1. 06 Oct, 2006 3 commits
    • Wim Taymans's avatar
      Activate pads before adding. · 3adedd4f
      Wim Taymans authored
      Original commit message from CVS:
      * ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads), (gst_dvdemux_chain):
      * gst/auparse/gstauparse.c: (gst_au_parse_add_srcpad):
      Activate pads before adding.
      3adedd4f
    • Wim Taymans's avatar
      gst/multipart/multipartdemux.c: Activate pads before adding. · 09328ad0
      Wim Taymans authored
      Original commit message from CVS:
      * gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
      (gst_multipart_find_pad_by_mime):
      Activate pads before adding.
      * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
      BOILERPLATE sets parent_class for us.
      09328ad0
    • 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
  2. 05 Oct, 2006 1 commit
    • Tim-Philipp Müller's avatar
      Printf format fixes. · 82f5a350
      Tim-Philipp Müller authored
      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.
      82f5a350
  3. 04 Oct, 2006 4 commits
    • Wim Taymans's avatar
      gst/rtsp/Makefile.am: Dist new .h file too. · a0ff313a
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/Makefile.am:
      Dist new .h file too.
      a0ff313a
    • Wim Taymans's avatar
      gst/rtsp/: Factor out extension in separate module. · 63c87f18
      Wim Taymans authored
      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.
      63c87f18
    • Thomas Vander Stichele's avatar
      docs/plugins/: added v4l2 stubs · c85684e2
      Thomas Vander Stichele authored
      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:
      added v4l2 stubs
      * gst-plugins-good.spec.in:
      add v4l2
      c85684e2
    • Tim-Philipp Müller's avatar
      gst/apetag/gstapedemux.c: Extract disc/album/medium number and count and try... · 424c5cb6
      Tim-Philipp Müller authored
      gst/apetag/gstapedemux.c: Extract disc/album/medium number and count and try harder to extract track number/count.
      
      Original commit message from CVS:
      * gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
      Extract disc/album/medium number and count and try harder
      to extract track number/count.
      424c5cb6
  4. 03 Oct, 2006 1 commit
  5. 29 Sep, 2006 5 commits
    • Tim-Philipp Müller's avatar
      tests/check/Makefile.am: Disable autodetect test temporarily, so that the... · 475aed8a
      Tim-Philipp Müller authored
      tests/check/Makefile.am: Disable autodetect test temporarily, so that the build bots update -bad and the ranks of unr...
      
      Original commit message from CVS:
      * tests/check/Makefile.am:
      Disable autodetect test temporarily, so that the build bots
      update -bad and the ranks of unreliable video sinks in there.
      * tests/check/elements/autodetect.c: (GST_START_TEST):
      Skip test if no usable videosink is found.
      475aed8a
    • Wim Taymans's avatar
      gst/rtsp/URLS: Add some more URLs. · 6e085503
      Wim Taymans authored
      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.
      6e085503
    • Wim Taymans's avatar
      gst/udp/gstudpsrc.*: Added property to post a message on timeout. · fcd901a5
      Wim Taymans authored
      Original commit message from CVS:
      * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
      (gst_udpsrc_create), (gst_udpsrc_set_property),
      (gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop):
      * gst/udp/gstudpsrc.h:
      Added property to post a message on timeout.
      Updated docs.
      When restarting the select, initialize the fdsets again.
      Init control sockets so we don't accidentally close a random socket.
      API: GstUDPSrc::timeout property
      fcd901a5
    • 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
    • Antoine Tremblay's avatar
      gst/udp/gstudpsrc.c: Fix possible infinite loop when shutting down, a read can... · 1a86fdc6
      Antoine Tremblay authored and Wim Taymans's avatar Wim Taymans committed
      gst/udp/gstudpsrc.c: Fix possible infinite loop when shutting down, a read can also return 0 to indicate no more mess...
      
      Original commit message from CVS:
      Patch by: Antoine Tremblay <hexa00 at gmail dot com>
      * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
      Fix possible infinite loop when shutting down, a read can also return
      0 to indicate no more messages are available. Fixes #358156.
      1a86fdc6
  6. 25 Sep, 2006 3 commits
  7. 23 Sep, 2006 2 commits
    • 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
    • Wim Taymans's avatar
      gst/rtp/: Fix klass typos. · af6e4da9
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
      * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
      * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
      * gst/rtp/gstrtpdepay.c:
      * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
      * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
      * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
      * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
      * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
      (gst_rtp_mp2t_depay_plugin_init):
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
      * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
      * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
      * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
      * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
      * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
      * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
      * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
      Fix klass typos.
      Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.
      af6e4da9
  8. 22 Sep, 2006 5 commits
    • Tim-Philipp Müller's avatar
      configure.ac: Need -base CVS for gst_base_rtp_depayload_push_ts(). · 3da33640
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * configure.ac:
      Need  -base CVS for gst_base_rtp_depayload_push_ts().
      3da33640
    • Wim Taymans's avatar
      gst/avi/gstavidemux.c: Don't check for a tag that is never there and check if... · aeec395c
      Wim Taymans authored
      gst/avi/gstavidemux.c: Don't check for a tag that is never there and check if we read the correct tag. Fixes seeking ...
      
      Original commit message from CVS:
      * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
      Don't check for a tag that is never there and check if we read the
      correct tag. Fixes seeking again.
      We must post an error when all pads are unlinked.
      aeec395c
    • Wim Taymans's avatar
      gst/rtp/: More fixage, set endoder-params correctly in the payloader. · 25a44f8e
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/Makefile.am:
      * gst/rtp/gstrtp.c: (plugin_init):
      * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
      * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
      (gst_rtp_vorbis_pay_reset_packet),
      (gst_rtp_vorbis_pay_init_packet),
      (gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
      (gst_rtp_vorbis_pay_handle_buffer):
      More fixage, set endoder-params correctly in the payloader.
      25a44f8e
    • Tim-Philipp Müller's avatar
      gst/autodetect/: Make static pad templates static to appease valgrind's leak detector. · e4ba5018
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst/autodetect/gstautoaudiosink.c:
      (gst_auto_audio_sink_base_init):
      * gst/autodetect/gstautovideosink.c:
      (gst_auto_video_sink_base_init):
      Make static pad templates static to appease valgrind's leak
      detector.
      * tests/check/Makefile.am:
      * tests/check/elements/.cvsignore:
      * tests/check/elements/autodetect.c: (GST_START_TEST),
      (autodetect_suite):
      Add simple test for the ghostpad lockup on shutdown fixed in core
      CVS (audio bit disabled because it would need dozens of alsa
      suppressions and I'm too lazy to add those now).
      e4ba5018
    • Wim Taymans's avatar
      gst/rtp/: Small cleanups. · 8dbf0334
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
      Small cleanups.
      * gst/rtp/Makefile.am:
      * gst/rtp/gstrtp.c: (plugin_init):
      * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
      (gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
      (gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
      (gst_rtp_vorbis_depay_process),
      (gst_rtp_vorbis_depay_set_property),
      (gst_rtp_vorbis_depay_get_property),
      (gst_rtp_vorbis_depay_change_state),
      (gst_rtp_vorbis_depay_plugin_init):
      * gst/rtp/gstrtpvorbisdepay.h:
      * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
      (gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
      (gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
      (gst_rtp_vorbis_pay_flush_packet),
      (gst_rtp_vorbis_pay_append_buffer),
      (gst_rtp_vorbis_pay_handle_buffer),
      (gst_rtp_vorbis_pay_plugin_init):
      * gst/rtp/gstrtpvorbispay.h:
      Add experimental vorbis pay and depayloaders.
      8dbf0334
  9. 21 Sep, 2006 3 commits
    • Wim Taymans's avatar
      gst/rtp/gstrtpmp4gpay.c: Fix profile-level-id parsing and setup. · 3b5584f8
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):
      Fix profile-level-id parsing and setup.
      3b5584f8
    • Wim Taymans's avatar
      gst/udp/: Update README, simple cleanup. · edd6b7ec
      Wim Taymans authored
      Original commit message from CVS:
      * gst/udp/README:
      * gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
      Update README, simple cleanup.
      edd6b7ec
    • Wim Taymans's avatar
      gst/rtp/README: Update README with some examples. · 46d9a8a5
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/README:
      Update README with some examples.
      * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
      (gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
      (gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
      (gst_rtp_mp4g_pay_setcaps):
      * gst/rtp/gstrtpmp4gpay.h:
      Make optional RTP parameters of type STRING, as required by the
      application/x-rtp caps specification.
      46d9a8a5
  10. 20 Sep, 2006 4 commits
    • Philippe Khalaf's avatar
      gst/rtp/: Correctly calculate size of each H263+ RTP buffer taking into account MTU and · f1533c55
      Philippe Khalaf authored
      Original commit message from CVS:
      * gst/rtp/gstrtph263pdepay.c:
      * gst/rtp/gstrtph263ppay.c:
      Correctly calculate size of each H263+ RTP buffer taking into account MTU and
      RTP header.
      f1533c55
    • Wim Taymans's avatar
      gst/rtp/Makefile.am: And makefile too. · e28d3b2a
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/Makefile.am:
      And makefile too.
      e28d3b2a
    • Wim Taymans's avatar
      gst/rtp/: Added preliminary ASF depayloader. · 93c0a73c
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtp.c: (plugin_init):
      * gst/rtp/gstrtpasfdepay.c: (gst_rtp_asf_depay_base_init),
      (gst_rtp_asf_depay_class_init), (gst_rtp_asf_depay_init),
      (decode_base64), (gst_rtp_asf_depay_setcaps),
      (gst_rtp_asf_depay_process), (gst_rtp_asf_depay_set_property),
      (gst_rtp_asf_depay_get_property), (gst_rtp_asf_depay_change_state),
      (gst_rtp_asf_depay_plugin_init):
      * gst/rtp/gstrtpasfdepay.h:
      Added preliminary ASF depayloader.
      * gst/rtp/gstrtph264depay.c: (decode_base64):
      Fix base64 decoding.
      93c0a73c
    • 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
  11. 19 Sep, 2006 3 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation. · a7d7309e
      Wim Taymans authored
      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.
      a7d7309e
    • Wim Taymans's avatar
      gst/rtsp/test.c: Fix build. · cdbd5ca1
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/test.c: (main):
      Fix build.
      cdbd5ca1
    • Wim Taymans's avatar
      gst/wavparse/gstwavparse.c: Add ms-gsm to the src template. · db4d1f89
      Wim Taymans authored
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c:
      Add ms-gsm to the src template.
      db4d1f89
  12. 18 Sep, 2006 5 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
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Export sometimes source pad with correct caps on the... · 108dbd54
      Wim Taymans authored
      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.
      108dbd54
    • Lutz Mueller's avatar
      gst/rtsp/gstrtspsrc.*: Make sure we can never set an invalid location. · cac807b6
      Lutz Mueller authored and Wim Taymans's avatar Wim Taymans committed
      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.
      cac807b6
    • Lutz Mueller's avatar
      gst/rtsp/gstrtspsrc.*: Use boilerplate. · afd156ad
      Lutz Mueller authored and Wim Taymans's avatar Wim Taymans committed
      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.
      afd156ad
    • Thijs Vermeir's avatar
      gst/rtsp/: Small cleanups. when multicast is selected as the transport, create... · 7484c92d
      Thijs Vermeir authored and Wim Taymans's avatar Wim Taymans committed
      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. 16 Sep, 2006 1 commit
    • Stefan Kost's avatar
      More G_OBJECT macro fixing. · eb1b7236
      Stefan Kost authored
      Original commit message from CVS:
      * ext/flac/gstflactag.c:
      * gst/alpha/gstalpha.c:
      * gst/debug/breakmydata.c:
      * gst/debug/negotiation.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/multipart/multipartdemux.c:
      * gst/multipart/multipartmux.c:
      * gst/videobox/gstvideobox.c:
      * gst/videofilter/gstgamma.c:
      * gst/videofilter/gstvideotemplate.c:
      * gst/videomixer/videomixer.c:
      * sys/sunaudio/gstsunaudiosrc.h:
      More G_OBJECT macro fixing.
      eb1b7236