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Example GStreamer Pipelines

1,109 bytes added, 10:16, 31 December 2013
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Loopback: Audio
=== Loopback: Audio ===
For setting the ALSA channels and amplification correctly see [http://www.igep.es/index.php?option=com_kunena&Itemid=97&func=view&catid=58&id=276&limit=6&limitstart=12#3912 this] link.
<br>
In order to have access to the alsasrc and alsasink plugins perform a 'apt-get install gstreamer0.10-alsa' on the igep board.
<br>
<pre>mv /etc/asound.conf /etc/asound.conf.orig
</pre>
to move the ALSA configuration file out of the way.
=== Loopback: Audio+Video ===
<pre>gst-launch udpsrc port=5555 caps="application/x-rtp"&nbsp;! queue&nbsp;! rtppcmudepay&nbsp;! mulawdec&nbsp;! audioconvert&nbsp;! alsasink
</pre>
The above example experienced dropped audio, please update pipeline when you get it working properly.
<br> Case 2: sending audio from Ubuntu host to target (BeagleBoard)
<pre>gst-launch udpsrc port=5555 caps="application/x-rtp"&nbsp;! queue&nbsp;! rtppcmudepay&nbsp;! mulawdec&nbsp;! audioconvert&nbsp;! alsasink
</pre>
The above example experienced dropped audio, please update pipeline when you get it working properly.<br>I had the same problem using my IGEP WLAN interface. After direct connect with Ethernet cable the dropped audio problem was solved. I also used these pipelines:<br>On host:<pre>gst-launch filesrc location=DownUnder.mp3 ! mad ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16,\rate=44100 ! rtpL16pay ! udpsink host=192.168.2.8 port=5000</pre>On target:<pre>gst-launch udpsrc port=5000 ! “application/x-rtp, media=(string)audio, clock-rate=44100, width=16, height=16, \encoding-name=(string)L16,encoding-params=(string)1, channels=(int)1, channel-position=(int)1, payload=(int)96” ! \gstrtpjitterbuffer do-lost=true ! rtpL16depay ! audioconvert ! alsasink sync=false</pre> And if you want to multicast your stream to multiple computers try this one: On host:<pre>gst-launch filesrc location=DownUnder.mp3 ! mad ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16, \rate=44100 ! rtpL16pay ! udpsink host=224.0.0.15 port=5000</pre> On multiple targets:<pre>gst-launch-0.10 udpsrc port=5000 ! "application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16, height=16,\ encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96" ! \gstrtpjitterbuffer do-lost=true ! rtpL16depay ! audioconvert ! alsasink sync=false</pre>
==== H.264 RTP Streaming ====
*gst-launch -v filesrc location=sample.aac ! faad ! audioconvert ! audioresample ! alsasink
*gst-launch -v filesrc location=sample.aac ! TIAuddec1 codecName=aachedec engineName=codecServer ! dmaiperf engine-name=codecServer ! alsasink sync=false  [[Category:DMAI_GStreamer_Plug-InGstreamer]] [[Category:GstreamerSoftware applications]]
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