Fixes playback of Windows Media RTSP streams and other non-Real RTSP
streams where the server errors out because it can't handle the
Real-specific 'Required: com.real.retain-entity-for-setup' header
we've been adding unconditionally in the recent past.
For reference:
rtsp://66.111.34.191:601/broadcast/alnour.rm
rtsp://195.134.224.231/snowboard_100.wmv
On win32, we're required to link to all the libraries used - including
ones only indirectly used by other libs. So, add gstaudio, gsttag, and
(for windows only) winsock.
Parse the ETag from the describe method and pass the sessionid as the value for
the If-Match header is subsequent setup calls.
Fixes support for more RealMedia RTSP streams.
Don't introduce glitches in the output by a) relaxing the threshold for
taking upstream timestamps in preference to our calculated timestamps and
b) only set the discont flag on outgoing buffers in response to an incoming
discont buffer.
Fixes: #575046
Don't allow a change in sample rate/channels/layer/version unless we can
see another frame at the correct offset. Prevents accidently flipping
due to simple single-bit corruption.
Since SEEK event handling might perform some conversion
from TIME to BYTES, do not let upstream fool application
into (TIME) seeking not being possible.
Integer underflow made accurate seeks to near zero fail and seek to
completely the wrong place. Fix by clamping to zero, since we can't seek
to negative times anyway.
Add a new utri handler for pnm:// that for now just redirects to the same uri
with the rtsp:// protocol, which usually works nowadays.
Separate the registration of the various plugins into a separate source file.
Drop packets with an invalid replicated data length
instead of continuing with an invalid timestamp
and uninitialized payload metadata.
All other code assumes that the timestamps are valid.
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/asfdemux/gstasfdemux.c: (gst_asf_demux_chain):
Remove duplicate and broken code for the streaming case and simply reuse
the much better working pull based code. Fixes#560348.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_video_stream):
Only copy sane aspect ratio values on the caps. Fixes#559682.
Original commit message from CVS:
Patch by: Tal Shalif <tshalif at nargila dot org>
* gst/mpegstream/gstdvddemux.c:
(gst_dvd_demux_get_subpicture_stream):
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream):
Fix memmory corruption due to not storing the new updated pointer
after a g_renew(). Fixes#558896.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream),
(gst_rmdemux_descramble_mp4a_audio),
(gst_rmdemux_handle_scrambled_packet):
Add suport for mpeg4 and aac audio. See #556714.
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/asfdemux/gstasfdemux.c: (gst_asf_demux_loop):
Fix aggregated GST_FLOW_RETURN check for when to send an error message
on the bus.
Re-fixes #546859
Original commit message from CVS:
* gst/realmedia/rdtdepay.c: (gst_rdt_depay_init),
(gst_rdt_depay_setcaps), (gst_rdt_depay_sink_event),
(create_segment_event), (gst_rdt_depay_push),
(gst_rdt_depay_handle_data), (gst_rdt_depay_change_state):
* gst/realmedia/rdtdepay.h:
Parse other values from the incomming caps.
Add event handler to handle flushing and segments.
Create segment events.
* gst/realmedia/rdtjitterbuffer.c: (rdt_jitter_buffer_insert):
Do skew correction based on RDT timestamps.
* gst/realmedia/rdtmanager.c: (activate_session),
(gst_rdt_manager_parse_caps), (gst_rdt_manager_setcaps),
(create_recv_rtp):
Parse caps to get the clockrate needed for the jitterbuffer.
* gst/realmedia/rmdemux.c: (gst_rmdemux_parse_video_packet):
Apply timestamp fixup after correcting for initial timestamp and
internal base timestamp corrections.
Original commit message from CVS:
* gst/realmedia/rdtdepay.c: (gst_rdt_depay_handle_data),
(gst_rdt_depay_change_state):
* gst/realmedia/rdtdepay.h:
Check seqnum gaps and drop duplicate packets or mark outgoing buffers
with a DISCONT flag when needed.
* gst/realmedia/rdtmanager.c: (gst_rdt_manager_query_src):
Report the configure latency instead of a hardcoded value.
Original commit message from CVS:
* gst/realmedia/rdtmanager.c: (create_session), (activate_session),
(free_session), (gst_rdt_manager_query_src),
(gst_rdt_manager_src_activate_push),
(gst_rdt_manager_handle_data_packet), (gst_rdt_manager_chain_rdt),
(gst_rdt_manager_loop), (create_recv_rtp):
Include the new rdt jitterbuffer in the session manager.
Original commit message from CVS:
* gst/realmedia/rdtdepay.c: (gst_rdt_depay_class_init),
(gst_rdt_depay_finalize), (gst_rdt_depay_setcaps),
(gst_rdt_depay_push), (gst_rdt_depay_handle_data),
(gst_rdt_depay_chain), (gst_rdt_depay_change_state):
* gst/realmedia/rdtdepay.h:
Use new RDT parsing helper functions.
Copy discont flags correctly.
Push the header from the chain function instead of the setcaps function.
Copy incomming timestamp to the output buffers instead of doing magic
with the RDT timestamps.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_init),
(find_seek_offset_time), (gst_rmdemux_reset), (gst_rmdemux_chain),
(gst_rmdemux_parse_mdpr), (gst_rmdemux_descramble_cook_audio),
(gst_rmdemux_descramble_dnet_audio),
(gst_rmdemux_parse_video_packet), (gst_rmdemux_parse_audio_packet):
* gst/realmedia/rmdemux.h:
Keep track of the first timestamp of the stream and add this to the
outgoing buffer timestamps so that we can handle live streams.
Set discont flag on the first buffers and after a seek.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
Properly aggregate flow returns for both push and pull mode, so we shut
down if all pads are unlinked.
Fixes#546859.
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/dvdread/dvdreadsrc.c: (plugin_init):
* ext/lame/gstlame.c: (plugin_init):
* gst/asfdemux/gstasf.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
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:
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
(gst_dvd_demux_get_audio_stream),
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_parse_packhead), (gst_dvd_demux_reset):
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_init),
(gst_mpeg_demux_process_event), (gst_mpeg_demux_init_stream),
(gst_mpeg_demux_parse_packhead), (gst_mpeg_demux_reset):
* gst/mpegstream/gstmpegdemux.h:
Resend tags event after a FLUSH (seek) to support prerolling
a partial pipeline.
Original commit message from CVS:
* configure.ac:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_object):
Use correct error code for encrypted streams.
Original commit message from CVS:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_subbuffer),
(gst_mpeg_demux_sync_stream_to_time):
Bridge gaps in stream by NEWSEGMENT sending. Fixes#540194.
Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c: (gst_dvd_read_src_read),
(gst_dvd_read_src_create), (gst_dvd_read_src_handle_seek_event):
Allow and implement non-flushing and/or segment seek
(mainly in TIME and chapter format).
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_event),
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_synchronise_pads),
(gst_dvd_demux_sync_stream_to_time):
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_process_event),
(gst_mpeg_demux_send_subbuffer),
(gst_mpeg_demux_sync_stream_to_time),
(gst_mpeg_streams_reset_cur_ts):
* gst/mpegstream/gstmpegdemux.h:
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_process_event),
(gst_mpeg_parse_pad_added), (gst_mpeg_parse_handle_src_query):
Delegate a query to upstream if it can't be handled.
Make segment stop aware.
Fix (subtitle) stream synchronization.
Add some debug statements.
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/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.
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.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_emit_frame):
Fix a calculation that was causing mp3parse to drop every incoming
frame when upstream delivered a segment in TIME format, breaking
playback of all mpeg system streams.
