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author | Tim-Philipp Müller <tim@centricular.com> | 2015-05-10 11:05:00 +0100 |
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committer | Tim-Philipp Müller <tim@centricular.com> | 2015-05-10 11:05:00 +0100 |
commit | 2e412a447a66c3a44aa660c122657db5056c623c (patch) | |
tree | 6fef500980e37009ea4de9e1f599bc6851d754de | |
parent | 497cfc83f6ebe3741e90d09e6a239004381837d5 (diff) | |
download | gstreamer-plugins-good-2e412a447a66c3a44aa660c122657db5056c623c.tar.gz |
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.
-rw-r--r-- | ext/shout2/gstshout2.c | 2 | ||||
-rw-r--r-- | ext/vpx/gstvp8dec.c | 2 | ||||
-rw-r--r-- | ext/vpx/gstvp8enc.c | 2 | ||||
-rw-r--r-- | ext/vpx/gstvp9dec.c | 2 | ||||
-rw-r--r-- | ext/vpx/gstvp9enc.c | 2 | ||||
-rw-r--r-- | gst/rtp/gstrtpL16depay.c | 2 | ||||
-rw-r--r-- | gst/rtp/gstrtpL16pay.c | 2 | ||||
-rw-r--r-- | gst/rtp/gstrtpL24depay.c | 2 | ||||
-rw-r--r-- | gst/rtp/gstrtpL24pay.c | 2 | ||||
-rw-r--r-- | gst/rtp/gstrtpac3pay.c | 2 | ||||
-rw-r--r-- | gst/rtp/gstrtpamrpay.c | 2 | ||||
-rw-r--r-- | gst/rtpmanager/gstrtpmux.c | 2 | ||||
-rw-r--r-- | tests/check/pipelines/wavenc.c | 2 | ||||
-rw-r--r-- | tests/examples/rtp/client-PCMA.c | 4 | ||||
-rw-r--r-- | tests/examples/rtp/server-alsasrc-PCMA.c | 2 |
15 files changed, 16 insertions, 16 deletions
diff --git a/ext/shout2/gstshout2.c b/ext/shout2/gstshout2.c index 613bc7ad6..2384ebc9a 100644 --- a/ext/shout2/gstshout2.c +++ b/ext/shout2/gstshout2.c @@ -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 diff --git a/ext/vpx/gstvp8dec.c b/ext/vpx/gstvp8dec.c index 7bc660a90..4b05f9969 100644 --- a/ext/vpx/gstvp8dec.c +++ b/ext/vpx/gstvp8dec.c @@ -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> */ diff --git a/ext/vpx/gstvp8enc.c b/ext/vpx/gstvp8enc.c index a3dd6f0b7..ee1cbb9f1 100644 --- a/ext/vpx/gstvp8enc.c +++ b/ext/vpx/gstvp8enc.c @@ -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> diff --git a/ext/vpx/gstvp9dec.c b/ext/vpx/gstvp9dec.c index db65c9f0d..a11c84810 100644 --- a/ext/vpx/gstvp9dec.c +++ b/ext/vpx/gstvp9dec.c @@ -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> */ diff --git a/ext/vpx/gstvp9enc.c b/ext/vpx/gstvp9enc.c index e64f5af5e..532b59ebd 100644 --- a/ext/vpx/gstvp9enc.c +++ b/ext/vpx/gstvp9enc.c @@ -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> diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c index 5bbbedc18..667a64a6c 100644 --- a/gst/rtp/gstrtpL16depay.c +++ b/gst/rtp/gstrtpL16depay.c @@ -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> diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c index 5d92c25e8..a8ed36fd2 100644 --- a/gst/rtp/gstrtpL16pay.c +++ b/gst/rtp/gstrtpL16pay.c @@ -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> diff --git a/gst/rtp/gstrtpL24depay.c b/gst/rtp/gstrtpL24depay.c index 1925ce082..7b5ef0c8b 100644 --- a/gst/rtp/gstrtpL24depay.c +++ b/gst/rtp/gstrtpL24depay.c @@ -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> diff --git a/gst/rtp/gstrtpL24pay.c b/gst/rtp/gstrtpL24pay.c index d2612e9c3..1bb37cb3d 100644 --- a/gst/rtp/gstrtpL24pay.c +++ b/gst/rtp/gstrtpL24pay.c @@ -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> diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c index a6ef9140a..e283afd0d 100644 --- a/gst/rtp/gstrtpac3pay.c +++ b/gst/rtp/gstrtpac3pay.c @@ -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> diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c index ed319cc03..ead9f9404 100644 --- a/gst/rtp/gstrtpamrpay.c +++ b/gst/rtp/gstrtpamrpay.c @@ -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> diff --git a/gst/rtpmanager/gstrtpmux.c b/gst/rtpmanager/gstrtpmux.c index a3759c210..db3585bc1 100644 --- a/gst/rtpmanager/gstrtpmux.c +++ b/gst/rtpmanager/gstrtpmux.c @@ -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 ! \ diff --git a/tests/check/pipelines/wavenc.c b/tests/check/pipelines/wavenc.c index b79fbf6ff..ee30c1863 100644 --- a/tests/check/pipelines/wavenc.c +++ b/tests/check/pipelines/wavenc.c @@ -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. * ... diff --git a/tests/examples/rtp/client-PCMA.c b/tests/examples/rtp/client-PCMA.c index ee8366f53..664b33664 100644 --- a/tests/examples/rtp/client-PCMA.c +++ b/tests/examples/rtp/client-PCMA.c @@ -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 */ diff --git a/tests/examples/rtp/server-alsasrc-PCMA.c b/tests/examples/rtp/server-alsasrc-PCMA.c index 4690eed45..af8f7de5a 100644 --- a/tests/examples/rtp/server-alsasrc-PCMA.c +++ b/tests/examples/rtp/server-alsasrc-PCMA.c @@ -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 \ |