Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_caps_create),
(gst_mp3parse_chain):
Don't set new caps on the srcpad everytime the bitrate or MPEG
version changes but calculate new spf value when the MPEG version
changes.
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:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
Add the real and rtsp elements and update the lists.
* docs/plugins/inspect/plugin-amrnb.xml:
* docs/plugins/inspect/plugin-asf.xml:
* docs/plugins/inspect/plugin-dvdlpcmdec.xml:
* docs/plugins/inspect/plugin-dvdsub.xml:
* docs/plugins/inspect/plugin-mpegaudioparse.xml:
* docs/plugins/inspect/plugin-mpegstream.xml:
* docs/plugins/inspect/plugin-realmedia.xml:
* docs/plugins/inspect/plugin-siddec.xml:
* docs/plugins/inspect/plugin-synaesthesia.xml:
Regenerate docs.
* gst/iec958/ac3_padder.c:
* gst/iec958/ac3_padder.h:
Do not use gtk-doc style comments for non gtk-doc comments. Note -
there are functions defined using extern in the .c file - does that
make sense?
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_handle_first_frame), (mp3parse_time_to_bytepos),
(mp3parse_bytepos_to_time):
Interpolate the VBRI seek table entries to get better results,
support 3 byte seek table entries and prevent overflows in the
seek table by adding the relative offsets when using the seek
table in a large enough data type.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_reset),
(gst_mp3parse_handle_first_frame), (mp3parse_time_to_bytepos),
(mp3parse_bytepos_to_time):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Add support for seeking based on the VBRI seek table. Might make
sense to use interpolation in the table later to get hopefully a
bit more accurate values.
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/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_reset),
(gst_mp3parse_emit_frame), (gst_mp3parse_handle_first_frame),
(mp3parse_total_bytes), (mp3parse_total_time):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Add initial support for reading VBRI headers as found in VBR files
created by some Fraunhofer encoders. Currently we only read the
number of frames and bytes (and calculate duration, etc from this)
but there is also a seek table that we currently don't use.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_handle_first_frame):
Guard against 0 values in the Xing header as frame count and
byte count and calculate the bitrate when we have all values
we need and not before.
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:
* ext/mad/gstmad.c: (mpg123_parse_xing_header):
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_handle_first_frame):
Make sure that the Xing TOC starts with 0 and the entries
are increasing over time. Otherwise it's broken and should
be skipped. Fixes bug #507821.
Original commit message from CVS:
* gst/realmedia/rdtmanager.c:
(gst_rdt_manager_marshal_VOID__UINT_UINT),
(gst_rdt_manager_class_init):
* gst/realmedia/rdtmanager.h:
Implement some more signals that rtspsrc connects to.
Fixes#504671.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event), (mp3parse_handle_seek):
Don't post SEGMENT_START messages on the bus, only the element
driving the pipeline should do that.
Original commit message from CVS:
2007-11-20 Julien MOUTTE <julien@moutte.net>
* gst/realmedia/rtspreal.c: (rtsp_ext_real_parse_sdp): Fix build
on Mac OS X.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
Restore the segment handling logic.
Please don't do behavioural changes under the heading of 'leak fixes'
or 'whitespace changes', people.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_ext_content_desc):
Convert tags that come as string into the type required by
GstTagList.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_emit_frame):
Remove some more broken code, it seems to clip even when it should not.
See #491305.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_emit_frame):
When the element is not driving the streaming thread it is not supposed
to emit EOS or post SEGMENT done. It is allowed to return UNEXPECTED
upstream when it detects EOS. See #491305.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <mnauw at users.sourceforge.net>
* gst/dvdsub/Makefile.am:
* gst/dvdsub/gstdvdsubdec.c:
* gst/dvdsub/gstdvdsubparse.c:
* gst/dvdsub/gstdvdsubparse.h:
Add dvd subtitle parser, which just packetizes the input
stream. This is needed to mux dvd subtitles into matroska
files, since the muxer expects unfragmented and properly
timestamped input (#415754).
Original commit message from CVS:
* gst/realmedia/asmrules.c: (gst_asm_scan_parse_expression),
(gst_asm_scan_parse_condition):
* gst/realmedia/rmdemux.c: (gst_rmdemux_parse_video_packet):
Fix some compiler warnings shown on Forte.
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:
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3parse_time_to_bytepos),
(mp3parse_bytepos_to_time):
Use gst_util_guint64_to_gdouble for conversions.
* win32/vs6/libgstmad.dsp:
Add a link to libgstaudio.
Original commit message from CVS:
* gst/iec958/ac3iec.c:
Chainup in finalize.
* tests/check/Makefile.am:
* tests/check/generic/states.c:
Improved state change unit test.
Original commit message from CVS:
* gst/dvdlpcmdec/gstdvdlpcmdec.c:
Add other allowed rates to the pad templates.
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_dispose):
Reset the parser to release memory in dispose.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event), (gst_mp3parse_emit_frame):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Queue segment event and push it after we know the caps on the pad or
else an autoplugger might not have plugged the element yet and the
segment is lost.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_reset),
(gst_rmdemux_chain), (gst_rmdemux_parse_mdpr),
(gst_rmdemux_fix_timestamp), (gst_rmdemux_parse_video_packet),
(gst_rmdemux_parse_audio_packet), (gst_rmdemux_parse_packet):
Do fragment collection in the demuxer so that we can now work with
both ffmpeg and realvideodec to decoder real video content.
Original commit message from CVS:
* gst/realmedia/rtspreal.c: (rtsp_ext_real_get_transports),
(rtsp_ext_real_parse_sdp), (rtsp_ext_real_stream_select):
Disable UDP transport for now.
Original commit message from CVS:
* gst/asfdemux/Makefile.am:
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstrtspwms.c: (gst_rtsp_wms_before_send),
(gst_rtsp_wms_after_send), (gst_rtsp_wms_parse_sdp),
(gst_rtsp_wms_configure_stream), (_do_init),
(gst_rtsp_wms_base_init), (gst_rtsp_wms_class_init),
(gst_rtsp_wms_init), (gst_rtsp_wms_finalize),
(gst_rtsp_wms_change_state), (gst_rtsp_wms_extension_init):
* gst/asfdemux/gstrtspwms.h:
Move WMS RTSP extension from -good to here.
Port it to the new pluggable extension interface.
Original commit message from CVS:
* configure.ac:
Sync liboil check with plugins-base. Add libm check.
* gst/synaesthesia/Makefile.am:
Link against libm. We're using sqrt here.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_reset),
(gst_mp3parse_emit_frame), (mp3parse_handle_seek):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Save some memory for each frame by only saving the start timestamp
and start byte position instead of additionally the stop timestamp
and stop byte position. This requires us to use a doubly-linked list
but still saves 8-12 bytes per frame.
