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* gst/rtsp/URLS: Add example H264 rtsp url.Wim Taymans2007-02-162-18/+26
| | | | | | | | | | Original commit message from CVS: * gst/rtsp/URLS: Add example H264 rtsp url. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): Don't convert values to lowercase or we might mess up base64 encoded properties.
* gst/rtsp/sdpmessage.c: Clear stack allocated SDPMedia struct before calling ↵Wim Taymans2007-02-141-0/+7
| | | | | | | | | | _init() on it. Original commit message from CVS: * gst/rtsp/sdpmessage.c: (sdp_parse_line): As spotted by: Peter Kjellerstedt <pkj at axis com>: Clear stack allocated SDPMedia struct before calling _init() on it. Clarify this in the docs as well.
* gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793.jp.liu2007-02-142-40/+204
| | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init), (sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init), (sdp_key_init), (sdp_attribute_init), (sdp_message_init), (sdp_message_uninit), (sdp_message_free), (sdp_media_init), (sdp_media_uninit), (sdp_media_free), (sdp_message_add_media), (sdp_parse_line): * gst/rtsp/sdpmessage.h: Based on patch by: jp.liu <jp_liu at astrocom dot cn> Fix memory management of SDP messages. Fixes #407793.
* gst/rtsp/rtspurl.c: Fix parsing of password field in url. Fixes #407797.jp.liu2007-02-141-1/+1
| | | | | | | Original commit message from CVS: Patch by: jp.liu <jp_liu at astrocom dot cn> * gst/rtsp/rtspurl.c: (rtsp_url_parse): Fix parsing of password field in url. Fixes #407797.
* gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 ↵Sébastien Moutte2007-02-114-6/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | seems to do not support it. Original commit message from CVS: * gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it. * gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp): Use gst_guint64_to_gdouble for conversion. * gst/rtsp/rtspconnection.c:(rtsp_connection_send): Move variables declaration before the first instruction. * gst/rtsp/rtspdefs.c:(rtsp_strresult): Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported. And don't include netdb.h for G_OS_WIN32 * gst/rtsp/sdpmessage.c:(sdp_parse_line): This initialization SDPMedia nmedia = {.media = NULL }; is not supported by VS6 then use an other way to initialize SDPMedia structure. * gst/udp/gstdynudpsink.h: * gst/udp/gstdynudpnetutils.h: Do not include <sys/time.h> for G_OS_WIN32 * gst/udp/gstudpsrc.c: Define socklen_t as int for G_OS_WIN32 * win/common/config.h.in: Undef HAVE_NETINET_IN_H * win32/vs6/gst_plugins_good.dsw: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: * win32/vs6/libgstautogen.dsp: * win32/vs6/libgstaudiofx.dsp: * win32/vs6/libgstudp.dsp: Add and update project files. * win32/common/gstudp-enumtypes.c: * win32/common/gstudp-enumtypes.h: Add a copy of udp enumtypes to win32/common as in core and base.
* gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the ↵Wim Taymans2007-01-251-7/+19
| | | | | | | | | | rules in the SDP to caps document. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_activate_streams): Convert SDP fields to upper/lowercase following the rules in the SDP to caps document.
* gst/rtsp/gstrtspsrc.*: Only unblock the udp pads when we linked and ↵Wim Taymans2007-01-242-9/+22
| | | | | | | | | | | | activated them all. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp): * gst/rtsp/gstrtspsrc.h: Only unblock the udp pads when we linked and activated them all. Fixes #395688.
* gst/rtsp/gstrtspsrc.c: Unblock pads after adding the pads to the element so ↵Wim Taymans2007-01-241-4/+7
| | | | | | | | | that autopluggers get a change to link so... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked): Unblock pads after adding the pads to the element so that autopluggers get a change to link something. Possibly fixes #395688.
* gst/rtsp/gstrtspsrc.c: Check for stream pad before activating.Lutz Mueller2007-01-111-5/+7
| | | | | | | Original commit message from CVS: Patch by: Lutz Mueller <lutz at topfrose dot de> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams): Check for stream pad before activating.
* gst/rtsp/: Allow url to be NULL to be able to use it for server connections.Peter Kjellerstedt2007-01-1010-299/+644
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/COPYING.MIT: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_send), (read_line), (parse_request_line), (parse_line), (rtsp_connection_read), (rtsp_connection_close): * gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult), (rtsp_method_as_text), (rtsp_header_as_text), (rtsp_status_as_text), (rtsp_find_header_field), (rtsp_find_method): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send), (rtsp_ext_wms_configure_stream): * gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init), (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_get_header), (rtsp_message_set_body), (rtsp_message_get_body), (dump_mem), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_media_get_attribute_val_n), (read_string), (read_string_del), (sdp_parse_line), (sdp_message_parse_buffer), (print_media), (sdp_message_dump): Allow url to be NULL to be able to use it for server connections. Can now send responses as well as requests. No longer hangs in an endless loop if EOF is received. Can now convert a status code to a text string. Return RTSP_HDR_INVALID for unknown headers. Return RTSP_INVALID for unknown methods. Copy CSeq and Session headers from the request. Only free memory corresponding to the currently set message type. Added const to function arguments as appropriate. Avoid a compiler warning when initializing nmedia. Use guint rather than gint to avoid compiler warnings. Fix crasher in wms extension. Factor out stream setup from open_connection. Delay activation of streams when actual data is received from the server, this prepares us to do proper protocol switching. Added new license. Fixes #380895.
* ext/jpeg/: These libjpeg callbacks should return a 'boolean' (unsigned char ↵Vincent Torri2007-01-081-2/+9
| | | | | | | | | | | | | | | | | apparently) and not a 'gboolean' (which m... Original commit message from CVS: Patch by: Vincent Torri <vtorri at univ-evry fr> * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/smokecodec.c: These libjpeg callbacks should return a 'boolean' (unsigned char apparently) and not a 'gboolean' (which maps to gint). Fixes warnings when compiling with MingW (#393427). * gst/rtsp/rtspconnection.c: (rtsp_connection_read): Use ioctlsocket on win32. * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Some printf format fixes for win32.
* gst/rtsp/: Add method so that extensions can choose to disable the setup of ↵Wim Taymans2006-11-284-9/+43
| | | | | | | | | | | | | | a stream. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspext.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream), (rtsp_ext_wms_get_context): Add method so that extensions can choose to disable the setup of a stream. Make the WMS extension skip setup of x-wms-rtx streams. Fixes #377792.
* gst/rtsp/rtspconnection.c: Don't set a data pointer to NULL and a size > 0 ↵Wim Taymans2006-11-152-1/+4
| | | | | | | | | | | | | | when we deal with empty packets. Original commit message from CVS: * gst/rtsp/rtspconnection.c: (read_body): Don't set a data pointer to NULL and a size > 0 when we deal with empty packets. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_take_body): Check that we can't create invalid empty packets.
* gst/rtsp/: Reuse already existing enum for lower transport.Wim Taymans2006-10-185-34/+56
| | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type), (gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_open), (gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri): * gst/rtsp/rtspconnection.c: (rtsp_connection_create): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspurl.c: (rtsp_url_parse): * gst/rtsp/rtspurl.h: Reuse already existing enum for lower transport. Add rtspt and rtspu protocols. Send redirect to rtspt when udp times out.
* Fix a bunch of problems discovered by the Forte compiler, mostly type mixups ↵Josep Torra Valles2006-10-162-2/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | and pointer arithmetics with void pointe... Original commit message from CVS: Patch by: Josep Torra Valles <josep at fluendo com> * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform): * ext/esd/esdsink.c: (gst_esdsink_write): * ext/flac/gstflacdec.c: (gst_flac_dec_length), (gst_flac_dec_read_seekable), (gst_flac_dec_chain), (gst_flac_dec_send_newsegment): * ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback), (gst_flac_enc_tell_callback): * ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode), (smokecodec_parse_header), (smokecodec_decode): * gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index): * gst/debug/efence.c: (gst_fenced_buffer_alloc): * gst/goom/Makefile.am: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward): * gst/rtsp/gstrtspsrc.c: * gst/rtsp/rtspconnection.c: (rtsp_connection_read): * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_change_state): * sys/sunaudio/gstsunaudiomixertrack.h: Fix a bunch of problems discovered by the Forte compiler, mostly type mixups and pointer arithmetics with void pointers. Fixes #362603.
* gst/rtsp/URLS: Added some other URL.Wim Taymans2006-10-116-14/+135
| | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/URLS: Added some other URL. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp), (gst_rtspsrc_handle_request), (gst_rtspsrc_send), (gst_rtspsrc_open), (gst_rtspsrc_play), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Work on fallback to TCP connection when the UDP socket times out. Handler server requests, just reply with OK for now. * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: Added some more Real extension headers. * gst/rtsp/rtspurl.c: (rtsp_url_parse): Fix parsing of urls with a ':' that is not part of the hostname:port part of the url.
* gst/rtsp/gstrtspsrc.c: Activate pads before adding them to the source.Tim-Philipp Müller2006-10-071-0/+2
| | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport): Activate pads before adding them to the source.
* gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try ↵Wim Taymans2006-10-069-241/+552
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | to share channels and udp ports. 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_media_to_caps), (gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_stream_configure_transport), (find_stream_by_channel), (gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_configure_transports), (gst_rtspsrc_open), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Rework how the transport string is constructed, try to share channels and udp ports. Make most of the stuff less dependant on RTP as we are also going to use it for RDT. Add support for transport specific session managers. * gst/rtsp/rtspconnection.c: (rtsp_connection_flush): Implement _flush(). * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: Add generic error return code. * gst/rtsp/rtspext.h: Add support for pluggable tranport strings. * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send), (rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp), (rtsp_ext_wms_get_context): Detect WMServer and activate the extension. * gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime), (rtsp_transport_get_manager), (rtsp_transport_parse): * gst/rtsp/rtsptransport.h: Added methods to get mime/manager for certain transports.
* Printf format fixes.Tim-Philipp Müller2006-10-051-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_update_font_height): * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps): * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain): * ext/libpng/gstpngdec.c: (user_endrow_callback): * gst/auparse/gstauparse.c: (gst_au_parse_parse_header): * gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_stream_data): * gst/cutter/gstcutter.c: (gst_cutter_chain): * gst/debug/efence.c: (gst_efence_buffer_alloc), (gst_fenced_buffer_copy): * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame): * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream): * gst/matroska/matroska-mux.c: (gst_matroska_mux_start): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_handle_message): * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): * sys/ximage/ximageutil.c: (ximageutil_xcontext_get): Printf format fixes.
* gst/rtsp/Makefile.am: Dist new .h file too.Wim Taymans2006-10-041-1/+1
| | | | | | Original commit message from CVS: * gst/rtsp/Makefile.am: Dist new .h file too.
* gst/rtsp/: Factor out extension in separate module.Wim Taymans2006-10-0411-67/+425
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtsp/URLS: Add some more URLs.Wim Taymans2006-09-293-68/+188
| | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/URLS: Add some more URLs. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add timeout property to control UDP timeouts. Fix error messages. Also start a loop function when operating in UDP mode so that we can do some more stuff async. Handle element messages from udpsrc to detect timeouts. If a timeout happens we currently generate an error. API: rtspsrc::timeout property. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create): Really implement the timeout in microseconds and not milliseconds.
* gst/rtsp/gstrtspsrc.c: Fix flag registration.Wim Taymans2006-09-292-4/+4
| | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type): Fix flag registration. * gst/rtsp/rtspconnection.c: (rtsp_connection_read): Reading 0 also means 'no more commands'
* gst/: Include stdlib.h in some more places, makes things compile with uClibc ↵Peter Kjellerstedt2006-09-251-0/+1
| | | | | | | | | | | | | | and -Werror (#357592). Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/alpha/gstalpha.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsrc.c: * gst/videomixer/videomixer.c: Include stdlib.h in some more places, makes things compile with uClibc and -Werror (#357592).
* gst/rtsp/: Improve error reporting.Wim Taymans2006-09-234-23/+89
| | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_open): * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: Improve error reporting.
* gst/rtsp/URLS: Added some test URLS.Wim Taymans2006-09-2022-201/+946
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation.Wim Taymans2006-09-193-84/+172
| | | | | | | | | | | | | | | | | | | | 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.
* gst/rtsp/test.c: Fix build.Wim Taymans2006-09-191-3/+3
| | | | | | Original commit message from CVS: * gst/rtsp/test.c: (main): Fix build.
* gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation.Wim Taymans2006-09-189-306/+426
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtsp/gstrtspsrc.*: Export sometimes source pad with correct caps on the ↵Wim Taymans2006-09-182-19/+68
| | | | | | | | | | | | | | | | 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.
* gst/rtsp/gstrtspsrc.*: Make sure we can never set an invalid location.Lutz Mueller2006-09-185-14/+70
| | | | | | | | | | | | | | | 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.
* gst/rtsp/gstrtspsrc.*: Use boilerplate.Lutz Mueller2006-09-182-100/+31
| | | | | | | | | | | | | 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.
* gst/rtsp/: Small cleanups. when multicast is selected as the transport, ↵Thijs Vermeir2006-09-182-84/+115
| | | | | | | | | | | | | | | | | 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.
* docs/plugins/gst-plugins-good-plugins-docs.sgml: There is no taglibmux ↵Tim-Philipp Müller2006-08-221-1/+1
| | | | | | | | | | | 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.
* Small documentation updates.Wim Taymans2006-08-222-4/+15
| | | | | | | | | | | 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.
* gst/rtsp/gstrtpdec.c: Add pads after setting them up.Wim Taymans2006-08-163-69/+205
| | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtsp/rtspconnection.c: Remove unwanted DEBUG line.Wim Taymans2006-07-241-1/+0
| | | | | | Original commit message from CVS: * gst/rtsp/rtspconnection.c: (rtsp_connection_send): Remove unwanted DEBUG line.
* gst/rtsp/gstrtspsrc.c: Don't try doing state changes on a NULL pointer.Tim-Philipp Müller2006-07-151-4/+6
| | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state): Don't try doing state changes on a NULL pointer.
* gst/rtsp/: replaced closesocket and close in code with one CLOSE_SOCKET.Wim Taymans2006-07-103-22/+30
| | | | | | | | | | Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/rtspconnection.c: (rtsp_connection_send), (rtsp_connection_close): * gst/rtsp/rtspdefs.h: replaced closesocket and close in code with one CLOSE_SOCKET. Some more cleanups. Fixes #345301.
* Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) plus two minor macro ↵Tim-Philipp Müller2006-06-222-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* Added documentation for the rtsp plugin. Fixes #345393.Wim Taymans2006-06-203-9/+62
| | | | | | | | | | | 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.
* gst/rtsp/rtspconnection.c: Use better G_OS_* macros. Fixes #345301 some more.Wim Taymans2006-06-201-7/+18
| | | | | | | Original commit message from CVS: * gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send), (rtsp_connection_close), (rtsp_connection_free): Use better G_OS_* macros. Fixes #345301 some more.
* gst/rtsp/rtspconnection.c: Make RTSP plugin compile on windows. Fixes #345301.Joni Valtanen2006-06-201-1/+49
| | | | | | | | | | Original commit message from CVS: Patch by: Joni Valtanen <joni dot valtanen at movial dot fi> * gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send), (rtsp_connection_close): Make RTSP plugin compile on windows. Fixes #345301. Some changes to original patch to catch errors better. use ifdef WIN32 instead of ifndef.
* Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClassStefan Kost2006-06-012-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * ext/aalib/gstaasink.h: * ext/annodex/gstcmmldec.h: * ext/cairo/gsttimeoverlay.h: * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.h: * ext/esd/esdmon.h: * ext/esd/esdsink.h: * ext/flac/gstflacenc.h: * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstgconfaudiosrc.h: * ext/gconf/gstgconfvideosink.h: * ext/gconf/gstgconfvideosrc.h: * ext/gdk_pixbuf/gstgdkanimation.h: * ext/gdk_pixbuf/pixbufscale.h: * ext/hal/gsthalaudiosink.h: * ext/hal/gsthalaudiosrc.h: * ext/jpeg/gstjpegenc.h: * ext/jpeg/gstsmokedec.h: * ext/jpeg/gstsmokeenc.h: * ext/libcaca/gstcacasink.h: * ext/libmng/gstmngdec.h: * ext/libmng/gstmngenc.h: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.h: * ext/raw1394/gstdv1394src.h: * ext/speex/gstspeexenc.h: * gst/autodetect/gstautoaudiosink.h: * gst/autodetect/gstautovideosink.h: * gst/avi/gstavidemux.h: * gst/cutter/gstcutter.h: * gst/debug/efence.h: * gst/debug/gstnavigationtest.h: * gst/debug/gstnavseek.h: * gst/flx/gstflxdec.h: * gst/goom/gstgoom.h: * gst/icydemux/gsticydemux.h: * gst/id3demux/gstid3demux.h: * gst/law/alaw-decode.h: * gst/law/alaw-encode.h: * gst/law/mulaw-decode.h: * gst/law/mulaw-encode.h: * gst/matroska/matroska-mux.h: * gst/median/gstmedian.h: * gst/oldcore/gstaggregator.h: * gst/oldcore/gstfdsink.h: * gst/oldcore/gstmd5sink.h: * gst/oldcore/gstmultifilesrc.h: * gst/oldcore/gstpipefilter.h: * gst/oldcore/gstshaper.h: * gst/oldcore/gststatistics.h: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.h: * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtspsrc.h: * gst/smpte/gstsmpte.h: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.h: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideoflip.h: * sys/oss/gstossdmabuffer.h: * sys/oss/gstossmixerelement.h: * sys/oss/gstosssink.h: * sys/oss/gstosssrc.h: * sys/osxvideo/osxvideosink.h: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiosink.h: * sys/ximage/gstximagesrc.h: Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
* Const-ify GEnumValue arrays.Tim-Philipp Müller2006-05-101-1/+1
| | | | | | | | | | | | | | | | | | 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.
* gst/rtsp/rtspurl.c: Make parsing of urls suck slightly less.Wim Taymans2006-05-081-9/+16
| | | | | | Original commit message from CVS: * gst/rtsp/rtspurl.c: (rtsp_url_parse): Make parsing of urls suck slightly less.
* Define GstElementDetails as const and also static (when defined as global)Stefan Kost2006-04-252-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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)
* Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)Stefan Kost2006-04-082-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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)
* gst/rtsp/gstrtspsrc.c: the OPTIONS request result is optional so don't fail ↵Wim Taymans2006-03-211-6/+5
| | | | | | | | | 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.
* Fix memleak with gst_static_pad_template_get().Edward Hervey2006-03-151-8/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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