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Ast rtp read rtp codec 96 received from
Ast rtp read rtp codec 96 received from









ast rtp read rtp codec 96 received from

Loopback: Video gst-launch -v videotestsrc ! TIDmaiVideoSink videoStd=VGA videoOutput=LCD accelFrameCopy=FALSE sync=false If you use IGEP GST FRAMEWORK 2.00.20 you can use "omapdmaifbsink" instead of "TIDmaiVideoSink" to display the video inside the X windowing system. This section covers pipelines for common use cases for the OMAP3530 or DM3730 processor.Įxport GST_REGISTRY=/tmp/gst_registry.binĮxport LD_LIBRARY_PATH=/opt/gstreamer/libĮxport GST_PLUGIN_PATH=/opt/gstreamer/lib/gstreamer-0.10 Run the command on your host computer.įfmpeg -i tropic_thunder-tlr1a_720p.mov -r 60 -b 6000000 -vcodec mpeg2video -ab 48000000 -acodec libmp3lame -s 1280x544 tropic.aviįollowing are a list of supported platforms, with links that jump directly to pipeline examples for each platform. mov file (say from the Apple movie trailers site) and make an AVI file. You should be able to use any audio and video media file that conforms to the appropriate standard.

ast rtp read rtp codec 96 received from

If you find an error in a pipeline please correct it. Refer to this Gstreamer article for more information on downloading and building TI Gstreamer elements.Ĭurrently these pipelines have not undergone any extensive testing. Some of the pipelines may need modification for things such as file names, ip addresses, etc. This page provides example pipelines that can be copied to the command line to demonstrate various GStreamer operations. If you do not do this you will get an error like gst-launch-0.10: BufTab.c:440: BufTab_getNumBufs: Assertion `hBufTab' failed Purpose You may also make a link to the codecserver in the directory were you execute your command.

ast rtp read rtp codec 96 received from

Gst-launch command in the directory were the codec server (cs.圆4P) is present. When using the pipelines that use the TI codecs on the DSP, make sure you execute the Perhaps also install gstreamer0.10-plugins-good and gstreamer0.10-plugins-bad.įor all examples I had to perform gst-launch-0.10 in stead of gst-launch. gstreamer0.10-alsa for alsasrc and alsasink.In order to make the next examples work I had to install (apt-get install ) the following packages on the target 7.2.2 Generic network audio streaming example.7.2.1 Controlling the sample rate and bit depth.See the LICENSE fileĢ4 : * \brief SIP SDP media stream handlingĥ3 : #include "asterisk/linkedlists.h" /* for AST_LIST_NEXT */ĥ8 : #include "asterisk/res_pjsip_session. Please do not directly contactġ1 : * any of the maintainers of this project for assistance ġ2 : * the project provides a web site, mailing lists and IRCġ5 : * This program is free software, distributed under the terms ofġ6 : * the GNU General Public License Version 2. Line data Source code 1 : /* 2 : * Asterisk - An open source telephony toolkit.ġ0 : * the Asterisk project. Top level - res - res_pjsip_sdp_rtp.c (source / functions) LCOV - Asterisk GIT-16-d8ac5bf1a5 - res/res_pjsip_sdp_rtp.c LCOV - code coverage report











Ast rtp read rtp codec 96 received from