Commit graph

2582 commits

Author SHA1 Message Date
Jan Schmidt
b26667c5b2 sys/v4l2/gstv4l2xoverlay.*: Fix build as per the patch in #338818 comment 36.
Original commit message from CVS:
* sys/v4l2/gstv4l2xoverlay.c:
* sys/v4l2/gstv4l2xoverlay.h:
Fix build as per the patch in #338818 comment 36.
2006-10-09 07:01:19 +00:00
Tim-Philipp Müller
58b341970f gst/rtsp/gstrtspsrc.c: Activate pads before adding them to the source.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
Activate pads before adding them to the source.
2006-10-07 21:15:40 +00:00
Wim Taymans
3adedd4f60 Activate pads before adding.
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.
2006-10-06 17:00:14 +00:00
Wim Taymans
09328ad010 gst/multipart/multipartdemux.c: Activate pads before adding.
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.
2006-10-06 16:03:23 +00:00
Wim Taymans
a600d31120 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.
2006-10-06 12:55:53 +00:00
Tim-Philipp Müller
82f5a3508c Printf format fixes.
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.
2006-10-05 16:37:33 +00:00
Wim Taymans
a0ff313ab7 gst/rtsp/Makefile.am: Dist new .h file too.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
Dist new .h file too.
2006-10-04 17:53:12 +00:00
Wim Taymans
63c87f1899 gst/rtsp/: Factor out extension in separate module.
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.
2006-10-04 17:24:40 +00:00
Thomas Vander Stichele
c85684e2d8 docs/plugins/: added v4l2 stubs
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
2006-10-04 10:29:11 +00:00
Tim-Philipp Müller
424c5cb6f7 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.
2006-10-04 10:24:49 +00:00
Thomas Vander Stichele
0b5d4c7787 add build stuff for v4l2, needs --enable-experimental until the last bits are resolved
Original commit message from CVS:
* configure.ac:
* sys/Makefile.am:
add build stuff for v4l2, needs --enable-experimental until
the last bits are resolved
2006-10-03 18:15:58 +00:00
Tim-Philipp Müller
475aed8af6 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.
2006-09-29 15:39:41 +00:00
Wim Taymans
6e08550345 gst/rtsp/URLS: Add some more URLs.
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.
2006-09-29 15:37:29 +00:00
Wim Taymans
fcd901a5bf gst/udp/gstudpsrc.*: Added property to post a message on timeout.
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
2006-09-29 11:09:40 +00:00
Wim Taymans
e8c59d9da3 gst/rtsp/gstrtspsrc.c: Fix flag registration.
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'
2006-09-29 08:15:05 +00:00
Antoine Tremblay
1a86fdc6e3 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.
2006-09-29 08:09:24 +00:00
Wim Taymans
9cadd004a8 gst/autodetect/: Small cleanups. don't try to set "sync" property when it is not available.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
(gst_auto_audio_sink_find_best):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
Small cleanups.
don't try to set "sync" property when it is not available.
2006-09-25 13:55:44 +00:00
Peter Kjellerstedt
b234d9b0f9 gst/: Include stdlib.h in some more places, makes things compile with uClibc 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).
2006-09-25 11:47:42 +00:00
Tim-Philipp Müller
ca1f196979 ext/jpeg/gstjpegdec.c: our code should handle that fine. Some of the buttons 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).
2006-09-25 09:15:10 +00:00
Wim Taymans
23ec2eb189 gst/rtsp/: Improve error reporting.
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.
2006-09-23 15:31:56 +00:00
Wim Taymans
af6e4da92e gst/rtp/: Fix klass typos.
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.
2006-09-23 15:30:40 +00:00
Tim-Philipp Müller
3da33640bb configure.ac: Need -base CVS for gst_base_rtp_depayload_push_ts().
Original commit message from CVS:
* configure.ac:
Need  -base CVS for gst_base_rtp_depayload_push_ts().
2006-09-22 17:53:48 +00:00
Wim Taymans
aeec395c22 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.
2006-09-22 17:22:34 +00:00
Wim Taymans
25a44f8e02 gst/rtp/: More fixage, set endoder-params correctly in the payloader.
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.
2006-09-22 15:15:13 +00:00
Tim-Philipp Müller
e4ba501855 gst/autodetect/: Make static pad templates static to appease valgrind's leak 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).
2006-09-22 12:12:10 +00:00
Wim Taymans
8dbf033420 gst/rtp/: Small cleanups.
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.
2006-09-22 12:08:14 +00:00
Wim Taymans
3b5584f8d1 gst/rtp/gstrtpmp4gpay.c: Fix profile-level-id parsing and setup.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):
Fix profile-level-id parsing and setup.
2006-09-21 13:33:16 +00:00
Wim Taymans
edd6b7ec72 gst/udp/: Update README, simple cleanup.
Original commit message from CVS:
* gst/udp/README:
* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
Update README, simple cleanup.
2006-09-21 09:50:41 +00:00
Wim Taymans
46d9a8a5e6 gst/rtp/README: Update README with some examples.
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.
2006-09-21 09:35:13 +00:00
Philippe Kalaf
f1533c5504 gst/rtp/: Correctly calculate size of each H263+ RTP buffer taking into 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.
2006-09-20 19:37:45 +00:00
Wim Taymans
e28d3b2a92 gst/rtp/Makefile.am: And makefile too.
Original commit message from CVS:
* gst/rtp/Makefile.am:
And makefile too.
2006-09-20 16:41:48 +00:00
Wim Taymans
93c0a73ce0 gst/rtp/: Added preliminary ASF depayloader.
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.
2006-09-20 16:09:03 +00:00
Wim Taymans
a365a29c77 gst/rtsp/URLS: Added some test URLS.
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.
2006-09-20 16:06:27 +00:00
Wim Taymans
a7d7309e18 gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation.
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.
2006-09-19 17:25:15 +00:00
Wim Taymans
cdbd5ca170 gst/rtsp/test.c: Fix build.
Original commit message from CVS:
* gst/rtsp/test.c: (main):
Fix build.
2006-09-19 10:53:56 +00:00
Wim Taymans
db4d1f89f6 gst/wavparse/gstwavparse.c: Add ms-gsm to the src template.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Add ms-gsm to the src template.
2006-09-19 10:14:52 +00:00
Wim Taymans
a437e9f0ed gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation.
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.
2006-09-18 17:37:46 +00:00
Wim Taymans
108dbd54cf 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.
2006-09-18 14:00:41 +00:00
Lutz Mueller
cac807b641 gst/rtsp/gstrtspsrc.*: Make sure we can never set an invalid location.
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.
2006-09-18 11:29:12 +00:00
Lutz Mueller
afd156ad0c gst/rtsp/gstrtspsrc.*: Use boilerplate.
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.
2006-09-18 10:42:52 +00:00
Thijs Vermeir
7484c92dfe 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.
2006-09-18 08:59:17 +00:00
Stefan Kost
eb1b7236f3 More G_OBJECT macro fixing.
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.
2006-09-16 21:57:29 +00:00
Yves Lefebvre
805b8ba808 gst/avi/gstavimux.c: Correctly set the dwLength in strh.
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
2006-09-16 14:30:59 +00:00
Darren Kenny
c8acc74c5e sys/sunaudio/gstsunaudiomixerctrl.c: Set the output track as the MASTER so 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.
2006-09-15 17:10:22 +00:00
Wim Taymans
00256ae0a9 gst/multipart/multipartdemux.c: Fix documentation, it is not possible to 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.
2006-09-15 16:01:48 +00:00
Tim-Philipp Müller
dcba7c77ef gst/: Don't interpret a first buffer with an offset of NONE as 'from the 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).
2006-09-14 11:05:35 +00:00
Tim-Philipp Müller
e73ddd490e gst/icydemux/gsticydemux.*: When we merge/collect multiple incoming buffers 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.
2006-09-14 10:38:42 +00:00
Stefan Kost
13a332da30 gst/avi/gstavidemux.c: More code reuse and better logging in _peek_chunk(). 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.
2006-09-13 13:26:15 +00:00
Stefan Kost
b507a3e175 gst/level/gstlevel.*: Fix type mixup in level->interval (gdouble<->guint64). 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
2006-09-11 20:38:41 +00:00
Stefan Kost
4b7c760e11 gst/avi/gstavidemux.c: Revert one change to fix streaming avi (adapter size != 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).
2006-09-06 09:05:33 +00:00