. Now I do it with 3
processes of gst-launch, but for optimization
of CPU usage I think I can do it in one
process.<br class="">
After several tries I get 3 videos with 3
resolutions, but i can't get audio for all
threads, only for one.<br class="">
I'm using gstreamer 1.00+ version:<br class="">
$ gst-launch-1.0 --version<br class="">
gst-launch-1.0 version 1.8.3<br class="">
GStreamer 1.8.3<br class="">
<br class="">
My configuration:<br class="">
<code class="java plain">gst-launch-</code><code class="java value">1.0</code> <code class="java plain">udpsrc uri=udp:</code><code class="java comments">//239.50.50.50:1234 !
queue ! tsparse ! queue ! tsdemux name=demux
\<br class="">
demux. ! queue ! mpegvideoparse ! queue !
decodebin ! queue ! deinterlace ! \<br class="">
tee name=tee1 ! queue ! videoscale method=0
! videorate ! video/x-raw, width=640,
height=512, framerate=25/1 ! x264enc !
h264parse ! mpegtsmux name=muxer alignment=7
\<br class="">
demux. ! queue ! decodebin ! queue !
audioconvert ! audioresample ! queue !
audio/x-raw, rate=48000, channels=1 !
audiorate ! voaacenc bitrate=32000 ! queue !
aacparse ! queue ! tee name=tee2 ! queue !
muxer. \<br class="">
muxer. ! rtpmp2tpay ! udpsink host=127.0.0.1
port=1111 \<br class="">
tee1. ! queue ! videoscale method=0 !
videorate ! video/x-raw, width=320,
height=256, framerate=25/1 ! x264enc !
h264parse ! mpegtsmux name=muxer1
alignment=7 muxer1. ! rtpmp2tpay ! udpsink
host=127.0.0.1 port=1112 \<br class="">
tee1. ! queue ! videoscale method=0 !
videorate ! video/x-raw, width=160,
height=128, framerate=25/1 ! x264enc !
h264parse ! mpegtsmux name=muxer2
alignment=7 muxer2. ! rtpmp2tpay ! udpsink
host=127.0.0.1 port=1113</code><br class="">
<br class="">
I can't join mpedtsmux and tee.<br class="">
<br class="">
So I want to decode video, get 3 copies, each
copy scale to some resolution, encode, decode
+ encode audio and join to each copy of video,
output each copy in udp unicast to some port.<br class="">
<br class="">
Help, maybe someone have working solution for
my task.<br class="">
</div>
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</blockquote>
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</blockquote>
<br class="">
<pre class="moz-signature" cols="72">--
BR, Dmitriy Novash
Wnet IPTV/Media team
tel: +380(44) 5-900-800 (доп. 1132)
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://wnet.ua/">http://wnet.ua</a></pre>
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</pre>
</blockquote>
<br class="">
<pre class="moz-signature" cols="72">--
BR, Dmitriy Novash
Wnet IPTV/Media team
tel: +380(44) 5-900-800 (доп. 1132)
<a class="moz-txt-link-freetext" href="http://wnet.ua/">http://wnet.ua</a></pre>
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