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137 commits

Author SHA1 Message Date
Wim Taymans
a7d7309e18 gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Reorganize stream parsing and creation.
Detect container formats in interleaved mode.
Keep more state about the streams.
Assume a server also supports PLAY if it does not say.
Add unicast and interleaved properties to TCP transport requests to make
some servers happy (WMServer).
* gst/rtsp/sdpmessage.h:
Add some defines for the standard Bandwidth types.
 2006-09-19 17:25:15 +00:00
Wim Taymans
a437e9f0ed gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Small cleanups, added documentation.
Try to clean up the requests and responses.
Refactor parsing the supported methods.
* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
(rtsp_connection_create), (rtsp_connection_send),
(parse_response_status), (parse_request_line),
(rtsp_connection_receive), (rtsp_connection_close),
(rtsp_connection_free):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (rtsp_transport_parse),
(rtsp_transport_free):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
(sdp_message_clean), (sdp_message_free), (sdp_media_new),
(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
Use g_return_val some more.
* gst/rtsp/rtspdefs.h:
Add more enum values to track initial states.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
(rtsp_message_init_request), (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free),
(rtsp_message_add_header), (rtsp_message_remove_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_take_body), (rtsp_message_get_body),
(rtsp_message_steal_body), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Reorder arguments, object goes as the first one.
Use g_return_val some more.
 2006-09-18 17:37:46 +00:00
Wim Taymans
108dbd54cf gst/rtsp/gstrtspsrc.*: Export sometimes source pad with correct caps on the template, create the ghostpad from the te... 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Export sometimes source pad with correct caps on the template, create
the ghostpad from the template.
Remove RTCP template as we never expose RTCP.
Protect against invalid body size.
Avoid memcpy when creating the output buffer.
Properly post an error and send EOS when the loop function is shut down.
 2006-09-18 14:00:41 +00:00
Lutz Mueller
cac807b641 gst/rtsp/gstrtspsrc.*: Make sure we can never set an invalid location. 
Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Make sure we can never set an invalid location.
* gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
* gst/rtsp/rtspmessage.h:
Added _steal_body method for future use.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
Make freeing of NULL url return immediatly.
 2006-09-18 11:29:12 +00:00
Lutz Mueller
afd156ad0c gst/rtsp/gstrtspsrc.*: Use boilerplate. 
Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Use boilerplate.
Make rtspsrc subclass GstBin to make state changes easier.
Add Range header field on the PLAY request.
 2006-09-18 10:42:52 +00:00
Thijs Vermeir
7484c92dfe gst/rtsp/: Small cleanups. when multicast is selected as the transport, create UDP sources and connect to the multica... 
Original commit message from CVS:
Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
* gst/rtsp/rtspconnection.c: (inet_aton):
Small cleanups.
when multicast is selected as the transport, create UDP sources and
connect to the multicast group.
Move parsing and setting of caps to a common place.
Fixes #349894.
 2006-09-18 08:59:17 +00:00
Tim-Philipp Müller
04895ee2ca docs/plugins/gst-plugins-good-plugins-docs.sgml: There is no taglibmux element ... 
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
There is no taglibmux element ...
* gst/rtsp/gstrtspsrc.c:
Use '%' rather than '&perc;' in gtk-doc blurb, docs build
was complaining about unknown entity here.
 2006-08-22 17:20:41 +00:00
Wim Taymans
0f38451f20 Small documentation updates. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
* sys/oss/gstosssink.c: (gst_oss_sink_open),
(gst_oss_sink_prepare), (gst_oss_sink_unprepare):
Small documentation updates.
 2006-08-22 16:45:37 +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
4965e7c58d gst/rtsp/gstrtspsrc.c: Don't try doing state changes on a NULL pointer. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state):
Don't try doing state changes on a NULL pointer.
 2006-07-15 15:25:05 +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
Tim-Philipp Müller
11cb7a31b4 Const-ify GEnumValue arrays. 
Original commit message from CVS:
* ext/esd/esdmon.c: (gst_esdmon_depths_get_type),
(gst_esdmon_channels_get_type):
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_profile_get_type):
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_method_get_type):
* ext/libcaca/gstcacasink.c: (gst_cacasink_dither_get_type):
* ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type):
* gst/alpha/gstalpha.c: (gst_alpha_method_get_type):
* gst/rtp/gstrtpilbcdepay.c: (gst_ilbc_mode_get_type):
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
* gst/videobox/gstvideobox.c: (gst_video_box_fill_get_type):
* gst/videofilter/gstvideoflip.c: (gst_video_flip_method_get_type):
* gst/videomixer/videomixer.c:
(gst_video_mixer_background_get_type):
Const-ify GEnumValue arrays.
 2006-05-10 10:29:54 +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
Wim Taymans
6d9a3ecc4c gst/rtsp/gstrtspsrc.c: the OPTIONS request result is optional so don't fail on it. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
the OPTIONS request result is optional so don't
fail on it.
 2006-03-21 16:19:37 +00:00
Wim Taymans
d465618d5b gst/rtsp/README: Updated README. 
Original commit message from CVS:
* gst/rtsp/README:
Updated README.

* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp):
* gst/rtsp/gstrtspsrc.h:
Make sure the RTP port is an even port an try to allocate
another if not.
Added retry property to control max retries for port allocation.
Make sure RTCP port is RTP port+1.
Cleanup when port allocation fails.
Fixes #319183.
 2006-02-16 10:42:25 +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
Thomas Vander Stichele
3ecf433432 expand tabs 
Original commit message from CVS:
expand tabs
 2005-12-06 19:44:58 +00:00
Wim Taymans
001a51dba2 gst/rtsp/gstrtspsrc.c: Strip spaces for key/value pairs. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_send):
Strip spaces for key/value pairs.
 2005-09-21 19:41:45 +00:00
Wim Taymans
485d25aef1 gst/rtsp/gstrtspsrc.c: More SDP parsing and caps setting. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_change_state):
More SDP parsing and caps setting.
Do NO_PREROLL differently.
add pads only after negotiated.

* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_getcaps):
Implement the getcaps function.
 2005-09-21 17:53:26 +00:00
Wim Taymans
9dd3929730 gst/rtp/README: Update README 
Original commit message from CVS:
* gst/rtp/README:
Update README

* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps):
Make extra params as strings.

* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send):
Make state change return NO_PREROLL as this is a live
source.

* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
Don't unref old caps when NULL.
 2005-09-21 12:19:24 +00:00
Wim Taymans
eb20f045f8 gst/rtsp/: Add URI handler. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send),
(gst_rtspsrc_open), (gst_rtspsrc_uri_get_type),
(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_get_uri),
(gst_rtspsrc_uri_set_uri), (gst_rtspsrc_uri_handler_init):
* gst/rtsp/sdpmessage.c: (sdp_media_get_format):
* gst/rtsp/sdpmessage.h:
Add URI handler.
Parse SDP and create caps.
 2005-09-20 17:35:11 +00:00
Andy Wingo
7ebd7b97d4 All plugins updated for element state changes. 
Original commit message from CVS:
2005-09-02  Andy Wingo  <wingo@pobox.com>

* All plugins updated for element state changes.
 2005-09-02 15:44:50 +00:00
Wim Taymans
f48c4cbe42 ext/amrnb/: Update caps with audio/AMR. 
Original commit message from CVS:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c: (gst_amrnbenc_setcaps):
* ext/amrnb/amrnbparse.c:
Update caps with audio/AMR.

* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain),
(gst_rtpamrdec_change_state):
* gst/rtp/gstrtpamrdec.h:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
Dont set FT headers twice, it was already in the encoded
bitstream.

* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
* gst/rtsp/rtspconnection.c: (parse_line):
Cleanups

* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property):
* gst/udp/gstudpsrc.h:
Added caps property, we need this soon to type the buffers.
 2005-08-19 12:44:35 +00:00
Wim Taymans
3e064477cf gst/rtp/gstrtpamrdec.c: Fix up amr depayloader a bit. 
Original commit message from CVS:
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_chain):
Fix up amr depayloader a bit.

* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Look for options result in Public and Allow header fields..
spec says Allow but some servers return Public...
 2005-08-18 20:14:46 +00:00
Wim Taymans
519f58bb6b gst/rtp/gstrtpamrenc.*: Added payload_type and ssrc properties to the payloader. 
Original commit message from CVS:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain),
(gst_rtpamrenc_set_property), (gst_rtpamrenc_get_property):
* gst/rtp/gstrtpamrenc.h:
Added payload_type and ssrc properties to the payloader.

* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Options need to be stripped and are in the Public header field.

* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix url / parsing...
 2005-08-18 19:47:38 +00:00
Wim Taymans
c831aef495 gst/rtsp/: Handle RTSP defaults better. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_method_as_text), (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Handle RTSP defaults better.
Issue OPTIONS request to figure out what we are allowed to do.
Make the methods a bitfield so we can easily collect supported
options.
Fix rtsp_find_method.
Do proper RTSP connection shutdown.
 2005-08-18 16:53:14 +00:00
Wim Taymans
bf2d582666 gst/rtsp/gstrtspsrc.c: Support absolute control urls too. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open), (gst_rtspsrc_play):
Support absolute control urls too.
 2005-08-17 10:30:55 +00:00
Wim Taymans
bae4e2ad4a gst/rtsp/gstrtspsrc.c: Fix for core changes. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_add_element),
(gst_rtspsrc_play):
Fix for core changes.
 2005-07-18 15:40:36 +00:00
Andy Wingo
2d109a18fb configure.ac (GST_CFLAGS): GCC strikes back!!! Let the build breakage ensue!!! 
Original commit message from CVS:
2005-06-29  Andy Wingo  <wingo@pobox.com>

* configure.ac (GST_CFLAGS): GCC strikes back!!! Let the build
breakage ensue!!!

* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_loop, gst_rtspsrc_open):
Signedness, unused var fixes.
(gst_rtspsrc_close): Unused?

* gst/realmedia/rmdemux.c (re_hexdump_bytes): Unused.

* gst/law/mulaw-encode.c (gst_mulawenc_chain): Signeness fix.

* gst/law/alaw-encode.c (alawenc_getcaps): Remove unneeded
declarations. Typo (probably crasher) fix.

* gst/law/mulaw-encode.c (mulawdec_getcaps):
* gst/law/mulaw-encode.c (mulawenc_getcaps):
* gst/law/alaw-decode.c (alawdec_getcaps): Same crasher fix.

* gst/goom/gstgoom.c (gst_goom_init): Hook up the event function.

* gst/effectv/gstwarp.c (gst_warptv_setup): Signedness fix.

* gst/effectv/gstdice.c (gst_dicetv_draw): Um, deferencing
uninitialized pointer not good.

* gst/videofilter/gstvideoexample.c (plugin_init):
* gst/videofilter/Makefile.am (libgstvideoexample_la_LIBADD): Link
to libgstvideofilter instead of gst_library_load.

* gst/alpha/gstalpha.c (gst_alpha_chroma_key_i420)
(gst_alpha_chroma_key_ayuv): Signedness fixen.
 2005-06-29 16:14:30 +00:00
Wim Taymans
8f2272a99d gst/rtsp/gstrtspsrc.c: Fix case where outpad could not be decided. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
Fix case where outpad could not be decided.
 2005-06-29 16:09:55 +00:00
Wim Taymans
c51c1a4c38 gst/rtsp/: Make rtspsrc a live source. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Make rtspsrc a live source.
Don't try to parse NULL urls.
 2005-06-23 14:30:13 +00:00
Wim Taymans
ca1f865eaa gst/rtsp/: RTSP cleanups. 
Original commit message from CVS:
* gst/rtsp/README:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (parse_mode), (parse_range),
(rtsp_transport_parse), (rtsp_transport_free):
RTSP cleanups.
 2005-06-02 13:26:36 +00:00
Wim Taymans
63177e0731 gst/rtsp/: Added README 
Original commit message from CVS:
* gst/rtsp/README:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play):
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_send), (read_line), (parse_request_line),
(parse_line), (read_body), (rtsp_connection_receive),
(rtsp_connection_free):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.c: (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspmessage.c: (rtsp_message_set_body),
(rtsp_message_take_body):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtspstream.h:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
Added README
Some cleanups.
 2005-05-11 12:01:10 +00:00
Wim Taymans
91ce2b294e gst/rtsp/gstrtspsrc.*: Setup UDP sources correctly, receives raw data from RTSP compliant servers now. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
(gst_rtspsrc_create_stream), (gst_rtspsrc_add_element),
(gst_rtspsrc_set_state), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Setup UDP sources correctly, receives raw data from RTSP
compliant servers now.
 2005-05-11 09:18:25 +00:00
Wim Taymans
6f0ea35883 Ported to 0.9. 
Original commit message from CVS:
Ported to 0.9.
Set up transports, init UDP ports, init RTP session managers.
 2005-05-11 07:44:44 +00:00