mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-18 15:51:11 +00:00
Code Issues 1 Projects Releases Wiki Activity
10219 commits 44 branches 1509 tags 620 MiB
Commit graph

23 commits

Author SHA1 Message Date
Stefan Kost
d8167e3071 various (gst): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 18:00:28 +03:00
Benjamin Otte
cccfeaa59c gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:32:00 +01:00
Benjamin Otte
3342b1679e Add -Wmissing-declarations -Wmissing-prototypes warning flags 
And fix all the warnings.
 2010-03-17 18:23:28 +01:00
Stefan Kost
e0cdd879b4 build: fprintf, sprintf, sscanf need stdio.h 2009-10-07 14:03:20 +03:00
Krzysztof Błaszkowski
9fbdfefc56 rtpdec: fix some buffer leaks 2009-06-25 13:18:14 +02:00
Edward Hervey
0cb5b42d54 Remove trivial unused variables detected by CLang static analyzer. 2009-04-18 18:51:28 +02:00
Stefan Kost
a99d3f8769 Update and add documentation for plugins with no deps (gst). 
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
 2009-01-28 12:32:59 +02:00
Wim Taymans
41f0496738 gst/rtsp/gstrtpdec.*: Add (dummy) SSRC management signals. 
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
(gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Add (dummy) SSRC management signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
(on_timeout), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add connection-speed property.
Add find_stream helper functions.
Handle stream EOS based on BYE messages or SSRC timeout.
Returns SUCCESS from the state change function as we hide our async
elements from the parent.
 2007-08-16 11:47:19 +00:00
Wim Taymans
321a79d484 gst/rtsp/gstrtpdec.*: Added signal for backwards compat. 
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Added signal for backwards compat.
 2007-05-21 10:03:42 +00:00
Wim Taymans
6937be1a09 gst/rtsp/gstrtpdec.*: Add dummy latency property to be backwards compat with rtpbin. 
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property):
* gst/rtsp/gstrtpdec.h:
Add dummy latency property to be backwards compat with rtpbin.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_parse_rtpinfo):
* gst/rtsp/gstrtspsrc.h:
Add latency property and configure in the session manager.
Don't set invalid clock-base and seqnum-base on caps, some servers
sometimes don't send them.
 2007-04-25 15:55:32 +00:00
Wim Taymans
b752470823 docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs. 
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Fix docs.
* gst/rtsp/URLS:
Add some more example urls.
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_chain_rtp):
Better debugging.
* gst/rtsp/gstrtspsrc.c: (request_pt_map),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_parse_rtpinfo):
Remove unused code.
 2007-04-13 09:32:21 +00:00
Wim Taymans
86a4c1c6b0 gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals. 
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
* gst/rtsp/gstrtpdec.h:
Make backward compat with rtpbin by adding the request-pt-map signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (request_pt_map),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams):
* gst/rtsp/gstrtspsrc.h:
Implement request-pt-map signals instead of setting caps on the buffers
for the session manager.
 2007-04-12 08:21:28 +00:00
Wim Taymans
f80444aaec gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback. 
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
 2007-04-06 12:54:16 +00:00
Wim Taymans
dc212cdb3d gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's what it will be in the future and rtspsrc... 
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
Make state change to PAUSED NO_PREROLL because that's what it will be in
the future and rtspsrc relies on it.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_change_state):
Don't error out when we don't get an error from the state change
function.
 2007-03-01 09:29:34 +00:00
Wim Taymans
63c87f1899 gst/rtsp/: Factor out extension in separate module. 
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
 2006-10-04 17:24:40 +00:00
Wim Taymans
a365a29c77 gst/rtsp/URLS: Added some test URLS. 
Original commit message from CVS:
* gst/rtsp/URLS:
Added some test URLS.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
When creating streams, give access to the complete SDP.
Fix some leaks.
Collect and merge global stream properties in stream caps.
Preliminary support for WMServer.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Make connection interruptable.
Refactor to make it reconnectable.
Don't fail on short reads when reading data packets.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
(rtsp_url_get_port):
* gst/rtsp/rtspurl.h:
Add methods for getting/setting the port.
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_message_get_attribute_val), (sdp_media_get_attribute),
(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
(sdp_media_get_format), (sdp_parse_line),
(sdp_message_parse_buffer):
Fix headers.
Add methods for getting multiple attributes with the same name.
Increase buffer size when parsing.
Fix parsing of a=foo fields.
* gst/rtsp/test.c: (main):
Update to new connection API.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
* gst/rtsp/rtsptransport.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/gstrtsp.c:
* gst/rtsp/gstrtsp.h:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Dual licensed under MIT and LGPL now.
 2006-09-20 16:06:27 +00:00
Wim Taymans
6eedcfbc8c gst/rtsp/gstrtpdec.c: Add pads after setting them up. 
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
 2006-08-16 09:48:26 +00:00
Tim-Philipp Müller
05eaedc496 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) plus two minor macro fixes. 
Original commit message from CVS:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/annodex/gstcmmlparser.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/gdk_pixbuf/pixbufscale.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalphacolor.c:
* gst/cutter/gstcutter.c:
* gst/debug/gstnavigationtest.c:
* gst/icydemux/gsticydemux.c:
* gst/level/gstlevel.c:
* gst/multipart/multipart.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstvideoflip.c:
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
plus two minor macro fixes.
 2006-06-22 19:31:04 +00:00
Wim Taymans
bfd2b35dda Added documentation for the rtsp plugin. Fixes #345393. 
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
Added documentation for the rtsp plugin. Fixes #345393.
 2006-06-20 14:57:09 +00:00
Stefan Kost
27f2c9b255 Define GstElementDetails as const and also static (when defined as global) 
Original commit message from CVS:
* ext/aalib/gstaasink.c:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/cairo/gsttextoverlay.c:
* ext/cairo/gsttimeoverlay.c:
* ext/cdio/gstcdiocddasrc.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
* ext/gdk_pixbuf/pixbufscale.c:
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libcaca/gstcacasink.c:
* ext/libmng/gstmngdec.c:
* ext/libmng/gstmngenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/mikmod/gstmikmod.c:
* ext/raw1394/gstdv1394src.c:
* ext/shout2/gstshout2.c: (gst_shout2send_init):
* ext/shout2/gstshout2.h:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalpha.c:
* gst/alpha/gstalphacolor.c:
* gst/apetag/gstapedemux.c:
* gst/auparse/gstauparse.c:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
* gst/avi/gstavimux.c: (gst_avimux_base_init):
* gst/cutter/gstcutter.c:
* gst/debug/breakmydata.c:
* gst/debug/efence.c:
* gst/debug/gstnavigationtest.c:
* gst/debug/gstnavseek.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/testplugin.c:
* gst/effectv/gstaging.c:
* gst/effectv/gstdice.c:
* gst/effectv/gstedge.c:
* gst/effectv/gstquark.c:
* gst/effectv/gstrev.c:
* gst/effectv/gstshagadelic.c:
* gst/effectv/gstvertigo.c:
* gst/effectv/gstwarp.c:
* gst/flx/gstflxdec.c:
* gst/goom/gstgoom.c:
* gst/icydemux/gsticydemux.c:
* gst/id3demux/gstid3demux.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/law/alaw-decode.c: (gst_alawdec_base_init):
* gst/law/alaw-encode.c: (gst_alawenc_base_init):
* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
* gst/level/gstlevel.c:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
* gst/median/gstmedian.c:
* gst/monoscope/gstmonoscope.c:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
* gst/oldcore/gstaggregator.c:
* gst/oldcore/gstfdsink.c:
* gst/oldcore/gstmd5sink.c:
* gst/oldcore/gstmultifilesrc.c:
* gst/oldcore/gstpipefilter.c:
* gst/oldcore/gstshaper.c:
* gst/oldcore/gststatistics.c:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/smpte/gstsmpte.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideoflip.c:
* gst/videofilter/gstvideotemplate.c:
(gst_videotemplate_base_init):
* gst/videomixer/videomixer.c:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_dispose),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_chain), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
* sys/oss/gstossmixerelement.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiosink.c:
Define GstElementDetails as const and also static (when defined as
global)
 2006-04-25 21:39:46 +00:00
Stefan Kost
b5af832d7b Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) 
Original commit message from CVS:
* ext/aalib/gstaasink.c: (gst_aasink_class_init):
* ext/esd/esdsink.c: (gst_esdsink_class_init):
* ext/flac/gstflactag.c: (gst_flac_tag_class_init):
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init):
* ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
* ext/libmng/gstmngdec.c: (gst_mngdec_class_init):
* ext/libmng/gstmngenc.c: (gst_mngenc_class_init):
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init):
* ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
* ext/mikmod/gstmikmod.c: (gst_mikmod_class_init):
* ext/shout2/gstshout2.c: (gst_shout2send_class_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
* gst/alpha/gstalpha.c: (gst_alpha_class_init):
* gst/avi/gstavimux.c: (gst_avimux_class_init):
* gst/debug/efence.c: (gst_efence_class_init):
* gst/debug/negotiation.c: (gst_negotiation_class_init):
* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
* gst/goom/gstgoom.c: (gst_goom_class_init):
* gst/id3demux/gstid3demux.c: (gst_id3demux_class_init):
* gst/interleave/deinterleave.c: (deinterleave_class_init):
* gst/interleave/interleave.c: (interleave_class_init):
* gst/law/alaw-decode.c: (gst_alawdec_class_init):
* gst/law/alaw-encode.c: (gst_alawenc_class_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_class_init):
* gst/median/gstmedian.c: (gst_median_class_init):
* gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init):
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init):
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init):
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init):
* gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init):
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init):
* gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init):
* gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init):
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init):
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init):
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init):
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init):
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init):
* gst/smpte/gstsmpte.c: (gst_smpte_class_init):
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
* gst/udp/gstudpsink.c: (gst_udpsink_class_init):
* gst/videomixer/videomixer.c: (gst_videomixer_class_init):
* gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
* sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init):
* sys/oss/gstosssink.c: (gst_oss_sink_class_init):
* sys/osxaudio/gstosxaudioelement.c:
(gst_osxaudioelement_class_init):
* sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init):
* sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init):
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
 2006-04-08 21:21:45 +00:00
Edward Hervey
50f4eb876a Fix memleak with gst_static_pad_template_get(). 
Original commit message from CVS:
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_init):
* ext/dv/gstdvdemux.c: (gst_dvdemux_init), (gst_dvdemux_add_pads):
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init):
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
(gst_jpeg_dec_setcaps):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_init):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_init):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init):
* ext/libmng/gstmngdec.c: (gst_mngdec_init),
(gst_mngdec_src_getcaps):
* ext/libpng/gstpngdec.c: (gst_pngdec_init),
(gst_pngdec_caps_create_and_set):
* ext/libpng/gstpngenc.c: (gst_pngenc_init):
* ext/mikmod/gstmikmod.c: (gst_mikmod_init):
* ext/speex/gstspeexdec.c: (gst_speex_dec_init):
* gst/alpha/gstalpha.c: (gst_alpha_init):
* gst/auparse/gstauparse.c: (gst_au_parse_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_init),
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream):
* gst/cutter/gstcutter.c: (gst_cutter_init):
* gst/debug/efence.c: (gst_efence_init), (gst_efence_getrange),
(gst_efence_checkgetrange):
* gst/debug/negotiation.c: (gst_negotiation_init):
* gst/flx/gstflxdec.c: (gst_flxdec_init):
* gst/goom/gstgoom.c: (gst_goom_init):
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_init):
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init):
* gst/rtp/gstrtpdepay.c: (gst_rtp_depay_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_init):
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init):
* gst/smpte/gstsmpte.c: (gst_smpte_init):
* gst/wavparse/gstwavparse.c: (gst_wavparse_init),
(gst_wavparse_create_sourcepad):
Fix memleak with gst_static_pad_template_get().
This uses gst_pad_new_from_static_template() instead.
Fixes #333512
 2006-03-15 16:17:12 +00:00
Wim Taymans
946e1e4363 gst/rtsp/: Resurected rtpdec to make rtspsrc happy again. 
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_get_type),
(gst_rtpdec_class_init), (gst_rtpdec_init), (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp),
(gst_rtpdec_set_property), (gst_rtpdec_get_property),
(gst_rtpdec_change_state):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
* gst/rtsp/rtspconnection.c: (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspmessage.c: (rtsp_message_dump):
Resurected rtpdec to make rtspsrc happy again.
Skip attributes from the session id.
Don't crash when dumping a message with an empty body.
 2006-02-09 14:20:14 +00:00