Original commit message from CVS:
* gst/mpegaudioparse/gstxingmux.c: (generate_xing_header):
Fix alignment issues that caused SIGBUS on some architectures.
Original commit message from CVS:
* gst/ac3parse/gstac3parse.c: (gst_ac3parse_chain):
Fix alignment issue which isn't really an issue at all because
the plugin hasn't been ported to 0.10 yet.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_parse_video_packet):
Properly aggregate GstFlowReturn from downstream in order to properly
stop, and doing that as early as possible.
Fixes#532807
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_parse_mdpr),
(gst_rmdemux_fix_timestamp), (gst_rmdemux_parse_video_packet),
(gst_rmdemux_parse_audio_packet), (gst_rmdemux_parse_packet):
Fix video timestamps by adjusting it with the first timestamp found.
Don't assume we have a complete fragment when flushing the adapter,
packets might have been lost or the stream might just be broken.
Original commit message from CVS:
* gst/realmedia/rdtmanager.c: (gst_rdt_manager_plugin_init):
Set Rank to NONE so that we don't accidentally try to autoplug the
rdtmanager.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_emit_frame):
Send a new duration message if the average bitrate changed and
we don't know the duration from the Xing or VBRI header.
Fixes bug #321857.
Original commit message from CVS:
* gst/realmedia/rtspreal.c: (rtsp_ext_real_before_send),
(rtsp_ext_real_parse_sdp), (rtsp_ext_real_stream_select):
* gst/realmedia/rtspreal.h:
Move assembly rule parsing to the place where we parse the SDP as it's
also there that we create the MDPR and we need the currently selected
asmrule in order to select the right MTLI.
Fixes#529359.
Original commit message from CVS:
* gst/realmedia/realhash.c:
* gst/realmedia/rtspreal.c:
Include generated "_stdint.h" instead of <stdint.h> which might not
exist on some systems.
Original commit message from CVS:
* configure.ac:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(mpeg_audio_seek_entry_free):
* gst/mpegaudioparse/gstxingmux.c: (gst_xing_seek_entry_free):
Depend on GLib 2.12 and use it unconditionally as we do in other
modules too already.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(mpeg_audio_seek_entry_new), (mpeg_audio_seek_entry_free),
(gst_mp3parse_reset), (gst_mp3parse_emit_frame):
* gst/mpegaudioparse/gstxingmux.c: (gst_xing_seek_entry_new),
(gst_xing_seek_entry_free), (gst_xing_mux_finalize), (xing_reset),
(gst_xing_mux_chain):
Use GSlice for allocating the seek table entries if we compile with
GLib 2.10 or newer.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_stream_props):
Guard against division by 0 and fall back to 25/1 framerate.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_video_stream),
(gst_asf_demux_process_ext_stream_props):
Instead of adding a fixes 25/1 framerate to the video caps, use the
average frame duration in the extended properties of the video stream as
the framerate. Fixes#524346.
Original commit message from CVS:
* gst/realmedia/asmrules.c: (gst_asm_scan_string), (main):
make ) also a delimiter for rules.
Skip \\ when scanning strings.
Add new testcase for these problems.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event):
Don't take the stream lock when caching events. This is not necessary
and results in a deadlock when seeking with rhythmbox (but not with
totem or banshee for some reason).
Original commit message from CVS:
Patch by: Pizpot Gargravarr <pgargravarr at siriuscybernetics dot org>
* gst/realmedia/rtspreal.c: (rtsp_ext_real_parse_sdp):
Add the version field when creating the CONT chunk resulting in
the Author, Comment and Copyright tags not being parsed correctly.
Fixes#521459.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event), (gst_mp3parse_chain):
Remove trailing newlines from debug statements.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_sink_event):
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event):
Push EOS, FLUSH_STOP and NEWSEGMENT immediately instead
of dropping and leaking them.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_dispose), (gst_mad_sink_event),
(gst_mad_chain):
* ext/mad/gstmad.h:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_dispose),
(gst_mp3parse_sink_event), (gst_mp3parse_emit_frame):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Cache all events except EOS if we still have to send a NEWSEGMENT
event. This will let TAG events be forwarded until after decodebin
to an encoder for example as decodebin only links the pads
after NEWSEGMENT. Fixes bug #518933.
Original commit message from CVS:
* gst/mpegaudioparse/gstxingmux.c: (get_xing_offset):
Write Xing header at the correct position in the MP3 frame for
stereo files. Fixes bug #518676.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_emit_frame), (gst_mp3parse_chain):
Try a bit harder to get valid timestamps, especially if upstream
gives us one and we are at the first frame or resyncing.
Return UNEXPECTED if we get a valid timestamp that is outside of
our configured segment. After all changes done so far this doesn't
seem to cause any regression, please test.
Original commit message from CVS:
Patch by:
Hans de Goede <j dot w dot r dot degoede at hhs dot nl>
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_seek_event):
If we don't have the position to seek to in our index first try
to convert from TIME to BYTES upstream and only if that fails
too use the old hack to simply seek to an earlier position
and let the sink drop everything before segment start.
Partially fixes bug #469930.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_emit_frame):
Handler buffers without valid timestamp more correctly: Don't drop
them and don't use the invalid timestamp to calculate the next
timestamp. Fixes bug #516811.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_emit_frame):
Return GST_FLOW_UNEXPECTED if we get data that is after our
configured segment. This makes upstream go EOS immediately instead
of sending us the complete stream. Also improve debugging a bit.
Original commit message from CVS:
* gst/mpegaudioparse/gstxingmux.c: (generate_xing_header):
Correctly write the size in bytes on big endian systems.
Fixes bug #515725.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:(mp3parse_time_to_bytepos):
Use gst_guint64_to_gdouble for conversion
* win32/vs6/libgstasfdemux.dsp:
* win32/vs6/libgstdvdsub.dsp:
* win32/vs6/libgstrealmedia.dsp:
Update project dependencies and add new source files
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.
