docs: update example pipelines in element docs

Mostly gst-launch -> gst-launch-1.0, but also
use autoaudiosink/autovideosink in more places
and update pipelines a little or flesh out
descriptions.

diff --git a/ext/a52dec/gsta52dec.c b/ext/a52dec/gsta52dec.c
index b987482d37caf804610e0e7985772b350801d777..6a9267e14ce65b00f2ab1c5fa5cbaa2bce9b9bb5 100644 (file)
--- a/ext/a52dec/gsta52dec.c
+++ b/ext/a52dec/gsta52dec.c
@@ -25,11 +25,11 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioresample ! audioconvert ! alsasink
- * ]| Play audio track from a dvd.
+ * gst-launch-1.0 dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioconvert ! audioresample ! autoaudiosink
+ * ]| Play audio part of a dvd title.
  * |[
- * gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink
- * ]| Decode a stand alone file and play it.
+ * gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
+ * ]| Decode and play a stand alone AC-3 file.
  * </refsect2>
  */
 

diff --git a/ext/amrnb/amrnbdec.c b/ext/amrnb/amrnbdec.c
index 2388a951f71dc4ce6e8c64b55279cde00b9f82ed..6bf617035638994dd246fd316d559df83410a49f 100644 (file)
--- a/ext/amrnb/amrnbdec.c
+++ b/ext/amrnb/amrnbdec.c
@@ -27,7 +27,7 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch filesrc location=abc.amr ! amrparse ! amrnbdec ! audioresample ! audioconvert ! alsasink
+ * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrnbdec ! audioconvert ! audioresample ! autoaudiosink
  * ]|
  * </refsect2>
  */

diff --git a/ext/amrnb/amrnbenc.c b/ext/amrnb/amrnbenc.c
index e0f21c2747b6780c919c94ae304c69a455d26258..0b5d755fa9aa12b76392b8738168e32e26d16560 100644 (file)
--- a/ext/amrnb/amrnbenc.c
+++ b/ext/amrnb/amrnbenc.c
@@ -27,7 +27,7 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! filesink location=abc.amr
+ * gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr
  * ]|
  * Please note that the above stream misses the header, that is needed to play
  * the stream.

diff --git a/ext/amrwbdec/amrwbdec.c b/ext/amrwbdec/amrwbdec.c
index b6e5f9854c19bd8a8bcd45e40d10e5bc83014021..be089e7d99c542b0f7d4e762a9994eb5e57d4aba 100644 (file)
--- a/ext/amrwbdec/amrwbdec.c
+++ b/ext/amrwbdec/amrwbdec.c
@@ -27,7 +27,7 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! audioresample ! audioconvert ! alsasink
+ * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink
  * ]|
  * </refsect2>
  */

diff --git a/ext/cdio/gstcdiocddasrc.c b/ext/cdio/gstcdiocddasrc.c
index a13d9085783f89a916f02ba4f395f4e591c3304b..601ee98da52522df134612184635554dae0ec2aa 100644 (file)
--- a/ext/cdio/gstcdiocddasrc.c
+++ b/ext/cdio/gstcdiocddasrc.c
@@ -62,7 +62,7 @@
  * <title>Example launch line</title>
  * <para>
  * <programlisting>
- * gst-launch cdiocddasrc track=5 device=/dev/cdrom ! audioconvert ! vorbisenc ! oggmux ! filesink location=track5.ogg
+ * gst-launch-1.0 cdiocddasrc track=5 device=/dev/cdrom ! audioconvert ! vorbisenc ! oggmux ! filesink location=track5.ogg
  * </programlisting>
  * This pipeline extracts track 5 of the audio CD and encodes it into an
  * Ogg/Vorbis file.

diff --git a/ext/lame/gstlamemp3enc.c b/ext/lame/gstlamemp3enc.c
index 8fcd00ac87420fefe9f0ab7731ef8df8ac294386..7ed53b34098ca1b08c85145aec3a2008dcf5bb2d 100644 (file)
--- a/ext/lame/gstlamemp3enc.c
+++ b/ext/lame/gstlamemp3enc.c
@@ -41,19 +41,19 @@
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
+ * gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
  * ]| Encode a test sine signal to MP3.
  * |[
- * gst-launch -v alsasrc ! audioconvert ! lamemp3enc target=bitrate bitrate=192 ! filesink location=alsasrc.mp3
+ * gst-launch-1.0 -v autoaudiosrc ! audioconvert ! lamemp3enc target=bitrate bitrate=192 ! filesink location=alsasrc.mp3
  * ]| Record from a sound card using ALSA and encode to MP3 with an average bitrate of 192kbps
  * |[
- * gst-launch -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lamemp3enc target=quality quality=0 ! id3v2mux ! filesink location=music.mp3
+ * gst-launch-1.0 -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lamemp3enc target=quality quality=0 ! id3v2mux ! filesink location=music.mp3
  * ]| Transcode from a .wav file to MP3 (the id3v2mux element is optional) with best VBR quality
  * |[
- * gst-launch -v cdda://5 ! audioconvert ! lamemp3enc target=bitrate cbr=true bitrate=192 ! filesink location=track5.mp3
+ * gst-launch-1.0 -v cdda://5 ! audioconvert ! lamemp3enc target=bitrate cbr=true bitrate=192 ! filesink location=track5.mp3
  * ]| Encode Audio CD track 5 to MP3 with a constant bitrate of 192kbps
  * |[
- * gst-launch -v audiotestsrc num-buffers=10 ! audio/x-raw,rate=44100,channels=1 ! lamemp3enc target=bitrate cbr=true bitrate=48 ! filesink location=test.mp3
+ * gst-launch-1.0 -v audiotestsrc num-buffers=10 ! audio/x-raw,rate=44100,channels=1 ! lamemp3enc target=bitrate cbr=true bitrate=48 ! filesink location=test.mp3
  * ]| Encode to a fixed sample rate
  * </refsect2>
  *

diff --git a/ext/mad/gstmad.c b/ext/mad/gstmad.c
index 8de38a89bc721325e5a372b8055f30e0a05afeba..3d706a68e9112f06368f830f48c1366d9e28ed24 100644 (file)
--- a/ext/mad/gstmad.c
+++ b/ext/mad/gstmad.c
@@ -21,13 +21,22 @@
  * SECTION:element-mad
  * @see_also: lame
  *
- * MP3 audio decoder.
+ * MP3 audio decoder. Note that while the mad plugin code is licensed under
+ * the LGPL, the libmad library itself is GPL licensed, so the effective
+ * runtime license of using the mad element is GPL.
+ *
+ * The mpg123audiodec and avdec_mp3 decoder elements are LGPL licensed and
+ * also tend to use less CPU for decoding.
+ *
+ * In general it doesn't matter which MP3 decoder you use, the main advantage
+ * of the mad decoder is that it also supports so-called 'freeform' mp3s which
+ * are mp3s with a bitrate higher than what the standard usually allows.
  *
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch filesrc location=music.mp3 ! mpegaudioparse ! mad ! audioconvert ! audioresample ! autoaudiosink
- * ]| Decode and play the mp3 file
+ * gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mad ! audioconvert ! audioresample ! autoaudiosink
+ * ]| Decode and play an mp3 file
  * </refsect2>
  */
 

diff --git a/ext/sidplay/gstsiddec.cc b/ext/sidplay/gstsiddec.cc
index a7cac36830b25c0c23e091117c033e2b578a08c8..c7a0905394a8b27362af80567b303e65d03bfbd2 100644 (file)
--- a/ext/sidplay/gstsiddec.cc
+++ b/ext/sidplay/gstsiddec.cc
@@ -33,8 +33,8 @@
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch -v filesrc location=Hawkeye.sid ! siddec ! audioconvert ! alsasink
- * ]| Decode a sid file and play back the audio using alsasink.
+ * gst-launch-1.0 -v filesrc location=Hawkeye.sid ! siddec ! audioconvert ! audioresample ! autoaudiosink
+ * ]| Decode a sid file and play it back.
  * </refsect2>
  */
 

diff --git a/ext/twolame/gsttwolamemp2enc.c b/ext/twolame/gsttwolamemp2enc.c
index 63f1fd33c0ea3cab1e3f9124ff215861bdc1b2a1..85438769cf39881fc3bc7b4b983b70292c4743de 100644 (file)
--- a/ext/twolame/gsttwolamemp2enc.c
+++ b/ext/twolame/gsttwolamemp2enc.c
@@ -33,16 +33,16 @@
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! twolame ! filesink location=sine.mp2
+ * gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! twolame ! filesink location=sine.mp2
  * ]| Encode a test sine signal to MP2.
  * |[
- * gst-launch -v alsasrc ! audioconvert ! twolame bitrate=192 ! filesink location=alsasrc.mp2
+ * gst-launch-1.0 -v alsasrc ! audioconvert ! twolame bitrate=192 ! filesink location=alsasrc.mp2
  * ]| Record from a sound card using ALSA and encode to MP2
  * |[
- * gst-launch -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! twolame bitrate=192 ! id3v2mux ! filesink location=music.mp2
+ * gst-launch-1.0 -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! twolame bitrate=192 ! id3v2mux ! filesink location=music.mp2
  * ]| Transcode from a .wav file to MP2 (the id3v2mux element is optional)
  * |[
- * gst-launch -v cdda://5 ! audioconvert ! twolame bitrate=192 ! filesink location=track5.mp2
+ * gst-launch-1.0 -v cdda://5 ! audioconvert ! twolame bitrate=192 ! filesink location=track5.mp2
  * ]| Encode Audio CD track 5 to MP2
  * </refsect2>
  *

diff --git a/ext/x264/gstx264enc.c b/ext/x264/gstx264enc.c
index 1161653272a288a9c829b1f75def9712c01e5442..c52bf1a8612ceba2cf5efb9292a537012d25fe32 100644 (file)
--- a/ext/x264/gstx264enc.c
+++ b/ext/x264/gstx264enc.c
@@ -56,31 +56,35 @@
  * non-x264enc streams/branches filling up and blocking upstream. They can
  * be fixed by relaxing the default time/size/buffer limits on the queue
  * elements in the non-x264 branches, or using a (single) multiqueue element
- * for all branches. Also see the last example below.
+ * for all branches. Also see the last example below. You can also work around
+ * this problem by setting the tune=zerolatency property, but this will affect
+ * overall encoding quality so may not be appropriate for your use case.
  * </note>
  *
  * <refsect2>
  * <title>Example pipeline</title>
  * |[
- * gst-launch -v videotestsrc num-buffers=1000 ! x264enc qp-min=18 ! \
+ * gst-launch-1.0 -v videotestsrc num-buffers=1000 ! x264enc qp-min=18 ! \
  *   avimux ! filesink location=videotestsrc.avi
  * ]| This example pipeline will encode a test video source to H264 muxed in an
  * AVI container, while ensuring a sane minimum quantization factor to avoid
- * some (excessive) waste.
+ * some (excessive) waste. You should ideally never put H264 into an AVI
+ * container (or really anything else, for that matter) - use Matroska or
+ * MP4/QuickTime or MPEG-TS instead.
  * |[
- * gst-launch -v videotestsrc num-buffers=1000 ! x264enc pass=quant ! \
- *   matroskamux ! filesink location=videotestsrc.avi
+ * gst-launch-1.0 -v videotestsrc num-buffers=1000 ! x264enc pass=quant ! \
+ *   matroskamux ! filesink location=videotestsrc.mkv
  * ]| This example pipeline will encode a test video source to H264 using fixed
  * quantization, and muxes it in a Matroska container.
  * |[
- * gst-launch -v videotestsrc num-buffers=1000 ! x264enc pass=5 quantizer=25 speed-preset=6 ! video/x-h264, profile=baseline ! \
+ * gst-launch-1.0 -v videotestsrc num-buffers=1000 ! x264enc pass=5 quantizer=25 speed-preset=6 ! video/x-h264, profile=baseline ! \
  *   qtmux ! filesink location=videotestsrc.mov
  * ]| This example pipeline will encode a test video source to H264 using
  * constant quality at around Q25 using the 'medium' speed/quality preset and
  * restricting the options used so that the output is H.264 Baseline Profile
  * compliant and finally multiplexing the output in Quicktime mov format.
  * |[
- * gst-launch -v videotestsrc num-buffers=1000 ! tee name=t ! queue ! xvimagesink \
+ * gst-launch-1.0 -v videotestsrc num-buffers=1000 ! tee name=t ! queue ! videoconvert ! autovideosink \
  *   t. ! queue ! x264enc rc-lookahead=5 ! fakesink
  * ]| This example pipeline will encode a test video source to H264 while
  * displaying the input material at the same time.  As mentioned above,

diff --git a/gst/realmedia/rademux.c b/gst/realmedia/rademux.c
index 130f7a847fec1f80ad7cc8c9d098fe4dac187302..5aa92ac434b6a9c304c9ca1b42333dc334f49b0c 100644 (file)
--- a/gst/realmedia/rademux.c
+++ b/gst/realmedia/rademux.c
@@ -25,13 +25,13 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch filesrc location=interview.ra ! rademux ! ffdec_real_288 ! audioconvert ! audioresample ! alsasink
+ * gst-launch-1.0 filesrc location=interview.ra ! rademux ! avdec_real_288 ! audioconvert ! audioresample ! autoaudiosink
  * ]| Read a RealAudio file and decode it and output it to the soundcard using
  * the ALSA element. The .ra file is assumed to contain RealAudio version 2.
  * |[
- * gst-launch gnomevfssrc location=http://www.example.org/interview.ra ! rademux ! a52dec ! audioconvert ! audioresample ! alsasink
+ * gst-launch-1.0 souphttpsrc location=http://www.example.org/interview.ra ! rademux ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
  * ]| Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
- * and output it to the soundcard using the ALSA element.
+ * and output it to the soundcard.
  * </refsect2>
  */
 

diff --git a/gst/xingmux/gstxingmux.c b/gst/xingmux/gstxingmux.c
index fd09ac078e06ab2712f8c9043ccc9e9323be3de7..26c6196180eb2b84b11500b40922910707549ee1 100644 (file)
--- a/gst/xingmux/gstxingmux.c
+++ b/gst/xingmux/gstxingmux.c
@@ -33,9 +33,9 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch audiotestsrc num-buffers=1000 ! audioconvert ! lamemp3enc ! xingmux ! filesink location=test.mp3
- * gst-launch filesrc location=test.mp3 ! xingmux ! filesink location=test2.mp3
- * gst-launch filesrc location=test.mp3 ! mp3parse ! xingmux ! filesink location=test2.mp3
+ * gst-launch-1.0 audiotestsrc num-buffers=1000 ! audioconvert ! lamemp3enc ! xingmux ! filesink location=test.mp3
+ * gst-launch-1.0 filesrc location=test.mp3 ! xingmux ! filesink location=test2.mp3
+ * gst-launch-1.0 filesrc location=test.mp3 ! mp3parse ! xingmux ! filesink location=test2.mp3
  * ]|
  * </refsect2>
  */