Commit 2e412a44 authored by Tim-Philipp Müller's avatar Tim-Philipp Müller 🐠

docs: update example pipelines in element docs

Mostly gst-launch -> gst-launch-1.0
Use autovideosink/autoaudiosink more often.
Sprinkle some converters here and there.
parent 497cfc83
......@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch uridecodebin uri=file:///path/to/audiofile ! audioconvert ! vorbisenc ! oggmux ! shout2send mount=/stream.ogg port=8000 username=source password=somepassword ip=server_IP_address_or_hostname
* gst-launch-1.0 uridecodebin uri=file:///path/to/audiofile ! audioconvert ! vorbisenc ! oggmux ! shout2send mount=/stream.ogg port=8000 username=source password=somepassword ip=server_IP_address_or_hostname
* ]| This pipeline demuxes, decodes, re-encodes and re-muxes an audio
* media file into oggvorbis and sends the resulting stream to an Icecast
* server. Properties mount, port, username and password are all server-config
......
......@@ -32,7 +32,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v filesrc location=videotestsrc.webm ! matroskademux ! vp8dec ! xvimagesink
* gst-launch-1.0 -v filesrc location=videotestsrc.webm ! matroskademux ! vp8dec ! videoconvert ! videoscale ! autovideosink
* ]| This example pipeline will decode a WebM stream and decodes the VP8 video.
* </refsect2>
*/
......
......@@ -40,7 +40,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v videotestsrc num-buffers=1000 ! vp8enc ! webmmux ! filesink location=videotestsrc.webm
* gst-launch-1.0 -v videotestsrc num-buffers=1000 ! vp8enc ! webmmux ! filesink location=videotestsrc.webm
* ]| This example pipeline will encode a test video source to VP8 muxed in an
* WebM container.
* </refsect2>
......
......@@ -32,7 +32,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v filesrc location=videotestsrc.webm ! matroskademux ! vp9dec ! xvimagesink
* gst-launch-1.0 -v filesrc location=videotestsrc.webm ! matroskademux ! vp9dec ! videoconvert ! videoscale ! autovideosink
* ]| This example pipeline will decode a WebM stream and decodes the VP9 video.
* </refsect2>
*/
......
......@@ -40,7 +40,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v videotestsrc num-buffers=1000 ! vp9enc ! webmmux ! filesink location=videotestsrc.webm
* gst-launch-1.0 -v videotestsrc num-buffers=1000 ! vp9enc ! webmmux ! filesink location=videotestsrc.webm
* ]| This example pipeline will encode a test video source to VP9 muxed in an
* WebM container.
* </refsect2>
......
......@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL16pay example to create the RTP stream.
* </refsect2>
......
......@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
* gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
* ]| This example pipeline will payload raw audio. Refer to
* the rtpL16depay example to depayload and play the RTP stream.
* </refsect2>
......
......@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink
* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL24pay example to create the RTP stream.
* </refsect2>
......
......@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink
* gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink
* ]| This example pipeline will payload raw audio. Refer to
* the rtpL24depay example to depayload and play the RTP stream.
* </refsect2>
......
......@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
* gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
* ]| This example pipeline will encode and payload AC3 stream. Refer to
* the rtpac3depay example to depayload and decode the RTP stream.
* </refsect2>
......
......@@ -27,7 +27,7 @@
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink
* gst-launch-1.0 -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink
* ]| This example pipeline will encode and payload an AMR stream. Refer to
* the rtpamrdepay example to depayload and decode the RTP stream.
* </refsect2>
......
......@@ -35,7 +35,7 @@
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch rtpmux name=mux ! udpsink host=127.0.0.1 port=8888 \
* gst-launch-1.0 rtpmux name=mux ! udpsink host=127.0.0.1 port=8888 \
* alsasrc ! alawenc ! rtppcmapay ! \
* application/x-rtp, payload=8, rate=8000 ! mux.sink_0 \
* audiotestsrc is-live=1 ! \
......
......@@ -58,7 +58,7 @@ bus_handler (GstBus * bus, GstMessage * message, gpointer data)
}
/*
* gst-launch \
* gst-launch-1.0 \
* audiotestsrc freq=440 num-buffers=100 ! interleave name=i ! audioconvert ! wavenc ! filesink location=/tmp/mc.wav \
* audiotestsrc freq=880 num-buffers=100 ! i.
* ...
......
......@@ -116,9 +116,9 @@ pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
/* build a pipeline equivalent to:
*
* gst-launch -v rtpbin name=rtpbin \
* gst-launch-1.0 -v rtpbin name=rtpbin \
* udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
* rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
* rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
* rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false
*/
......
......@@ -108,7 +108,7 @@ print_stats (GstElement * rtpbin)
/* build a pipeline equivalent to:
*
* gst-launch -v rtpbin name=rtpbin \
* gst-launch-1.0 -v rtpbin name=rtpbin \
* $AUDIO_SRC ! audioconvert ! audioresample ! $AUDIO_ENC ! $AUDIO_PAY ! rtpbin.send_rtp_sink_0 \
* rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \
* rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
......
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