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rtpmanager: rename gstrtp* -> rtp*

Browse source 
This was done in 0.10 to avoid conflict with the rtp elements in
farsight, but the gst-prefixing is no longer needed in 0.11
This commit is contained in:
Tim-Philipp Müller 2011-11-24 00:52:40 +00:00
parent c5cad2aff2
commit 09ca5fa910
10 changed files with 62 additions and 64 deletions
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12
docs/plugins/gst-plugins-good-plugins-docs.sgml
View file

@ -83,11 +83,6 @@
<xi:include href="xml/element-gdkpixbufsink.xml" />
<xi:include href="xml/element-goom.xml" />
<xi:include href="xml/element-goom2k1.xml" />
<xi:include href="xml/element-gstrtpbin.xml" />
<xi:include href="xml/element-gstrtpjitterbuffer.xml" />
<xi:include href="xml/element-gstrtpptdemux.xml" />
<xi:include href="xml/element-gstrtpsession.xml" />
<xi:include href="xml/element-gstrtpssrcdemux.xml" />
<xi:include href="xml/element-halaudiosink.xml" />
<xi:include href="xml/element-halaudiosrc.xml" />
<xi:include href="xml/element-hdv1394src.xml" />
@ -145,6 +140,11 @@
<xi:include href="xml/element-rtpj2kpay.xml" />
<xi:include href="xml/element-rtpjpegpay.xml" />
<xi:include href="xml/element-rtspsrc.xml" />
<xi:include href="xml/element-rtpbin.xml" />
<xi:include href="xml/element-rtpjitterbuffer.xml" />
<xi:include href="xml/element-rtpptdemux.xml" />
<xi:include href="xml/element-rtpsession.xml" />
<xi:include href="xml/element-rtpssrcdemux.xml" />
<xi:include href="xml/element-shagadelictv.xml" />
<xi:include href="xml/element-shapewipe.xml" />
<xi:include href="xml/element-smokedec.xml" />
@ -233,7 +233,7 @@
<xi:include href="xml/plugin-pulseaudio.xml" />
<xi:include href="xml/plugin-replaygain.xml" />
<xi:include href="xml/plugin-rtp.xml" />
<xi:include href="xml/plugin-gstrtpmanager.xml" />
<xi:include href="xml/plugin-rtpmanager.xml" />
<xi:include href="xml/plugin-rtsp.xml" />
<xi:include href="xml/plugin-shapewipe.xml" />
<xi:include href="xml/plugin-shout2send.xml" />

20
docs/plugins/gst-plugins-good-plugins-sections.txt
View file

@ -957,8 +957,8 @@ GST_IS_GOOM_CLASS
</SECTION>
<SECTION>
<FILE>element-gstrtpbin</FILE>
<TITLE>gstrtpbin</TITLE>
<FILE>element-rtpbin</FILE>
<TITLE>rtpbin</TITLE>
GstRtpBin
<SUBSECTION Standard>
GstRtpBinPrivate
@ -972,8 +972,8 @@ GST_IS_RTP_BIN_CLASS
</SECTION>
<SECTION>
<FILE>element-gstrtpjitterbuffer</FILE>
<TITLE>gstrtpjitterbuffer</TITLE>
<FILE>element-rtpjitterbuffer</FILE>
<TITLE>rtpjitterbuffer</TITLE>
GstRtpJitterBuffer
<SUBSECTION Standard>
GstRtpJitterBufferClass
@ -987,8 +987,8 @@ GST_IS_RTP_JITTER_BUFFER_CLASS
</SECTION>
<SECTION>
<FILE>element-gstrtpptdemux</FILE>
<TITLE>gstrtpptdemux</TITLE>
<FILE>element-rtpptdemux</FILE>
<TITLE>rtpptdemux</TITLE>
GstRtpPtDemux
<SUBSECTION Standard>
GstRtpPtDemuxClass
@ -1002,8 +1002,8 @@ GST_IS_RTP_PT_DEMUX_CLASS
</SECTION>
<SECTION>
<FILE>element-gstrtpsession</FILE>
<TITLE>gstrtpsession</TITLE>
<FILE>element-rtpsession</FILE>
<TITLE>rtpsession</TITLE>
GstRtpSession
<SUBSECTION Standard>
GstRtpSessionClass
@ -1018,8 +1018,8 @@ GST_RTP_SESSION_CAST
</SECTION>
<SECTION>
<FILE>element-gstrtpssrcdemux</FILE>
<TITLE>gstrtpssrcdemux</TITLE>
<FILE>element-rtpssrcdemux</FILE>
<TITLE>rtpssrcdemux</TITLE>
GstRtpSsrcDemux
<SUBSECTION Standard>
GstRtpSsrcDemuxClass

12
docs/plugins/inspect/plugin-gstrtpmanager.xml → docs/plugins/inspect/plugin-rtpmanager.xml
View file

@ -1,5 +1,5 @@
<plugin>
<name>gstrtpmanager</name>
<name>rtpmanager</name>
<description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename>
@ -10,7 +10,7 @@
<origin>Unknown package origin</origin>
<elements>
<element>
<name>gstrtpbin</name>
<name>rtpbin</name>
<longname>RTP Bin</longname>
<class>Filter/Network/RTP</class>
<description>Real-Time Transport Protocol bin</description>
@ -55,7 +55,7 @@
</pads>
</element>
<element>
<name>gstrtpjitterbuffer</name>
<name>rtpjitterbuffer</name>
<longname>RTP packet jitter-buffer</longname>
<class>Filter/Network/RTP</class>
<description>A buffer that deals with network jitter and other transmission faults</description>
@ -82,7 +82,7 @@
</pads>
</element>
<element>
<name>gstrtpptdemux</name>
<name>rtpptdemux</name>
<longname>RTP Demux</longname>
<class>Demux/Network/RTP</class>
<description>Parses codec streams transmitted in the same RTP session</description>
@ -103,7 +103,7 @@
</pads>
</element>
<element>
<name>gstrtpsession</name>
<name>rtpsession</name>
<longname>RTP Session</longname>
<class>Filter/Network/RTP</class>
<description>Implement an RTP session</description>
@ -154,7 +154,7 @@
</pads>
</element>
<element>
<name>gstrtpssrcdemux</name>
<name>rtpssrcdemux</name>
<longname>RTP SSRC Demux</longname>
<class>Demux/Network/RTP</class>
<description>Splits RTP streams based on the SSRC</description>

45
gst/rtpmanager/gstrtpbin.c
View file

@ -555,10 +555,10 @@ create_session (GstRtpBin * rtpbin, gint id)
GstElement *session, *demux;
GstState target;
if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
goto no_session;
if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
goto no_demux;
sess = g_new0 (GstRtpBinSession, 1);
@ -615,13 +615,13 @@ create_session (GstRtpBin * rtpbin, gint id)
/* ERRORS */
no_session:
{
g_warning ("gstrtpbin: could not create gstrtpsession element");
g_warning ("rtpbin: could not create gstrtpsession element");
return NULL;
}
no_demux:
{
gst_object_unref (session);
g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
g_warning ("rtpbin: could not create gstrtpssrcdemux element");
return NULL;
}
}
@ -1403,11 +1403,11 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
rtpbin = session->bin;
if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
goto no_jitterbuffer;
if (!rtpbin->ignore_pt)
if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
goto no_demux;
@ -1471,13 +1471,13 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
/* ERRORS */
no_jitterbuffer:
{
g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
g_warning ("gstrtpbin: could not create gstrtpptdemux element");
g_warning ("rtpbin: could not create gstrtpptdemux element");
return NULL;
}
}
@ -2646,7 +2646,7 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */
no_name:
{
g_warning ("gstrtpbin: invalid name given");
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
@ -2656,12 +2656,12 @@ create_error:
}
pad_failed:
{
g_warning ("gstrtpbin: failed to get session pad");
g_warning ("rtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
g_warning ("gstrtpbin: failed to link pads");
g_warning ("rtpbin: failed to link pads");
return NULL;
}
}
@ -2757,7 +2757,7 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
/* ERRORS */
no_name:
{
g_warning ("gstrtpbin: invalid name given");
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
@ -2767,12 +2767,12 @@ create_error:
}
pad_failed:
{
g_warning ("gstrtpbin: failed to get session pad");
g_warning ("rtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
g_warning ("gstrtpbin: failed to link pads");
g_warning ("rtpbin: failed to link pads");
return NULL;
}
}
@ -2858,7 +2858,7 @@ create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */
no_name:
{
g_warning ("gstrtpbin: invalid name given");
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
@ -2868,13 +2868,12 @@ create_error:
}
pad_failed:
{
g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
g_warning ("rtpbin: failed to get session pad for session %d", sessid);
return NULL;
}
no_srcpad:
{
g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
sessid);
g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
return NULL;
}
}
@ -2944,17 +2943,17 @@ create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */
no_name:
{
g_warning ("gstrtpbin: invalid name given");
g_warning ("rtpbin: invalid name given");
return NULL;
}
no_session:
{
g_warning ("gstrtpbin: session with id %d does not exist", sessid);
g_warning ("rtpbin: session with id %d does not exist", sessid);
return NULL;
}
pad_failed:
{
g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
return NULL;
}
}
@ -3088,7 +3087,7 @@ wrong_template:
{
g_free (pad_name);
GST_RTP_BIN_UNLOCK (rtpbin);
g_warning ("gstrtpbin: this is not our template");
g_warning ("rtpbin: this is not our template");
return NULL;
}
}
@ -3138,7 +3137,7 @@ gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
unknown_pad:
{
GST_RTP_BIN_UNLOCK (rtpbin);
g_warning ("gstrtpbin: %s:%s is not one of our request pads",
g_warning ("rtpbin: %s:%s is not one of our request pads",
GST_DEBUG_PAD_NAME (pad));
return;
}

13
gst/rtpmanager/gstrtpmanager.c
View file

@ -30,23 +30,22 @@
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "gstrtpbin", GST_RANK_NONE,
GST_TYPE_RTP_BIN))
if (!gst_element_register (plugin, "rtpbin", GST_RANK_NONE, GST_TYPE_RTP_BIN))
return FALSE;
if (!gst_element_register (plugin, "gstrtpjitterbuffer", GST_RANK_NONE,
if (!gst_element_register (plugin, "rtpjitterbuffer", GST_RANK_NONE,
GST_TYPE_RTP_JITTER_BUFFER))
return FALSE;
if (!gst_element_register (plugin, "gstrtpptdemux", GST_RANK_NONE,
if (!gst_element_register (plugin, "rtpptdemux", GST_RANK_NONE,
GST_TYPE_RTP_PT_DEMUX))
return FALSE;
if (!gst_element_register (plugin, "gstrtpsession", GST_RANK_NONE,
if (!gst_element_register (plugin, "rtpsession", GST_RANK_NONE,
GST_TYPE_RTP_SESSION))
return FALSE;
if (!gst_element_register (plugin, "gstrtpssrcdemux", GST_RANK_NONE,
if (!gst_element_register (plugin, "rtpssrcdemux", GST_RANK_NONE,
GST_TYPE_RTP_SSRC_DEMUX))
return FALSE;
@ -55,6 +54,6 @@ plugin_init (GstPlugin * plugin)
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"gstrtpmanager",
"rtpmanager",
"RTP session management plugin library",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

10
tests/check/elements/rtpbin.c
View file

@ -29,7 +29,7 @@ GST_START_TEST (test_cleanup_send)
GObject *session;
gint count = 2;
rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
while (count--) {
/* request session 0 */
@ -231,7 +231,7 @@ GST_START_TEST (test_cleanup_recv)
init_data (&data);
rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added_cb, &data);
g_signal_connect (rtpbin, "pad-removed", (GCallback) pad_removed_cb, &data);
@ -306,7 +306,7 @@ GST_START_TEST (test_cleanup_recv2)
init_data (&data);
rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added_cb, &data);
g_signal_connect (rtpbin, "pad-removed", (GCallback) pad_removed_cb, &data);
@ -382,7 +382,7 @@ GST_START_TEST (test_request_pad_by_template_name)
GstElement *rtpbin;
GstPad *rtp_sink1, *rtp_sink2, *rtp_sink3;
rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
rtp_sink1 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_%u");
fail_unless (rtp_sink1 != NULL);
fail_unless_equals_string (GST_PAD_NAME (rtp_sink1), "recv_rtp_sink_0");
@ -417,7 +417,7 @@ GST_END_TEST;
static Suite *
gstrtpbin_suite (void)
{
Suite *s = suite_create ("gstrtpbin");
Suite *s = suite_create ("rtpbin");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);

2
tests/examples/rtp/client-PCMA.c
View file

@ -184,7 +184,7 @@ main (int argc, char *argv[])
g_assert (res == TRUE);
/* the rtpbin element */
rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
g_assert (rtpbin);
gst_bin_add (GST_BIN (pipeline), rtpbin);

4
tests/examples/rtp/client-PCMA.py
View file

@ -35,7 +35,7 @@ RTP_RECV_PORT = 5002
RTCP_RECV_PORT = 5003
RTCP_SEND_PORT = 5007
#gst-launch -v gstrtpbin name=rtpbin \
#gst-launch -v rtpbin name=rtpbin \
# udpsrc caps=$AUDIO_CAPS port=$RTP_RECV_PORT ! rtpbin.recv_rtp_sink_0 \
# rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
# udpsrc port=$RTCP_RECV_PORT ! rtpbin.recv_rtcp_sink_0 \
@ -84,7 +84,7 @@ pipeline.add(audiodepay, audiodec, audioconv, audiores, audiosink)
res = gst.element_link_many(audiodepay, audiodec, audioconv, audiores, audiosink)
# the rtpbin element
rtpbin = gst.element_factory_make('gstrtpbin', 'rtpbin')
rtpbin = gst.element_factory_make('rtpbin', 'rtpbin')
pipeline.add(rtpbin)

4
tests/examples/rtp/server-alsasrc-PCMA.c
View file

@ -104,7 +104,7 @@ print_stats (GstElement * rtpbin)
/* build a pipeline equivalent to:
*
* gst-launch -v gstrtpbin name=rtpbin \
* gst-launch -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 \
@ -150,7 +150,7 @@ main (int argc, char *argv[])
}
/* the rtpbin element */
rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
g_assert (rtpbin);
gst_bin_add (GST_BIN (pipeline), rtpbin);

4
tests/examples/rtp/server-alsasrc-PCMA.py
View file

@ -4,7 +4,7 @@ import gobject, pygst
pygst.require("0.10")
import gst
#gst-launch -v gstrtpbin name=rtpbin audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \
#gst-launch -v rtpbin name=rtpbin audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \
# rtpbin.send_rtp_src_0 ! udpsink port=10000 host=xxx.xxx.xxx.xxx \
# rtpbin.send_rtcp_src_0 ! udpsink port=10001 host=xxx.xxx.xxx.xxx sync=false async=false \
# udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0
@ -37,7 +37,7 @@ pipeline.add(audiosrc, audioconv, audiores, audioenc, audiopay)
res = gst.element_link_many(audiosrc, audioconv, audiores, audioenc, audiopay)
# the rtpbin element
rtpbin = gst.element_factory_make('gstrtpbin', 'rtpbin')
rtpbin = gst.element_factory_make('rtpbin', 'rtpbin')
pipeline.add(rtpbin)