Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(copy_source), (rtp_session_create_sources),
(rtp_session_get_property):
Add G_PARAM_STATIC_STRINGS.
Add property to return a GValueArray of all known RTPSources in the
session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_create_sdes), (rtp_source_set_property),
(rtp_source_get_property):
Remove properties to set the various SDES items, an application is never
supposed to change the RTPSource data.
Change the SDES getter properties to one SDES property that returns all
SDES items in a GstStructure.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (GST_START_TEST):
Make the unit test a bit faster to prevent timeouts, especially
with valgrind.
Original commit message from CVS:
* gst/mxf/mxfdemux.c: (gst_mxf_demux_push_src_event),
(gst_mxf_demux_handle_header_metadata_update_streams):
* gst/mxf/mxfparse.c: (gst_mxf_ul_hash),
(mxf_partition_pack_parse), (mxf_primer_pack_parse),
(mxf_metadata_preface_parse), (mxf_metadata_content_storage_parse),
(mxf_metadata_generic_package_parse),
(mxf_metadata_sequence_parse),
(mxf_metadata_generic_descriptor_parse),
(mxf_metadata_multiple_descriptor_parse):
Some more format string fixes and usage of guint instead of gint
where negative values don't make sense.
Original commit message from CVS:
* gst/mxf/mxfaes-bwf.c:
(mxf_metadata_wave_audio_essence_descriptor_parse):
* gst/mxf/mxfaes-bwf.h:
* gst/mxf/mxfdemux.c: (gst_mxf_demux_pull_range),
(gst_mxf_demux_pull_klv_packet),
(gst_mxf_demux_parse_footer_metadata),
(gst_mxf_demux_handle_klv_packet),
(gst_mxf_demux_pull_and_handle_klv_packet), (gst_mxf_demux_chain):
* gst/mxf/mxfmpeg.c: (mxf_metadata_mpeg_video_descriptor_parse):
* gst/mxf/mxfmpeg.h:
* gst/mxf/mxfparse.c: (mxf_timestamp_parse), (mxf_fraction_parse),
(mxf_utf16_to_utf8), (mxf_product_version_parse),
(mxf_partition_pack_parse), (mxf_primer_pack_parse),
(mxf_local_tag_parse), (mxf_metadata_preface_parse),
(mxf_metadata_identification_parse),
(mxf_metadata_content_storage_parse),
(mxf_metadata_essence_container_data_parse),
(mxf_metadata_generic_package_parse), (mxf_metadata_track_parse),
(mxf_metadata_sequence_parse),
(mxf_metadata_structural_component_parse),
(mxf_metadata_generic_descriptor_parse),
(mxf_metadata_file_descriptor_parse),
(mxf_metadata_generic_sound_essence_descriptor_parse),
(mxf_metadata_generic_picture_essence_descriptor_parse),
(mxf_metadata_cdci_picture_essence_descriptor_parse),
(mxf_metadata_multiple_descriptor_parse),
(mxf_metadata_locator_parse):
* gst/mxf/mxfparse.h:
Use guint instead of guint64 or gsize for all buffer sizes and
use correct format strings for them. Only local tag set sizes
are still guint16 as they can't be larger.
Only allow KLV packets of sizes below 1<<32 as GStreamer only uses
guint for buffer sizes. The MXF standard allows packet sizes up
to 1<<64.
Original commit message from CVS:
* gst/dccp/gstdccp.c: (gst_dccp_socket_write):
Use G_GSIZE_FORMAT instead of "%u" for a size_t variable in
the format string to prevent a compiler warning.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
Release the right pads on rtpbin. Fixes#561752.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_set_property),
(gst_speex_resample_get_property):
Add a "filter-length" property that maps to the quality values
for compatibilty with audioresample.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (get_current_times),
(rtcp_thread), (gst_rtp_session_chain_recv_rtp):
Pass the running time to the session when processing RTP packets.
Improve the time function to provide more info.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (update_arrival_stats),
(rtp_session_process_rtp), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (session_start_rtcp),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Mark the internal source with a flag.
Use running_time instead of the more useless timestamp.
Validate a source when a valid SDES has been received.
Pass the current system time when processing SR packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_create_stats),
(rtp_source_get_property), (rtp_source_send_rtp),
(rtp_source_process_rb), (rtp_source_get_new_rb),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add property to get source stats.
Mark params as STATIC_STRINGS.
Calculate the bitrate at the sender SSRC.
Avoid negative values in the round trip time calculations.
* gst/rtpmanager/rtpstats.h:
Update some docs and change some variable name to more closely reflect
what it contains.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain_rtcp):
Initialize return value to fix compiler warning about uninitialized
variable.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream), (free_stream),
(new_ssrc_pad_found):
Remove internal sync pad, use signals instead to get lip-sync
notifications.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink),
(remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad),
(gst_rtp_jitter_buffer_release_pad),
(gst_rtp_jitter_buffer_sink_rtcp_event),
(gst_rtp_jitter_buffer_chain_rtcp),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Make it possible to send SR packets to the jitterbuffer.
Check if the SR timestamps are valid by comparing them to the RTP
timestamps.
Signal the SR packet and the timing information to listeners.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query):
Remove some unused code.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last seen RTP timestamp so that we can filter out
invalid SR packets.
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Fix GST_DEBUG call to only have as many arguments as required
by the format string. Fixes a compiler warning.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found):
Do not try to keep track of the clock-rate ourselves but simply get the
value from the jitterbuffer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add some debug info.
Pass the clock-rate to the jitterbuffer.
Also pass the clock-rate along with the rtp timestamp when getting the
sync parameters.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix some debug.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of clock-rate changes and return the clock-rate together with
the rtp timestamps used for sync.
Don't try to construct timestamps when we have no base_time.
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Request a new clock-rate when the payload type changes.
Reset the jitter calculation when the clock-rate changes.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Guard against a NULL dereference I somehow encountered -
with a FLUSH_STOP arriving either before basetransform _start(),
or after _stop().
* gst/typefind/gsttypefindfunctions.c:
Make sure we never jump backwards when typefinding corrupt mov files.
Original commit message from CVS:
* ext/x264/gstx264enc.c: (gst_x264_enc_set_src_caps):
Construct source caps in more conventional (and correct) manner.
Original commit message from CVS:
* gst-libs/gst/play/.cvsignore:
* gst-libs/gst/play/play.h:
* gst-libs/gst/play/play.vcproj:
Remove cruft. This is not entered by make and its not even compilable.
Original commit message from CVS:
* ext/dirac/gstdiracenc.cc:
Set pixel-aspect-ratio correctly in the encoder API, as well
as some default gstreamerish colorspace properties. Also,
apparently, change a bunch of indentation.
Original commit message from CVS:
* ext/jp2k/gstjasperdec.c: (gst_jasper_dec_init),
(gst_jasper_dec_reset), (gst_jasper_dec_negotiate),
(gst_jasper_dec_get_picture):
* ext/jp2k/gstjasperdec.h:
Make pad template caps reflect the supported formats.
Add or modify some debug statements, and slightly simplify image
passing to encoding library.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain):
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew):
Small cleanups and some more debug info.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/aacparse.c:
* tests/check/elements/amrparse.c:
Add unit tests for new parsers.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/qtmux.c: (setup_src_pad),
(teardown_src_pad), (setup_qtmux), (cleanup_qtmux),
(check_qtmux_pad), (GST_START_TEST), (qtmux_suite), (main):
Add unit test for qtmux.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
Also configure the next expected output seqnum when we get a seqnum-base
on the caps.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst/h264parse/gsth264parse.c:
Wim, you're a bad boy. You don't want people to contact you or what?
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace2_class_init), (gst_deinterlace2_init),
(gst_deinterlace2_set_property), (gst_deinterlace2_get_property):
Bring properties into this century.
