* gst-launch-1.0 filesrc location=test.dv ! dvdemux name=demux ! dvdec ! xvimagesink
* ]| This pipeline decodes and renders the raw DV stream to a videosink.
* </refsect2>
- *
- * Last reviewed on 2006-02-28 (0.10.3)
*/
#ifdef HAVE_CONFIG_H
* gst-launch-1.0 filesrc location=test.dv ! dvdemux name=demux ! queue ! audioconvert ! alsasink demux. ! queue ! dvdec ! xvimagesink
* ]| This pipeline decodes and renders the raw DV stream to an audio and a videosink.
* </refsect2>
- *
- * Last reviewed on 2006-02-27 (0.10.3)
*/
/* DV output has two modes, normal and wide. The resolution is the same in both
* gst-launch-1.0 audiotestsrc ! jackaudiosink
* ]| Play a sine wave to using jack.
* </refsect2>
- *
- * Last reviewed on 2006-11-30 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
* gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0
* ]| Get audio input into gstreamer from jack.
* </refsect2>
- *
- * Last reviewed on 2008-07-22 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
* ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the
* documentation of speexenc.
* </refsect2>
- *
- * Last reviewed on 2006-04-05 (0.10.2)
*/
#ifdef HAVE_CONFIG_H
* compressed audio or video data, this will only work if you have the
* right decoder elements/plugins installed.
* </refsect2>
- *
- * Last reviewed on 2006-12-29 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
* This plays an avi file with a video and subtitle stream.
* </para>
* </refsect2>
- *
- * Last reviewed on 2008-02-01
*/
/* example of a subtitle chunk in an avi file
* </programlisting>
* </para>
* </refsect2>
- *
- * Documentation last reviewed on 2011-04-21
*/
#ifdef HAVE_CONFIG_H
* Records a video stream captured from a v4l2 device, encodes it into H.264
* and muxes it into an mp4 file.
* </refsect2>
- *
- * Documentation last reviewed on 2011-04-21
*/
/* ============================= 3gppmux ==================================== */
* ]|
* Records a video stream captured from a v4l2 device and muxes it into a qt file.
* </refsect2>
- *
- * Last reviewed on 2010-12-03
*/
/*
* compressed audio or video data, this will only work if you have the
* right decoder elements/plugins installed.
* </refsect2>
- *
- * Last reviewed on 2006-12-29 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
* gst-launch-1.0 videotestsrc ! multifilesink post-messages=true filename="frame%d"
* ]|
* </refsect2>
- *
- * Last reviewed on 2009-09-11 (0.10.17)
*/
#ifdef HAVE_CONFIG_H
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL16pay example to create the RTP stream.
* </refsect2>
- *
- * Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
* ]| This example pipeline will payload raw audio. Refer to
* the rtpL16depay example to depayload and play the RTP stream.
* </refsect2>
- *
- * Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
* ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to
* the rtpac3pay example to create the RTP stream.
* </refsect2>
- *
- * Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
* ]| This example pipeline will encode and payload AC3 stream. Refer to
* the rtpac3depay example to depayload and decode the RTP stream.
* </refsect2>
- *
- * Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
* ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to
* the rtpamrpay example to create the RTP stream.
* </refsect2>
- *
- * Last reviewed on 2013-04-25 (1.1.0)
*/
/*
* ]| This example pipeline will encode and payload an AMR stream. Refer to
* the rtpamrdepay example to depayload and decode the RTP stream.
* </refsect2>
- *
- * Last reviewed on 2013-04-25 (1.1.0)
*/
/* references:
*
* Extract BroadcomVoice audio from RTP packets according to RFC 4298.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
- *
- * Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
*
* Payload BroadcomVoice audio into RTP packets according to RFC 4298.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
- *
- * Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
* Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
* on port 5007.
* </refsect2>
- *
- * Last reviewed on 2007-08-30 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
* inserted into the pipeline to smooth out network jitter and to reorder the
* out-of-order RTP packets.
* </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
* generated, both are encoded into different payload types and muxed together
* so they can be sent on the same port.
* </refsect2>
- *
- * Last reviewed on 2010-09-30 (0.10.21)
*/
#ifdef HAVE_CONFIG_H
* ]| Takes an RTP stream and send the RTP packets with the first detected
* payload type to fakesink, discarding the other payload types.
* </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
*/
/*
* You can play with the drop-probability value for one or both streams.
* You should hear a clear sound. (after a few seconds the two streams wave feel synchronized)
* </refsect2>
- *
- * Last reviewed on 2013-11-08 (1.x)
*/
#ifdef HAVE_CONFIG_H
* packets are sent in the PAUSED state). Applications should manually set and
* keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
* </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
* ]| Takes an RTP stream and send the RTP packets with the first detected SSRC
* to fakesink, discarding the other SSRCs.
* </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
* SECTION:element-rtpdec
*
* A simple RTP session manager used internally by rtspsrc.
- *
- * Last reviewed on 2006-06-20 (0.10.4)
*/
/* #define HAVE_RTCP */
* ]| Establish a connection to an RTSP server and send the raw RTP packets to a
* fakesink.
* </refsect2>
- *
- * Last reviewed on 2006-08-18 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" />
* </programlisting></informalexample>
* </refsect2>
- *
- * Last reviewed on 2011-03-10 (0.10.29)
*/
#ifdef HAVE_CONFIG_H
* gst-launch-1.0 -v udpsrc port=0 ! fakesink
* ]| read udp packets from a free port.
* </refsect2>
- *
- * Last reviewed on 2007-09-20 (0.10.7)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
* gst-launch-1.0 videotestsrc ! gamma gamma=0.5 ! videoconvert ! ximagesink
* ]| This pipeline will make the image "darker".
* </refsect2>
- *
- * Last reviewed on 2010-04-18 (0.10.22)
*/
#ifdef HAVE_CONFIG_H
* ]| This pipeline converts the image to black and white by setting the
* saturation to 0.0.
* </refsect2>
- *
- * Last reviewed on 2010-04-18 (0.10.22)
*/
#ifdef HAVE_CONFIG_H
* gst-launch-1.0 videotestsrc ! videoflip method=clockwise ! videoconvert ! ximagesink
* ]| This pipeline flips the test image 90 degrees clockwise.
* </refsect2>
- *
- * Last reviewed on 2010-04-18 (0.10.22)
*/
* gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
* ]| Stream data from a network url.
* </refsect2>
- *
- * Last reviewed on 2007-02-14 (0.10.6)
*/
/*
* gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink
* ]| Play an Ogg/Vorbis file.
* </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
*/
#ifdef HAVE_CONFIG_H