diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index 7815a15fa3..82d27d9e7d 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -78,6 +78,7 @@ EXTRA_HFILES = \ $(top_srcdir)/ext/aalib/gstaasink.h \ $(top_srcdir)/ext/annodex/gstcmmldec.h \ $(top_srcdir)/ext/annodex/gstcmmlenc.h \ + $(top_srcdir)/ext/cairo/gsttextoverlay.h \ $(top_srcdir)/ext/cairo/gsttimeoverlay.h \ $(top_srcdir)/ext/dv/gstdvdec.h \ $(top_srcdir)/ext/dv/gstdvdemux.h \ @@ -85,6 +86,10 @@ EXTRA_HFILES = \ $(top_srcdir)/ext/flac/gstflacdec.h \ $(top_srcdir)/ext/flac/gstflacenc.h \ $(top_srcdir)/ext/flac/gstflactag.h \ + $(top_srcdir)/ext/gconf/gstgconfaudiosrc.h \ + $(top_srcdir)/ext/gconf/gstgconfaudiosink.h \ + $(top_srcdir)/ext/gconf/gstgconfvideosrc.h \ + $(top_srcdir)/ext/gconf/gstgconfvideosink.h \ $(top_srcdir)/ext/gdk_pixbuf/gstgdkpixbufsink.h \ $(top_srcdir)/ext/hal/gsthalaudiosink.h \ $(top_srcdir)/ext/hal/gsthalaudiosrc.h \ @@ -103,6 +108,8 @@ EXTRA_HFILES = \ $(top_srcdir)/ext/pulse/pulsesink.h \ $(top_srcdir)/ext/pulse/pulsesrc.h \ $(top_srcdir)/ext/pulse/pulsemixer.h \ + $(top_srcdir)/ext/speex/gstspeexenc.h \ + $(top_srcdir)/ext/speex/gstspeexdec.h \ $(top_srcdir)/ext/wavpack/gstwavpackdec.h \ $(top_srcdir)/ext/wavpack/gstwavpackenc.h \ $(top_srcdir)/ext/wavpack/gstwavpackparse.h \ diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml index 010acb2ff9..3c96461c0e 100644 --- a/docs/plugins/gst-plugins-good-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml @@ -40,6 +40,7 @@ + @@ -48,7 +49,6 @@ - @@ -59,11 +59,16 @@ + + + + + @@ -106,6 +111,8 @@ + + diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index 6b9e1b9a47..0c81487186 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -103,7 +103,6 @@ gst_audio_amplify_get_type GstAudioChebBand GstAudioChebBandClass -GstAudioChebBandProcessFunc GST_AUDIO_CHEB_BAND GST_AUDIO_CHEB_BAND_CLASS GST_AUDIO_CHEB_BAND_GET_CLASS @@ -119,7 +118,6 @@ gst_audio_cheb_band_get_type GstAudioChebLimit GstAudioChebLimitClass -GstAudioChebLimitProcessFunc GST_AUDIO_CHEB_LIMIT GST_AUDIO_CHEB_LIMIT_CLASS GST_AUDIO_CHEB_LIMIT_GET_CLASS @@ -243,7 +241,6 @@ gst_audio_panorama_get_type GstAudioWSincBand GstAudioWSincBandClass -GstAudioWSincBandProcessFunc GST_AUDIO_WSINC_BAND GST_AUDIO_WSINC_BAND_CLASS GST_IS_AUDIO_WSINC_BAND @@ -258,7 +255,6 @@ gst_audio_wsincband_get_type GstAudioWSincLimit GstAudioWSincLimitClass -GstAudioWSincLimitProcessFunc GST_AUDIO_WSINC_LIMIT GST_AUDIO_WSINC_LIMIT_CLASS GST_IS_AUDIO_WSINC_LIMIT @@ -396,6 +392,20 @@ GST_IS_CACASINK_CLASS gst_cacasink_get_type +
+element-cairotextoverlay +cairotextoverlay +GstCairoTextOverlay + +GstCairoTextOverlayClass +GST_TYPE_CAIRO_TEXT_OVERLAY +GST_CAIRO_TEXT_OVERLAY +GST_CAIRO_TEXT_OVERLAY_CLASS +GST_IS_CAIRO_TEXT_OVERLAY +GST_IS_CAIRO_TEXT_OVERLAY_CLASS +gst_text_overlay_get_type +
+
element-cairotimeoverlay cairotimeoverlay @@ -447,7 +457,6 @@ GstCutter GST_CUTTER GST_CUTTER_CLASS -GST_CUTTER_GET_CLASS GST_IS_CUTTER GST_IS_CUTTER_CLASS GST_TYPE_CUTTER @@ -697,6 +706,62 @@ GST_IS_GAMMA_CLASS gst_gamma_get_type
+
+element-gconfaudiosrc +gconfaudiosrc +GstGConfAudioSrc + +GstGConfAudioSrcClass +GST_GCONF_AUDIO_SRC +GST_IS_GCONF_AUDIO_SRC +GST_TYPE_GCONF_AUDIO_SRC +GST_GCONF_AUDIO_SRC_CLASS +GST_IS_GCONF_AUDIO_SRC_CLASS +gst_gconf_audio_src_get_type +
+ +
+element-gconfaudiosink +gconfaudiosink +GstGConfAudioSink + +GstGConfAudioSinkClass +GST_GCONF_AUDIO_SINK +GST_IS_GCONF_AUDIO_SINK +GST_TYPE_GCONF_AUDIO_SINK +GST_GCONF_AUDIO_SINK_CLASS +GST_IS_GCONF_AUDIO_SINK_CLASS +gst_gconf_audio_sink_get_type +
+ +
+element-gconfvideosrc +gconfvideosrc +GstGConfVideoSrc + +GstGConfVideoSrcClass +GST_GCONF_VIDEO_SRC +GST_IS_GCONF_VIDEO_SRC +GST_TYPE_GCONF_VIDEO_SRC +GST_GCONF_VIDEO_SRC_CLASS +GST_IS_GCONF_VIDEO_SRC_CLASS +gst_gconf_video_src_get_type +
+ +
+element-gconfvideosink +gconfvideosink +GstGConfVideoSink + +GstGConfVideoSinkClass +GST_GCONF_VIDEO_SINK +GST_IS_GCONF_VIDEO_SINK +GST_TYPE_GCONF_VIDEO_SINK +GST_GCONF_VIDEO_SINK_CLASS +GST_IS_GCONF_VIDEO_SINK_CLASS +gst_gconf_video_sink_get_type +
+
element-gdkpixbufsink gdkpixbufsink @@ -881,7 +946,6 @@ GST_MATROSKA_MUX GST_MATROSKA_MUX_CLASS GST_IS_MATROSKA_MUX GST_IS_MATROSKA_MUX_CLASS -gst_matroska_mux_get_type gst_matroska_mux_plugin_init
@@ -897,7 +961,6 @@ GST_MATROSKA_DEMUX GST_MATROSKA_DEMUX_CLASS GST_IS_MATROSKA_DEMUX GST_IS_MATROSKA_DEMUX_CLASS -gst_matroska_demux_get_type gst_matroska_demux_plugin_init @@ -1271,7 +1334,7 @@ GST_IS_RTP_JPEG_PAY GST_TYPE_RTP_JPEG_PAY GST_RTP_JPEG_PAY_CLASS GST_IS_RTP_JPEG_PAY_CLASS -gst_rtp_jpeg_pay_get_type +gst_rtp_jpeg_pay_plugin_init
@@ -1375,8 +1438,6 @@ GST_IS_SMOKEENC_CLASS GstSpectrum GstSpectrumClass -GstSpectrumFFTFreeFunc -GstSpectrumProcessFunc GST_SPECTRUM GST_SPECTRUM_CLASS GST_IS_SPECTRUM @@ -1385,6 +1446,34 @@ GST_TYPE_SPECTRUM gst_spectrum_get_type
+
+element-speexdec +speexdec +GstSpeexDec + +GstSpeexDecClass +GST_TYPE_SPEEX_DEC +GST_SPEEX_DEC +GST_SPEEX_DEC_CLASS +GST_IS_SPEEX_DEC +GST_IS_SPEEX_DEC_CLASS +gst_speex_dec_get_type +
+ +
+element-speexenc +speexenc +GstSpeexEnc + +GstSpeexEncClass +GST_TYPE_SPEEX_ENC +GST_SPEEX_ENC +GST_SPEEX_ENC_CLASS +GST_IS_SPEEX_ENC +GST_IS_SPEEX_ENC_CLASS +gst_speex_enc_get_type +
+
element-taginject taginject @@ -1446,12 +1535,6 @@ gst_udpsink_get_type videobox GstVideoBox -GstVideoBoxClass -GST_IS_VIDEO_BOX -GST_IS_VIDEO_BOX_CLASS -GST_VIDEO_BOX -GST_VIDEO_BOX_CLASS -GST_TYPE_VIDEO_BOX
@@ -1474,9 +1557,7 @@ GST_TYPE_VIDEO_CROP aspectratiocrop GstAspectRatioCrop -GstAspectRatioCropClass -AspectRatioCropPixelFormat -GstAspectRatioCropImageDetails +GstAspectRatioCropClass GST_IS_ASPECT_RATIO_CROP GST_IS_ASPECT_RATIO_CROP_CLASS GST_ASPECT_RATIO_CROP @@ -1552,7 +1633,6 @@ GST_V4L2_MIN_BUFFERS GST_V4L2_MAX_SIZE GstV4l2BufferPool GstV4l2Buffer -GstV4l2Src GstV4l2SrcClass GST_V4L2SRC GST_IS_V4L2SRC @@ -1594,7 +1674,6 @@ GST_IS_WAVENC GST_TYPE_WAVENC GST_WAVENC_CLASS GST_IS_WAVENC_CLASS -gst_wavenc_get_type
diff --git a/ext/aalib/gstaasink.c b/ext/aalib/gstaasink.c index 8ecc90b7a0..8e07568fec 100644 --- a/ext/aalib/gstaasink.c +++ b/ext/aalib/gstaasink.c @@ -20,23 +20,16 @@ * SECTION:element-aasink * @see_also: #GstCACASink * - * - * * Displays video as b/w ascii art. - * + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! aasink - * - * This pipeline renders a video to ascii art into a separate window. - * - * - * + * ]| This pipeline renders a video to ascii art into a separate window. + * |[ * gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! aasink driver=curses - * - * This pipeline renders a video to ascii art into the current terminal. - * + * ]| This pipeline renders a video to ascii art into the current terminal. * */ diff --git a/ext/annodex/gstcmmldec.c b/ext/annodex/gstcmmldec.c index 75038c61c7..765a0a12a5 100644 --- a/ext/annodex/gstcmmldec.c +++ b/ext/annodex/gstcmmldec.c @@ -25,17 +25,16 @@ * SECTION:element-cmmldec * @see_also: cmmlenc, oggdemux * - * - * * Cmmldec extracts a CMML document from a CMML bitstream.CMML is * an XML markup language for time-continuous data maintained by the Annodex Foundation. - * + * + * * Example pipeline - * + * |[ * gst-launch -v filesrc location=annotated.ogg ! oggdemux ! cmmldec ! filesink location=annotations.cmml - * + * ]| * */ diff --git a/ext/annodex/gstcmmlenc.c b/ext/annodex/gstcmmlenc.c index 9b2c03c6ca..f3b65e557c 100644 --- a/ext/annodex/gstcmmlenc.c +++ b/ext/annodex/gstcmmlenc.c @@ -25,16 +25,16 @@ * SECTION:element-cmmlenc * @see_also: cmmldec, oggmux * - * - * Cmmlenc encodes a CMML document into a CMML stream. CMML is * an XML markup language for time-continuous data maintained by the Annodex Foundation. - * + * + * * Example pipeline - * + * |[ * gst-launch -v filesrc location=annotations.cmml ! cmmlenc ! oggmux name=mux ! filesink location=annotated.ogg - * + * ]| * */ diff --git a/ext/cairo/gsttextoverlay.c b/ext/cairo/gsttextoverlay.c index 6ac993f1d9..6bdd6fd9ee 100644 --- a/ext/cairo/gsttextoverlay.c +++ b/ext/cairo/gsttextoverlay.c @@ -17,6 +17,18 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-cairotextoverlay + * + * cairotextoverlay renders the text on top of the video frames. + * + * + * Example launch line + * |[ + * gst-launch videotestsrc ! cairotextoverlay text="hello" ! autovideosink + * ]| + * + */ #ifdef HAVE_CONFIG_H #include diff --git a/ext/cairo/gsttimeoverlay.c b/ext/cairo/gsttimeoverlay.c index 8f1d03d16d..5a2f016f4b 100644 --- a/ext/cairo/gsttimeoverlay.c +++ b/ext/cairo/gsttimeoverlay.c @@ -21,17 +21,14 @@ /** * SECTION:element-cairotimeoverlay * - * - * * cairotimeoverlay renders the buffer timestamp for each frame on top of * the frame. - * + * + * * Example launch line - * - * - * gst-launch -v -m videotestsrc ! cairotimeoverlay ! autovideosink - * - * + * |[ + * gst-launch videotestsrc ! cairotimeoverlay ! autovideosink + * ]| * */ diff --git a/ext/dv/gstdvdec.c b/ext/dv/gstdvdec.c index 958aa51852..ade979665f 100644 --- a/ext/dv/gstdvdec.c +++ b/ext/dv/gstdvdec.c @@ -21,25 +21,21 @@ /** * SECTION:element-dvdec * - * - * * dvdec decodes DV video into raw video. The element expects a full DV frame * as input, which is 120000 bytes for NTSC and 144000 for PAL video. - * - * - * This element can perform simple frame dropping with the drop-factor + * + * This element can perform simple frame dropping with the #GstDVDec:drop-factor * property. Setting this property to a value N > 1 will only decode every * Nth frame. - * + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=test.dv ! dvdemux name=demux ! dvdec ! xvimagesink - * - * This pipeline decodes and renders the raw DV stream to a videosink. - * - * Last reviewed on 2006-02-28 (0.10.3) + * ]| This pipeline decodes and renders the raw DV stream to a videosink. * + * + * Last reviewed on 2006-02-28 (0.10.3) */ #ifdef HAVE_CONFIG_H diff --git a/ext/dv/gstdvdemux.c b/ext/dv/gstdvdemux.c index 3934011643..7c7509feb8 100644 --- a/ext/dv/gstdvdemux.c +++ b/ext/dv/gstdvdemux.c @@ -30,24 +30,20 @@ /** * SECTION:element-dvdemux * - * - * * dvdemux splits raw DV into its audio and video components. The audio will be * decoded raw samples and the video will be encoded DV video. - * - * - * This element can operate in both push and pull mode depending on the capabilities - * of the upstream peer. - * + * + * This element can operate in both push and pull mode depending on the + * capabilities of the upstream peer. + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=test.dv ! dvdemux name=demux ! queue ! audioconvert ! alsasink demux. ! queue ! dvdec ! xvimagesink - * - * This pipeline decodes and renders the raw DV stream to an audio and a videosink. - * - * Last reviewed on 2006-02-27 (0.10.3) + * ]| This pipeline decodes and renders the raw DV stream to an audio and a videosink. * + * + * Last reviewed on 2006-02-27 (0.10.3) */ /* DV output has two modes, normal and wide. The resolution is the same in both diff --git a/ext/esd/esdmon.c b/ext/esd/esdmon.c index 880af9d877..25b1201ea3 100644 --- a/ext/esd/esdmon.c +++ b/ext/esd/esdmon.c @@ -20,9 +20,10 @@ * Boston, MA 02111-1307, USA. */ /** - * SECTION:element-esdmod + * SECTION:element-esdmon + * @see_also: #GstAlsaSrc, #GstAutoAudioSrc * - * This element outputs sound to an already-running Enlightened Sound Daemon + * This element records sound from an already-running Enlightened Sound Daemon * (ESound Daemon, esd). Note that a sound daemon will never be auto-spawned * through this element (regardless of the system configuration), since this * is actively prevented by the element. If you must use esd, you need to @@ -38,6 +39,7 @@ * ]| Record from audioinput * */ + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/ext/esd/esdsink.c b/ext/esd/esdsink.c index ed34d793ce..ddc84e6a9d 100644 --- a/ext/esd/esdsink.c +++ b/ext/esd/esdsink.c @@ -26,24 +26,20 @@ * SECTION:element-esdsink * @see_also: #GstAlsaSink, #GstAutoAudioSink * - * - * * This element outputs sound to an already-running Enlightened Sound Daemon * (ESound Daemon, esd). Note that a sound daemon will never be auto-spawned * through this element (regardless of the system configuration), since this * is actively prevented by the element. If you must use esd, you need to * make sure it is started automatically with your session or otherwise. - * - * + * * TODO: insert some comments about how sucky esd is and that all the cool * kids use pulseaudio or whatever these days. - * - * - * Simple example pipeline that plays an Ogg/Vorbis file via esd: - * + * + * + * Example launch line + * |[ * gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! audioresample ! esdsink - * - * + * ]| play an Ogg/Vorbis audio file via esd * */ diff --git a/ext/flac/gstflacdec.c b/ext/flac/gstflacdec.c index 8244e79a93..dedbd6176f 100644 --- a/ext/flac/gstflacdec.c +++ b/ext/flac/gstflacdec.c @@ -21,26 +21,20 @@ /** * SECTION:element-flacdec - * @seealso: flacenc + * @see_also: #GstFlacEnc * - * - * * flacdec decodes FLAC streams. * FLAC * is a Free Lossless Audio Codec. - * + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=media/small/dark.441-16-s.flac ! flacdec ! audioconvert ! audioresample ! autoaudiosink - * - * - * Another example launch line - * - * + * ]| + * |[ * gst-launch gnomevfssrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacdec ! audioconvert ! audioresample ! queue min-threshold-buffers=10 ! autoaudiosink - * - * + * ]| * */ diff --git a/ext/flac/gstflacenc.c b/ext/flac/gstflacenc.c index e5a7804bd9..3fade91096 100644 --- a/ext/flac/gstflacenc.c +++ b/ext/flac/gstflacenc.c @@ -16,6 +16,21 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-flacenc + * @see_also: #GstFlacDec + * + * flacenc encodes FLAC streams. + * FLAC + * is a Free Lossless Audio Codec. + * + * + * Example launch line + * |[ + * gst-launch audiotestsrc num-buffers=100 ! flacenc ! filesink location=beep.flac + * ]| + * + */ /* TODO: - We currently don't handle discontinuities in the stream in a useful * way and instead rely on the developer plugging in audiorate if diff --git a/ext/gconf/gstgconfaudiosink.c b/ext/gconf/gstgconfaudiosink.c index a8bf330566..19dee86c71 100644 --- a/ext/gconf/gstgconfaudiosink.c +++ b/ext/gconf/gstgconfaudiosink.c @@ -17,6 +17,19 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-gconfaudiosink + * + * This element outputs sound to the audiosink that has been configured in + * GConf by the user. + * + * + * Example launch line + * |[ + * gst-launch filesrc location=foo.ogg ! decodebin ! audioconvert ! audioresample ! gconfaudiosink + * ]| Play on configured audiosink + * + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/gconf/gstgconfaudiosrc.c b/ext/gconf/gstgconfaudiosrc.c index babd653d7d..77c3730563 100644 --- a/ext/gconf/gstgconfaudiosrc.c +++ b/ext/gconf/gstgconfaudiosrc.c @@ -17,6 +17,20 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-gconfaudiosrc + * @see_also: #GstAlsaSrc, #GstAutoAudioSrc + * + * This element records sound from the audiosink that has been configured in + * GConf by the user. + * + * + * Example launch line + * |[ + * gst-launch gconfaudiosrc ! audioconvert ! wavenc ! filesink location=record.wav + * ]| Record from configured audioinput + * + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/gconf/gstgconfvideosink.c b/ext/gconf/gstgconfvideosink.c index ec04cd86ba..4090cc9114 100644 --- a/ext/gconf/gstgconfvideosink.c +++ b/ext/gconf/gstgconfvideosink.c @@ -16,6 +16,19 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-gconfvideosink + * + * This element outputs video to the videosink that has been configured in + * GConf by the user. + * + * + * Example launch line + * |[ + * gst-launch filesrc location=foo.ogg ! decodebin ! ffmpegcolorspace ! gconfvideosink + * ]| Play on configured videosink + * + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/gconf/gstgconfvideosrc.c b/ext/gconf/gstgconfvideosrc.c index f56f43a259..fe177d8639 100644 --- a/ext/gconf/gstgconfvideosrc.c +++ b/ext/gconf/gstgconfvideosrc.c @@ -17,6 +17,20 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-gconfvideosrc + * @see_also: #GstAlsaSrc, #GstAutoVideoSrc + * + * This element records video from the videosink that has been configured in + * GConf by the user. + * + * + * Example launch line + * |[ + * gst-launch gconfvideosrc ! theoraenc ! oggmux ! filesink location=record.ogg + * ]| Record from configured videoinput + * + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/gdk_pixbuf/gstgdkpixbufsink.c b/ext/gdk_pixbuf/gstgdkpixbufsink.c index f4d318a17a..3c988f92d2 100644 --- a/ext/gdk_pixbuf/gstgdkpixbufsink.c +++ b/ext/gdk_pixbuf/gstgdkpixbufsink.c @@ -19,20 +19,14 @@ /** * SECTION:element-gdkpixbufsink - * @short_description: video sink that converts RGB images to GdkPixbufs. - * @see_also: + * @Since: 0.10.8 * - * - * * This sink element takes RGB or RGBA images as input and wraps them into - * GdkPixbuf objects, for easy saving to file via the - * - * GdkPixbuf library API or displaying in Gtk+ applications (e.g. using - * the - * GtkImage widget). - * - * - * There are two ways to use this element and obtain the GdkPixbuf objects + * #GdkPixbuf objects, for easy saving to file via the + * GdkPixbuf library API or displaying in Gtk+ applications (e.g. using + * the #GtkImage widget). + * + * There are two ways to use this element and obtain the #GdkPixbuf objects * created: * * @@ -43,34 +37,30 @@ * contained in them. * * - * Retrieving the current pixbuf via the "last-pixbuf" - * property when needed. + * Retrieving the current pixbuf via the #GstGdkPixbufSink:last-pixbuf property + * when needed. * * - * - * - * The primary purpose of this element is to abstract away the GstBuffer to - * GdkPixbuf conversion. Other than that it's very similar to the fakesink + * + * The primary purpose of this element is to abstract away the #GstBuffer to + * #GdkPixbuf conversion. Other than that it's very similar to the fakesink * element. - * - * + * * This element is meant for easy no-hassle video snapshotting. It is not * suitable for video playback or video display at high framerates. Use * ximagesink, xvimagesink or some other suitable video sink in connection - * with the GstXOverlay interface instead if you want to do video playback. - * + * with the #GstXOverlay interface instead if you want to do video playback. + * + * * Message details - * * As mentioned above, this element will by default post element messages * containing structures named "preroll-pixbuf" * or "pixbuf" on the bus (this - * can be disabled by setting the - * "send-messages" - * property to #FALSE though). The element message's structure - * will have the following fields: + * can be disabled by setting the #GstGdkPixbufSink:send-messages property + * to #FALSE though). The element message structure has the following fields: * * - * "pixbuf": the GdkPixbuf object + * "pixbuf": the #GdkPixbuf object * * * "pixel-aspect-ratio": the pixel aspect @@ -78,30 +68,25 @@ * PAR is usually 1:1 for images, but is often something non-1:1 in the case * of video input. In this case the image may be distorted and you may need * to rescale it accordingly before saving it to file or displaying it. This - * can easily be done using the - * - * GdkPixbuf library API (the reason this is not done automatically - * is that the application will often scale the image anyway according to the - * size of the output window, in which case it is much more efficient to only - * scale once rather than twice). You can put a videoscale element and a - * capsfilter element with + * can easily be done using gdk_pixbuf_scale() (the reason this is not done + * automatically is that the application will often scale the image anyway + * according to the size of the output window, in which case it is much more + * efficient to only scale once rather than twice). You can put a videoscale + * element and a capsfilter element with * video/x-raw-rgb,pixel-aspect-ratio=(fraction)1/1 caps * in front of this element to make sure the pixbufs always have a 1:1 PAR. * * - * - * Example pipeline - * - * - * gst-launch -m -v videotestsrc num-buffers=1 ! gdkpixbufsink - * - * Process one single test image as pixbuf (note that the output you see will - * be slightly misleading. The message structure does contain a valid pixbuf - * object even if the structure string says '(NULL)'). - * * * - * Since: 0.10.8 + * + * Example pipeline + * |[ + * gst-launch -m -v videotestsrc num-buffers=1 ! gdkpixbufsink + * ]| Process one single test image as pixbuf (note that the output you see will + * be slightly misleading. The message structure does contain a valid pixbuf + * object even if the structure string says '(NULL)'). + * */ #ifdef HAVE_CONFIG_H diff --git a/ext/hal/gsthalaudiosink.c b/ext/hal/gsthalaudiosink.c index de4d8844c6..e010ea4e00 100644 --- a/ext/hal/gsthalaudiosink.c +++ b/ext/hal/gsthalaudiosink.c @@ -21,27 +21,21 @@ /** * SECTION:element-halaudiosink * - * - * * HalAudioSink allows access to output of sound devices by specifying the * corresponding persistent Unique Device Id (UDI) from the Hardware Abstraction - * Layer (HAL) in the udi property. + * Layer (HAL) in the #GstHalAudioSink:udi property. * It currently always embeds alsasink or osssink as HAL doesn't support other * sound systems yet. You can also specify the UDI of a device that has ALSA or * OSS subdevices. If both are present ALSA is preferred. - * + * + * * Examples - * - * To list the UDIs of all your ALSA output devices : - * + * |[ * hal-find-by-property --key alsa.type --string playback - * - * Here is a pipeline to test your sound output : - * + * ]| list the UDIs of all your ALSA output devices + * |[ * gst-launch -v audiotestsrc ! halaudiosink udi=/org/freedesktop/Hal/devices/pci_8086_27d8_alsa_playback_0 - * - * This pipeline produces a test signal on the specified sound device. - * + * ]| test your soundcard by playing a test signal on the specified sound device. * */ diff --git a/ext/hal/gsthalaudiosrc.c b/ext/hal/gsthalaudiosrc.c index 626e9d354c..bd0a70b5eb 100644 --- a/ext/hal/gsthalaudiosrc.c +++ b/ext/hal/gsthalaudiosrc.c @@ -22,28 +22,22 @@ /** * SECTION:element-halaudiosrc * - * - * * HalAudioSrc allows access to input of sound devices by specifying the * corresponding persistent Unique Device Id (UDI) from the Hardware Abstraction - * Layer (HAL) in the udi property. + * Layer (HAL) in the #GstHalAudioSrc:udi property. * It currently always embeds alsasrc or osssrc as HAL doesn't support other * sound systems yet. You can also specify the UDI of a device that has ALSA or * OSS subdevices. If both are present ALSA is preferred. - * + * + * * Examples - * - * To list the UDIs of all your ALSA input devices : - * + * |[ * hal-find-by-property --key alsa.type --string capture - * - * Here is a pipeline to test your sound input : - * + * ]| list the UDIs of all your ALSA input devices + * |[ * gst-launch -v halaudiosrc udi=/org/freedesktop/Hal/devices/pci_8086_27d8_alsa_capture_0 ! autoaudiosink - * - * You should now hear yourself with a small delay if you have a microphone + * ]| You should now hear yourself with a small delay if you have a microphone * connected to the specified sound device. - * * */ diff --git a/ext/hal/hal.c b/ext/hal/hal.c index d23a91a677..b0e1d823aa 100644 --- a/ext/hal/hal.c +++ b/ext/hal/hal.c @@ -40,7 +40,7 @@ GST_DEBUG_CATEGORY_EXTERN (hal_debug); #define LIBHAL_FREE_DBUS_ERROR(e) dbus_error_free (e) #endif -/** +/* * gst_hal_get_alsa_element: * @ctx: a #LibHalContext which should be used for querying HAL. * @udi: a #gchar corresponding to the UDI you want to get. @@ -130,7 +130,7 @@ gst_hal_get_alsa_element (LibHalContext * ctx, const gchar * udi, return string; } -/** +/* * gst_hal_get_oss_element: * @ctx: a #LibHalContext which should be used for querying HAL. * @udi: a #gchar corresponding to the UDI you want to get. @@ -203,7 +203,7 @@ gst_hal_get_oss_element (LibHalContext * ctx, const gchar * udi, return string; } -/** +/* * gst_hal_get_string: * @udi: a #gchar corresponding to the UDI you want to get. * @device_type: a #GstHalDeviceType specifying the wanted device type. diff --git a/ext/jpeg/gstjpegenc.c b/ext/jpeg/gstjpegenc.c index 1161c07562..b1aae3a5cc 100644 --- a/ext/jpeg/gstjpegenc.c +++ b/ext/jpeg/gstjpegenc.c @@ -16,7 +16,11 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ - +/** + * SECTION:element-jpegenc + * + * Encodes jpeg images. + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/jpeg/gstsmokedec.c b/ext/jpeg/gstsmokedec.c index 665de92d34..b66a230827 100644 --- a/ext/jpeg/gstsmokedec.c +++ b/ext/jpeg/gstsmokedec.c @@ -17,6 +17,11 @@ * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-smokedec + * + * Decodes images in smoke format. + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/jpeg/gstsmokeenc.c b/ext/jpeg/gstsmokeenc.c index bb3d89e67a..31f7ed28da 100644 --- a/ext/jpeg/gstsmokeenc.c +++ b/ext/jpeg/gstsmokeenc.c @@ -16,7 +16,11 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ - +/** + * SECTION:element-smokeenc + * + * Encodes images in smoke format. + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/libcaca/gstcacasink.c b/ext/libcaca/gstcacasink.c index 1f6f4c20b8..c0a6f49f52 100644 --- a/ext/libcaca/gstcacasink.c +++ b/ext/libcaca/gstcacasink.c @@ -20,24 +20,17 @@ * SECTION:element-cacasink * @see_also: #GstAASink * - * - * * Displays video as color ascii art. - * + * + * * Example launch line - * - * + * |[ * CACA_GEOMETRY=160x60 CACA_FONT=5x7 gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! cacasink - * - * This pipeline renders a video to ascii art into a separate window using a + * ]| This pipeline renders a video to ascii art into a separate window using a * small font and specifying the ascii resolution. - * - * - * + * |[ * CACA_DRIVER=ncurses gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! cacasink - * - * This pipeline renders a video to ascii art into the current terminal. - * + * ]| This pipeline renders a video to ascii art into the current terminal. * */ @@ -50,6 +43,17 @@ #include "gstcacasink.h" +#define GST_CACA_DEFAULT_SCREEN_WIDTH 80 +#define GST_CACA_DEFAULT_SCREEN_HEIGHT 25 +#define GST_CACA_DEFAULT_BPP 24 +#define GST_CACA_DEFAULT_RED_MASK GST_VIDEO_BYTE1_MASK_32_INT +#define GST_CACA_DEFAULT_GREEN_MASK GST_VIDEO_BYTE2_MASK_32_INT +#define GST_CACA_DEFAULT_BLUE_MASK GST_VIDEO_BYTE3_MASK_32_INT + +//#define GST_CACA_DEFAULT_RED_MASK R_MASK_32_REVERSE_INT +//#define GST_CACA_DEFAULT_GREEN_MASK G_MASK_32_REVERSE_INT +//#define GST_CACA_DEFAULT_BLUE_MASK B_MASK_32_REVERSE_INT + /* elementfactory information */ static const GstElementDetails gst_cacasink_details = GST_ELEMENT_DETAILS ("A colored ASCII art video sink", diff --git a/ext/libcaca/gstcacasink.h b/ext/libcaca/gstcacasink.h index f955f63d95..548ca5dacb 100644 --- a/ext/libcaca/gstcacasink.h +++ b/ext/libcaca/gstcacasink.h @@ -34,17 +34,6 @@ extern "C" { #endif /* __cplusplus */ -#define GST_CACA_DEFAULT_SCREEN_WIDTH 80 -#define GST_CACA_DEFAULT_SCREEN_HEIGHT 25 -#define GST_CACA_DEFAULT_BPP 24 -#define GST_CACA_DEFAULT_RED_MASK GST_VIDEO_BYTE1_MASK_32_INT -#define GST_CACA_DEFAULT_GREEN_MASK GST_VIDEO_BYTE2_MASK_32_INT -#define GST_CACA_DEFAULT_BLUE_MASK GST_VIDEO_BYTE3_MASK_32_INT - -//#define GST_CACA_DEFAULT_RED_MASK R_MASK_32_REVERSE_INT -//#define GST_CACA_DEFAULT_GREEN_MASK G_MASK_32_REVERSE_INT -//#define GST_CACA_DEFAULT_BLUE_MASK B_MASK_32_REVERSE_INT - #define GST_TYPE_CACASINK \ (gst_cacasink_get_type()) #define GST_CACASINK(obj) \ diff --git a/ext/libpng/gstpngdec.c b/ext/libpng/gstpngdec.c index 8932e88bb7..d0f9d0d58c 100644 --- a/ext/libpng/gstpngdec.c +++ b/ext/libpng/gstpngdec.c @@ -12,7 +12,6 @@ * Boston, MA 02111-1307, USA. * */ - /** * SECTION:element-pngdec * diff --git a/ext/libpng/gstpngenc.c b/ext/libpng/gstpngenc.c index 160958c183..1d1de33d4d 100644 --- a/ext/libpng/gstpngenc.c +++ b/ext/libpng/gstpngenc.c @@ -15,6 +15,11 @@ * Boston, MA 02111-1307, USA. * */ +/** + * SECTION:element-pngenc + * + * Encodes png images. + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/pulse/pulsemixer.c b/ext/pulse/pulsemixer.c index 3a0de9dab0..5ced971b2b 100644 --- a/ext/pulse/pulsemixer.c +++ b/ext/pulse/pulsemixer.c @@ -21,16 +21,14 @@ /** * SECTION:element-pulsemixer - * @short_description: Element to control sound input and output levels for the PulseAudio sound server * @see_also: pulsesrc, pulsesink * - * - * * This element lets you adjust sound input and output levels for the * PulseAudio sound server. It supports the GstMixer interface, which can be * used to obtain a list of available mixer tracks. Set the mixer element to * READY state before using the GstMixer interface on it. - * + * + * * Example pipelines * * pulsemixer can't be used in a sensible way in gst-launch. diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index df774990ca..b41dba50c9 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -21,28 +21,20 @@ /** * SECTION:element-pulsesink - * @short_description: Output audio to a PulseAudio sound server * @see_also: pulsesrc, pulsemixer * - * - * - * This element outputs audio to a PulseAudio sound server. - * - * Example pipelines - * - * - * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink - * - * Play an Ogg/Vorbis file. - * - * - * - * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink - * - * Play a 440Hz sine wave. - * - * + * This element outputs audio to a + * PulseAudio sound server. * + * + * Example pipelines + * |[ + * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink + * ]| Play an Ogg/Vorbis file. + * |[ + * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink + * ]| Play a 440Hz sine wave. + * */ #ifdef HAVE_CONFIG_H diff --git a/ext/pulse/pulsesrc.c b/ext/pulse/pulsesrc.c index 0e1ca20e17..08fec59547 100644 --- a/ext/pulse/pulsesrc.c +++ b/ext/pulse/pulsesrc.c @@ -21,20 +21,16 @@ /** * SECTION:element-pulsesrc - * @short_description: Capture audio from a PulseAudio sound server * @see_also: pulsesink, pulsemixer * + * This element captures audio from a + * PulseAudio sound server. + * * - * - * This element captures audio from a PulseAudio sound server. - * * Example pipelines - * - * + * |[ * gst-launch -v pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg - * - * Record from a sound card using ALSA and encode to Ogg/Vorbis. - * + * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis. * */ diff --git a/ext/raw1394/gstdv1394src.c b/ext/raw1394/gstdv1394src.c index 5db691aa55..af7b04546d 100644 --- a/ext/raw1394/gstdv1394src.c +++ b/ext/raw1394/gstdv1394src.c @@ -22,18 +22,14 @@ /** * SECTION:element-dv1394src * - * - * * Read DV (digital video) data from firewire port. - * + * + * * Example launch line - * - * - * gst-launch dv1394src ! dvdemux name=d ! queue ! dvdec ! xvimagesink d. ! queue ! alsasink - * - * This pipeline captures from the firewire port and displays it (might need + * |[ + * gst-launch dv1394src ! queue ! dvdemux name=d ! queue ! dvdec ! xvimagesink d. ! queue ! alsasink + * ]| This pipeline captures from the firewire port and displays it (might need * format converters for audio/video). - * * */ diff --git a/ext/raw1394/gsthdv1394src.c b/ext/raw1394/gsthdv1394src.c index 8293b4a03f..cbb02136f1 100644 --- a/ext/raw1394/gsthdv1394src.c +++ b/ext/raw1394/gsthdv1394src.c @@ -16,6 +16,21 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-hdv1394src + * + * Read MPEG-TS data from firewire port. + * + * + * Example launch line + * |[ + * gst-launch hdv1394src ! queue ! decodebin name=d ! queue ! xvimagesink d. ! queue ! alsasink + * ]| captures from the firewire port and plays the streams. + * |[ + * gst-launch hdv1394src ! queue ! filesink location=mydump.ts + * ]| capture to a disk file + * + */ #ifdef HAVE_CONFIG_H #include "config.h" @@ -776,6 +791,7 @@ gst_hdv1394src_uri_get_type (void) { return GST_URI_SRC; } + static gchar ** gst_hdv1394src_uri_get_protocols (void) { @@ -783,6 +799,7 @@ gst_hdv1394src_uri_get_protocols (void) return protocols; } + static const gchar * gst_hdv1394src_uri_get_uri (GstURIHandler * handler) { diff --git a/ext/soup/gstsouphttpsrc.c b/ext/soup/gstsouphttpsrc.c index ad7433c133..580d06b627 100644 --- a/ext/soup/gstsouphttpsrc.c +++ b/ext/soup/gstsouphttpsrc.c @@ -14,65 +14,48 @@ /** * SECTION:element-souphttpsrc - * @short_description: Read from an HTTP/HTTPS/WebDAV/Icecast/Shoutcast - * location. * - * - * * This plugin reads data from a remote location specified by a URI. * Supported protocols are 'http', 'https'. - * - * + * * An HTTP proxy must be specified by its URL. * If the "http_proxy" environment variable is set, its value is used. * If built with libsoup's GNOME integration features, the GNOME proxy * configuration will be used, or failing that, proxy autodetection. - * The element-souphttpsrc::proxy property can be used to override the - * default. - * - * - * In case the element-souphttpsrc::iradio-mode property is set and the - * location is an HTTP resource, souphttpsrc will send special Icecast HTTP - * headers to the server to request additional Icecast meta-information. If - * the server is not an Icecast server, it will behave as if the - * element-souphttpsrc::iradio-mode property were not set. If it is, - * souphttpsrc will output data with a media type of application/x-icy, - * in which case you will need to use the #ICYDemux element as follow-up - * element to extract the Icecast metadata and to determine the underlying - * media type. - * - * - * Example pipeline: - * + * The #GstSoupHTTPSrc:proxy property can be used to override the default. + * + * In case the #GstSoupHTTPSrc:iradio-mode property is set and the location is + * an HTTP resource, souphttpsrc will send special Icecast HTTP headers to the + * server to request additional Icecast meta-information. + * If the server is not an Icecast server, it will behave as if the + * #GstSoupHTTPSrc:iradio-mode property were not set. If it is, souphttpsrc will + * output data with a media type of application/x-icy, in which case you will + * need to use the #ICYDemux element as follow-up element to extract the Icecast + * metadata and to determine the underlying media type. + * + * + * Example launch line + * |[ * gst-launch -v souphttpsrc location=https://some.server.org/index.html * ! filesink location=/home/joe/server.html - * - * The above pipeline reads a web page from a server using the HTTPS protocol + * ]| The above pipeline reads a web page from a server using the HTTPS protocol * and writes it to a local file. - * - * - * Another example pipeline: - * + * |[ * gst-launch -v souphttpsrc user-agent="FooPlayer 0.99 beta" * automatic-redirect=false proxy=http://proxy.intranet.local:8080 * location=http://music.foobar.com/demo.mp3 ! mad ! audioconvert * ! audioresample ! alsasink - * - * The above pipeline will read and decode and play an mp3 file from a + * ]| The above pipeline will read and decode and play an mp3 file from a * web server using the HTTP protocol. If the server sends redirects, * the request fails instead of following the redirect. The specified * HTTP proxy server is used. The User-Agent HTTP request header * is set to a custom string instead of "GStreamer souphttpsrc." - * - * - * Yet another example pipeline: - * + * |[ * gst-launch -v souphttpsrc location=http://10.11.12.13/mjpeg * do-timestamp=true ! multipartdemux * ! image/jpeg,width=640,height=480 ! matroskamux * ! filesink location=mjpeg.mkv - * - * The above pipeline reads a motion JPEG stream from an IP camera + * ]| The above pipeline reads a motion JPEG stream from an IP camera * using the HTTP protocol, encoded as mime/multipart image/jpeg * parts, and writes a Matroska motion JPEG file. The width and * height properties are set in the caps to provide the Matroska @@ -81,9 +64,7 @@ * These are used by the mime/multipart demultiplexer to emit timestamps * on the JPEG-encoded video frame buffers. This allows the Matroska * multiplexer to timestamp the frames in the resulting file. - * * - * */ #ifdef HAVE_CONFIG_H diff --git a/ext/speex/gstspeexdec.c b/ext/speex/gstspeexdec.c index 0987d865e1..b70c45c63f 100644 --- a/ext/speex/gstspeexdec.c +++ b/ext/speex/gstspeexdec.c @@ -20,23 +20,19 @@ /** * SECTION:element-speexdec - * @short_description: a decoder that decodes Speex to raw audio * @see_also: speexenc, oggdemux * - * - * * This element decodes a Speex stream to raw integer audio. * Speex is a royalty-free * audio codec maintained by the Xiph.org * Foundation. - * + * + * * Example pipelines - * - * + * |[ * gst-launch -v filesrc location=speex.ogg ! oggdemux ! speexdec ! audioconvert ! audioresample ! alsasink - * - * Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the documentation of speexenc. - * + * ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the + * documentation of speexenc. * * * Last reviewed on 2006-04-05 (0.10.2) diff --git a/ext/speex/gstspeexenc.c b/ext/speex/gstspeexenc.c index 963d560706..631d90b911 100644 --- a/ext/speex/gstspeexenc.c +++ b/ext/speex/gstspeexenc.c @@ -17,6 +17,22 @@ * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-speexenc + * @see_also: speexdec, oggmux + * + * This element encodes audio as a Speex stream. + * Speex is a royalty-free + * audio codec maintained by the Xiph.org + * Foundation. + * + * + * Example pipelines + * |[ + * gst-launch audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg + * ]| Encode an Ogg/Speex file. + * + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/ext/taglib/gstapev2mux.cc b/ext/taglib/gstapev2mux.cc index b7a61bff4a..6857e2530f 100644 --- a/ext/taglib/gstapev2mux.cc +++ b/ext/taglib/gstapev2mux.cc @@ -23,28 +23,23 @@ * SECTION:element-apev2mux * @see_also: #GstTagSetter * - * - * * This element adds APEv2 tags to the beginning of a stream using the taglib * library. - * - * + * * Applications can set the tags to write using the #GstTagSetter interface. * Tags sent by upstream elements will be picked up automatically (and merged * according to the merge mode set via the tag setter interface). - * - * - * Here is a simple pipeline that transcodes a file from Ogg/Vorbis to mp3 - * format with an APEv2 that contains the same as the the Ogg/Vorbis file: - * + * + * + * Example pipelines + * |[ * gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! apev2mux ! filesink location=foo.mp3 - * - * Make sure the Ogg/Vorbis file actually has comments to preserve. - * You can verify the tags were written using: - * + * ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an + * APEv2 that contains the same as the the Ogg/Vorbis file. Make sure the + * Ogg/Vorbis file actually has comments to preserve. + * |[ * gst-launch -m filesrc location=foo.mp3 ! apedemux ! fakesink silent=TRUE 2> /dev/null | grep taglist - * - * + * ]| Verify that tags have been written. * */ diff --git a/ext/taglib/gstid3v2mux.cc b/ext/taglib/gstid3v2mux.cc index cf1762797b..fd15836ea8 100644 --- a/ext/taglib/gstid3v2mux.cc +++ b/ext/taglib/gstid3v2mux.cc @@ -22,30 +22,25 @@ * SECTION:element-id3v2mux * @see_also: #GstID3Demux, #GstTagSetter * - * - * * This element adds ID3v2 tags to the beginning of a stream using the taglib * library. More precisely, the tags written are ID3 version 2.4.0 tags (which * means in practice that some hardware players or outdated programs might not * be able to read them properly). - * - * + * * Applications can set the tags to write using the #GstTagSetter interface. * Tags sent by upstream elements will be picked up automatically (and merged * according to the merge mode set via the tag setter interface). - * - * - * Here is a simple pipeline that transcodes a file from Ogg/Vorbis to mp3 - * format with an ID3v2 that contains the same as the the Ogg/Vorbis file: - * + * + * + * Example pipelines + * |[ * gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! id3v2mux ! filesink location=foo.mp3 - * - * Make sure the Ogg/Vorbis file actually has comments to preserve. - * You can verify the tags were written using: - * + * ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an + * ID3v2 that contains the same as the the Ogg/Vorbis file. Make sure the + * Ogg/Vorbis file actually has comments to preserve. + * |[ * gst-launch -m filesrc location=foo.mp3 ! id3demux ! fakesink silent=TRUE 2> /dev/null | grep taglist - * - * + * ]| Verify that tags have been written. * */ diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index 70c80f0b71..41000273e1 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -24,19 +24,17 @@ /** * SECTION:element-wavpackdec * - * * WavpackDec decodes framed (for example by the WavpackParse element) * Wavpack streams and decodes them to raw audio. * Wavpack is an open-source * audio codec that features both lossless and lossy encoding. + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink - * - * This pipeline decodes the Wavpack file test.wv into raw audio buffers and + * ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers and * tries to play it back using an automatically found audio sink. - * * */ diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c index 8033764784..647800c56b 100644 --- a/ext/wavpack/gstwavpackenc.c +++ b/ext/wavpack/gstwavpackenc.c @@ -22,32 +22,24 @@ /** * SECTION:element-wavpackenc * - * * WavpackEnc encodes raw audio into a framed Wavpack stream. * Wavpack is an open-source * audio codec that features both lossless and lossy encoding. + * + * * Example launch line - * - * + * |[ * gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv - * - * This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed + * ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed * as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits). - * - * - * + * |[ * gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv - * - * This pipeline encodes audio from an audio CD into a Wavpack file using + * ]| This pipeline encodes audio from an audio CD into a Wavpack file using * lossless encoding (the file output will be fairly large). - * - * - * + * |[ * gst-launch cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv - * - * This pipeline encodes audio from an audio CD into a Wavpack file using + * ]| This pipeline encodes audio from an audio CD into a Wavpack file using * lossy encoding at a certain bitrate (the file will be fairly small). - * * */ diff --git a/ext/wavpack/gstwavpackparse.c b/ext/wavpack/gstwavpackparse.c index 3f85205d37..8fea90b0c6 100644 --- a/ext/wavpack/gstwavpackparse.c +++ b/ext/wavpack/gstwavpackparse.c @@ -24,19 +24,17 @@ /** * SECTION:element-wavpackparse * - * * WavpackParse takes raw, unframed Wavpack streams and splits them into * single Wavpack chunks with information like bit depth and the position * in the stream. * Wavpack is an open-source * audio codec that features both lossless and lossy encoding. + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! fakesink - * - * This pipeline decodes the Wavpack file test.wv into raw audio buffers. - * + * ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers. * */ diff --git a/gst/matroska/matroska-mux.h b/gst/matroska/matroska-mux.h index 4ca0035fad..22ee93520f 100644 --- a/gst/matroska/matroska-mux.h +++ b/gst/matroska/matroska-mux.h @@ -137,8 +137,6 @@ typedef struct _GstMatroskaMuxClass { GstElementClass parent; } GstMatroskaMuxClass; -GType gst_matroska_mux_get_type (void); - gboolean gst_matroska_mux_plugin_init (GstPlugin *plugin); G_END_DECLS diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c index 0da694610a..bcdbc35961 100644 --- a/gst/udp/gstudpsrc.c +++ b/gst/udp/gstudpsrc.c @@ -29,7 +29,7 @@ * #GstUDPSrc:port property to 0. After setting the udpsrc to PAUSED, the * allocated port can be obtained by reading the port property. * - * udpsrc can read from multicast groups by setting the #GstUDPSrc:multicast_group + * udpsrc can read from multicast groups by setting the #GstUDPSrc:multicast-group * property to the IP address of the multicast group. * * Alternatively one can provide a custom socket to udpsrc with the #GstUDPSrc:sockfd