Original commit message from CVS:
* gst/flv/gstflvparse.c: (FLV_GET_STRING),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_type), (gst_flv_parse_header):
Properly check everywhere that we have enough data to parse and
don't read outside the allocated memory region.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
If the caps change during playback and negotiation fails error out
instead of trying to continue.
Original commit message from CVS:
* gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_request_new_pad), (gst_flv_mux_write_buffer),
(gst_flv_mux_collected):
* gst/flv/gstflvmux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate):
Add support for Speex audio and allow buffers without valid
timestamp in the muxer.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_loop),
(gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push),
(gst_flv_demux_handle_seek_pull):
Don't post an error message on the bus if sending EOS downstream
didn't work. Fixes bug #550454.
Fix seek event handling to look at the flags of the seek event
instead of assuming some random flags, don't send segment-start
messages when operating in push mode and push seek events upstream
if we couldn't handle them.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_create_index),
(gst_flv_demux_loop):
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_timestamp):
* gst/flv/gstflvparse.h:
In pull mode we create our own index before doing anything else
and don't use the index provided by some files (which are more than
often incorrect and cause failed seeks).
For push mode we still use the index provided by the file and extend it
while doing the playback.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_push_src_event),
(gst_flv_demux_loop), (gst_flv_demux_handle_seek_pull),
(gst_flv_demux_sink_event):
Instead of using gst_pad_event_default() use a small
gst_pad_push_event() wrapper that only does what we want and is much
more simple.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_change_state),
(gst_flv_demux_set_index), (gst_flv_demux_init):
* gst/flv/gstflvdemux.h:
If our index was created by the element and not provided from the
outside we should destroy it when starting a new stream to get
all old entries removed.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range):
Improve debugging a bit when pulling a buffer from upstream fails.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_handle_seek_pull), (gst_flv_demux_dispose):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Close the currently playing segment from the streaming thread
instead of the thread where the seek event is handled.
Original commit message from CVS:
Patch by: David Härdeman <david at hardeman dot nu>
* gst/mpegdemux/mpegtspacketizer.c: (mpegts_packetizer_parse_nit):
Add support for the frequency list descriptor, which provides
additional frequencies that should be scanned by a DVB application.
Fixes bug #557814.
Original commit message from CVS:
Patch by: vanista <vanista at gmail dot com>
* gst/mpegtsmux/mpegtsmux.c: (mpegtsmux_choose_best_stream):
Fix EOS logic by correctly popping the collect pad buffers only
when we've chosen to use them instead of popping them always and
storing them in a private queue.
Before the pipeline would deadlock if all pads go EOS at the same
time. Fixes bug #557763.
Original commit message from CVS:
Patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
* gst/mpegdemux/gstmpegtsdemux.h:
Properly handle some resync cases in the optimised
buffering strategy.
Original commit message from CVS:
* gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_write_buffer):
Don't set video_codec to the value that actually should go
into audio codec, otherwise we create invalid files.
Fixes bug #556564.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix problem with using the output seqnum counter to check for input
seqnum discontinuities.
Improve gap detection and recovery, reset and flush the jitterbuffer on
seqnum restart. Fixes#556520.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert):
Fix wrong G_LIKELY.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src):
Install event handler on the rtcp_src pad, make LATENCY event return
TRUE.
Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_send_data):
Make sure the mpegpsdemux element creates valid newsegment events.
Fixes#556428
Original commit message from CVS:
patch by: Sebastian Pölsterl
* gst/mpegdemux/mpegtspacketizer.c:
Fixes segfault in get_encoding_and_convert.
Fixes#556482
Original commit message from CVS:
patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
Fixes a segfault in the adaptation buffer size strategy.
Fixes#556440
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_input_selector_event),
(gst_input_selector_query):
Gracefully handle the cases when we dont' have otherpad.
Fixes#556430
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Use gst_pad_alloc_buffer_and_set_caps() to make sure we get
a buffer with caps that we can work with (i.e. the pad's caps).
Add non-keyframe video frames to the index too but without the
keyframe flag.
Add audio frames to the index only if we have no video stream.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Create pads from the pad templates, use fixed caps on them
and only activate them after the caps are set.
Original commit message from CVS:
2008-10-10 Jan Schmidt <jan.schmidt@sun.com>
* gst/flacparse/gstbaseparse.c (gst_base_parse_push_buffer),
(gst_base_parse_update_upstream_durations):
Fix compiler warning on OS/X about parameters not matching
the debug format string.
Original commit message from CVS:
* gst/deinterlace2/tvtime/tomsmocomp.c:
(gst_deinterlace_method_tomsmocomp_class_init):
Fix unused variable compiler warning when not building
X86 assembly.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_loop):
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_timestamp):
* gst/flv/gstflvparse.h:
Get an approximate duration of the file by looking at the timestamp
of the last tag in pull mode. If we get (maybe better) duration from
metadata later we'll use that instead.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range),
(gst_flv_demux_pull_tag), (gst_flv_demux_pull_header):
Refactor _pull_range() logic with checks into a seperate function
to make things a bit more readable.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain),
(gst_flv_demux_base_init):
Use gst_element_class_set_details_simple().
If we get GST_FLOW_NOT_LINKED in the parse loop but at least
one of the pads is linked continue the loop.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate),
(gst_flv_parse_tag_audio), (gst_flv_parse_video_negotiate):
Correct caps for video codec id 5: It's On2 VP6 with alpha channel
which needs a different decoder and has different caps.
Add support for audio codec id 14, which is MP3 with 8kHz sampling
rate.
Fix endianness and signedness for raw audio codec ids.
Add support for alaw and mulaw audio.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain):
Go out of the parse loop as soon as we get an error instead
of parsing until the GstAdapter is empty.
Add some explanations about the header and tag size.
Don't print synchronizing message if everything is fine.
Original commit message from CVS:
* gst/flv/Makefile.am:
* gst/flv/gstflvdemux.c: (plugin_init):
* gst/flv/gstflvmux.c: (gst_flv_mux_base_init),
(gst_flv_mux_class_init), (gst_flv_mux_init),
(gst_flv_mux_finalize), (gst_flv_mux_reset),
(gst_flv_mux_handle_src_event), (gst_flv_mux_handle_sink_event),
(gst_flv_mux_video_pad_setcaps), (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_request_new_pad), (gst_flv_mux_release_pad),
(gst_flv_mux_write_header), (gst_flv_mux_write_buffer),
(gst_flv_mux_collected), (gst_flv_mux_change_state):
* gst/flv/gstflvmux.h:
Add first version of a FLV muxer. The only missing feature is writing
of stream metadata.
Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_send_data),
(gst_flups_demux_parse_pack_start):
Prevent a division by zero if last mux rate was zero.
If we're going to send a NEWSEGMENT event but the segment start
and the current buffer timestamp differ by more than a second we
will start the NEWSEGMENT at the buffer timestamp.
This fixes playback of the tv2-1_25.mpg file, which has 0 as first SCR
but the first PTS are around 1 hour and 40 minutes.
Fixes bug #553755.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin-marshal.list:
Add marshaller for new action signal.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add action signal to retrieve the internal RTPSession object.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_get_property), (gst_rtp_session_release_pad):
Add property to access the internal RTPSession object.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(check_collision):
* gst/rtpmanager/rtpsession.h:
Add action signal to retrieve an RTPSource object by SSRC.
See #555396.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps):
Only update the seqnum-base when it was not already configured for the
streams.
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.
Original commit message from CVS:
* gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c),
(deinterlace_line_mmx), (gst_deinterlace_method_vfir_class_init):
Implement the VFIR deinterlacing method as simple method.
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_deinterlace_simple_method_class_init),
(gst_deinterlace_simple_method_init):
* gst/deinterlace2/gstdeinterlace2.h:
Add a GstDeinterlaceSimpleMethod subclass of GstDeinterlaceMethod that
can be used by simple deinterlacing methods. They only have to provide
a function for interpolating a scanline or copying a scanline.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_chain):
Respect the latency of the deinterlacing algorithm for the timestamps
of every buffer.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedyh.asm:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
Add the MMX registers to the clobbered registers only if __MMX__ is
defined.
Original commit message from CVS:
* gst/deinterlace2/Makefile.am:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method),
(gst_deinterlace2_class_init):
Enable tomsmocomp again as the C port will be ready for the next
release.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_init),
(gst_greatest_common_divisor), (gst_fraction_double),
(gst_deinterlace2_getcaps), (gst_deinterlace2_setcaps):
Don't use proxy_getcaps() but implement our own getcaps() function
that doubles/halfs the framerate if all fields should be sent out.
Original commit message from CVS:
* gst/festival/gstfestival.c:
Guard unistd.h with HAVE_UNISTD_H
* gst/modplug/libmodplug/load_it.cpp:
* gst/modplug/libmodplug/sndfile.cpp:
* gst/modplug/libmodplug/sndfile.h:
* gst/modplug/libmodplug/stdafx.h:
Support for compiling with MSVC: use _MSC_VER for detecting MSVC instead
of MSC_VER. Make CanPackSamples take the type it's passed. Change scope
of a variable in load_it.cpp to MSVC's scoping rules.
* gst/sdp/gstsdpdemux.c:
Guard unistd.h with HAVE_UNISTD_H
Fixes#544457.
Original commit message from CVS:
* gst/modplug/libmodplug/Makefile.am:
Use GST_CXXFLAGS instead of GST_CFLAGS for CXXFLAGS. GST_CFLAGS
can contain compiler parameters that are invalid for C++.
Fixes bug #543860.
