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you need connect to "pad-added" signal and in callback of that
signal you will get GstPad. from this pad you get caps and according
to them you link this pad to additional elements.<br>
<br>
<div class="moz-cite-prefix">Dňa 27.11.2014 o 08:05 Rajesh salumuri
napísal(a):<br>
</div>
<blockquote
cite="mid:CANQbXdjzr==AFkDNmVMi4SCD0EB6aKCMaDE3PAqr2FAiSfqnJg@mail.gmail.com"
type="cite">
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<div>
<div>
<div>Hi,<br>
<br>
</div>
I am trying to convert the following gst-launch command into
c:<br>
<br>
<b>gst-launch rtspsrc location=<a class="moz-txt-link-freetext" href="rtsp://ipaddress:port">rtsp://ipaddress:port</a>
name=demux demux. ! queue ! decodebin ! autovideosink
demux. ! queue ! rtppcmudepay ! autoaudiosink<br>
<br>
</b></div>
<div>I tried the above command it is able to stream both audio
and video.<b><br>
</b></div>
<div><br>
</div>
And I played this rtspurl in vlc media player it is showing
the properties are as follows:<br>
<p>Type: Video Codec: H264 - MPEG-4 AVC (part 10) (h264)
Resolution: 320x180 Decoded format: Planar 4:2:0 YUV</p>
<p>Type: Audio Codec: PCM MU-LAW (mlaw) Channels: Mono Sample
rate: 8000 Hz Bits per sample: 8</p>
<b>I am converting the above gst-launch command in c in brief
explained below</b><br>
<br>
pipeline = gst_pipeline_new("nice_pipeline");<br>
source = gst_element_factory_make("nicesrc","source");//thsi
source will take the stream from rtsp url<br>
<b> demuxer = gst_element_factory_make ("avidemux",
"avi-demuxer");</b><br>
decvd= gst_element_factory_make ("decodebin", "decodebin");<br>
vdsink = gst_element_factory_make ("autovideosink",
"video-sink");<br>
vdqueue = gst_element_factory_make ("queue", "video-queue");<br>
adqueue = gst_element_factory_make ("queue", "audio-queue");<br>
decad= gst_element_factory_make ("rtppcmudepay",
"rtppcmudepay");<br>
adsink = gst_element_factory_make ("autoaudiosink",
"audio-sink");<br>
<br>
bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));<br>
gst_bus_add_watch(bus, bus_call, NULL);<br>
gst_object_unref (bus);<br>
gst_bin_add_many(GST_BIN (pipeline), source,decvd,decad,
adqueue, vdqueue, vdsink, adsink, NULL);<br>
<br>
gst_element_link (source, demuxer);<br>
gst_element_link (decvd, vdqueue);<br>
gst_element_link (vdqueue, vdsink);<br>
gst_element_link (decad, adqueue);<br>
gst_element_link (adqueue, adsink);<br>
<br>
g_signal_connect (demuxer, "pad-added", G_CALLBACK
(on_pad_added), NULL);<br>
gst_element_set_state(pipeline, GST_STATE_PLAYING);<br>
<br>
</div>
<b>on_pad_added func defintion:</b><br>
<br>
GstCaps *caps;<br>
GstStructure *str;<br>
<br>
caps = gst_pad_get_caps (pad);<br>
g_assert (caps != NULL);<br>
str = gst_caps_get_structure (caps, 0);<br>
g_assert (str != NULL);<br>
<br>
if (g_strrstr (gst_structure_get_name (str), "video")) {<br>
g_debug ("Linking video pad to dec_vd");<br>
<br>
GstPad *targetsink = gst_element_get_pad (decvd,
"sink");<br>
g_assert (targetsink != NULL);<br>
gst_pad_link (pad, targetsink);<br>
gst_object_unref (targetsink);<br>
}<br>
<br>
if (g_strrstr (gst_structure_get_name (str), "audio")) {<br>
g_debug ("Linking audio pad to dec_ad");<br>
<br>
GstPad *targetsink = gst_element_get_pad (decad,
"sink");<br>
g_assert (targetsink != NULL);<br>
gst_pad_link (pad, targetsink);<br>
gst_object_unref (targetsink);<br>
}<br>
<br>
gst_caps_unref (caps);<br>
<br>
<div>
<div>
<div>My question is that what type of demuxer i have to
select for the properties of video and audio it is
streaming.<br>
</div>
<div><br>
</div>
<div>If any other solution is there for demuxing audio and
video from rtsp url please suggest or provide some sample
code for that.<br>
<br>
</div>
<div>Thanks.<br>
</div>
</div>
</div>
</div>
<br>
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