docs: fix some docs

from git grep for ffmpegcolorspace and x-raw-

diff --git a/docs/design/draft-klass.txt b/docs/design/draft-klass.txt
index ee1d8da..2fae20b 100644 (file)
--- a/docs/design/draft-klass.txt
+++ b/docs/design/draft-klass.txt
@@ -131,7 +131,7 @@ for wild (application specific) customisation.
    id3demux         : Extracter/Metadata
    udpsrc           : Source/Network/Protocol/Device
    videomixer       : Mixer/Video
-   ffmpegcolorspace : Filter/Video             (intended use to convert video with as little
+   videoconvert     : Filter/Video             (intended use to convert video with as little
                                                 visible change as possible)
    vertigotv        : Effect/Video             (intended use is to change the video)
    volume           : Effect/Audio             (intended use is to change the audio data)

diff --git a/docs/design/part-missing-plugins.txt b/docs/design/part-missing-plugins.txt
index f92d7e0..c467858 100644 (file)
--- a/docs/design/part-missing-plugins.txt
+++ b/docs/design/part-missing-plugins.txt
@@ -168,7 +168,7 @@ A three step process:
              e.g. urisource-http or urisource-mms
 
          - element-$(ELEMENT_REQUIRED),
-             e.g. element-ffmpegcolorspace
+             e.g. element-videoconvert
 
          - decoder-$(CAPS_REQUIRED)
              e.g. decoder-audio/x-vorbis or

diff --git a/docs/faq/using.xml b/docs/faq/using.xml
index 4ee7eed..939ff40 100644 (file)
--- a/docs/faq/using.xml
+++ b/docs/faq/using.xml
@@ -179,18 +179,18 @@ More examples can be found in the gst-launch man page.
 To automatically detect the right codec in a pipeline, try 
 <programlisting>
 gst-launch filesrc location=my-random-media-file.mpeg ! decodebin !
- audioconvert ! audioresample ! osssink
+ audioconvert ! pulsesink
 </programlisting>.
 or
 <programlisting>
 gst-launch filesrc location=my-random-media-file.mpeg ! decodebin !
- ffmpegcolorspace ! xvimagesink
+ videoconvert ! xvimagesink
 </programlisting>
 Something more complicated:
 <programlisting>
 gst-launch filesrc location=my-random-media-file.mpeg ! decodebin name=decoder
-   decoder. ! ffmpegcolorspace ! xvimagesink
-   decoder. ! audioconvert ! audioresample ! osssink
+   decoder. ! videoconvert ! xvimagesink
+   decoder. ! audioconvert ! pulsesink
 </programlisting>
         </para>
         <para>

diff --git a/docs/manual/advanced-dataaccess.xml b/docs/manual/advanced-dataaccess.xml
index 3781a36..be56564 100644 (file)
--- a/docs/manual/advanced-dataaccess.xml
+++ b/docs/manual/advanced-dataaccess.xml
@@ -98,9 +98,9 @@ main (gint   argc,
   filter = gst_element_factory_make ("capsfilter", "filter");
   g_assert (filter != NULL); /* should always exist */
 
-  csp = gst_element_factory_make ("ffmpegcolorspace", "csp");
+  csp = gst_element_factory_make ("videoconvert", "csp");
   if (csp == NULL)
-    g_error ("Could not create 'ffmpegcolorspace' element");
+    g_error ("Could not create 'videoconvert' element");
 
   sink = gst_element_factory_make ("xvimagesink", "sink");
   if (sink == NULL) {
@@ -111,13 +111,11 @@ main (gint   argc,
 
   gst_bin_add_many (GST_BIN (pipeline), src, filter, csp, sink, NULL);
   gst_element_link_many (src, filter, csp, sink, NULL);
-  filtercaps = gst_caps_new_simple ("video/x-raw-rgb",
+  filtercaps = gst_caps_new_simple ("video/x-raw",
+                          "format", G_TYPE_STRING, "RGB16",
                           "width", G_TYPE_INT, 384,
                           "height", G_TYPE_INT, 288,
                           "framerate", GST_TYPE_FRACTION, 25, 1,
-                          "bpp", G_TYPE_INT, 16,
-                          "depth", G_TYPE_INT, 16,
-                          "endianness", G_TYPE_INT, G_BYTE_ORDER,
                           NULL);
   g_object_set (G_OBJECT (filter), "caps", filtercaps, NULL);
   gst_caps_unref (filtercaps);
@@ -313,18 +311,16 @@ main (gint   argc,
   pipeline = gst_pipeline_new ("pipeline");
   fakesrc = gst_element_factory_make ("fakesrc", "source");
   flt = gst_element_factory_make ("capsfilter", "flt");
-  conv = gst_element_factory_make ("ffmpegcolorspace", "conv");
+  conv = gst_element_factory_make ("videoconvert", "conv");
   videosink = gst_element_factory_make ("xvimagesink", "videosink");
 
   /* setup */
   g_object_set (G_OBJECT (flt), "caps",
-               gst_caps_new_simple ("video/x-raw-rgb",
+               gst_caps_new_simple ("video/x-raw",
+                                    "format", G_TYPE_STRING, "RGB16",
                                     "width", G_TYPE_INT, 384,
                                     "height", G_TYPE_INT, 288,
                                     "framerate", GST_TYPE_FRACTION, 1, 1,
-                                    "bpp", G_TYPE_INT, 16,
-                                    "depth", G_TYPE_INT, 16,
-                                    "endianness", G_TYPE_INT, G_BYTE_ORDER,
                                     NULL), NULL);
   gst_bin_add_many (GST_BIN (pipeline), fakesrc, flt, conv, videosink, NULL);
   gst_element_link_many (fakesrc, flt, conv, videosink, NULL);

diff --git a/docs/manual/appendix-checklist.xml b/docs/manual/appendix-checklist.xml
index af9a239..f6e0143 100644 (file)
--- a/docs/manual/appendix-checklist.xml
+++ b/docs/manual/appendix-checklist.xml
@@ -132,7 +132,7 @@
     <para>
       &GStreamer; contains a bunch of conversion plugins that most
       applications will find useful. Specifically, those are videoscalers
-      (videoscale), colorspace convertors (ffmpegcolorspace), audio format
+      (videoscale), colorspace convertors (videoconvert), audio format
       convertors and channel resamplers (audioconvert) and audio samplerate
       convertors (audioresample). Those convertors don't do anything when not
       required, they will act in passthrough mode. They will activate when
@@ -155,7 +155,7 @@
         <command>gst-launch</command> is a simple script-like commandline
         application that can be used to test pipelines. For example, the
         command <command>gst-launch audiotestsrc ! audioconvert ! 
-        audio/x-raw-int,channels=2 ! alsasink</command> will run
+        audio/x-raw,channels=2 ! alsasink</command> will run
         a pipeline which generates a sine-wave audio stream and plays it
         to your ALSA audio card. <command>gst-launch</command> also allows
         the use of threads (will be used automatically as required or as queue
@@ -165,7 +165,7 @@
         or even omit the padname to automatically select a pad. Using
         all this, the pipeline
         <command>gst-launch filesrc location=file.ogg ! oggdemux name=d
-         d. ! queue ! theoradec ! ffmpegcolorspace ! xvimagesink
+         d. ! queue ! theoradec ! videoconvert ! xvimagesink
          d. ! queue ! vorbisdec ! audioconvert ! audioresample ! alsasink
         </command> will play an Ogg file
         containing a Theora video-stream and a Vorbis audio-stream. You can

diff --git a/docs/manual/appendix-programs.xml b/docs/manual/appendix-programs.xml
index 7e84a75..ca8eb11 100644 (file)
--- a/docs/manual/appendix-programs.xml
+++ b/docs/manual/appendix-programs.xml
@@ -21,7 +21,7 @@ gst-launch filesrc location=hello.mp3 ! mad ! audioresample ! osssink
     <screen>
 gst-launch filesrc location=redpill.vob ! dvddemux name=demux \
  demux.audio_00 ! queue ! a52dec ! audioconvert ! audioresample ! osssink \
- demux.video_00 ! queue ! mpeg2dec ! ffmpegcolorspace ! xvimagesink
+ demux.video_00 ! queue ! mpeg2dec ! videoconvert ! xvimagesink
     </screen>
 
     </para>

diff --git a/docs/manual/basics-pads.xml b/docs/manual/basics-pads.xml
index a56ee00..96459b7 100644 (file)
--- a/docs/manual/basics-pads.xml
+++ b/docs/manual/basics-pads.xml
@@ -245,23 +245,22 @@ link_to_multiplexer (GstPad     *tolink_pad,
         <quote>audio/x-vorbis</quote>. The source pad will be used
         to send raw (decoded) audio samples to the next element, with
         a raw audio mime-type (in this case,
-       <quote>audio/x-raw-float</quote>).  The source pad will also
+       <quote>audio/x-raw</quote>).  The source pad will also
         contain properties for the audio samplerate and the amount of
         channels, plus some more that you don't need to worry about
         for now.
       </para>
       <programlisting>
+
 Pad Templates:
   SRC template: 'src'
     Availability: Always
     Capabilities:
-      audio/x-raw-float
-                   rate: [ 8000, 50000 ]
-               channels: [ 1, 2 ]
-             endianness: 1234
-                  width: 32
-          buffer-frames: 0
- 
+      audio/x-raw
+                 format: F32LE
+                   rate: [ 1, 2147483647 ]
+               channels: [ 1, 256 ]
+
   SINK template: 'sink'
     Availability: Always
     Capabilities:
@@ -435,7 +434,7 @@ Pad Templates:
           You can do caps filtering by inserting a capsfilter element into
           your pipeline and setting its <quote>caps</quote> property. Caps
           filters are often placed after converter elements like audioconvert,
-          audioresample, ffmpegcolorspace or videoscale to force those
+          audioresample, videoconvert or videoscale to force those
           converters to convert data to a specific output format at a
           certain point in a stream.
         </para>
@@ -528,8 +527,8 @@ link_elements_with_filter (GstElement *element1, GstElement *element2)
   gboolean link_ok;
   GstCaps *caps;
 
-  caps = gst_caps_new_simple ("video/x-raw-yuv",
-             "format", GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'),
+  caps = gst_caps_new_simple ("video/x-raw",
+             "format", G_TYPE_STRING, "I420",
              "width", G_TYPE_INT, 384,
              "height", G_TYPE_INT, 288,
              "framerate", GST_TYPE_FRACTION, 25, 1,
@@ -569,12 +568,12 @@ link_elements_with_filter (GstElement *element1, GstElement *element2)
   GstCaps *caps;
                                                                                 
   caps = gst_caps_new_full (
-      gst_structure_new ("video/x-raw-yuv",
+      gst_structure_new ("video/x-raw",
                         "width", G_TYPE_INT, 384,
                         "height", G_TYPE_INT, 288,
                         "framerate", GST_TYPE_FRACTION, 25, 1,
                         NULL),
-      gst_structure_new ("video/x-raw-rgb",
+      gst_structure_new ("video/x-bayer",
                         "width", G_TYPE_INT, 384,
                         "height", G_TYPE_INT, 288,
                         "framerate", GST_TYPE_FRACTION, 25, 1,

diff --git a/docs/pwg/advanced-negotiation.xml b/docs/pwg/advanced-negotiation.xml
index 8a4e90f..1bae8b2 100644 (file)
--- a/docs/pwg/advanced-negotiation.xml
+++ b/docs/pwg/advanced-negotiation.xml
@@ -97,9 +97,8 @@
     </para>
     <programlisting>
 [..]
-  caps = gst_caps_new_simple ("audio/x-raw-float",
-      "width", G_TYPE_INT, 32,
-      "endianness", G_TYPE_INT, G_BYTE_ORDER,
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE(F32),
       "buffer-frames", G_TYPE_INT, &lt;bytes-per-frame&gt;,
       "rate", G_TYPE_INT, &lt;samplerate&gt;,
       "channels", G_TYPE_INT, &lt;num-channels&gt;, NULL);

diff --git a/docs/pwg/building-boiler.xml b/docs/pwg/building-boiler.xml
index 537723f..3476005 100644 (file)
--- a/docs/pwg/building-boiler.xml
+++ b/docs/pwg/building-boiler.xml
@@ -374,10 +374,8 @@ GST_STATIC_PAD_TEMPLATE (
   GST_PAD_SINK,
   GST_PAD_ALWAYS,
   GST_STATIC_CAPS (
-    "audio/x-raw-int, "
-      "width = (int) 16, "
-      "depth = (int) 16, "
-      "endianness = (int) BYTE_ORDER, "
+    "audio/x-raw, "
+      "format = (string) " GST_AUDIO_NE (S16) ", "
       "channels = (int) { 1, 2 }, "
       "rate = (int) [ 8000, 96000 ]"
   )

diff --git a/docs/pwg/building-pads.xml b/docs/pwg/building-pads.xml
index 81f9df8..06b3273 100644 (file)
--- a/docs/pwg/building-pads.xml
+++ b/docs/pwg/building-pads.xml
@@ -145,9 +145,9 @@ gst_my_filter_setcaps (GstPad  *pad,
    * and from that audio type, we need to get the samplerate and
    * number of channels. */
   mime = gst_structure_get_name (structure);
-  if (strcmp (mime, "audio/x-raw-int") != 0) {
+  if (strcmp (mime, "audio/x-raw") != 0) {
     GST_WARNING ("Wrong mimetype %s provided, we only support %s",
-                mime, "audio/x-raw-int");
+                mime, "audio/x-raw");
     return FALSE;
   }
 

diff --git a/docs/pwg/other-ntoone.xml b/docs/pwg/other-ntoone.xml
index 5cb1f35..2196b4e 100644 (file)
--- a/docs/pwg/other-ntoone.xml
+++ b/docs/pwg/other-ntoone.xml
@@ -80,7 +80,7 @@ gst_avi_mux_audio_link (GstPad        *pad,
      * stream structure (strh/strf). */
     [..]
     return GST_PAD_LINK_OK;
-  } else if !strcmp (str, "audio/x-raw-int")) {
+  } else if !strcmp (str, "audio/x-raw")) {
     /* See above, but now with the raw audio tag (0x0001). */
     [..]
     return GST_PAD_LINK_OK;
@@ -122,7 +122,7 @@ gst_avi_mux_loop (GstElement *element)
                        NULL);
             break;
           case 0x0001: /* pcm */
-            caps = gst_caps_new_simple ("audio/x-raw-int",
+            caps = gst_caps_new_simple ("audio/x-raw",
                                        [..]);
             break;
           [..]

diff --git a/libs/gst/base/gstbasetransform.c b/libs/gst/base/gstbasetransform.c
index 94396c3..21e7945 100644 (file)
--- a/libs/gst/base/gstbasetransform.c
+++ b/libs/gst/base/gstbasetransform.c
@@ -70,7 +70,7 @@
  *   <itemizedlist>
  *   <title>Example elements</title>
  *     <listitem>Level</listitem>
- *     <listitem>Videoscale, audioconvert, ffmpegcolorspace, audioresample in
+ *     <listitem>Videoscale, audioconvert, videoconvert, audioresample in
  *     certain modes.</listitem>
  *   </itemizedlist>
  * </listitem>
@@ -102,7 +102,7 @@
  *     <listitem>Volume</listitem>
  *     <listitem>Audioconvert in certain modes (signed/unsigned
  *     conversion)</listitem>
- *     <listitem>ffmpegcolorspace in certain modes (endianness
+ *     <listitem>videoconvert in certain modes (endianness
  *     swapping)</listitem>
  *   </itemizedlist>
  *  </listitem>
@@ -141,7 +141,7 @@
  *   </itemizedlist>
  *   <itemizedlist>
  *   <title>Example elements</title>
- *     <listitem>Videoscale, ffmpegcolorspace, audioconvert when doing
+ *     <listitem>Videoscale, videoconvert, audioconvert when doing
  *     scaling/conversions</listitem>
  *   </itemizedlist>
  * </listitem>

diff --git a/plugins/elements/gstcapsfilter.c b/plugins/elements/gstcapsfilter.c
index 1c585d9..5fa2ab3 100644 (file)
--- a/plugins/elements/gstcapsfilter.c
+++ b/plugins/elements/gstcapsfilter.c
@@ -28,7 +28,7 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch videotestsrc ! video/x-raw-gray ! ffmpegcolorspace ! autovideosink
+ * gst-launch videotestsrc ! video/x-raw,format=GRAY8 ! videoconvert ! autovideosink
  * ]| Limits acceptable video from videotestsrc to be grayscale.
  * </refsect2>
  */
@@ -273,8 +273,7 @@ gst_capsfilter_transform_ip (GstBaseTransform * base, GstBuffer * buf)
  * This ensures that outgoing buffers have caps if we can, so
  * that pipelines like:
  *   gst-launch filesrc location=rawsamples.raw !
- *       audio/x-raw-int,width=16,depth=16,rate=48000,channels=2,
- *       endianness=4321,signed='(boolean)'true ! alsasink
+ *       audio/x-raw,format=S16LE,rate=48000,channels=2 ! alsasink
  * will work.
  */
 static GstFlowReturn

diff --git a/plugins/elements/gsttee.c b/plugins/elements/gsttee.c
index 3c7f019..a00c1d1 100644 (file)
--- a/plugins/elements/gsttee.c
+++ b/plugins/elements/gsttee.c
@@ -37,7 +37,7 @@
  * <refsect2>
  * <title>Example launch line</title>
  * |[
- * gst-launch filesrc location=song.ogg ! decodebin2 ! tee name=t ! queue ! autoaudiosink t. ! queue ! audioconvert ! goom ! ffmpegcolorspace ! autovideosink
+ * gst-launch filesrc location=song.ogg ! decodebin2 ! tee name=t ! queue ! autoaudiosink t. ! queue ! audioconvert ! goom ! videoconvert ! autovideosink
  * ]| Play a song.ogg from local dir and render visualisations using the goom
  * element.
  * </refsect2>

diff --git a/tests/benchmarks/caps.c b/tests/benchmarks/caps.c
index 000e46f..32d1725 100644 (file)
--- a/tests/benchmarks/caps.c
+++ b/tests/benchmarks/caps.c
@@ -25,15 +25,20 @@
 
 #define NUM_CAPS 10000
 
+#define AUDIO_FORMATS_ALL " { S8, U8, " \
+    "S16LE, S16BE, U16LE, U16BE, " \
+    "S24_32LE, S24_32BE, U24_32LE, U24_32BE, " \
+    "S32LE, S32BE, U32LE, U32BE, " \
+    "S24LE, S24BE, U24LE, U24BE, " \
+    "S20LE, S20BE, U20LE, U20BE, " \
+    "S18LE, S18BE, U18LE, U18BE, " \
+    "F32LE, F32BE, F64LE, F64BE }"
 
 #define GST_AUDIO_INT_PAD_TEMPLATE_CAPS \
-  "audio/x-raw-int, " \
+  "audio/x-raw, " \
+  "format = (string) " AUDIO_FORMATS_ALL ", " \
   "rate = (int) [ 1, MAX ], " \
-  "channels = (int) [ 1, MAX ], " \
-  "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
-  "width = (int) { 8, 16, 24, 32 }, " \
-  "depth = (int) [ 1, 32 ], " \
-  "signed = (boolean) { true, false }"
+  "channels = (int) [ 1, MAX ]"
 
 
 gint

diff --git a/tests/benchmarks/capsnego.c b/tests/benchmarks/capsnego.c
index 3d05d14..7db7e91 100644 (file)
--- a/tests/benchmarks/capsnego.c
+++ b/tests/benchmarks/capsnego.c
@@ -50,12 +50,12 @@ enum
 
 static const gchar *factories[NUM_FLAVOURS][NUM_ELEM] = {
   {"audiotestsrc", "adder", "volume", "audioconvert"},
-  {"videotestsrc", "videomixer", "videoscale", "ffmpegcolorspace"}
+  {"videotestsrc", "videomixer", "videoscale", "videoconvert"}
 };
 
 static const gchar *sink_pads[NUM_FLAVOURS][NUM_ELEM] = {
-  {NULL, "sink%d", NULL, NULL},
-  {NULL, "sink_%d", NULL, NULL}
+  {NULL, "sink_%u", NULL, NULL},
+  {NULL, "sink_%u", NULL, NULL}
 };
 
 

diff --git a/tests/check/gst/gststructure.c b/tests/check/gst/gststructure.c
index 9150872..6427481 100644 (file)
--- a/tests/check/gst/gststructure.c
+++ b/tests/check/gst/gststructure.c
@@ -35,7 +35,7 @@ GST_START_TEST (test_from_string_int)
     "video/x-raw, red_mask = (int) 0x80000000",
     "video/x-raw, red_mask = (int) 0xFF000000",
     /* result from
-     * gst-launch ... ! "video/x-raw-rgb, red_mask=(int)0xFF000000" ! ... */
+     * gst-launch ... ! "video/x-raw, red_mask=(int)0xFF000000" ! ... */
     "video/x-raw,\\ red_mask=(int)0xFF000000",
   };
   gint results[] = {

diff --git a/tools/gst-launch.1.in b/tools/gst-launch.1.in
index 43c5c92..92a9811 100644 (file)
--- a/tools/gst-launch.1.in
+++ b/tools/gst-launch.1.in
@@ -212,7 +212,7 @@ Likewise, "xvimagesink" can be substituted with "ximagesink", "sdlvideosink",
 "osxvideosink", or "aasink". Keep in mind though that different sinks might
 accept different formats and even the same sink might accept different formats
 on different machines, so you might need to add converter elements like
-audioconvert and audioresample (for audio) or ffmpegcolorspace (for video)
+audioconvert and audioresample (for audio) or videoconvert (for video)
 in front of the sink to make things work.
 
 .B Audio playback
@@ -310,7 +310,7 @@ an SDL window
 Play both video and audio portions of an MPEG movie
 
 .B
-        gst\-launch filesrc location=movie.mpg ! mpegdemux name=demuxer  demuxer. ! queue ! mpeg2dec ! ffmpegcolorspace ! sdlvideosink   demuxer. ! queue ! mad ! audioconvert ! audioresample ! osssink
+        gst\-launch filesrc location=movie.mpg ! mpegdemux name=demuxer demuxer. ! queue ! mpeg2dec ! videoconvert ! sdlvideosink   demuxer. ! queue ! mad ! audioconvert ! audioresample ! osssink
 .br
 Play an AVI movie with an external text subtitle stream
 
@@ -318,7 +318,7 @@ This example also shows how to refer to specific pads by name if an element
 (here: textoverlay) has multiple sink or source pads.
 
 .B
-        gst\-launch textoverlay name=overlay ! ffmpegcolorspace ! videoscale ! autovideosink   filesrc location=movie.avi ! decodebin2 ! ffmpegcolorspace ! overlay.video_sink   filesrc location=movie.srt ! subparse ! overlay.text_sink
+        gst\-launch textoverlay name=overlay ! videoconvert ! videoscale !  autovideosink   filesrc location=movie.avi ! decodebin2 ! videoconvert ! overlay.video_sink   filesrc location=movie.srt ! subparse ! overlay.text_sink
 
 .br
 Play an AVI movie with an external text subtitle stream using playbin2
@@ -331,7 +331,7 @@ Play an AVI movie with an external text subtitle stream using playbin2
 Stream video using RTP and network elements.
 
 .B
-        gst\-launch v4l2src ! video/x-raw-yuv,width=128,height=96,format='(fourcc)'UYVY ! ffmpegcolorspace ! ffenc_h263 ! video/x-h263 ! rtph263ppay pt=96 ! udpsink host=192.168.1.1 port=5000 sync=false
+        gst\-launch v4l2src ! video/x-raw,width=128,height=96,format=UYVY ! videoconvert ! ffenc_h263 ! video/x-h263 ! rtph263ppay pt=96 ! udpsink host=192.168.1.1 port=5000
 .br
 This command would be run on the transmitter
 
@@ -371,7 +371,7 @@ to get a working pipeline.
 Play any supported audio format
 
 .B
-        gst\-launch filesrc location=videofile ! decodebin name=decoder  decoder. ! queue ! audioconvert ! audioresample ! osssink   decoder. ! ffmpegcolorspace ! xvimagesink
+        gst\-launch filesrc location=videofile ! decodebin name=decoder decoder. ! queue ! audioconvert ! audioresample ! osssink   decoder. !  videoconvert ! xvimagesink
 .br
 Play any supported video format with video and audio output. Threads are used
 automatically. To make this even easier, you can use the playbin element:
@@ -386,12 +386,12 @@ automatically. To make this even easier, you can use the playbin element:
 These examples show you how to use filtered caps.
 
 .B
-        gst\-launch videotestsrc ! 'video/x-raw-yuv,format=(fourcc)YUY2;video/x-raw-yuv,format=(fourcc)YV12' ! xvimagesink
+        gst\-launch videotestsrc ! 'video/x-raw,format=YUY2;video/x-raw,format=YV12' ! xvimagesink
 .br
 Show a test image and use the YUY2 or YV12 video format for this.
 
 .B
-        gst\-launch osssrc ! 'audio/x-raw-int,rate=[32000,64000],width=[16,32],depth={16,24,32},signed=(boolean)true' ! wavenc ! filesink location=recording.wav
+        gst\-launch osssrc !  'audio/x-raw,rate=[32000,64000],format={S16LE,S24LE,S32LE}' ! wavenc ! filesink location=recording.wav
 .br
 record audio and write it to a .wav file. Force usage of signed 16 to 32 bit
 samples and a sample rate between 32kHz and 64KHz.