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.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_transform_ip):
Removed cruft code that was just commented out. Removed some obsolete
debug logs statements.
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.
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.
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.
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
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'
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.
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.
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).
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.
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.
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).
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.
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.
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.
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.
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.
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.
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.
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.
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
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.
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).
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.
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.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_start), (gst_spectrum_stop), (gst_spectrum_event):
Implements stop() to clear the adapter and event() to clear the
adapter on FLUSH_STOP and EOS.
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
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_set_property):
* gst/spectrum/gstspectrum.h:
Fix type mixup in spectrum->interval (gdouble<->guint64). Spotted by
René Stadler
Original commit message from CVS:
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (main):
Use more defines
* gst/spectrum/gstspectrum.c: (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_caps),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Apply some of the spectrum cleanup changes suggested in #348085.
Original commit message from CVS:
* configure.ac:
Bump requirements of -base (videocrop test case needs this).
* gst/videocrop/gstvideocrop.c:
Document sloppy handling of subsampled chroma planes if
left/top cropping is an odd number.
* tests/check/elements/videocrop.c: (handoff_cb),
(videocrop_test_cropping_init_context),
(videocrop_test_cropping_deinit_context),
(videocrop_test_cropping), (check_1x1_buffer), (GST_START_TEST),
(videocrop_suite), (main):
Add another unit test that crops the input to 1x1 (and checks
that that pixel has the expected values in a number of formats).
Original commit message from CVS:
* gst/videocrop/Makefile.am:
* gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init),
(gst_video_crop_transform_packed),
(gst_video_crop_transform_planar):
Some quick tests indicate that it doesn't make a great deal
of sense to use liboil here, at least not for the memcpy()s
we do, so remove liboil usage until there is clear evidence
it actually makes a positive difference somewhere.
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).
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)
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.
Original commit message from CVS:
* configure.ac:
* gst/videocrop/Makefile.am:
* gst/videocrop/gstvideocrop.c: (gst_video_crop_base_init),
(gst_video_crop_class_init), (gst_video_crop_init),
(gst_video_crop_get_image_details_from_caps),
(gst_video_crop_get_unit_size), (gst_video_crop_transform_packed),
(gst_video_crop_transform_planar), (gst_video_crop_transform),
(gst_video_crop_transform_dimension),
(gst_video_crop_transform_dimension_value),
(gst_video_crop_transform_caps), (gst_video_crop_set_caps),
(gst_video_crop_set_property), (gst_video_crop_get_property),
(plugin_init):
Port/rewrite videocrop from scratch for GStreamer-0.10, and make
it support all formats videoscale supports (#345653).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
(gst_qtdemux_do_seek):
Reset each streams last_flow to GST_FLOW_OK.
(gst_qtdemux_activate_segment):
Removing mystic modifications for good.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(qtdemux_parse_tree):
put back 'segment start<=stop' change that was mystically reverted by
the last commit
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(gst_qtdemux_add_stream), (qtdemux_parse_trak),
(qtdemux_video_caps):
Make sure segment start<=stop in weird quicktime files.
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.
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)
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
Original commit message from CVS:
* gst/audiofxgood/audiopanorama.c:
(gst_audio_panorama_transform_m2s):
Fix docs & debug category. Add Fixme for volume pan levels.
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
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.
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.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_do_seek), (gst_avi_demux_handle_seek),
(gst_avi_demux_process_next_entry):
* gst/avi/gstavidemux.h:
Mark DISCONT.
Remove old unused fields and reorder the struct a bit.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_index_entry_for_time),
(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
(gst_avi_demux_stream_init), (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header), (gst_avi_demux_do_seek),
(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
(gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Precalc most of the duration query for each stream.
Make seeking more correct.
Use GstSegment to track position and duration.
Code cleanups and leak fixes.
Calculate correct total duration based on index length.
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
(parse_insert_string_field):
If strings in text fields are marked ISO8859-1, but contain
valid UTF-8 already, then handle them as UTF-8 and ignore
the encoding. (#351794)
Original commit message from CVS:
* gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
Don't unref buffers of which we've already given away
ownership to the adapter.
Original commit message from CVS:
* gst/audiopanorama/.cvsignore:
* gst/audiopanorama/Makefile.am:
* gst/audiopanorama/audiofx.c:
* gst/audiopanorama/audiopanorama.c:
* gst/audiopanorama/audiopanorama.h:
die! die! die! you should never have been there
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse),
(qtdemux_node_dump_foreach), (qtdemux_parse_trak),
(qtdemux_video_caps), (qtdemux_audio_caps):
Some more constification.
Fix some paletted data formats again.
Fix ulaw/alaw in qt.
Set correct caps for raw RGB.
Add support for yuv2, which is like Yuv2.
Add support for raw audio with the NONE fourcc, which is like raw.
Original commit message from CVS:
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size):
* gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size):
use g_assert in _get_unit_size
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/inspect/plugin-audiofxgood.xml:
cleanup -unused.txt to make it useful, add previously missing docs
* ext/Makefile.am:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/esd/gstesd.c: (plugin_init):
reflow to get rid of two external symbols
* gst/audiofxgood/audiofx.c: (plugin_init):
re-add
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/multipart/multipartdemux.c: (multipart_parse_header):
Accept leading whitespace before the boundary
This patch makes the demuxer allow some whitespace before the actual
boundary. This makes the demuxer work with the ``old'' gstreamer
multipartmuxer again (which placed an extra \n before the start
of the stream) Fixes#349068.
Original commit message from CVS:
* configure.ac:
Require CVS of GStreamer core and -base (for
GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).
* ext/taglib/gstid3v2mux.cc:
Write extended comment tags properly (#348762).
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame):
Extract COMM frames into extended comments, which makes it
easier to properly retain the description bit of the tag
and maintain this information when re-tagging (#348762).
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_find_best):
When we can't find a usable audiosink, don't error out,
but use a fake sink instead and post a warning message
on the bus (#341278).
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpmp4gdepay.c:
Caps extra properties must be defined as strings for
depayloaders because they are generated from an SDP.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init),
(gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init),
(gst_rtp_h264_depay_finalize), (decode_base64),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
(gst_rtp_h264_depay_set_property),
(gst_rtp_h264_depay_get_property),
(gst_rtp_h264_depay_change_state),
(gst_rtp_h264_depay_plugin_init):
* gst/rtp/gstrtph264depay.h:
Added basic, not completely functional RFC 3984 H264 depayloader.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_redirects_sort_func),
(qtdemux_process_redirects), (qtdemux_parse_tree):
Extract all references/redirections if there is more
than one and sort them; also extract minimum required
bitrate information if available. (#350399)
Original commit message from CVS:
Patch by: Edward Hervey <edward@fluendo.com>
* configure.ac:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_data):
Send the newsegment event in the streaming thread.
Fixes#347529
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
(gst_multipart_demux_finalize), (get_line_end),
(multipart_parse_header), (multipart_find_boundary),
(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
(gst_multipart_set_property), (gst_multipart_get_property):
Uses GstAdapter instead of own buffering.
Actually parses the mime-type correctly (In tests the mime-type was
always "" with the old version).
Uses the Content-length header if available to speed up things.
Reliably autoscans the boundary name by default.
Fixes#349068.
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
Don't start the stream with a \n.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_massage_index):
* gst/avi/gstavidemux.h:
Whitespace fixes and more debug
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_create_element_with_pretty_name),
(gst_auto_audio_sink_find_best),
(gst_auto_audio_sink_change_state):
Get rid of old and unused magic sound-server properties stuff.
Add suffix to child sink's name that makes it easy to see from
the name alone which type it actually is (alsa, oss, esd, etc.).
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_set_property), (gst_udpsrc_get_property),
(gst_udpsrc_start):
* gst/udp/gstudpsrc.h:
Rename "buffer" to "buffer-size" to make clear it is a size we set and
not some sort of feature we enable.
