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.

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

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>
  */

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>

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>
  */

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>

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>

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>

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>

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>

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>

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>

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 !                                \

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.
  * ...

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
  */

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 \