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:
* 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.
Original commit message from CVS:
* gst/mpegdemux/gstmpegtsdemux.c:
Make private section pads have a caps set so they are not tried
to be linked in parse_launch for example.
Original commit message from CVS:
patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
No need to reclaculate flush in this case.
Fixes some bad decode errors introduced.
Original commit message from CVS:
patch by: Sebastian Pölsterl
* gst/mpegdemux/gstmpegdesc.c:
Length should be a guint8 not a gint.
* gst/mpegdemux/mpegtspacketizer.c:
Convert text to utf8 for each descriptor separately and not
concatenate them first and convert after.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
Add TODO at the top of the file for enabling SSE/ARM specific
optimizations and choosing the fastest implementation at runtime.
Add g_assert_not_reached() at two places that should really never
be reached.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_check_discont):
Fix format string and arguments.
* gst/speexresample/resample_sse.h:
Add missing file.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
(gst_speex_resample_convert_buffer), (_benchmark_int_float),
(_benchmark_int_int), (_benchmark_integer_resampling),
(plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add missing headers to Makefile.am.
Update copyright, years and my mail address.
Benchmark the integer resampling implementation against the
float implementation and use the faster one for 8/16 bit integer
input. On most recent systems the floating point version is faster.
Original commit message from CVS:
* configure.ac:
Use AC_TRY_COMPILE instead of AC_TRY_RUN as the result of the linking
is what is interesting, not that it actually runs.
Fixes cross-compilation and fixes bug #558639.
Original commit message from CVS:
* sys/qtwrapper/audiodecoders.c:
Add ALAC support.
Fix decode of mono AAC files created by itunes.
Set output format correctly (don't ask quicktime to
resample for us).
Use a larger decode buffer to avoid problems with large
ALAC packets.
Fix decode to loop until we have all output data.
* sys/qtwrapper/qtutils.c:
Fix includes so we compile on more OSes.
Original commit message from CVS:
* configure.ac:
Require at least Gtk 2.8.0 for the demos (that's the oldest I can
test with; I'm fairly certain Gtk 2.0.0 is not good enough any
longer); clean up some unused Gtk-related configure cruft.
* examples/scaletempo/demo-gui.c:
Define Gtk 2.12 function to noop when compiling against older Gtk.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
The length for the buffer conversion function is the number of
audio frames, i.e. we need to multiply it by the number of channels
to get the number of values. Also spotted by the unit test after
running in valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_funcs),
(gst_speex_resample_transform_size),
(gst_speex_resample_convert_buffer),
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add support for int8, int24 and int32 input by converting internally
to/from int16 or double.
Original commit message from CVS:
* ext/resindvd/resindvdsrc.c:
* ext/resindvd/resindvdsrc.h:
Better fix for #546319 and similar cases by explicitly
registering when we're in playing state or not.
Original commit message from CVS:
* ext/ladspa/gstladspa.c:
Whitespace.
* ext/ladspa/gstsignalprocessor.c:
Add a FIXME:. not sure if this code does the forwarding correctly.
Original commit message from CVS:
* gst/audiobuffer/Makefile.am:
* gst/audiobuffer/gstaudioringbuffer.c:
(gst_int_ring_buffer_acquire), (gst_int_ring_buffer_release),
(gst_int_ring_buffer_start), (gst_int_ring_buffer_base_init),
(gst_int_ring_buffer_class_init), (gst_int_ring_buffer_init),
(gst_int_ring_buffer_new), (gst_audio_ringbuffer_get_type),
(gst_audio_ringbuffer_class_init), (gst_audio_ringbuffer_init),
(gst_audio_ringbuffer_finalize), (gst_audio_ringbuffer_getcaps),
(gst_audio_ringbuffer_setcaps), (gst_audio_ringbuffer_bufferalloc),
(gst_audio_ringbuffer_handle_sink_event),
(gst_audio_ringbuffer_render), (gst_audio_ringbuffer_chain),
(gst_audio_ringbuffer_handle_src_event),
(gst_audio_ringbuffer_handle_src_query),
(gst_audio_ringbuffer_get_range),
(gst_audio_ringbuffer_src_checkgetrange_function),
(gst_audio_ringbuffer_sink_activate_push),
(gst_audio_ringbuffer_src_activate_push),
(gst_audio_ringbuffer_src_activate_pull),
(gst_audio_ringbuffer_change_state),
(gst_audio_ringbuffer_set_property),
(gst_audio_ringbuffer_get_property), (plugin_init):
Add first version of an audioringbuffer element that can be inserted in
the pipeline to convert push-based upstream into a pull-based
downstream.
Original commit message from CVS:
Patch by: Robin Stocker <robin at nibor dot org>
* gst/real/gstrealvideodec.c: (gst_real_video_dec_setcaps):
A RealVideo video inside a container (for example MKV) should use the
PAR which is specified on the sinkpad caps. Fixes#558416.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
Original commit message from CVS:
* ext/resindvd/resindvdsrc.c:
Make sure to start the NAV packet processing when changing
state to PLAYING by passing a flag that indicates the state
change is in progress.
Fixes: #546319
Original commit message from CVS:
* ext/resindvd/resin-play:
Remove $@ to fix parse_launch warning
* ext/resindvd/resin-play2:
Add a version that uses deinterlace and xvimagesink.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_loop), (gst_flv_demux_handle_seek_push),
(gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event),
(gst_flv_demux_dispose), (gst_flv_demux_init):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video), (gst_flv_parse_tag_timestamp):
Put the GstSegment directly into the instance struct instead of
allocating and free'ing it again.
Push tags already if only one pad was added, no need to wait for
the second one.
When generating our index set has_video and has_audio if we find
video or audio in case the FLV header has incorrect data.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
(gst_speex_resample_process):
Fixate to the nearest supported rate instead of the first one.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_pull_header),
(gst_flv_demux_create_index):
* 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_parse_tag_type),
(gst_flv_parse_header):
* gst/flv/gstflvparse.h:
Don't memcpy() all data we want to push downstream, instead just
create subbuffers and push them downstream.
Fix some minor memory leaks.
Original commit message from CVS:
* gst/flv/Makefile.am:
Fix (non-critical) syntax error and add all required CFLAGS and LIBS.
* gst/flv/gstflvparse.c: (FLV_GET_STRING),
(gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_timestamp), (gst_flv_parse_tag_type):
Rewrite the script tag parsing to make sure we don't try to read
more data than we have. Also use GST_READ_UINT24_BE directly and
fix some minor memory leaks.
This should make all crashes on fuzzed FLV files disappear.
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:
* ext/apexsink/gstapexplugin.c: (plugin_init):
Set apexsink's rank to NONE so it doesn't get used by
autoaudiosink (there's no point really). (#556588)
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:
2008-10-16 Michael Smith <msmith@songbirdnest.com>
* sys/acmenc/Makefile.am:
Remove incorrect use of DIRECTSOUND_LDFLAGS
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.
