Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c:
Actually copy the structure passed in when assigning it because
it gets freed straight after the function call.
Re: pat_info and pmt_info GstStructures.
Original commit message from CVS:
Patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
Fix wrong firing of critical introduced by previous optimisation.
Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_finalize),
(gst_base_parse_class_init), (gst_base_parse_push_buffer),
(gst_base_parse_change_state), (gst_base_parse_set_index),
(gst_base_parse_get_index):
Add support for GstIndex.
Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_class_init),
(gst_base_parse_push_buffer),
(gst_base_parse_update_upstream_durations),
(gst_base_parse_convert), (gst_base_parse_frame_in_segment):
* gst/flacparse/gstbaseparse.h:
Provide a vfunc for the subclass to decide whether a frame is inside
the segment or not and add a default implementation.
Fix approximate bitrate calculations.
Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_class_init),
(gst_base_parse_init), (gst_base_parse_push_buffer),
(gst_base_parse_update_upstream_durations), (gst_base_parse_chain),
(gst_base_parse_loop), (gst_base_parse_activate),
(gst_base_parse_convert), (gst_base_parse_query):
Approximate the average bitrate, duration and size if possible
and add a default conversion function which uses this for
time<->byte conversions.
* gst/flacparse/gstflacparse.c: (gst_flac_parse_get_frame_size):
Fix parsing if upstream gives -1 as duration.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes),
(on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout):
Ref the rtpsource object before we release the session lock when we emit
the signals.
Original commit message from CVS:
Patch from: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
* gst/mpegdemux/gstmpegtsdemux.h:
Use a preallocated buffer per stream for PES packets sent on src pads.
Adaptively adjust buffer size appropriately.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
Do not try to adjust the offset of streams for which we have not yet
seen an SR packet. Avoids large ts-offsets in some cases.
Original commit message from CVS:
* gst/aiffparse/aiffparse.c:
Support chunks in AIFF in any order in pull mode, and any order so
long as we get COMM before the actual data (SSND) in push mode.
Fixes playback of AIFC files.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_reset),
(gst_input_selector_reset), (gst_input_selector_change_state):
Reset the selector state when going to READY.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_parse_pack_start):
* gst/mpegdemux/gstmpegtsdemux.c: (gst_fluts_demux_data_cb):
Fix build on macosx.
Original commit message from CVS:
* ext/resindvd/plugin.c: (plugin_init):
* ext/resindvd/resindvdsrc.c:
* ext/twolame/gsttwolame.c: (plugin_init):
* gst/aiffparse/aiffparse.c: (plugin_init):
Enable/fix up translations for these plugins.
* po/LINGUAS:
Add 'ca' to LINGUAS.
* po/POTFILES.in:
* po/POTFILES.skip:
Add more files for translation and more files which tools
should skip.
Original commit message from CVS:
* gst/mpegtsmux/mpegtsmux_aac.c: (mpegtsmux_prepare_aac):
Allocate a fixed size buffer on the stack instead of using malloc().
* gst/mpegtsmux/tsmux/tsmux.c: (tsmux_new), (tsmux_free),
(tsmux_program_new), (tsmux_program_free):
* gst/mpegtsmux/tsmux/tsmuxstream.c: (tsmux_stream_new),
(tsmux_stream_free), (tsmux_stream_consume),
(tsmux_stream_add_data):
Use GSlice.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_input_selector_init),
(gst_input_selector_event), (gst_input_selector_query):
Reuse the get_linked_pads for both source and sinkpads because they are
the same.
Implement a custum event handler and get the internally linked pad
directly instead of relying on the default (slower) implementation.
Original commit message from CVS:
* gst/dccp/gstdccp.c:
* gst/dccp/gstdccpclientsrc.c:
Fix compilation on Solaris by including filio.h as needed.
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
Fix compilation with Forte - apparently it hates concatenating a
macro argument that starts with an underscore??
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
(gst_rtp_session_event_send_rtp_sink):
Send EOS when the session object instructs us to.
* gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible for the session manager to instruct us to send EOS. We
currently will EOS when the session is a sender and when the sender part
goes EOS. This is not entirely correct behaviour because the session
could still participate as a receiver.
Fixes#549409.
Original commit message from CVS:
* gst/aiffparse/aiffparse.c:
Read size of chunks preceeding the audio data with the
correct endianness. Fixes playback of some files.
Fixes#538500
Original commit message from CVS:
* configure.ac:
* gst/aiffparse/Makefile.am:
* gst/aiffparse/aiffparse.c:
* gst/aiffparse/aiffparse.h:
Add an AIFF parsing element, heavily based on wavparse.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_input_selector_init),
(gst_input_selector_query):
Implement the LATENCY query in a better way by taking the latency of all
sinkpads and taking the min/max instead of just taking a random pad.
Original commit message from CVS:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
* gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
Unroll the loop to handle two bytes at once. This should give
a small speedup and makes it possible to handle chroma and luma
different which is needed later.
Original commit message from CVS:
* gst/dccp/gstdccpserversink.c:
* gst/dccp/gstdccpserversink.h:
Don't put globals only used by one '.c' file in a header !
Declare it as static, fixes build on macosx.
Original commit message from CVS:
* gst/dccp/gstdccp.c: (gst_dccp_read_buffer),
(gst_dccp_send_buffer), (gst_dccp_set_sock_windowsize):
size_t's size varies by platform/architecture. Use glib convenience
macro instead. Fixes build on macosx.
Remove ending '\n' in debug statements.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace_method_class_init):
* gst/deinterlace2/gstdeinterlace2.h:
* gst/deinterlace2/tvtime/tomsmocomp.c:
(gst_deinterlace_method_tomsmocomp_class_init):
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
* gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
* gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
First part of the C implementation of the tomsmocomp deinterlacing
algorithm. This only supports search-effort=0 currently, is painfully
slow and needs some cleanup later when all search-effort settings
are implemented in C.
Original commit message from CVS:
* gst/pcapparse/gstpcapparse.c:
* sys/winscreencap/gstdx9screencapsrc.c:
* sys/winscreencap/gstgdiscreencapsrc.c:
Added documentation blobs. Thanks to Stefan for noticing!
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
Reset rtp timestamp interpollation when we detect a gap when the
clock_base changed.
Don't try to adjust the ts-offset when it's too big (> 3seconds)
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
* gst/rtpmanager/gstrtpsession.h:
Add method to set session SSRC.
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Added debugging for the collision checks.
Add method to change the internal SSRC of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Reset the clock base when we detect large jumps in the seqnums.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Print the pad-name in debug log.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
Use "-" instead of "_" in property names. Can we call them just
"device" like everywhere else?
Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Make the buffer metadata writable before inserting it in the
jitterbuffer because the jitterbuffer will modify the timestamps.
* gst/rtpmanager/rtpjitterbuffer.c:
Update method comment about requiring writable metadata on buffers.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_rtcp):
Make the RTCP buffer metadata writable because we want to modify the
metadata.
Fixes#546312.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_input_selector_getcaps),
(gst_input_selector_activate_sinkpad):
Move the select-all logic into the activation of the currently selected
pad. We want to remember the last pad with activity in select-all mode.
Fix the getcaps function, we can produce the union of the upstream caps
in select-all mode, not the intersection like proxy_getcaps() does.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Fix debug by logging the right seqnum.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (get_pt_map):
Release lock before emitting the request-pt-map signal.
Fixes#543480.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace_simple_method_interpolate_scanline),
(gst_deinterlace_simple_method_copy_scanline),
(gst_deinterlace_simple_method_deinterlace_frame):
* gst/deinterlace2/tvtime/greedy.c: (deinterlace_frame_di_greedy):
* gst/deinterlace2/tvtime/greedyh.c:
(deinterlace_frame_di_greedyh):
* gst/deinterlace2/tvtime/scalerbob.c:
(deinterlace_scanline_scaler_bob):
* gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy):
* gst/deinterlace2/tvtime/weave.c: (deinterlace_scanline_weave),
(copy_scanline):
* gst/deinterlace2/tvtime/weavebff.c: (deinterlace_scanline_weave),
(copy_scanline):
* gst/deinterlace2/tvtime/weavetff.c: (deinterlace_scanline_weave),
(copy_scanline):
Use oil_memcpy() instead of memcpy() as it's faster for the sizes that
are usually used here.
