Some Xing headers apparently start the TOC at byte 1 instead of 0. Don't
reject them because of it, just subtract the initial offset when reading
the table.
Be more lenient about what we accept as changing bits in a header - basically,
only require that the mp3 sync marker is present, for the mpeg version,
layer and samplerate.
Fixes: #581464
Some mp3 streams have an offset in timestamps, requiring us to push the
frame *AFTER* segment.stop in order for the decoder to be able to push
all data up to the segment.stop position.
This also makes timestamps (more) consistent before and after a possible
seek, and moreover makes for reasonable position reporting in live stream
(whose payload timestamps should not be taken for granted).
* Improve newsegment handling, e.g. upstream might live in TIME.
* Only send newsegment if we have needed info.
* Avoid reading past end of data section.
The problem that happens is the following:
* A packet with multiple payloads comes in
* Those payloads get handled one by one
* The first payload contains the first audio payload with timestamp A
* The second payload contains the first video (key)frame with timestamp V (where V < A)
With the previous code, the following would happen:
* the first payload gets processed, then passed to queue_for_stream
* queue_for_stream detects it's the first valid timestamp received and stores
first_ts = A
* the second payload gets processed, then pass to queue_for_stream
* queue_for_stream detects the timestamp is lower than first_ts... and
discards it... resulting in losing the first keyframe of the video stream
We've been having this issue for *ages*... it's just that nobody noticed it
that much with playbin. But with playbin2's aggresive multiqueue handling, this
will result in multiqueue not being able to preroll (because the video decoder will
be dropping a ton of buffers before (maybe) receiving the next keyframe).
Tested with over 200 asf files, and they all play the first frame correctly now,
even the most braindead ones.
This might be caused by entering the if() line 1214 and then not having
any activated_streams.. resulting in reaching line 1267 without having
any valid flow value.
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.
