Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (channel_mode_class),
(GST_TYPE_MP3_CHANNEL_MODE), (mp3_type_frame_length_from_header),
(gst_mp3parse_emit_frame), (mp3parse_get_query_types):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Do an initial class_ref on an internal enum type from within the
class_init function so that there aren't any issues when multiple
mp3parse elements are started in separate threads at the same
time. (Why we use an enum type here if the tag is registered as
a string type, I don't know). Also remove custom UNUSED macro
and use GLib's instead.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
Calculate samples per frame correctly for "MPEG 2.5" layer 3.
Fixes skipping on these files.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event):
Post a GST_ELEMENT_ERROR if we get EOS before seeing any valid
frames. Partially fixes bug #552237.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event), (gst_mp3parse_emit_frame),
(mp3parse_total_time), (mp3parse_bytepos_to_time):
Don't recurse from mp3parse_bytepos_to_time() to mp3parse_total_time()
if we're called from there already. Otherwise we end up in a endless
recursion and crash with a stack overflow.
This can happen when a Xing or VBRI header with TOC exists but it
doesn't contain the total time. Fixes bug #545370.
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_setcaps):
* gst/mpegaudioparse/gstmpegaudioparse.c:
(mp3_type_frame_length_from_header), (mp3_caps_create),
(gst_mp3parse_chain):
Put the MPEG audio version into the caps as "mpegaudioversion".
This is different from "mpegversion".
Original commit message from CVS:
* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
* ext/a52dec/gsta52dec.c:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c:
* ext/amrnb/amrnbparse.c:
* ext/lame/gstlame.c:
* ext/mad/gstmad.c:
* ext/sidplay/gstsiddec.cc:
* gst/asfdemux/gstrtspwms.c:
* gst/mpegaudioparse/gstxingmux.c:
* gst/realmedia/rademux.c:
* gst/realmedia/rdtmanager.c:
* gst/realmedia/rtspreal.c:
* gst/synaesthesia/gstsynaesthesia.c:
Add missing elements to docs. Restore alphabetical order in section
file. Document mad (it was included in docs already).
Fix doc-markup: use convinience syntax for examples
(produces valid docbook), add several refsec2 when we have several
titles. Fix some types.
Original commit message from CVS:
* ext/lame/gstlame.c:
* ext/sidplay/gstsiddec.cc:
* gst/mpegaudioparse/gstxingmux.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (head_check):
Don't mark MPEG headers with emphasis == 0x2 as invalid. This
emphasis value is reserved but unfortunately files with that
value exist and the information is not important for the decoder
anyway. Fixes bug #537235.
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/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:
* 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/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:
* 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:
* 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:
* 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/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:
* 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/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: 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/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.
