diff options
| -rw-r--r-- | ChangeLog | 9 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-good-plugins.args | 38 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-alpha.xml | 2 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-rtp.xml | 20 | ||||
| -rw-r--r-- | gst/level/gstlevel.c | 2 |
5 files changed, 57 insertions, 14 deletions
@@ -1,5 +1,14 @@ 2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org> + * docs/plugins/gst-plugins-good-plugins.args: + * docs/plugins/inspect/plugin-alpha.xml: + * docs/plugins/inspect/plugin-rtp.xml: + * gst/level/gstlevel.c: (gst_level_set_caps), + (gst_level_transform_ip): + updating docs + +2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org> + * Makefile.am: * check/elements/level.c: (GST_START_TEST): * gst/level/Makefile.am: diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args index 45ebd632..fba4f7a9 100644 --- a/docs/plugins/gst-plugins-good-plugins.args +++ b/docs/plugins/gst-plugins-good-plugins.args @@ -4239,13 +4239,23 @@ </ARG> <ARG> +<NAME>GstRtpMP4VEnc::send-config</NAME> +<TYPE>gboolean</TYPE> +<RANGE></RANGE> +<FLAGS>rw</FLAGS> +<NICK>Send Config</NICK> +<BLURB>Send the config parameters in RTP packets as well.</BLURB> +<DEFAULT>FALSE</DEFAULT> +</ARG> + +<ARG> <NAME>GstLevel::interval</NAME> -<TYPE>gdouble</TYPE> -<RANGE>[0.01,100]</RANGE> +<TYPE>guint64</TYPE> +<RANGE>>= 1</RANGE> <FLAGS>rw</FLAGS> <NICK>Interval</NICK> -<BLURB>Interval between posts (in seconds).</BLURB> -<DEFAULT>0.1</DEFAULT> +<BLURB>Interval of time between message posts (in nanoseconds).</BLURB> +<DEFAULT>100000000</DEFAULT> </ARG> <ARG> @@ -4254,7 +4264,7 @@ <RANGE></RANGE> <FLAGS>rw</FLAGS> <NICK>mesage</NICK> -<BLURB>Post a level message for each interval.</BLURB> +<BLURB>Post a level message for each passed interval.</BLURB> <DEFAULT>TRUE</DEFAULT> </ARG> @@ -4270,12 +4280,12 @@ <ARG> <NAME>GstLevel::peak-ttl</NAME> -<TYPE>gdouble</TYPE> -<RANGE>[0,100]</RANGE> +<TYPE>guint64</TYPE> +<RANGE></RANGE> <FLAGS>rw</FLAGS> <NICK>Peak TTL</NICK> -<BLURB>Time To Live of decay peak before it falls back.</BLURB> -<DEFAULT>0.3</DEFAULT> +<BLURB>Time To Live of decay peak before it falls back (in nanoseconds).</BLURB> +<DEFAULT>300000000</DEFAULT> </ARG> <ARG> @@ -6498,3 +6508,13 @@ <DEFAULT>2000000000</DEFAULT> </ARG> +<ARG> +<NAME>GstRtpGSMParse::frequency</NAME> +<TYPE>gint</TYPE> +<RANGE></RANGE> +<FLAGS>rw</FLAGS> +<NICK>frequency</NICK> +<BLURB>frequency.</BLURB> +<DEFAULT>8000</DEFAULT> +</ARG> + diff --git a/docs/plugins/inspect/plugin-alpha.xml b/docs/plugins/inspect/plugin-alpha.xml index d6a986d7..ee501667 100644 --- a/docs/plugins/inspect/plugin-alpha.xml +++ b/docs/plugins/inspect/plugin-alpha.xml @@ -1,6 +1,6 @@ <plugin> <name>alpha</name> - <description>resizes a video by adding borders or cropping</description> + <description>adds an alpha channel to video</description> <filename>../../gst/alpha/.libs/libgstalpha.so</filename> <basename>libgstalpha.so</basename> <version>0.9.1.1</version> diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml index b04402f9..5e04186c 100644 --- a/docs/plugins/inspect/plugin-rtp.xml +++ b/docs/plugins/inspect/plugin-rtp.xml @@ -13,14 +13,14 @@ <name>rtpamrdec</name> <longname>RTP packet parser</longname> <class>Codec/Parser/Network</class> - <description>Extracts MPEG audio from RTP packets</description> + <description>Extracts AMR audio from RTP packets (RFC 3267)</description> <author>Wim Taymans <wim@fluendo.com></author> </element> <element> <name>rtpamrenc</name> <longname>RTP packet parser</longname> <class>Codec/Parser/Network</class> - <description>Encode AMR audio into RTP packets</description> + <description>Encode AMR audio into RTP packets (RFC 3267)</description> <author>Wim Taymans <wim@fluendo.com></author> </element> <element> @@ -31,6 +31,20 @@ <author>Wim Taymans <wim@fluendo.com></author> </element> <element> + <name>rtpgsmenc</name> + <longname>RTP GSM Audio Encoder</longname> + <class>Codec/Encoder/Network</class> + <description>Encodes GSM audio into a RTP packet</description> + <author>Zeeshan Ali <zak147@yahoo.com></author> + </element> + <element> + <name>rtpgsmparse</name> + <longname>RTP packet parser</longname> + <class>Codec/Parser/Network</class> + <description>Extracts GSM audio from RTP packets</description> + <author>Zeeshan Ali <zak147@yahoo.com></author> + </element> + <element> <name>rtph263pdec</name> <longname>RTP packet parser</longname> <class>Codec/Parser/Network</class> @@ -41,7 +55,7 @@ <name>rtph263penc</name> <longname>RTP packet parser</longname> <class>Codec/Parser/Network</class> - <description>Extracts H263+ video from RTP packets</description> + <description>Encodes H263+ video in RTP packets (RFC 2429)</description> <author>Wim Taymans <wim@fluendo.com></author> </element> <element> diff --git a/gst/level/gstlevel.c b/gst/level/gstlevel.c index f7d22edb..b1d63441 100644 --- a/gst/level/gstlevel.c +++ b/gst/level/gstlevel.c @@ -25,7 +25,7 @@ * <refsect2> * <para> * Level analyses incoming audio buffers and, if the - * <link linkend="GstLevel--message">message property</link> is #TRUE. + * <link linkend="GstLevel--message">message property</link> is #TRUE, * generates an application message named * <classname>"level"</classname>: * after each interval of time given by the |