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.
