Timestamps should start at the segment start, rather than 0, so
we need to not subtract the first timestamp. This makes the sink
correctly account for running time when switching PMTs where a
stream starts not quite at zero, causing timing offsets that can
become noticeable and causing dropped frames after a few times.
A new program object is created to replace an existing one
in the programs hash table, so its refcount needs to match.
With the default of 0 refcount on creation, the next PAT
change will cause that refcount to be both incremented and
decremented (assuming the new PAT references that stream too),
which will cause the program to be destroyed.
https://bugzilla.gnome.org/show_bug.cgi?id=748412
If the stream which is about to be removed still has a ref on a tag list we
should drop it.
Fix a leak which was occasionally happening with the
validate.file.playback.change_state_intensive.tron_en_ge_aac_h264_ts scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=748576
Such seeks are used to change playback rate and we do not want
to alter the position in that case, so we bypass the flush/seek
logic, and set things up so a new segment is scheduled to be
regenerated.
https://bugzilla.gnome.org/show_bug.cgi?id=735100
This will happen when the PMT changes, replacing streams with
new ones. In that case, we need to accumulate the running time
from the previous chain in the segment base.
https://bugzilla.gnome.org/show_bug.cgi?id=745102
Always update the segment and not only for accurate seeking and always
send a new segment event after seeks.
For non-accurate force a reset of our segment info to start from
where our seek led us as we don't need to be accurate
https://bugzilla.gnome.org/show_bug.cgi?id=743363
The flush is called on discont and we shouldn't output a new segment
each time a discont happens. So this commit remove the mark for a new
segment when flushing streams by propagating the 'hard' flag passed
on the flusing from the base class.
https://bugzilla.gnome.org/show_bug.cgi?id=743363
When dealing with random-access content (such as files), we initially
search for the last PCR in order to figure out duration and to handle
other position estimation such as those used in seeking.
Previously, the code looking for that last PCR would search in the last
640kB of the file going forward, and stop at the first PCR encountered.
The problem with that was two-fold:
* It wouldn't really be the last PCR (it would be the first one within
those last 640kB. In case of VBR files, this would put off duration
and seek code slightly.
* It would fail on files with bitrates higher than 52Mbit/s (not common)
Instead this patch modifies that code by:
* Scanning over the last 2048kB (allows to cope with streams up to 160Mbit/s)
* Starts by the end of the file, going over chunks of 300 MPEG-TS packets
* Doesn't stop at the first PCR detected in a chunk, but instead records all
of them, and only stop searching if there was "at least" one PCR within
that chunk
This should improve duration reporting and seeking operations on VBR files
https://bugzilla.gnome.org/show_bug.cgi?id=708532
As a consequence, tsdemux won't remove its pads anymore on EOS.
Fixes the case when mpegtsbase is not able to process new packets
after EOS as the corresponding pids aren't known anymore because
the programs were removed and the pes/psi were kept, preventing the
PAT to be parsed again.
https://bugzilla.gnome.org/show_bug.cgi?id=738695
Assume that small backward PCR jumps are just from upstream packet
mis-ordering and don't reset timestamp tracking state - assuming that
things will be OK again shortly.
Make the threshold for detecting discont between sequential buffers
configurable and match the smoothing-latency setting on tsparse
to better cope with data bursts.
When the set-timestamps property is set, use PCRs on the provided
(or autodetected) pcr-pid to apply (or replace) timestamps on the
output buffers, using piece-wise linear interpolation.
This allows tsparse to be used to stream an arbitrary mpeg-ts file,
or to smooth jittery reception timestamps from a network stream.
The reported latency is increased to match the smoothing latency if
necessary.
Signal sparse streams properly in stream-start event and force sending
of pending sticky events which have been stored on the pad already and
which otherwise would only be sent on the first buffer or serialized
event (which means very late in case of subtitle streams). Playsink in
playbin waits for stream-start or another serialized event, and if we
don't do this it will wait for the multiqueue to run full before
starting playback, which might take a couple of seconds.
https://bugzilla.gnome.org/show_bug.cgi?id=734040
All pads of a stream are now added at the beginning. In order to cope with
streams that don't get any data (forever or for a long time) we detect gaps
and push out GAP events when needed.
Cleanups and commenting by Jan Schmidt <jan@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=734040
If a discontinuity in the stream is detected, data is discarded until
a new PES starts. If the first packet after the discontinuity is also
the start of a PES, there is no reason to discard the packets.
https://bugzilla.gnome.org/show_bug.cgi?id=737569
packet_length is defined as a guint16 in the PESHeader structure. This
definition match the specification. But since we add 6 bytes to the
packet_length value (length of start_code + stream_id + packet_length),
we can overflow the guint16 when the value in the PES header is greater
than 65529.
So use a guint32 instead of a guint16 to avoid overflow.
https://bugzilla.gnome.org/show_bug.cgi?id=736490
32 bit integers are going to overflow, especially the PCR offset to
the first PCR will overflow after about 159 seconds. This makes playback
of streams stop at 159 seconds as suddenly the timestamps are starting
again from 0. Now we have a few more years time until it happens again
and 64 bits are too small.
It was previously a mix and match of both variants, introducing just too much
confusion.
The prefix are from now on:
* GstMpegts for structures and type names (and not GstMpegTs)
* gst_mpegts_ for functions (and not gst_mpeg_ts_)
* GST_MPEGTS_ for enums/flags (and not GST_MPEG_TS_)
* GST_TYPE_MPEGTS_ for types (and not GST_TYPE_MPEG_TS_)
The rationale for chosing that is:
* the namespace is shorter/direct (it's mpegts, not mpeg_ts nor mpeg-ts)
* the namespace is one word under Gst
* it's shorter (yah)
When wrapover/reset occur, we end up with a small window of time where
the PTS/DTS will still be using the previous/next time-range.
In order not to return bogus values, return GST_CLOCK_TIME_NONE if the
PTS/DTS value to convert differs by more than 15s against the last seen
PCR
https://bugzilla.gnome.org/show_bug.cgi?id=674536
