| Commit message (Collapse) | Author | Age | Files | Lines |
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Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/base64.c:
* gst/rtsp/base64.h:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
(gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
(gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
(gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
(gst_rtsp_ext_list_setup_media),
(gst_rtsp_ext_list_configure_stream),
(gst_rtsp_ext_list_get_transports),
(gst_rtsp_ext_list_stream_select):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
(gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspconnection.c:
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c:
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtspmessage.c:
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsprange.c:
* gst/rtsp/rtsprange.h:
* gst/rtsp/rtsptransport.c:
* gst/rtsp/rtsptransport.h:
* gst/rtsp/rtspurl.c:
* gst/rtsp/rtspurl.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.c:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/test.c:
Use shiny new RTSP and SDP library.
Implement RTSP extensions using the new interface.
Remove a lot of old code.
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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.
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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.
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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.
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Original commit message from CVS:
* ext/libpng/gstpngenc.c: (gst_pngenc_chain):
* gst/avi/gstavidemux.c: (gst_avi_demux_invert):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps):
* gst/rtsp/sdpmessage.h:
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_render):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_get_stats):
C89 fixes: declare variables at the beginning of a block and
make gcc-2.9x happy (#328264; patch by: Jens Granseuer
<jensgr at gmx dot net>).
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Original commit message from CVS:
expand tabs
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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.
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Original commit message from CVS:
Ported to 0.9.
Set up transports, init UDP ports, init RTP session managers.
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