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.
