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* ext/jpeg/gstjpegdec.c: our code should handle that fine. Some of the buttons ↵Tim-Philipp Müller2006-09-251-0/+7
| | | | | | | | | | on the apple trailer site are apparently... Original commit message from CVS: * ext/jpeg/gstjpegdec.c: Set minimum height to 8 (from 16), our code should handle that fine. Some of the buttons on the apple trailer site are apparently only 15 pixels high (see #357470).
* gst/rtsp/: Improve error reporting.Wim Taymans2006-09-231-0/+11
| | | | | | | | | | | | 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/rtp/: Fix klass typos.Wim Taymans2006-09-231-0/+23
| | | | | | | | | | | | | | | | | | | | | | | | 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.
* configure.ac: Need -base CVS for gst_base_rtp_depayload_push_ts().Tim-Philipp Müller2006-09-221-0/+5
| | | | | | Original commit message from CVS: * configure.ac: Need -base CVS for gst_base_rtp_depayload_push_ts().
* gst/avi/gstavidemux.c: Don't check for a tag that is never there and check ↵Wim Taymans2006-09-221-0/+7
| | | | | | | | | | 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.
* gst/rtp/: More fixage, set endoder-params correctly in the payloader.Wim Taymans2006-09-221-0/+12
| | | | | | | | | | | | | 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.
* gst/autodetect/: Make static pad templates static to appease valgrind's leak ↵Tim-Philipp Müller2006-09-221-0/+17
| | | | | | | | | | | | | | | | | | | detector. 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).
* gst/rtp/: Small cleanups.Wim Taymans2006-09-221-0/+27
| | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtp/gstrtpmp4gpay.c: Fix profile-level-id parsing and setup.Wim Taymans2006-09-211-0/+5
| | | | | | Original commit message from CVS: * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config): Fix profile-level-id parsing and setup.
* gst/udp/: Update README, simple cleanup.Wim Taymans2006-09-211-0/+6
| | | | | | | Original commit message from CVS: * gst/udp/README: * gst/udp/gstudpsrc.c: (gst_udpsrc_set_property): Update README, simple cleanup.
* gst/rtp/README: Update README with some examples.Wim Taymans2006-09-211-1/+14
| | | | | | | | | | | | | 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.
* gst/rtp/: Correctly calculate size of each H263+ RTP buffer taking into ↵Philippe Kalaf2006-09-201-0/+7
| | | | | | | | | | account MTU and 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.
* gst/rtp/Makefile.am: And makefile too.Wim Taymans2006-09-201-0/+5
| | | | | | Original commit message from CVS: * gst/rtp/Makefile.am: And makefile too.
* gst/rtp/: Added preliminary ASF depayloader.Wim Taymans2006-09-201-0/+15
| | | | | | | | | | | | | | | 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.
* gst/rtsp/URLS: Added some test URLS.Wim Taymans2006-09-201-0/+56
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Taymans2006-09-191-0/+20
| | | | | | | | | | | | | | | | | | | | 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 Taymans2006-09-191-0/+5
| | | | | | Original commit message from CVS: * gst/rtsp/test.c: (main): Fix build.
* gst/wavparse/gstwavparse.c: Add ms-gsm to the src template.Wim Taymans2006-09-191-0/+5
| | | | | | Original commit message from CVS: * gst/wavparse/gstwavparse.c: Add ms-gsm to the src template.
* gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation.Wim Taymans2006-09-181-0/+41
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Taymans2006-09-181-0/+13
| | | | | | | | | | | | | | | | 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 Mueller2006-09-181-0/+17
| | | | | | | | | | | | | | | 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 Mueller2006-09-181-0/+13
| | | | | | | | | | | | | 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 Vermeir2006-09-181-1/+16
| | | | | | | | | | | | | | | | | 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.
* More G_OBJECT macro fixing.Stefan Kost2006-09-161-0/+24
| | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/avi/gstavimux.c: Correctly set the dwLength in strh.Yves Lefebvre2006-09-161-0/+9
| | | | | | | | | Original commit message from CVS: Patch by: Yves Lefebvre <ivanohe at abacom dot com> * gst/avi/gstavimux.c: (gst_avi_mux_stop_file): Correctly set the dwLength in strh. With this patch, the file duration is now displayed correctly in window media player and the AVI plays completely. Fixes #356147
* sys/sunaudio/gstsunaudiomixerctrl.c: Set the output track as the MASTER so ↵Darren Kenny2006-09-151-0/+10
| | | | | | | | | | | | that the gnome-settings-daemon keybindings... Original commit message from CVS: Patch by: Darren Kenny <darren dot kenny at sun dot com> * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_build_list): Set the output track as the MASTER so that the gnome-settings-daemon keybindings for changing the volume using the keyboard works. Fixes #356142.
* gst/multipart/multipartdemux.c: Fix documentation, it is not possible to ↵Wim Taymans2006-09-151-0/+8
| | | | | | | | | | | control the framerate of jpegdec using filte... Original commit message from CVS: * gst/multipart/multipartdemux.c: (gst_multipart_demux_chain): Fix documentation, it is not possible to control the framerate of jpegdec using filtered caps yet. Fixes #355210. Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we stop when there is an error.
* gst/: Don't interpret a first buffer with an offset of NONE as 'from the ↵Tim-Philipp Müller2006-09-141-0/+8
| | | | | | | | | | | middle of the stream', but only a first buff... Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag): * gst/id3demux/gstid3demux.c: (gst_id3demux_chain): Don't interpret a first buffer with an offset of NONE as 'from the middle of the stream', but only a first buffer that has a valid buffer offset that's non-zero (see #345449).
* gst/icydemux/gsticydemux.*: When we merge/collect multiple incoming buffers ↵Tim-Philipp Müller2006-09-141-0/+15
| | | | | | | | | | | | | | | | | for typefinding purposes, keep an initial... Original commit message from CVS: * gst/icydemux/gsticydemux.c: (gst_icydemux_reset), (gst_icydemux_typefind_or_forward): * gst/icydemux/gsticydemux.h: When we merge/collect multiple incoming buffers for typefinding purposes, keep an initial 0 offset on the first outgoing buffer as well (otherwise id3demux won't work right). Fixes #345449. Also Make buffer metadata writable before setting buffer caps. * tests/check/elements/icydemux.c: (typefind_succeed), (cleanup_icydemux), (push_data), (GST_START_TEST), (icydemux_suite): Small test case for the above.
* gst/avi/gstavidemux.c: More code reuse and better logging in _peek_chunk(). ↵Stefan Kost2006-09-131-0/+11
| | | | | | | | | | | | | | Reintroduce check for chunk sizes before ... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_header_push), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): More code reuse and better logging in _peek_chunk(). Reintroduce check for chunk sizes before reading them (avoid oom). Better handling for invalid chunksizes when streaming.
* gst/level/gstlevel.*: Fix type mixup in level->interval (gdouble<->guint64). ↵Stefan Kost2006-09-111-0/+7
| | | | | | | | | | Spotted by Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_set_property): * gst/level/gstlevel.h: Fix type mixup in level->interval (gdouble<->guint64). Spotted by René Stadler
* gst/avi/gstavidemux.c: Revert one change to fix streaming avi (adapter size ↵Stefan Kost2006-09-061-0/+7
| | | | | | | | | | != data size). Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_sync), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data): Revert one change to fix streaming avi (adapter size != data size).
* gst/matroska/: Add support for VOBSUB subtitle tracks and zlib-compressed ↵Frédéric Riss2006-09-041-0/+14
| | | | | | | | | | | | | | | | tracks. Make sure we start on a keyframe af... Original commit message from CVS: Patch by: Frédéric Riss <frederic.riss at gmail dot com> * gst/matroska/matroska-demux.c: (gst_matroska_track_free), (gst_matroska_demux_reset), (gst_matroska_demux_read_track_encodings), (gst_matroska_demux_add_stream), (gst_matroska_decode_buffer), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_subtitle_caps): * gst/matroska/matroska-ids.h: Add support for VOBSUB subtitle tracks and zlib-compressed tracks. Make sure we start on a keyframe after a seek. (#343348)
* gst/matroska/: not perfect yet though, needs some tweaking in flacdec; also, ↵Tim-Philipp Müller2006-09-041-0/+13
| | | | | | | | | | | | | | | | seeking could be better. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf), (gst_matroska_demux_push_flac_codec_priv_data), (gst_matroska_demux_push_xiph_codec_priv_data), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps): * gst/matroska/matroska-ids.h: Add basic FLAC support (#311586), not perfect yet though, needs some tweaking in flacdec; also, seeking could be better. Do better bounds checking when deserialising vorbis stream headers to make sure we don't read beyond the end of the buffer on bad input.
* ext/annodex/gstcmmldec.c: Seeking back in a file containing a CMML stream ↵Alessandro Decina2006-09-041-0/+15
| | | | | | | | | | | | | | | errors out if the seek goes back up to the ... Original commit message from CVS: Patch by: Alessandro Decina <alessandro at nnva dot org> * ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain): Seeking back in a file containing a CMML stream errors out if the seek goes back up to the CMML headers. This is because after the seek the xml processing instruction <?xml ...?> is submitted to the xml parser again, which results in an error. The attached patch fixes the problem. Fixes #353908. * ext/annodex/gstcmmlenc.h: Fix authors name.
* ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle): New helper ↵Andy Wingo2006-08-281-0/+11
| | | | | | | | | | | | | | | | function to lessen the ifdefs. Original commit message from CVS: 2006-08-28 Andy Wingo <wingo@pobox.com> * ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle): New helper function to lessen the ifdefs. (GST_INFO_OBJECT): (gst_dv1394src_iso_receive): Use it. (gst_dv1394src_create): Also use the control sockets in iec61883 mode. (gst_dv1394src_start, gst_dv1394src_stop): Always use a separate handle for AVC operations; fixes #348233.
* Rename again (audiofxgood -> audiofx).Stefan Kost2006-08-271-0/+15
| | | | | | | | | | | | | | | | Original commit message from CVS: * configure.ac: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audiofxgood.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: * gst/audiofxgood/.cvsignore: * gst/audiofxgood/Makefile.am: * gst/audiofxgood/audiofx.c: * gst/audiofxgood/audiopanorama.c: * gst/audiofxgood/audiopanorama.h: Rename again (audiofxgood -> audiofx).
* gst/avi/gstavidemux.c: Initialze variables.Stefan Kost2006-08-271-0/+6
| | | | | | | Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan): Initialze variables.
* gst/avi/gstavidemux.*: More attempts to turn this into readable code.Wim Taymans2006-08-251-0/+21
| | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_class_init), (gst_avi_demux_init), (gst_avi_demux_finalize), (gst_avi_demux_reset), (gst_avi_demux_index_last), (gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_peek_tag), (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_loop), (gst_avi_demux_chain), (gst_avi_demux_sink_activate), (gst_avi_demux_change_state): * gst/avi/gstavidemux.h: More attempts to turn this into readable code. Don't leak adapters. Calculate duration according to index more efficiently. Don't try to act like we drive the pipeline in chain mode.
* ext/annodex/gstcmmlutils.c: Fix build.Wim Taymans2006-08-251-0/+5
| | | | | | Original commit message from CVS: * ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt): Fix build.
* ext/annodex/gstannodex.c: Do some extra sanity checks.Alessandro Decina2006-08-251-0/+40
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: Patch by: Alessandro Decina <alessandro at nnva dot org> * ext/annodex/gstannodex.c: (gst_annodex_granule_to_time): Do some extra sanity checks. Fixes #350340. * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state), (gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip), (gst_cmml_enc_push_clip), (gst_cmml_enc_push): Check if clip->start_time is valid before adding the clip to the track list. Reset enc->preamble going from PAUSED to READY. Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is only used for EOS. Only post an error message if we were the one that created the fatal GstFlowReturn value. * ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt), (gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip): Parse the seconds field of the npt-sec time format using %llu rather than %d and check that the value scaled by GST_SECOND doesn't overflow. Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos. Lookup a clip's track with clip->track rather than clip->id which makes no sense. Identify a clip by its track and start time and not its xml id. do some more input checking and make sure we don't do undefined shifts. * tests/check/elements/cmmldec.c: (setup_cmmldec), (teardown_cmmldec), (check_output_buffer_is_equal), (push_data), (cmml_tag_message_pop), (check_headers), (push_clip_full), (push_clip), (push_empty_clip), (check_output_clip), (GST_START_TEST), (cmmldec_suite): * tests/check/elements/cmmlenc.c: (setup_cmmlenc), (teardown_cmmlenc), (check_output_buffer_is_equal), (push_data), (check_headers), (push_clip), (check_clip_times), (check_clip), (check_empty_clip), (GST_START_TEST), (cmmlenc_suite): Added some more checks.
* Make also the pan-property float (saves scaling and yields better resolution)Stefan Kost2006-08-241-0/+14
| | | | | | | | | | | | | | | Original commit message from CVS: * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_class_init), (gst_audio_panorama_set_property), (gst_audio_panorama_get_property), (gst_audio_panorama_transform_m2s_int), (gst_audio_panorama_transform_s2s_int), (gst_audio_panorama_transform_m2s_float), (gst_audio_panorama_transform_s2s_float): * gst/audiofxgood/audiopanorama.h: * tests/check/elements/audiopanorama.c: (GST_START_TEST): Make also the pan-property float (saves scaling and yields better resolution)
* gst/audiofxgood/audiopanorama.c: ChangeLog surgery to add cymax's real nameStefan Kost2006-08-241-1/+11
| | | | | | | | Original commit message from CVS: * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps), (gst_audio_panorama_transform_m2s_float), (gst_audio_panorama_transform_s2s_float): ChangeLog surgery to add cymax's real name
* gst/audiofxgood/audiopanorama.*: Added float support (thanks cymax)Stefan Kost2006-08-241-0/+11
| | | | | | | | | | | | Original commit message from CVS: * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps), (gst_audio_panorama_transform_m2s_int), (gst_audio_panorama_transform_s2s_int), (gst_audio_panorama_transform_m2s_float), (gst_audio_panorama_transform_s2s_float), (gst_audio_panorama_transform): * gst/audiofxgood/audiopanorama.h: Added float support (thanks cymax)
* gst/audiofxgood/audiopanorama.c: Fix docs & debug category. Add Fixme for ↵Stefan Kost2006-08-241-0/+6
| | | | | | | | | volume pan levels. Original commit message from CVS: * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_transform_m2s): Fix docs & debug category. Add Fixme for volume pan levels.
* gst/avi/gstavidemux.c: unbreak AVI index handling, some more debug, remove ↵Stefan Kost2006-08-241-0/+10
| | | | | | | | | | | | | an obsolete adapter_flush that caused stre... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_sync), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_header_pull), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_chain): unbreak AVI index handling, some more debug, remove an obsolete adapter_flush that caused streaming to wander off in the wild
* gst/avi/gstavidemux.*: Some more cleanups.Wim Taymans2006-08-241-0/+15
| | | | | | | | | | | | | | | | Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_odml), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_header_pull): * gst/avi/gstavidemux.h: Some more cleanups. Fix totalFrames parsing in ODML. Disable use of index for length calculation in case of ODML as this is broken now.
* ext/flac/gstflacdec.c: Use libgsttag helper function here too.Tim-Philipp Müller2006-08-241-0/+5
| | | | | | Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_update_metadata): Use libgsttag helper function here too.
* gst/avi/gstavidemux.*: Initial streaming support for avidemux (fixes #336465)Stefan Kost2006-08-231-0/+23
| | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_class_init), (gst_avi_demux_init), (gst_avi_demux_dispose), (gst_avi_demux_reset), (gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_peek_chunk_info), (gst_avi_demux_peek_chunk), (gst_avi_demux_stream_init_push), (gst_avi_demux_stream_init_pull), (gst_avi_demux_parse_subindex), (gst_avi_demux_read_subindexes_push), (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream), (sort), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_peek_tag), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_header_pull), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (push_tag_lists), (gst_avi_demux_loop), (gst_avi_demux_chain), (gst_avi_demux_sink_activate), (gst_avi_demux_activate_push), (gst_avi_demux_change_state): * gst/avi/gstavidemux.h: Initial streaming support for avidemux (fixes #336465)
* docs/plugins/gst-plugins-good-plugins-docs.sgml: There is no taglibmux ↵Tim-Philipp Müller2006-08-221-0/+9
| | | | | | | | | | | 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.
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