Original commit message from CVS:
* ext/mpeg2enc/gstmpeg2encpicturereader.cc:
(GstMpeg2EncPictureReader.StreamPictureParams):
Fix compilation with libmjpegtools 1.8.x.
Original commit message from CVS:
* ext/ladspa/gstsignalprocessor.c:
(gst_signal_processor_class_add_pad_template):
Don't unref the pad template after adding it.
gst_element_class_add_pad_template takes ownership of it.
Original commit message from CVS:
* ext/mpeg2enc/gstmpeg2encpicturereader.cc:
Use the incoming pixel-aspect-ratio if provided to infer a
default aspect ratio, which can be overridden using the 'aspect'
property.
Fixes: #499008
Original commit message from CVS:
Patch by: Andrzej Mendel <andrzej dot mendel at gmail dot com>
* configure.ac:
Fix variable naming to make it possible to build the glimagesink
plugin. Fixes bug #514093.
Original commit message from CVS:
* ext/metadata/gstmetadatademux.c:
Demote metadatademux to GST_RANK_NONE for the release, it's not
ready to be autoplugged yet.
* tests/icles/metadata_editor.c:
Fix printf format warning for GType on ppc32 by removing it,
since it doesn't make sense to print the GType value anyway.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_event):
Don't leak event on pads that are not linked. Fixes#512826.
Original commit message from CVS:
* configure.ac:
Bump core/base requirements to released versions, to avoid confusion.
* gst/deinterlace/Makefile.am:
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_set_caps):
Use the new GstVideoFormat API to get strides, plane offsets etc..
For Y42B we still need to calculate these ourselves, since the lib
in -base doesn't know about this format yet and we can't bump the
requirement to CVS right now. Fix the Y42B stride, offset and size
calculations for odd widths and heights while we're at it though
(to match those in videotestsrc).
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
Really clean up the queue instead of just unreffing all buffers
in it.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_dispose), (gst_app_src_finalize):
Fix dispose/finalize.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_to_gst):
Fix compiler warning by making the function signature match what
everyone is passing in...
* tests/icles/Makefile.am:
Fix the build on Solaris by removing GNU ld specific flags that
look unnecessary.
Original commit message from CVS:
* configure.ac:
* ext/metadata/metadataxmp.c:
(metadatamux_xmp_for_each_tag_in_list):
Fix build with exempi >= 1.99.5 and fix the include
path for exempi.
Original commit message from CVS:
* ext/gio/gstgiobasesink.c: (close_stream_cb),
(gst_gio_base_sink_stop), (gst_gio_base_sink_event),
(gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesrc.c: (close_stream_cb),
(gst_gio_base_src_stop), (gst_gio_base_src_create),
(gst_gio_base_src_set_stream):
Use async variants of the close stream functions to prevent blocking
for a long time there and add some more sanity checks for a correct
stream.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_init):
Let the proxy property default to the content of the $http_proxy
environment variable.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* tests/check/test-cert.pem:
* tests/check/test-key.pem:
Add missing files for the unit test.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Try to get the new clock-rate from the buffer caps when we receive a new
payload type instead of always firing the signal. Fixes#512774.
Original commit message from CVS:
* ext/mpeg2enc/gstmpeg2enc.cc:
Define LOG_NONE and friends if they're not defined yet. mjpegtools
1.9.0rc3 removed their definitions but without it doesn't make much
sense to write a log handler.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.types:
Add base classes for metadata and equalizer (no introspection yet).
Original commit message from CVS:
* gst/h264parse/gsth264parse.c: (gst_h264_parse_chain_forward):
Try to avoid 'unused variable' compiler warning if debugging is
disabled (not bullet proof, but seems to do for now). (#512654)
Original commit message from CVS:
* ext/soundtouch/gstbpmdetect.cc:
Clean up a bit and only allocate a temporary buffer for the data
if processing stereo data as BPMDetect downmixes from stereo to
mono and stores the result in the input data. Thanks to
Stefan Kost for the suggestions.
Original commit message from CVS:
* ext/soundtouch/gstpitch.cc:
* ext/soundtouch/gstpitch.hh:
Implement LATENCY query and notify about latency changes.
Unfortunately we don't have a fixed latency but it changes
a bit with each buffer so we only send an LATENCY event with
the maximum latency if it changes.
Always calculate the timestamp, duration, etc from the sample
rate instead of using a pre-calculated duration for one sample
to prevent large rounding errors.
Original commit message from CVS:
Based on a patch by:
Hans de Goede <j dot w dot r dot degoede at hhs dot nl>
* configure.ac:
* ext/mpeg2enc/gstmpeg2encoder.cc:
* ext/mpeg2enc/gstmpeg2encpicturereader.cc:
* ext/mpeg2enc/gstmpeg2encpicturereader.hh:
Add support for building against mjpegtools 1.9 while keeping
compatiblity with older versions.
Original commit message from CVS:
* ext/soundtouch/Makefile.am:
* ext/soundtouch/gstbpmdetect.cc:
* ext/soundtouch/gstbpmdetect.hh:
* ext/soundtouch/plugin.c: (plugin_init):
Add BPM detection plugin based on SoundTouch's libBPM.
* ext/soundtouch/gstpitch.cc:
Allow sample rates until MAX instead of only 48kHz and remove the
buffer-frames field from that caps.
Clear the remaining samples completely when necessary to get into
a clean state again.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Don't implement get_unit_size() ourselves, the GstAudioFilter base
class already does this for us.
Original commit message from CVS:
* ext/soundtouch/gstpitch.cc:
Allow seeking only in TIME and DEFAULT format, other formats will
not work as expected. Also handle a stop position of -1 correctly
for seeks, newsegment events and the queries. This fixes playback
with the pitch element if upstream doesn't know the duration or has
-1 as stop position in NEWSEGMENT events for other reasons. Before
simply nothing was played as the segment was going from 0 to 0.
Send a GST_MESSAGE_DURATION whenever the rate or tempo is changed
so applications can update their cached duration. Fixes bug #503308.
Some random cleanup and memory leak closing.
Original commit message from CVS:
* ext/musepack/gstmusepackdec.h:
* ext/musepack/gstmusepackreader.c:
First include the libmpcdec headers before everything else as they
#define TRUE and FALSE unconditionally and we otherwise get conflicts
with the ones that GLib defines.
Original commit message from CVS:
* configure.ac:
* ext/soundtouch/gstpitch.cc:
Add support for libsoundtouch 1.3.1 and add an ugly workaround for
the header definined PACKAGE and other variables for which we need
our own values from config.h.
Original commit message from CVS:
* configure.ac:
Check for libglade-2.0, for the metadata-editor example.
* tests/icles/Makefile.am:
Only try to build the metadata-editor example if we have gtk and
glade (otherwise the build would just fail ...); fix build in
uninstalled setup.
* tests/icles/metadata_editor.c: (on_cell_edited), (ui_add_columns):
Fix compiler warnings (use GLib macros to cast pointer <-> int).
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(create_stream), (payload_type_change), (new_ssrc_pad_found):
Also handle lip-sync when the clock-rate is not provided with caps but
with a signal.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
(rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove the fixed clock-rate from the jitterbuffer and extend it so that
a clock-rate can be provided with each buffer instead. Fixes#511686.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Remove old unused variable.
Track pt on input buffers and get the clock-rate when it changes.
Ignore packets with unknown clock-rate. See #511686.
Original commit message from CVS:
* sys/dvb/gstdvbsrc.c:
Post bus message about adapter type and it's capabilities,
when opening the frontend.
After failing to read from the dvr, post a bus message to
inform the app.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/rtpsource.c: Fix unref of buffer using the
wrong function. Fixes#511920
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
Now we have full hierarchy.
* docs/plugins/inspect/plugin-metadata.xml:
Regenerate.
* ext/amrwb/gstamrwbdec.h:
Add doc blob for object instance.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/inspect/plugin-metadata.xml:
Update this too, hopefully fixes the docs build (does at least
for me, after make clean in docs/plugins).
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
Fix network name descriptor, the length is actually the
descriptor length not stored in the byte after.
Fix bounds checking to be more correct.
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
Parse and add to relevant bus messages the terrestrial delivery
system descriptor and the logical channel descriptor.
Do bounds checking on data stored in descriptor before use.
Original commit message from CVS:
* configure.ac:
* ext/dts/gstdtsdec.c:
Add support for building against libdca (with the libdts compat
header). Fixes bug #511530.
Should probably be ported to libdca as some points as it's the
successor of libdts.
Original commit message from CVS:
* sys/dvb/gstdvbsrc.c:
Do not go on forever if problem with reading from dvr, rather
return NULL.
Handle some cleanup issues of closing filedescriptors when
failing to tune or similar.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/xingheader/gstxingmux.c:
* gst/xingheader/gstxingmux.h:
Add documentation for the xingheader plugin.
* tests/check/elements/xingmux.c: (GST_START_TEST):
Set element state to PLAYING before doing something else.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/xingmux.c: (setup_xingmux),
(cleanup_xingmux), (GST_START_TEST), (xingmux_suite), (main):
* tests/check/elements/xingmux_testdata.h:
Add simple unit test for the xingmux element.
* gst/xingheader/gstxingmux.c: (generate_xing_header),
(gst_xing_mux_finalize), (xing_reset):
Fix a memleak and invalid seek tables with less than 100 MP3 frames.
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
Parsed the satellite delivery system descriptor and
added into nit's transport structure for delivery
over the bus.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
Remove leaks introduced by not freeing g_strndup'd strings.
Fix start_time and duration parsing in EIT.
Original commit message from CVS:
* gst/mpegtsparse/Makefile.am:
* gst/mpegtsparse/gstmpegdesc.c:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
Added descriptor searching infrastructure from Fluendo TS demuxer.
Add channel name and provider to the sdt structure sent in the
bus message.
Original commit message from CVS:
2008-01-22 Julien Moutte <julien@fluendo.com>
* gst/h264parse/gsth264parse.c: (gst_h264_parse_chain_forward):
Parse NAL units in forward mode to mark delta units flags.
Original commit message from CVS:
* docs/plugins/Makefile.am:
Add missing eol \
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
Place object names to standard sectionas plugin dont document those.
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-dvb.xml:
* docs/plugins/inspect/plugin-nuvdemux.xml:
regenerate.
* ext/ivorbis/vorbisdec.c:
* ext/ivorbis/vorbisdec.h:
Mark private vars and add short desc.
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
Add short desc.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/nuvdemux/gstnuvdemux.c:
One less to do. Its 'nuv' not 'nvu'. As an extra bonus I mention what
it actually is.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
Update lists again. Those whole can build ivorbisdec, mythtvsrc,
nvudemux and theoradecexp, please commit the inspect/plugin-xxx.xml.
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-rawparse.xml
* docs/plugins/inspect/plugin-videoparse.xml:
Replace videoparse with rawparse.
* gst/dvdspu/gstdvdspu.h:
Help gtk-doc to recognize the object struct.
Original commit message from CVS:
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
Don't use gtk-doc comment style for non gtk-doc comments.
Make one static function static.
Original commit message from CVS:
Patch by: Gabriel Bouvigne <bouvigne at mp3-tech dot org>
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init),
(gst_deinterlace_init), (gst_deinterlace_set_caps),
(gst_deinterlace_transform_ip), (gst_deinterlace_set_property),
(gst_deinterlace_get_property):
* gst/deinterlace/gstdeinterlace.h:
Provide 4:2:2 support
Also deinterlace chroma planes
Allow to turn on/off deinterlacing
Change of default thresholds, in order to provide acceptable results
with default params. Fixes#511001.
Original commit message from CVS:
* gst/dvdspu/gstdvdspu-render.c: (gst_dvd_spu_render_spu):
* gst/dvdspu/gstdvdspu.c: (dvdspu_debug), (GST_CAT_DEFAULT),
(subpic_sink_factory), (gst_dvd_spu_base_init),
(gst_dvd_spu_class_init), (gst_dvd_spu_init), (gst_dvd_spu_clear),
(gst_dvd_spu_dispose), (gst_dvd_spu_finalize),
(gst_dvd_spu_flush_spu_info), (gst_dvd_spu_buffer_alloc),
(gst_dvd_spu_src_event), (gst_dvd_spu_video_set_caps),
(gst_dvd_spu_video_proxy_getcaps), (gst_dvd_spu_video_event),
(gst_dvd_spu_video_chain), (dvspu_handle_vid_buffer),
(gst_dvd_spu_redraw_still), (gst_dvd_spu_parse_chg_colcon),
(gst_dvd_spu_exec_cmd_blk), (gst_dvd_spu_finish_spu_buf),
(gst_dvd_spu_setup_cmd_blk), (gst_dvd_spu_handle_new_spu_buf),
(gst_dvd_spu_handle_dvd_event), (gst_dvd_spu_advance_spu),
(gst_dvd_spu_check_still_updates), (gst_dvd_spu_subpic_chain),
(gst_dvd_spu_subpic_event), (gst_dvd_spu_change_state),
(gst_dvd_spu_plugin_init):
* gst/dvdspu/gstdvdspu.h: (GST_TYPE_DVD_SPU):
Fix up dvdspu element again after previous namespace mangling:
rename debug category variable to old name, matching that in
dvdspu-render.c, to avoid undefined symbol error when loading
the module; same for the _render function in dvdspu-render.c:
we must use the same name in both .c files; change functions
now called gstgst_* back to gst_* again; and while we're at it,
we may as well canonicalise the namespace properly, namely to
gst_dvd_spu_*.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* ext/theora/theoradec.c:
* ext/theora/theoradec.h:
Coherent namespace usage and adding symbold from unused to sections.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
Add symbols from -unused.txt to the right place.
* gst/dvdspu/gstdvdspu.c:
* gst/dvdspu/gstdvdspu.h:
Coherent namespace usage.
* gst/spectrum/gstspectrum.c:
Fix broken XML fragment in doc snippet even more.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (soup_got_headers):
Report the size of the stream as the total size instead of
the remaining Content-Length, which is wrong after a seek.
Original commit message from CVS:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_push_buffer),
(gst_raw_parse_loop):
Handle framesizes > 4096 with multiple frames per buffer correctly
in pull mode and handle short reads better.
Also put offset and offset_end on outgoing buffers.
Original commit message from CVS:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_loop):
Improve handling of unknown or too small upstream sizes in
pull mode.
Original commit message from CVS:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_loop),
(gst_raw_parse_handle_seek_push):
Improve debugging a bit and for handling multiple frames per buffer
in pull mode choose the next smallest multiply of framesize below
4096 instead of always handling 1024 frames.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (soup_got_headers):
Correctly set duration on the GstBaseSrc segment when we know it
to fix failing the duration query.
Original commit message from CVS:
* gst/h264parse/gsth264parse.c: (gst_h264_parse_flush_decode),
(gst_h264_parse_queue_buffer), (gst_h264_parse_chain_reverse):
Set timestamps more correctly.
Original commit message from CVS:
* tests/check/Makefile.am:
Enable spectrum test again.
* tests/check/gst-plugins-bad.supp:
Add suppressions for a singleton in GIO that can't be freed.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/equalizer.c: (setup_equalizer),
(cleanup_equalizer), (GST_START_TEST), (equalizer_suite), (main):
Add some minimal tests for the equalizer plugin.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_finalize):
Unparent all bands from the equalizer when finalizing to stop
leaking them.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init),
(gst_souphttp_src_init), (gst_souphttp_src_create),
(gst_souphttp_src_is_seekable), (gst_souphttp_src_do_seek),
(soup_add_range_header), (soup_got_headers), (soup_got_chunk):
* ext/soup/gstsouphttpsrc.h:
Add support for seeking to souphttpsrc. Fixes bug #502335.
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Update for API changes in GIO and require GIO 2.15.2 for this.
Original commit message from CVS:
* gst/xingheader/gstxingmux.c: (generate_xing_header):
Bitrate is 4 bits, not 8 so check for 0xe as maximum value instead
of 0xfe.
Original commit message from CVS:
* gst/xingheader/gstxingmux.c: (has_xing_header),
(generate_xing_header), (gst_xing_mux_chain),
(gst_xing_mux_sink_event):
Choose smallest possible frame size for the Xing header, properly
set the timestamp, duration and offset on the outgoing buffers,
only send NEWSEGMENT events in BYTE format downstream and also
drop VBRI headers if already existing.
Original commit message from CVS:
* gst/xingheader/Makefile.am:
* gst/xingheader/gstxingmux.c: (parse_header), (get_xing_offset),
(has_xing_header), (generate_xing_header),
(gst_xing_mux_base_init), (gst_xing_mux_finalize), (xing_reset),
(gst_xing_mux_init), (gst_xing_mux_chain),
(gst_xing_mux_sink_event), (gst_xing_mux_change_state):
* gst/xingheader/gstxingmux.h:
Major cleanup and rewrite of xingmux with less bugs and new features:
- Handles other layers as 3
- Write TOC
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
If we find the caps in the cache, use it to parse the clock-rate instead
of returning an error. Fixes a TODO as found by Youness Alaoui.
Original commit message from CVS:
Patch by: Youness Alaoui <youness dot alaoui at collabora dot co dot uk>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(rtp_session_set_process_rtp_callback),
(rtp_session_set_send_rtp_callback),
(rtp_session_set_send_rtcp_callback),
(rtp_session_set_sync_rtcp_callback),
(rtp_session_set_clock_rate_callback),
(rtp_session_set_reconsider_callback), (source_push_rtp),
(source_clock_rate), (rtp_session_process_bye),
(rtp_session_process_rtcp), (rtp_session_send_bye),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible to use different user_data for each of the callbacks.
Fixes#508587.
Original commit message from CVS:
* gst/mpegvideoparse/mpegpacketiser.c: (mpeg_util_find_start_code):
Small meaningless cleanup.
* gst/mpegvideoparse/mpegvideoparse.c: (gst_mpegvideoparse_flush),
(mpegvideoparse_drain_avail), (gst_mpegvideoparse_chain_forward),
(scan_keyframe), (gst_mpegvideoparse_flush_decode),
(gst_mpegvideoparse_chain_reverse), (gst_mpegvideoparse_chain),
(mpv_parse_sink_event), (gst_mpegvideoparse_change_state):
* gst/mpegvideoparse/mpegvideoparse.h:
Track segment events.
Do the first part of reverse playback by sending data between two
I-frames to the decoder.
Original commit message from CVS:
* autogen.sh:
Add -Wno-portability to the automake parameters to stop warnings
about GNU make extensions being used. We require GNU make in almost
every Makefile anyway.
* configure.ac:
Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
at the same time is required for per target flags.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (create_send_rtp_sink):
Don't set fixed caps, we can basically do everything the upsteam peer
pad can renegotiate to. Fixes#507940.
Original commit message from CVS:
* Makefile.am:
Include lcov.mak to allow building coverage reports. Add top-level
check targets similar to other gst packages.
Original commit message from CVS:
* ext/directfb/Makefile.am:
Add GST_CFLAGS. Otherwise we don't get -Wall -Werror.
* ext/directfb/dfbvideosink.c:
Getting tired of directfb's chatter. Quiet it.
Original commit message from CVS:
* configure.ac:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
* tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
Update to GMemoryInputStream API changes in GLib SVN and require
gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
We can also report the duration for every GSeekable, not only
GFileInputStream and GMemoryInputStream.
Original commit message from CVS:
* sys/glsink/glimagesink.c:
* sys/glsink/glimagesink.h:
* sys/glsink/gstglbuffer.h:
* sys/glsink/gstgldisplay.c:
* sys/glsink/gstgldisplay.h:
* sys/glsink/gstglupload.c:
Handle xoverlay exposes correctly. This means glimagesink works
correctly most of the time in totem (fullscreening being an
execption). Doesn't handle expose events directly to the GL
window.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Don't unref the popped buffer when we don't have ownership.
Fixes#507020.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable vcdsrc in states test because it takes too much time
to get to PLAYING if it can find a device.
Original commit message from CVS:
* ext/musicbrainz/gsttrm.c:
Don't emit signiture when going to READY, because it might
not be ready.
* ext/nas/nassink.c:
Remove useless call that sleeps for 5 seconds. Yup, it calls
sleep(1) 5 times. Go NAS.
* gst/librfb/gstrfbsrc.c:
* gst/librfb/rfbdecoder.c:
Initialize our debug categories properly.
* gst/rawparse/gstrawparse.c:
Don't register element details for a non-element. Be much more
rude when subclass doesn't set a pad template (assert!). Don't
unref the pad template; we don't own it.
* gst/videosignal/gstvideoanalyse.c:
Initialize debug category.
* tests/check/Makefile.am:
Ignore nassink element in tests because it has unavoidable
long timeouts.
Original commit message from CVS:
* configure.ac:
* sys/glsink/Makefile.am:
Switch to using pkgconfig to detect libGL. Since we use
recent features added to Mesa, there's no point in adding
a check for pre-pkgconfig versions.
Original commit message from CVS:
* ext/ladspa/gstladspa.c: (gst_ladspa_get_property):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_show_frame):
* gst/mve/gstmvemux.c: (gst_mve_mux_request_new_pad):
* sys/dvb/dvbbasebin.c: (dvb_base_bin_class_init):
Fix 'xyz may be used uninitialized' compiler warnings caused
by broken g_assert_not_reached() macro in GLib-2.15.x and don't
abort() in any case but properly report the error.
Original commit message from CVS:
* ext/soup/Makefile.am:
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_get_property),
(gst_souphttp_src_unicodify), (soup_got_headers):
Use gst_tag_freeform_string_to_utf8() and post radio station
info as tags on the bus.
Original commit message from CVS:
* sys/glsink/glimagesink.c:
* sys/glsink/gstglupload.c:
Change glimagesink over to using GL buffers. This breaks
glimagesink for normal operation, but should be fixed soon.
Original commit message from CVS:
* sys/glsink/gltestsrc.c:
* sys/glsink/gstglbuffer.c:
* sys/glsink/gstglbuffer.h:
* sys/glsink/gstgldownload.c:
* sys/glsink/gstglfilter.c:
* sys/glsink/gstglfilterexample.c:
* sys/glsink/gstgltestsrc.c:
* sys/glsink/gstglupload.c:
Convert gldownload to BaseTransform. Make glfilterexample
visually interesting. Add support for various formats to
downloading. Fix a few places where we leak GL state to
other elements (bad, but hard to prevent).
Original commit message from CVS:
* sys/glsink/BUGS:
* sys/glsink/Makefile.am:
* sys/glsink/gstglbuffer.c:
* sys/glsink/gstglbuffer.h:
* sys/glsink/gstglconvert.c:
* sys/glsink/gstgldisplay.c:
* sys/glsink/gstglfilter.c:
* sys/glsink/gstglfilter.h:
* sys/glsink/gstglfilterexample.c:
* sys/glsink/gstgltestsrc.c:
* sys/glsink/gstglupload.c:
* sys/glsink/gstopengl.c:
Remove code that handles non-texture buffers. Add a
GstGLBufferFormat type that corresponds to how to use the
texture, not the original video format. Convert gstflfilter.c
into a base class, add glfilterexample and glconvert elements.
* sys/glsink/color_matrix.c:
Minor ramblings about color conversion matrices.
Original commit message from CVS:
* sys/glsink/gstglbuffer.c:
* sys/glsink/gstgldownload.c:
* sys/glsink/gstglfilter.c:
Clean up code. Fix a few leaks.
Original commit message from CVS:
* sys/glsink/glimagesink.c:
* sys/glsink/glimagesink.h:
* sys/glsink/glvideo.c:
* sys/glsink/glvideo.h:
* sys/glsink/gstglbuffer.c:
* sys/glsink/gstglbuffer.h:
* sys/glsink/gstgldownload.c:
* sys/glsink/gstglfilter.c:
* sys/glsink/gstglupload.c:
Rewrite a bunch of code to use textures as the intermediate
instead of renderbuffers. upload, download, filtering all
work.
Original commit message from CVS:
* gst/videoparse/Makefile.am:
* gst/videoparse/README:
* gst/videoparse/gstvideoparse.c:
Remove videoparse element, because it was moved to gst/rawparse/
Original commit message from CVS:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_src_event):
Always seek on frame boundaries, will produce nothing useful
otherwise.
Original commit message from CVS:
* configure.ac:
* gst/rawparse/Makefile.am:
* gst/rawparse/README:
* gst/rawparse/gstaudioparse.c: (gst_audio_parse_format_get_type),
(gst_audio_parse_endianness_get_type), (gst_audio_parse_base_init),
(gst_audio_parse_class_init), (gst_audio_parse_init),
(gst_audio_parse_set_property), (gst_audio_parse_get_property),
(gst_audio_parse_update_frame_size), (gst_audio_parse_get_caps):
* gst/rawparse/gstaudioparse.h:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_base_init),
(gst_raw_parse_class_init), (gst_raw_parse_init),
(gst_raw_parse_dispose),
(gst_raw_parse_class_set_src_pad_template),
(gst_raw_parse_class_set_multiple_frames_per_buffer),
(gst_raw_parse_reset), (gst_raw_parse_chain),
(gst_raw_parse_convert), (gst_raw_parse_sink_event),
(gst_raw_parse_src_event), (gst_raw_parse_src_query_type),
(gst_raw_parse_src_query), (gst_raw_parse_set_framesize),
(gst_raw_parse_set_fps), (gst_raw_parse_get_fps),
(gst_raw_parse_is_negotiated):
* gst/rawparse/gstrawparse.h:
* gst/rawparse/gstvideoparse.c: (gst_video_parse_format_get_type),
(gst_video_parse_endianness_get_type), (gst_video_parse_base_init),
(gst_video_parse_class_init), (gst_video_parse_init),
(gst_video_parse_set_property), (gst_video_parse_get_property),
(gst_video_parse_format_to_fourcc),
(gst_video_parse_update_frame_size), (gst_video_parse_get_caps):
* gst/rawparse/gstvideoparse.h:
* gst/rawparse/plugin.c: (plugin_init):
Add new plugin rawparse that contains a base class for raw data
parsers and the two elements audioparse and videoparse that can
be used to parse raw audio and video. These are inspired by the
old videoparse element which the new rawparse plugin deprecates.
Original commit message from CVS:
* sys/glsink/glextensions.c:
* sys/glsink/glextensions.h:
* sys/glsink/glvideo.c:
* sys/glsink/glvideo.h:
* sys/glsink/gstglbuffer.c:
* sys/glsink/gstgldownload.c:
* sys/glsink/gstglupload.c:
A careful read of the documentation reveals that I can't use
renderbuffers as textures. Duh. Checkpoint because I'm about
to rewrite a bunch of code.
Original commit message from CVS:
* sys/glsink/Makefile.am:
* sys/glsink/glextensions.c:
* sys/glsink/glextensions.h:
* sys/glsink/glimagesink.c:
* sys/glsink/glimagesink.h:
* sys/glsink/glvideo.c:
* sys/glsink/glvideo.h:
* sys/glsink/gstglbuffer.c:
* sys/glsink/gstglbuffer.h:
* sys/glsink/gstgldownload.c:
* sys/glsink/gstglfilter.c:
* sys/glsink/gstglupload.c:
* sys/glsink/gstopengl.c:
Switch to using framebuffer_objects instead of GLXPixmaps,
because that's what my driver supports. Remove GLDrawable,
since GstGLDisplay now has a default drawable and context.
Original commit message from CVS:
2007-12-18 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch-marshal.list:
* gst/switch/gstswitch.h (struct _GstStreamSelectorClass):
* gst/switch/gstswitch.c (enum, gst_selector_pad_class_init)
(gst_selector_pad_get_property)
(gst_selector_pad_get_running_time)
(gst_stream_selector_class_init, gst_segment_get_timestamp)
(gst_segment_set_stop, gst_segment_set_start)
(gst_stream_selector_set_active_pad, gst_stream_selector_block)
(gst_stream_selector_push_pending_stop)
(gst_stream_selector_switch): Change so that the signals and
properties deal in running time, not buffer time. Document the
signals more. Change uint64 in API to int64, to reflect what's in
GstSegment.
Original commit message from CVS:
* Makefile.am:
Include common/win32.mak for CRLF check of win32 project
files (see #393626).
* configure.ac:
Bump requirements to -base CVS for libgstvideo additions in
glimagesink. Disable glimagesink until the missing files get
checked in.
Original commit message from CVS:
* sys/glsink/Makefile.am:
* sys/glsink/glimagesink.c:
* sys/glsink/glimagesink.h:
* sys/glsink/glvideo.c:
* sys/glsink/glvideo.h:
* sys/glsink/gstopengl.c:
* sys/glsink/gstglupload.c:
Use new GstVideoFormat checked into -base. Add new glupload
element to upload raw video into a GLXPixbuf. Untested. Will
likely crash your motorcycle if you try it.
* sys/glsink/gstvideo-common.c:
* sys/glsink/gstvideo-common.h:
Remove.
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstmultifilesrc.h:
When subsequent files are read, if the file doesn't exist, send
an EOS instead of causing an error.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_selector_pad_chain): Return OK when
a buffer is ignored, not NOT_LINKED. No sense in making a source
element error out; at least fdsrc considers NOT_LINKED to be a
fatal error. Patch 11/12. There is no patch 12/12. Foo.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch-marshal.list:
* gst/switch/gstswitch.h (struct _GstStreamSelectorClass):
* gst/switch/gstswitch.c (gst_stream_selector_class_init)
(gst_stream_selector_block): Make the block() signal return the
last stop time of the active pad. Patch 10/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_selector_pad_get_property)
(gst_selector_pad_class_init, gst_stream_selector_class_init)
(gst_stream_selector_get_property): Expose 'last-stop-time' as a
pad property, not an element property.
(gst_selector_pad_chain): Mark the last_stop time as timestamp +
duration, not timestamp. Patch 9/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_stream_selector_change_state)
(gst_stream_selector_block, gst_stream_selector_switch): Use the
cond mechanism instead of blocked pads. Patch 8/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelector):
* gst/switch/gstswitch.c (gst_stream_selector_wait)
(gst_selector_pad_chain, gst_stream_selector_init)
(gst_stream_selector_dispose): Add infrastructure for new blocking
mechanism that does not use gst_pad_set_blocked, which does not
work on sink pads. Patch 7/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelector): Add some
state variables.
* gst/switch/gstswitch.c (gst_stream_selector_push_pending_stop)
(gst_selector_pad_chain): Push any pending stop event.
(gst_stream_selector_set_active_pad)
(gst_stream_selector_set_property): Factor out setting the active
pad to a function. Close the segment of the previous active pad if
told to do so via a stop_time != GST_CLOCK_TIME_NONE.
(gst_stream_selector_switch): Implement switch vmethod. Patch 5/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_stream_selector_block): Implement
the block() signal. This implementation will be replaced in future
patches, however. Patch 4/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelectorClass):
* gst/switch/gstswitch.c (gst_stream_selector_class_init): Add
`block' and `switch' signals.
* gst/switch/Makefile.am:
* gst/switch/gstswitch-marshal.list: Add foo to generate a
marshaller for the `switch' signal. Patch 2/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h:
* gst/switch/gstswitch.c: Replace with files from
gststreamselector.[ch], registered as the "switch" plugin, with
"GstSwitch" types. Patch 1/12.
Original commit message from CVS:
* sys/glsink/Makefile.am:
* sys/glsink/glextensions.c:
* sys/glsink/glextensions.h:
* sys/glsink/glvideo.c:
Add vblank synchronization. Isn't really working on my
driver. :(
Original commit message from CVS:
* sys/glsink/Makefile.am:
* sys/glsink/glimagesink.c:
* sys/glsink/glvideo.c:
* sys/glsink/glvideo.h:
* sys/glsink/gstvideo-common.c:
* sys/glsink/gstvideo-common.h:
Add support for xRGB, xBGR, and AYUV. Re-add support for
power-of-2 textures.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_dispose),
(gst_video_parse_sink_event):
Free the adapter on dispose and correctly reset on newsegment events.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_sink_event),
(gst_video_parse_src_event), (gst_video_parse_src_query):
Improve duration query by first asking upstream and if it can't handle
the query try to get the duration in bytes from upstream and convert.
For seeks, try if upstream handles this already first and do our
conversion to byte format only if it doesn't and if we get a
newsegment event in time format keep it and only do our conversions
if the event has another format.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c:
(gst_video_parse_format_get_type),
(gst_video_parse_endianness_get_type),
(gst_video_parse_class_init), (gst_video_parse_init),
(gst_video_parse_set_property), (gst_video_parse_get_property),
(gst_video_parse_format_to_fourcc),
(gst_video_parse_update_block_size), (gst_video_parse_chain),
(gst_video_parse_sink_event):
Add support for video/x-raw-rgb and video/x-raw-gray. Also send
downstream elements downstream, not upstream.
Original commit message from CVS:
* sys/glsink/gstvideo-common.c:
* sys/glsink/gstvideo-common.h:
Pull together some common raw video functions into one location.
This should eventually move to -base.
* sys/glsink/Makefile.am:
* sys/glsink/glimagesink.c:
* sys/glsink/glimagesink.h:
* sys/glsink/glvideo.c:
* sys/glsink/glvideo.h:
* sys/glsink/gstopengl.c:
Use the new video-common.h stuff. Readd support for RGB video.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
Hash streams by pid again. Add a linked list inside each
stream with a list of sub_tables. Fix multiple sections
as it was borked with my last commit.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_init),
(gst_video_parse_src_event), (gst_video_parse_src_query_type):
Implement a query type function for the src pad, implement seeking
and use ANY caps for the sink pad as the element doesn't care what
caps the input has and everything is handled via properties.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_convert),
(gst_video_parse_sink_event):
Handle -1 values for the CONVERT query too.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_sink_event):
Add YV12 to the pad templates as it is supported too and allow
-1 as stop position for NEWSEGMENT events.
Original commit message from CVS:
* gst/videoparse/Makefile.am:
Add $(GST_PLUGINS_BASE_CFLAGS) to CFLAGS to fix the build.
* gst/videoparse/gstvideoparse.c: (gst_video_parse_init),
(gst_video_parse_set_property), (gst_video_parse_get_property):
Use g_value_[sg]et_enum() for enum properties, g_value_[sg]et_int()
gives a g_critical().
Original commit message from CVS:
* gst/videoparse/README:
* gst/videoparse/gstvideoparse.c:
Add a bunch of features: handle format specification, handle
queries and conversion. Works much like a normal parser now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset),
(gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query),
(gst_rtp_ssrc_demux_change_state):
Clean up the dynamic pads when going to READY.
Original commit message from CVS:
* ext/dts/gstdtsdec.c: (gst_dtsdec_init),
(gst_dtsdec_sink_setcaps), (gst_dtsdec_chain_raw),
(gst_dtsdec_chain):
* ext/dts/gstdtsdec.h:
Add support for "audio/x-private1-dts" as used by flupsparse. Most
changes adapted from a52dec.
Original commit message from CVS:
* sys/glsink/Makefile.am:
* sys/glsink/glimagesink.c:
* sys/glsink/glvideo.c:
* sys/glsink/glvideo.h:
Split out gl-related code into a separate file with a
sensible API. Major cleanup. Still crashes occasionally
due to different threads touching bits at the same time.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (_do_init),
(gst_souphttp_src_class_init), (gst_souphttp_src_init),
(gst_souphttp_src_dispose), (gst_souphttp_src_set_property),
(gst_souphttp_src_get_property), (unicodify),
(gst_souphttp_src_unicodify), (gst_souphttp_src_create),
(gst_souphttp_src_start), (gst_souphttp_src_stop),
(gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop),
(gst_souphttp_src_get_size), (gst_souphttp_src_is_seekable),
(soup_got_headers), (soup_got_body), (soup_finished),
(soup_got_chunk), (soup_response), (soup_parse_status),
(gst_souphttp_src_uri_get_type),
(gst_souphttp_src_uri_get_protocols),
(gst_souphttp_src_uri_get_uri), (gst_souphttp_src_uri_set_uri),
(gst_souphttp_src_uri_handler_init):
* ext/soup/gstsouphttpsrc.h:
Do not try to unpause I/O in the "queued" state.
Reorganise a bunch of things and cleanups.
Uses G_GUINT64_FORMAT instead of hard-coding %llu.
See #502335.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Don't strdup (and thus leak) codec name strings when passing
them to gst_tag_list_add().
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_handle_message):
* gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
(on_ssrc_sdes):
Post a message when the SDES infor changes for a source.
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
Update some comments.
Original commit message from CVS:
Based on patch by: <mutex at runbox dot com>
* gst/videoparse/gstvideoparse.c: (gst_video_parse_src_query):
Forward the query upstream, the default element event handler does
something different. Fixes#502879.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
* gst/rtpmanager/rtpjitterbuffer.c:
Update comment.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_set_property), (gst_rtp_session_get_property):
Define some GObject properties to set SDES and other configuration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_ssrc_sdes), (rtp_session_set_bandwidth),
(rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction),
(rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string),
(rtp_session_get_sdes_string), (obtain_source),
(rtp_session_get_internal_source), (rtp_session_process_sdes),
(rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes),
(is_rtcp_time):
* gst/rtpmanager/rtpsession.h:
Add signal when new SDES infor has been found for a source.
Create properties for SDES and other info.
Simplify the SDES API.
Add method for getting the internal source object of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_finalize), (rtp_source_set_property),
(rtp_source_get_property), (rtp_source_set_callbacks),
(rtp_source_get_ssrc), (rtp_source_set_as_csrc),
(rtp_source_is_as_csrc), (rtp_source_is_active),
(rtp_source_is_validated), (rtp_source_is_sender),
(rtp_source_received_bye), (rtp_source_get_bye_reason),
(rtp_source_set_sdes), (rtp_source_set_sdes_string),
(rtp_source_get_sdes), (rtp_source_get_sdes_string),
(rtp_source_get_new_sr), (rtp_source_get_new_rb):
* gst/rtpmanager/rtpsource.h:
Add GObject properties for various things.
Don't leak the bye reason.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Fix list of supported and known codecs.
Emit tag with the codec name so it gets properly reported in totem and
other applications.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
The transform() methods are not called in passthrough mode so
there's no need for checking if the element is in passthrough mode.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
Sync the GObject properties with the controller even in passthrough
mode to get consistent property values.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
A sub table is identified by the pair table_id and
sub_table_identifier, not by pid. So hash with that.
* sys/dvb/dvbbasebin.c:
Make sure initial pids are added properly to filter,
Original commit message from CVS:
2007-12-05 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_switch_set_property): Don't push
buffers from app thread when unsetting `queue-buffers', it's
dangerous and the chain function will do it for us anyway.
Original commit message from CVS:
* gst/mpegtsparse/Makefile.am:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
Remove signals for pat, pmt, nit, eit, sdt. Replace with bus
messages.
* sys/dvb/dvbbasebin.c:
Instead of attaching to signals, use the bus messages.
Also fix up so the dvbsrc starts only outputting the info tables
like PAT, CAT, NIT, SDT, EIT instead of the whole ts.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* configure.ac:
Bump libsoup requirement as libsoup does not support async client
operation prior to version 2.2.104 and it has some leaks.
* ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init),
(gst_souphttp_src_init), (gst_souphttp_src_dispose),
(gst_souphttp_src_set_property), (gst_souphttp_src_create),
(gst_souphttp_src_start), (gst_souphttp_src_stop),
(gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop),
(gst_souphttp_src_get_size), (soup_got_headers), (soup_got_body),
(soup_finished), (soup_got_chunk), (soup_response),
(soup_session_close):
* ext/soup/gstsouphttpsrc.h:
Implement unlock().
Picks up the size from the Content-Length header and emit a duration
message.
Don't leak the GMainContext object.
Fixes#500099.
Original commit message from CVS:
* ext/alsaspdif/alsaspdifsink.c: (alsaspdifsink_set_caps),
(alsaspdifsink_get_time), (alsaspdifsink_set_params),
(alsaspdifsink_find_pcm_device):
Don't free uninitialized data when we are in error.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/resample.c: (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double),
(speex_resampler_process_native), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_get_input_latency),
(speex_resampler_get_output_latency):
* gst/speexresample/speex_resampler.h:
Update speex resampler to latest SVN. We're now down to only the
changes noted in README again.
* gst/speexresample/speex_resampler_wrapper.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_query):
Adjust to API changes.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Output segment with proper 'stop' value, makes flvdemux 100% compatible
with gnonlin.
Original commit message from CVS:
patch by: Alessandro Decina
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
* gst/mpegtsparse/mpegtsparse.c:
* gst/mpegtsparse/mpegtsparse.h:
pat-info is now a signal not a GObject property that
gets notified.
pat-info, pmt-info now instead of passing a GObject as
a parameter, pass a GstStructure.
New signals: nit-info, sdt-info, eit-info for DVB SI information
* sys/dvb/camconditionalaccess.c:
* sys/dvb/camconditionalaccess.h:
* sys/dvb/camdevice.c:
* sys/dvb/camdevice.h:
* sys/dvb/camswclient.c:
* sys/dvb/camswclient.h:
* sys/dvb/camutils.c:
* sys/dvb/camutils.h:
Cam code now uses the pmt GstStructure passed from mpegtsparse
signals rather than the GObject.
* sys/dvb/dvbbasebin.c:
Use new signals in mpegtsparse and use GstStructures as per
mpegtsparse's modified API.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Only post the latency message if we have a resampler state already.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/speexresample/resample.c: (speex_resampler_get_latency),
(speex_resampler_drain_float), (speex_resampler_drain_int),
(speex_resampler_drain_interleaved_float),
(speex_resampler_drain_interleaved_int):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add functions to push the remaining samples and to get the latency
of the resampler. These will get added to Speex SVN in this or a
slightly changed form at some point too and should get merged then
again.
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
(gst_speex_resample_init_state),
(gst_speex_resample_transform_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_fix_output_buffer), (gst_speex_resample_process),
(gst_speex_resample_query), (gst_speex_resample_query_type):
Drop the prepending zeroes and output the remaining samples on EOS.
Also properly implement the latency query for this. speexresample
should be completely ready for production use now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
jitterbuffer can buffer an unlimited amount of time and thus has no
max_latency requirements.
Original commit message from CVS:
* gst/speexresample/README:
Add README explaining where the resampling code was taken from
and which changes were done.
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free):
Use g_malloc() and friends instead of malloc() to achieve higher
portability and define the functions inline.
* gst/speexresample/speex_resampler.h:
Add back some useless preprocessor stuff to keep the diff between
our version and the one from the Speex SVN repository lower.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_set_property):
* gst/filter/gstlpwsinc.c: (lpwsinc_set_property):
Post a GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* ext/faac/gstfaac.c: (gst_faac_sink_event):
Don't try to flush the decoder on EOS when it was not initialized.
Fixes#498667
Original commit message from CVS:
2007-11-21 Julien Moutte <julien@fluendo.com>
* ext/sdl/sdlaudiosink.c: (gst_sdlaudio_sink_write): Fix build
on Mac OS X. (missing format parameter)
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Remove preset iface again. We'll re-add this after its been released
in -good.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
Some small cleanup and addition of a TODO item.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
If the resampler gives less output samples than expected
adjust the output buffer and print a warning.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Activate preset iface and upload two presets here.
Original commit message from CVS:
Patch by: Wai-Ming Ho <webregbox at yahoo dot co dot uk>
* gst/mpegtsparse/mpegtspacketizer.c: (foreach_stream_clear),
(remove_all), (mpegts_packetizer_clear):
Ensure that the plugin does not crash when the property pat-info is
queried before a PAT is available. It also ensures that the PAT info is
cleared when the changing from PLAYING to READY.
Fixes#487892.
Original commit message from CVS:
Patch by: Michael Kötter <m dot koetter at oraise dot de>
* ext/alsaspdif/alsaspdifsink.c: (alsaspdifsink_set_caps),
(alsaspdifsink_get_time), (alsaspdifsink_open),
(alsaspdifsink_set_params), (alsaspdifsink_delay), (plugin_init):
Fix sample rate and clocking.
Remove buffer_time and period_time as this seems to break on some
hardware. Fixes#485462.
Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
don't forget to handle the offset's
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
precalculate some many used values
Original commit message from CVS:
patch by: Armando Taffarel Neto <taffarel@solis.coop.br>
* gst/librfb/gstrfbsrc.c:
Set the timestamp for the output buffers
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/replaygain/rganalysis.c: (yule_filter):
Avoid slowdown from denormals when processing near-silence input data.
Spotted by Gabriel Bouvigne. Fixes#494499.
Original commit message from CVS:
* gst/flv/gstflvparse.c:
Add mapping for Nellymoser ASAO audio codec.
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Make sure we
actually have data to read at the end of the tag. This avoids trying
to allocate negative buffers.
Original commit message from CVS:
* gst/equalizer/demo.c: (draw_spectrum):
* gst/spectrum/demo-audiotest.c: (draw_spectrum):
* gst/spectrum/demo-osssrc.c: (draw_spectrum):
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
Change the meaning of the magnitude values given in the
GstMessages by spectrum to decibel instead of
decibel+threshold.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* gst/equalizer/gstiirequalizernbands.c:
And continue to update docs. Also include some sample code
for the n-band equalizer in the docs.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_class_init):
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_class_init):
* gst/equalizer/gstiirequalizernbands.c:
Update docs and property ranges to the real values.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c:
Now do the scaling right for real. Also initialize a previously
uninitialized variable.
Original commit message from CVS:
* gst/equalizer/demo.c:
Make default volume a bit less. Improve layout by giving more space to
the slider with big-numbers and enable fill.
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Require GIO >= 0.1.2 and adjust unit test for an API change.
Original commit message from CVS:
* ext/gio/gstgio.h:
Add macro to check if a stream supports seeking.
* ext/gio/Makefile.am:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
(gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
(gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
(gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
(gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
(gst_gio_base_sink_render), (gst_gio_base_sink_query),
(gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
(gst_gio_base_src_class_init), (gst_gio_base_src_init),
(gst_gio_base_src_finalize), (gst_gio_base_src_start),
(gst_gio_base_src_stop), (gst_gio_base_src_get_size),
(gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
(gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
(gst_gio_base_src_create), (gst_gio_base_src_set_stream):
* ext/gio/gstgiobasesrc.h:
Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
base classes that only require a GInputStream or GOutputStream to
work.
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_init),
(gst_gio_sink_finalize), (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_init),
(gst_gio_src_finalize), (gst_gio_src_start):
* ext/gio/gstgiosrc.h:
Use the newly created base classes here.
* ext/gio/gstgio.c: (plugin_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
(gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
(gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
(gst_gio_stream_sink_get_property):
* ext/gio/gstgiostreamsink.h:
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
(gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
(gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
(gst_gio_stream_src_get_property):
* ext/gio/gstgiostreamsrc.h:
Implement GstGioStreamSink and GstGioStreamSrc that have a property
to set the GInputStream/GOutputStream that should be used.
* tests/check/Makefile.am:
* tests/check/pipelines/.cvsignore:
* tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
(gio_testsuite), (main):
Add unit test for giostreamsrc and giostreamsink.
Original commit message from CVS:
* ext/gio/gstgio.c: (plugin_init):
Remove nowadays unnecessary workaround for a crash.
* ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
(gst_gio_sink_start), (gst_gio_sink_stop),
(gst_gio_sink_unlock_stop):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
(gst_gio_src_stop), (gst_gio_src_unlock_stop):
* ext/gio/gstgiosrc.h:
Make the finalize function safer, clean up everything that could stay
around.
Reset the cancellable instead of creating a new one after cancelling
some operation.
Don't store the GFile in the element, it's only necessary for creating
the streams.
Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (main):
Use autoaudiosink instead of alsasink and use a sine wave.
* gst/spectrum/gstspectrum.c:
Fix the magnitude calculation.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
(CDshowFakeSink.CDshowFakeSink):
* gst-libs/gst/dshow/gstdshowfakesink.h: (CDshowFakeSink.m_hres):
Fix crasher in constructor due to the base class's constructor
not necessarily being NULL-safe (depends on the SDK version used
apparently; #492406).
* sys/dshowsrcwrapper/gstdshowaudiosrc.c: (gst_dshowaudiosrc_prepare):
* sys/dshowsrcwrapper/gstdshowvideosrc.c: (gst_dshowvideosrc_set_caps):
Fix a couple of MSVC compiler warnings (#492406).
Original commit message from CVS:
* gst/equalizer/demo.c: (main):
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_class_init), (setup_filter):
Allow setting 0 as bandwidth and handle this correctly.
Also handle a bandwidth of rate/2 properly.
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_class_init):
Make it possible to generate a N-band equalizer with 1 bands. The
previous limit of 2 was caused by a nowadays replaced calculation
doing a division by zero if number of bands was 1.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst/rtpmanager/gstrtpsession.c:
Fix bad function signatures (#492798).
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/demo.c: (on_window_destroy), (on_configure_event),
(on_gain_changed), (on_bandwidth_changed), (on_freq_changed),
(draw_spectrum), (message_handler), (main):
Add small demo application based on the spectrum demo applications
that gets white noise as input, pushes it through an equalizer and
paints the spectrum. For every equalizer band it's possible to set
gain, bandwidth and frequency.
* gst/equalizer/gstiirequalizer.c: (setup_filter):
Add some guarding against too large or too small frequencies and
bandwidths. Also improve debugging a bit.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init), (arg_to_scale),
(setup_filter), (gst_iir_equalizer_compute_frequencies):
Replace filters with a bit better filters for which we can actually
find documentation, which don't change anything on zero gain, etc.
Make the frequency property of the bands writable, rename the
band-width property to bandwidth and change the meaning to the
frequency difference between bandedges, change the meaning of the
gain property to dB instead of a weird scale between -1 and 1 that
has no real meaning.
Original commit message from CVS:
Changed kclass to "Parser/Extracter/Metadata", changed caps to "image/jpeg, tags-extract=true/false" and changed priority to GST_RANK_PRIMARY+1. Also, srcpad can only work in push mode until fixed to also work in pull mode.
Original commit message from CVS:
Created new plugin ('medadata') and element ('metadataparse') that extract metadata from images (look at bug #486659).
Original commit message from CVS:
* ext/faac/gstfaac.c: (gst_faac_profile_get_type),
(gst_faac_class_init), (gst_faac_init):
Fix bitrate ranges and change enum nick for low complexity
profile from LOW to LC for consistency (#490060).
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_request_new_pad):
Let's not call every request pad we create "sink%d", that'll
create problems if there's to be more than one pad. Fixes#490682.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/interleave.c:
Add unit test for the above.
Original commit message from CVS:
2007-10-27 Julien MOUTTE <julien@moutte.net>
* gst/mpeg4videoparse/mpeg4videoparse.c:
(gst_mpeg4vparse_align),
(gst_mpeg4vparse_drain), (gst_mpeg4vparse_chain),
(gst_mpeg4vparse_sink_setcaps), (gst_mpeg4vparse_sink_event),
(gst_mpeg4vparse_cleanup), (gst_mpeg4vparse_change_state),
(gst_mpeg4vparse_dispose), (gst_mpeg4vparse_base_init),
(gst_mpeg4vparse_class_init), (gst_mpeg4vparse_init),
(plugin_init):
* gst/mpeg4videoparse/mpeg4videoparse.h: Improved version not
damaging headers using a simple state machine.
Original commit message from CVS:
* sys/dvb/gstdvbsrc.c:
Actually use the code-rate-hp parameter for DVB-S.
It turns out setting to AUTO does not always work (
especially in diseq situations). Set by default to
FEC_AUTO.
Original commit message from CVS:
2007-10-22 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video), (gst_flv_parse_tag_type): Don't
emit no-more-pads for single pad scenarios as the header
is definitely not reliable. We emit them for 2 pads scenarios
though to speed up media discovery.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init), (gst_iir_equalizer_band_init),
(gst_iir_equalizer_band_get_type), (gst_iir_equalizer_class_init),
(setup_filter), (gst_iir_equalizer_setup):
* gst/equalizer/gstiirequalizer.h:
Move bandwidth property to the separate bands and add float64 support.
Original commit message from CVS:
Patch by: Richard Hult <richard imendio com>
* gst/dvdspu/Makefile.am:
Fix LIBS - we need to link against libgstreamer.
Original commit message from CVS:
patch by: Alessandro Decina
* sys/dvb/Makefile.am:
* sys/dvb/cam.c:
* sys/dvb/cam.h:
* sys/dvb/camapplication.c:
* sys/dvb/camapplication.h:
* sys/dvb/camapplicationinfo.c:
* sys/dvb/camapplicationinfo.h:
* sys/dvb/camconditionalaccess.c:
* sys/dvb/camconditionalaccess.h:
* sys/dvb/camdevice.c:
* sys/dvb/camdevice.h:
* sys/dvb/camresourcemanager.c:
* sys/dvb/camresourcemanager.h:
* sys/dvb/camsession.c:
* sys/dvb/camsession.h:
* sys/dvb/camswclient.c:
* sys/dvb/camswclient.h:
* sys/dvb/camtransport.c:
* sys/dvb/camtransport.h:
* sys/dvb/camutils.c:
* sys/dvb/camutils.h:
* sys/dvb/dvbbasebin.c:
* sys/dvb/dvbbasebin.h:
* sys/dvb/gstdvb.c:
* sys/dvb/gstdvbsrc.c:
* sys/dvb/gstdvbsrc.h:
Integrate SoC work done by Alessandro for the Freevo project.
Adds cam support to the dvb stack in GStreamer and a new
element (actually a bin) called dvbbasebin that integrates
dvbsrc and mpegtsparse to a) handle decryption and b) allow
acquiring multiple channels on same transponder without
knowing pid numbers.
Original commit message from CVS:
patch by: Alessandro Decina
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
* gst/mpegtsparse/mpegtsparse.h:
Add request pad for getting the full transport stream coming in.
Original commit message from CVS:
* configure.ac:
Update the highest allowed neon version from 0.26.99 to 0.27.99.
No code changes are required to work with the newest neon version.
Original commit message from CVS:
* configure.ac:
Require core CVS. This is implicit in the -base CVS
requirement already, so we might just well spell it
out. Also, we do need at least 0.10.14 for
gst_element_class_set_details_simple(). Make check
for gmyth a bit more restrictive so things don't break
if the next version changes API.
* ext/alsaspdif/alsaspdifsink.c:
Work around alsa alloca macros triggering 'always evaluates to
true' warnings with gcc-4.2 and fix compilation with gcc-4.2.
Also don't leak the device string.
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/soundtouch/gstpitch.cc:
* gst/modplug/gstmodplug.cc:
Fix compilation with g++4.2 and -Wall -Werror (also needs plugin
define fix from core CVS). Fixes#462737.
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Use GIO function to get a list of supported URI schemes instead of
hard coding something.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_next_timeout),
When reconsidering RTCP timeouts, set the next timeout against the last
report time instead of the current clock time so that we don't end up
reconsidering forever.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Only peek at the tail element instead of popping it off, which allows
us to greatly simplify things when the tail element changes.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_recv_rtp_sink):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_sink_event):
Forward FLUSH events instead of leaking them.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove the tail-changed callback in favour of a simple boolean when we
insert a buffer in the queue.
Add method to peek the tail of the buffer.
Original commit message from CVS:
Patch by: Gautier Portet <kassoulet at gmail dot com>
* gst/xingheader/gstxingmux.c:
The size of the Xing header is actually 417 as it's rounded to the
next smaller integer. Fixes#397759.
* gst/xingheader/gstxingmux.c: (xing_generate_header),
(xing_push_header):
Some random cleanup, add FIXMEs and TODOs and check if the newsegment
event to the beginning was successful before pushing the header again.
Original commit message from CVS:
Patch by: Wai-Ming Ho <webregbox at yahoo dot co dot uk>
* gst/mpegtsparse/mpegtspacketizer.c:
(mpegts_packetizer_stream_new):
Don't skip PAT with version number 0. Fixes#483400.
* gst/mpegtsparse/mpegtsparse.c: (mpegts_parse_apply_pat):
Make all values above 0 mark a referenced program as they can be
incremented and only 1 had marked a referenced program before, causing
actually referenced programs to be unreferenced.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state), (apply_offset),
(gst_rtp_jitter_buffer_loop):
Remove some old unused variables.
Don't add the latency to the skew corrected timestamp, latency is only
used to sync against the clock.
Improve debugging.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(rtp_jitter_buffer_reset_skew), (calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Handle case where server timestamp goes backwards or wildly jumps by
temporarily pausing the skew correction.
Improve debugging.
Original commit message from CVS:
Patch by: mutex at runbox dot com
* gst/mpegtsparse/mpegtspacketizer.c:
(mpegts_packetizer_parse_adaptation_field_control):
* gst/mpegtsparse/mpegtsparse.c: (mpegts_parse_base_init),
(mpegts_parse_init), (mpegts_parse_push):
* gst/mpegtsparse/mpegtsparse.h:
Remove useless src pad that only results in not linked errors,
fix a broken pointer dereference and make MAX_CONTINUITY constant
conform to the standard to stop outputting corrupted data.
Fixes#481276, #481279.
Original commit message from CVS:
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_set_property), (gst_gio_sink_render):
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_set_property):
Some minor cleanup and allow setting the location only when the
element is not playing or paused.
Original commit message from CVS:
* configure.ac:
Update gio's pkg-config file name as currently in SVN.
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_set_location):
Remove special casing for a NULL query string. g_strjoin won't add
the separator if there's only one string.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (free_client):
Fix crasher in dispose.
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Handle cases where input buffers have no timestamps so that no clock
skew can be calculated, in this case interpollate timestamps based on
rtp timestamp and assume a 0 clock skew.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c: (apply_latency),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query):
Remove jitter correction code, it's now in the lower level object.
Use new -core method for doing a peer query.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(calculate_skew), (rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Move jitter correction to the lowlevel jitterbuffer.
Increase the max window size.
When filling the window, already start estimating the skew using a
parabolic weighting factor so that we have a much better startup
behaviour that gets more accurate with the more samples we have.
Increase the default weighting factor for the steady state to get
smoother timestamps.
Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_set_proxy),
(gst_neonhttp_src_send_request_and_redirect):
Now that we require libneon >= 0.26 remove the neon 0.25 backward
compatibility stuff. Also fix the default location.
Original commit message from CVS:
* ext/xvid/gstxvidenc.c:
* ext/xvid/gstxvidenc.h:
Remove superfluous 'frame-encoded' signal (people can
use an upstream identity's 'handoff' signal or a pad
probe for this if they must know).
Original commit message from CVS:
2007-09-27 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): I got it wrong again, audio rate
was not detected correctly in all cases.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_dispose),
(gst_rtp_bin_finalize):
Fix cleanup crasher.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Dynamically adjust the skew calculation window so that we calculate it
over a period of around 2 seconds.
Original commit message from CVS:
2007-09-26 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): codec_data is needed for every tag
not just the first one. (Fix a stupid bug i introduced without
testing)
Original commit message from CVS:
2007-09-26 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): Fix bit masks operations to be
sure we detect the codec_tags and sample rates correctly.
Fix raw audio caps generation.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
Update hierarchy.
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.h:
Mark private fields of the instance structs private.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* configure.ac:
* ext/Makefile.am:
* ext/gio/Makefile.am:
* ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
(gst_gio_get_supported_protocols),
(gst_gio_uri_handler_get_type_sink),
(gst_gio_uri_handler_get_type_src),
(gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
(gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
(gst_gio_uri_handler_do_init), (plugin_init):
* ext/gio/gstgio.h:
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_init),
(gst_gio_sink_finalize), (gst_gio_sink_set_property),
(gst_gio_sink_get_property), (gst_gio_sink_start),
(gst_gio_sink_stop), (gst_gio_sink_unlock),
(gst_gio_sink_unlock_stop), (gst_gio_sink_event),
(gst_gio_sink_render), (gst_gio_sink_query):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_init),
(gst_gio_src_finalize), (gst_gio_src_set_property),
(gst_gio_src_get_property), (gst_gio_src_start),
(gst_gio_src_stop), (gst_gio_src_get_size),
(gst_gio_src_is_seekable), (gst_gio_src_unlock),
(gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
(gst_gio_src_create):
* ext/gio/gstgiosrc.h:
Add a GIO/GVFS plugin with source and sink elements. This will
only be enabled when --enable-experimental is given to configure
for now as the GIO API is not stable yet. Fixes#476916.
Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
Added offset-x, offset-y, width and height property
for selecting a region from the screen
Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
Minimum raw encoding is working now
* gst/librfb/rfbdecoder.c:
fix address while reading from stream
Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
raw encoding is working, but it looks like the
ffmpegcolorspace plugin can't handle high resolutions
Original commit message from CVS:
* ext/alsaspdif/alsaspdifsink.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst/mpegvideoparse/mpegvideoparse.c:
Fix memory leaks. More to come.
* tests/check/Makefile.am:
* tests/check/generic/states.c:
Improved state change unit test.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_set_property), (gst_app_sink_get_property),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_event), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add properties, signals and actions to access the element even without
linking to the library.
Fix some method names and signatures.
Original commit message from CVS:
Patch by: Jonathan Matthew <jonathan at kaolin wh9 net>
* ext/faad/gstfaad.c: (gst_faad_chanpos_to_gst),
(gst_faad_srcgetcaps), (gst_faad_update_caps):
Don't set channel positions on regular mono and stereo cases.
Fixes#476370.
Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
It is now possible to connect to a vncserver.
there are still some issues with the ouput of
the screen. Looks like some lines are confused
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(open_library), (gst_real_video_dec_init),
(gst_real_video_dec_set_property),
(gst_real_video_dec_get_property), (gst_real_video_dec_class_init):
* gst/real/gstrealvideodec.h:
Don't generate an error for occasional decoding errors.
Add max-errors property.
Error out when we receive max-errors in a row. Fixes#478159.
Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
Add password property (write only)
* gst/librfb/rfbdecoder.c:
Read the reason on failure
Use the password property for authentication
* gst/librfb/rfbdecoder.h:
Add defines for version checking
Original commit message from CVS:
* ext/directfb/dfbvideosink.c: (gst_dfbvideosink_surface_destroy),
(gst_dfbsurface_class_init):
When finalizing GstDfbSurface, a subclass of GstBuffer, correctly
chain up to the parent class to free everything, including caps.
Original commit message from CVS:
* gst/librfb/Makefile.am:
* gst/librfb/d3des.c:
* gst/librfb/d3des.h:
* gst/librfb/rfbdecoder.c:
* gst/librfb/vncauth.c:
* gst/librfb/vncauth.h:
VNC Authentication should be working now
temperaly with fake password 'testtest'
Original commit message from CVS:
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
Added some documentation about security handling
start implementing security handling for rfb 3.3
Original commit message from CVS:
* ChangeLog:
Add missing newline.
* gst/librfb/rfbdecoder.c:
Fix the build (missing stdlib.h).
* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
Use basetransform segment so that it is correctly managed on flushes
and start/stop. Report message timestamp as stream time, which is what
an application can understand. (Yes these are adapted from wim recent
level element changes)
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
Link to the right pads regardless of which one was created first in the
ssrc demuxer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsource.c: (calculate_jitter):
Improve debugging.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
(gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links):
* gst/rtpmanager/gstrtpssrcdemux.h:
Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
Use lock to protect variable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop):
Reconstruct GST timestamp from RTP timestamps based on measured clock
skew and sync offset.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(rtp_jitter_buffer_set_tail_changed),
(rtp_jitter_buffer_set_clock_rate),
(rtp_jitter_buffer_get_clock_rate), (calculate_skew),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek):
* gst/rtpmanager/rtpjitterbuffer.h:
Measure clock skew.
Add callback to be notfied when a new packet was inserted at the tail.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Remove clock skew detection, it's move to the jitterbuffer now.
Original commit message from CVS:
Patch by: Daniel Charles <dcharles at ti dot com>
* ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_bandmode_get_type),
(gst_amrwbenc_set_property), (gst_amrwbenc_get_property),
(gst_amrwbenc_class_init), (gst_amrwbenc_chain):
* ext/amrwb/gstamrwbenc.h:
Add property to control bandmode. Fixes#477306.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
Also set NTP base time on new sessions.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Use the right lock to protect our variables.
Fix some comment.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp),
(gst_rtp_session_chain_send_rtp), (create_send_rtp_sink):
Implement getcaps on the sender sinkpad so that payloaders can negotiate
the right SSRC.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base),
(gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp):
Calculate and configure the NTP base time so that we can generate better
NTP times in SR packets.
Set caps on new ghostpad.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Clean debug statement.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Add ntp-ns-base property to convert running_time to NTP time.
Handle NEWSEGMENT events on send and recv RTP pads so that we can
calculate the running time and thus NTP time of the packets.
Simplify getting the current NTP time using the pipeline clock.
Implement internal links functions.
Use the buffer timestamp to calculate the NTP time instead of the clock.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links),
(gst_rtp_ssrc_demux_src_query):
* gst/rtpmanager/gstrtpssrcdemux.h:
Implement internal links function.
Calculate the diff between different streams, this might be used later
to get the inter stream latency.
* gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp):
Simple cleanup.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr):
Make the clock skew window a little bigger.
Apply the clock skew to all buffers, not just one with a new timestamp.
Calculate and debug sender clock drift.
Use extended last timestamp to interpollate for SR reports.
Original commit message from CVS:
Patch by: Thomas Green <tom78999 gmail com>
* ext/neon/gstneonhttpsrc.c:
With libneon 2.6, we need to set the NE_SESSFLAG_ICYPROTO
flag if we want ICY streams to be handled too, otherwise
libneon will error out with a 'can't parse reponse' error.
Fixes#474696.
* tests/check/elements/neonhttpsrc.c:
Unit test for the above by Yours Truly.
Original commit message from CVS:
* configure.ac:
Use AC_TRY_COMPILE instead of AC_TRY_RUN for the faad and the
xvid configure checks, so they still work when cross-compiling.
Fixes#452009.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_transform_ip):
Use the correct parameter order for the memset calls.
Thanks to Christian Schaller for noticing.
Original commit message from CVS:
* configure.ac:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-audiotest.c: (draw_spectrum),
(message_handler), (main):
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_start),
(gst_spectrum_setup), (gst_spectrum_message_new),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Port GstSpectrum to GstAudioFilter and libgstfft, add support
for int32, float and double, use floats for the message contents,
average all FFTs done in one interval for better results, use
a better windowing function, allow posting the phase in the message
and actually do an FFT with the requested number of bands instead
of interpolating.
* tests/check/elements/spectrum.c: (GST_START_TEST),
(spectrum_suite):
Improve the units tests by checking for a 11025Hz sine wave
and add unit tests for all 4 supported sample types.
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(gst_real_video_dec_setcaps):
Add some more debugging.
Don't set LONG for width/height in caps.
Set correct output buffer size when caps changed.
The custom message sent to the decoder should not include the format and
subformat. Fixes#471554.
Original commit message from CVS:
2007-09-03 Johan Dahlin <johan@gnome.org>
* gst/nsf/gstnsf.c: (gst_nsfdec_finalize), (start_play_tune):
* gst/nsf/gstnsf.h:
Add support for (very) basic tagging.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream),
(gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found),
(create_recv_rtp), (create_recv_rtcp), (create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
Updated example pipelines in docs.
Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync.
Set the default latency correctly.
Add some more points where we can get caps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Add ts-offset property to control timestamping.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp),
(gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate),
(gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Various cleanups.
Feed rtpsession manager with NTP time based on pipeline clock when
handling RTP packets and RTCP timeouts.
Perform all RTCP with the system clock.
Set caps on RTCP outgoing buffers.
* gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc),
(create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain):
* gst/rtpmanager/gstrtpssrcdemux.h:
Also demux RTCP messages.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rb), (rtp_session_process_sr),
(rtp_session_process_rr), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_send_bye),
(session_start_rtcp), (session_report_blocks), (session_cleanup),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Remove the get_time callback, the GStreamer part will feed us with
enough timing information.
Split sync timing and RTCP timing information.
Factor out common RB handling for SR and RR.
Send out SR RTCP packets for lip-sync.
Move SR and RR packet info generation to the source.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp),
(rtp_source_process_sr), (rtp_source_process_rb),
(rtp_source_get_new_sr), (rtp_source_get_new_rb),
(rtp_source_get_last_sr):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Use caps on incomming buffers to get timing information when they are
there.
Calculate clock scew of the receiver compared to the sender and adjust
the rtp timestamps.
Calculate the round trip in sources.
Do SR and RR calculations in the source.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop):
Use extended timestamp to release buffers from the jitterbuffer so that
we can handle the rtp wraparound correctly.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Improve Comments.
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
(gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
(gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
(create_send_rtp_sink):
Also parse the sink caps for clock-rate instead of only relying on the
result of the signal.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Make sure we fetch the clock rate for payloads we are sending out so
that we can use it for SR reports.
Original commit message from CVS:
* gst/switch/gstswitch.c (gst_switch_chain, gst_switch_set_property):
If all information is known at time of setting start-time
property, send new segments then.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state),
(gst_rtp_session_event_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Distribute synchronisation parameters to the session manager so that it
can generate correct SR packets for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time),
(rtp_session_set_timestamp_sync), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Add methods for setting sync parameters.
Set correct RTP time in SR packets using the sync params.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Record last RTP <-> GST timestamp so that we can use them to convert NTP
to RTP timestamps in SR packets.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
Add some more advanced example pipelines.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_send_rtcp):
Add some debug and FIXME.
Release LOCK when performing session cleanup.
* gst/rtpmanager/rtpsession.c: (session_report_blocks):
Add some debug.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_send_rtp):
Make sure we always send RTP packets with the session SSRC.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query):
When synchronizing buffers, take peer latency into account.
Don't try to add our latency to invalid peer max latency values.
Original commit message from CVS:
2007-08-27 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_handle_seek_pull):
Make sure we initialize the seek result.
Original commit message from CVS:
* examples/switch/switcher.c (main):
* gst/switch/gstswitch.c (gst_switch_chain):
Make switch more reliable and also not lock up when
sink pad caps change.
Original commit message from CVS:
* gst/mpegtsparse/flutspatinfo.c:
* gst/mpegtsparse/flutspatinfo.h:
* gst/mpegtsparse/flutspmtinfo.c:
* gst/mpegtsparse/flutspmtinfo.h:
Update licences to reflect LGPL-ness of these files also.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.signals:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.h:
Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE
registers a GType that's different than the GstRTPFoo types that
farsight registers (luckily GType names are case sensitive). Should
finally fix#430664.
Original commit message from CVS:
* configure.ac:
* win32/common/config.h:
* win32/common/config.h.in:
Automatically generate win32/common/config.h via configure (this
ensures the win32 version of config.h is up-to-date when a release
is made, #433373). config.h.in file might need some more work.
Original commit message from CVS:
Patch by: Jonathan Matthew <jonathan at kaolin wh9 net>
* configure.ac:
* gst/festival/Makefile.am:
* gst/festival/gstfestival.c:
Port festival plugin to GStreamer-0.10 (#461377).
Original commit message from CVS:
2007-08-22 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_pull_tag):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script), (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): Handle pixel aspect ratio through
metadata tags like ASF does. Fluendo muxer supports this and
Flash players can support it as well this way.
Original commit message from CVS:
2007-08-22 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag):
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script), (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): Make sure we don't try filling up the
index if no times object was parsed. Fix the way we decide to
push
tags and emit no-more-pads. Fix some printf typing in debugging.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_set_property):
When drop-on-latency is set but we have no latency configured, just push
the buffer as fast as possible.
Fix typo in comment.
Original commit message from CVS:
* configure.ac:
* gst/mpegtsparse/Makefile.am:
* gst/mpegtsparse/flutspatinfo.c:
* gst/mpegtsparse/flutspatinfo.h:
* gst/mpegtsparse/flutspmtinfo.c:
* gst/mpegtsparse/flutspmtinfo.h:
* gst/mpegtsparse/flutspmtstreaminfo.c:
* gst/mpegtsparse/flutspmtstreaminfo.h:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
* gst/mpegtsparse/mpegtsparse.c:
* gst/mpegtsparse/mpegtsparse.h:
* gst/mpegtsparse/mpegtsparsemarshal.list:
Add mpeg transport stream parser written by:
Alessandro Decina. Includes a couple of files from the
Fluendo transport stream demuxer that Fluendo have
kindly allowed to be licenced under LGPL also.
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess net>
* ext/mythtv/gstmythtvsrc.c:
Add examples for live mythtv:// URIs to docs (#468039).
Also convert some tabs into spaces.
Original commit message from CVS:
* tests/check/elements/bpwsinc.c: (GST_START_TEST),
(bpwsinc_suite):
* tests/check/elements/lpwsinc.c: (GST_START_TEST),
(lpwsinc_suite):
Also test everything in 32 bit float mode.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_push_residue),
(bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
(bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
* gst/filter/gstbpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Reset residue length only when actually creating a residue.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
Override the preroll vmethod instead of overriding the render method
twice.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix EOS handling.
Convert some DEBUG into WARNINGs.
Pause task when flushing.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
Use system clock for RTCP session management timeouts.
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
Release the session lock when emiting signals.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/generic/.cvsignore:
* tests/check/generic/states.c:
Add generic state-change test suite to help to fi leaks.
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
patch by: <delete if not someone else's patch>
* ext/timidity/gstwildmidi.c:
* ext/timidity/gstwildmidi.h:
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
(gst_app_sink_class_init), (gst_app_sink_dispose),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
(gst_app_sink_render), (gst_app_sink_get_caps),
(gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
(gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Make love to appsink.
Make it support pulling of the preroll buffer.
Add docs and debug statements.
Fix some races wrt to EOS handling and stopping.
Implement getcaps.
Implement FLUSHING.
API: gst_app_sink_pull_preroll()
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_set_index),
(gst_flv_demux_get_index):
Fix locking and refcounting on the index.
Original commit message from CVS:
2007-08-14 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_adapter_flush), (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_do_seek),
(gst_flv_demux_handle_seek), (gst_flv_demux_sink_event),
(gst_flv_demux_src_event), (gst_flv_demux_query),
(gst_flv_demux_change_state), (gst_flv_demux_set_index),
(gst_flv_demux_get_index), (gst_flv_demux_dispose),
(gst_flv_demux_class_init): First method for seeking in pull
mode using the index built step by step or coming from metadata.
* gst/flv/gstflvdemux.h:
* 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): Parse
more metadata types and keyframes index.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/bpwsinc.c: (setup_bpwsinc),
(cleanup_bpwsinc), (GST_START_TEST), (bpwsinc_suite), (main):
Add unit tests for bpwsinc, testing fundamental functionality again.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/lpwsinc.c: (setup_lpwsinc),
(cleanup_lpwsinc), (GST_START_TEST), (lpwsinc_suite), (main):
Add unit tests for lpwsinc, testing fundamental functionality.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Improve debugging a bit.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_start):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(lpwsinc_start):
Reset the residue in BaseTransform::start to get a clean residue
on stream changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (process_32), (process_64):
* gst/filter/gstlpwsinc.c: (process_32), (process_64):
Fix processing with buffer sizes that are larger than the filter
kernel size.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
Fix a segfault with more than one channel and don't rebuild
the kernel & residue with every buffer.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_mode_get_type),
(gst_bpwsinc_window_get_type), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_build_kernel), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Add support for a bandreject mode and allow specifying the window
function that should be used.
* gst/filter/gstlpwsinc.c:
And another small formatting fix.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_setup), (bpwsinc_get_unit_size),
(bpwsinc_transform), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Apply the same changes to the bandpass filter:
- Support double input
- Fix processing for input with >1 channels
- Specify frequency in Hz
- Specify actual filter kernel length
- Use transform instead of transform_ip as we're working
out of place anyway
- Factor out filter kernel generation and update the filter
kernel when the properties are set
Fix bandpass filter kernel generation to actually generate
a bandpass filter by creating a highpass instead of a second
lowpass.
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Small formatting fix.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Specify the actual filter length instead of a weird
2N+1. Setting the property will round to the next odd number.
Also remove now obsolete FIXMEs.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_window_get_type),
(gst_lpwsinc_class_init), (gst_lpwsinc_init),
(lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Allow choosing between hamming and blackman window. The blackman
window provides a better stopband attenuation but a bit slower
rolloff.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (process_32), (process_64),
(lpwsinc_build_kernel):
Fix processing if the input has more than one channel.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
(gst_bpwsinc_init), (bpwsinc_setup), (bpwsinc_transform_ip),
(bpwsinc_set_property), (bpwsinc_get_property):
"this" is a C++ keyword, use "self" instead.
Add TODOs and FIXMEs and remove two wrong FIXMEs.
* gst/filter/gstlpwsinc.c:
Add FIXMEs and a new TODO.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
(gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32),
(process_64), (lpwsinc_build_kernel), (lpwsinc_setup),
(lpwsinc_get_unit_size), (lpwsinc_transform),
(lpwsinc_set_property), (lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
Add double support, replace "this" with "self" as the former
is a C++ keyword.
Implement the frequency property in Hz instead of fraction
of sampling frequency.
Remove some unecessary FIXMEs and add some TODOs, add some
required locking and refactor the kernel generation into a
separate function that is also called when the properties
change now.
And use BaseTransform::transform instead of transform_ip
as the convolution is done out of place anyway. Should
be done in place later.
Original commit message from CVS:
* configure.ac:
* gst/stereo/Makefile.am:
* gst/stereo/gststereo.c: (gst_stereo_base_init),
(gst_stereo_class_init), (gst_stereo_init),
(gst_stereo_transform_ip), (gst_stereo_set_property),
(gst_stereo_get_property):
* gst/stereo/gststereo.h:
Port the stereo element to GStreamer 0.10.
Original commit message from CVS:
* gst/filter/Makefile.am:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose),
(gst_bpwsinc_base_init), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_setup):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose),
(gst_lpwsinc_base_init), (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_setup):
* gst/filter/gstlpwsinc.h:
Use GstAudioFilter as base class and don't leak the memory
of the filter kernel and residue.
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(gst_real_video_dec_activate_push), (open_library),
(gst_real_video_dec_init), (gst_real_video_dec_finalize):
* gst/real/gstrealvideodec.h:
Remove some old unused vars.
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
Small cleanups.
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(open_library):
Remove fragment and timestamp correction code from the decoder to make
the caps and buffer contents compatible with matroska/ffdec_rvx0/...
Original commit message from CVS:
* po/POTFILES.skip:
Add POTFILES.skip with list of source files that aren't disted at the
moment but contain translatable strings. Should hopefully pacify
broken tools and make it clearer that these files are left out
intentionally (#461601 and others).
Original commit message from CVS:
Patch by: Ian Munro <imunro at netspace net au>
* gst/bayer/gstbayer2rgb.c:
Include our own "_stdint.h" instead of <stdint.h> (which may not
be available).
* gst/speed/gstspeed.h:
Native HP-UX compiler dosn't seem to like enum typedefs before the
actual enum was defined.
* gst/vmnc/vmncdec.c:
Fix wrong usage of GST_ELEMENT_ERROR macro (#461373).
Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
Use the proper context variable when setting the password !
LOG => WARNING for errors.
Give proper path when opening the codec (needs a '/' at the end).
Original commit message from CVS:
* ext/timidity/gsttimidity.c: (gst_timidity_init),
(gst_timidity_change_state), (plugin_init):
* ext/timidity/gsttimidity.h:
Don't initialize timidity in plugin_init for similar reason as below.
Original commit message from CVS:
* ext/timidity/gstwildmidi.c: (wildmidi_open_config),
(gst_wildmidi_init), (gst_wildmidi_change_state), (plugin_init):
* ext/timidity/gstwildmidi.h:
Don't initialize wildmidi in plugin_init as it also setups audio
filters which is slow.
