1. 23 Feb, 2007 2 commits
    • Jan Schmidt's avatar
      gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken... · 825cf238
      Jan Schmidt authored
      gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken from icecast to replace it. Relicensed fr...
      
      Original commit message from CVS:
      * gst/rtsp/Makefile.am:
      * gst/rtsp/rtspconnection.c: (append_auth_header),
      (rtsp_connection_send), (rtsp_connection_set_auth):
      g_base64_encode is a GLib 2.12 function. Use an equivalent taken
      from icecast to replace it. Relicensed from GPL courtesy of Mike
      Smith.
      825cf238
    • Jan Schmidt's avatar
      gst/rtsp/: Implement simple Basic Authentication support so that urls like... · 66df66da
      Jan Schmidt authored
      gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
      (gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
      (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
      (gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
      (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
      (gst_rtspsrc_uri_set_uri):
      * gst/rtsp/gstrtspsrc.h:
      * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
      (append_auth_header), (rtsp_connection_send),
      (rtsp_connection_free), (rtsp_connection_set_auth):
      * gst/rtsp/rtspconnection.h:
      * gst/rtsp/rtspdefs.h:
      * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
      * gst/rtsp/rtspurl.h:
      Implement simple Basic Authentication support so that urls like
      rtsp://user:pass@hostname/rtspstream work on hosts that require
      authentication.
      66df66da
  2. 11 Feb, 2007 1 commit
    • Sébastien Moutte's avatar
      gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it. · 9c8ea356
      Sébastien Moutte authored
      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.
      9c8ea356
  3. 10 Jan, 2007 1 commit
    • Peter Kjellerstedt's avatar
      gst/rtsp/: Allow url to be NULL to be able to use it for server connections. · 12ab127d
      Peter Kjellerstedt authored
      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.
      12ab127d
  4. 08 Jan, 2007 1 commit
    • Vincent Torri's avatar
      ext/jpeg/: These libjpeg callbacks should return a 'boolean' (unsigned char... · fd185066
      Vincent Torri authored
      ext/jpeg/: These libjpeg callbacks should return a 'boolean' (unsigned char 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.
      fd185066
  5. 15 Nov, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/rtspconnection.c: Don't set a data pointer to NULL and a size > 0... · 0cbacacb
      Wim Taymans authored
      gst/rtsp/rtspconnection.c: Don't set a data pointer to NULL and a size > 0 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.
      0cbacacb
  6. 18 Oct, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Reuse already existing enum for lower transport. · b14738fb
      Wim Taymans authored
      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.
      b14738fb
  7. 16 Oct, 2006 1 commit
    • Josep Torra Valles's avatar
      Fix a bunch of problems discovered by the Forte compiler, mostly type mixups... · c4e7ebfe
      Josep Torra Valles authored
      Fix a bunch of problems discovered by the Forte compiler, mostly type mixups 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.
      c4e7ebfe
  8. 06 Oct, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try to... · a600d311
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try 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.
      a600d311
  9. 29 Sep, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Fix flag registration. · e8c59d9d
      Wim Taymans authored
      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'
      e8c59d9d
  10. 23 Sep, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Improve error reporting. · 23ec2eb1
      Wim Taymans authored
      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.
      23ec2eb1
  11. 20 Sep, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/URLS: Added some test URLS. · a365a29c
      Wim Taymans authored
      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.
      a365a29c
  12. 18 Sep, 2006 2 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation. · a437e9f0
      Wim Taymans authored
      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.
      a437e9f0
    • Thijs Vermeir's avatar
      gst/rtsp/: Small cleanups. when multicast is selected as the transport, create... · 7484c92d
      Thijs Vermeir authored
      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.
      7484c92d
  13. 24 Jul, 2006 1 commit
  14. 10 Jul, 2006 1 commit
  15. 20 Jun, 2006 2 commits
  16. 09 Feb, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Resurected rtpdec to make rtspsrc happy again. · 946e1e43
      Wim Taymans authored
      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.
      946e1e43
  17. 16 Dec, 2005 1 commit
  18. 27 Nov, 2005 1 commit
  19. 21 Nov, 2005 1 commit
  20. 19 Aug, 2005 1 commit
    • Wim Taymans's avatar
      ext/amrnb/: Update caps with audio/AMR. · f48c4cbe
      Wim Taymans authored
      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.
      f48c4cbe
  21. 29 Jun, 2005 1 commit
    • Andy Wingo's avatar
      configure.ac (GST_CFLAGS): GCC strikes back!!! Let the build breakage ensue!!! · 2d109a18
      Andy Wingo authored
      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.
      2d109a18
  22. 11 May, 2005 2 commits
    • Wim Taymans's avatar
      gst/rtsp/: Added README · 63177e07
      Wim Taymans authored
      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.
      63177e07
    • Wim Taymans's avatar
      Ported to 0.9. · 6f0ea358
      Wim Taymans authored
      Original commit message from CVS:
      Ported to 0.9.
      Set up transports, init UDP ports, init RTP session managers.
      6f0ea358