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.
