... by seeking to target offset determined by new seek segment,
rather than that of the previous segment. The latter would typically
seek back to start for a non-accurate seek, and lead to a lot
of skipping in case of an accurate seek.
Currently tsdemux timestamps only the PTS, and only issues the DTS if
it's different. In that case, parsers tend to estimate the next DTS
based on the previous DTS and the duration, which can accumulate
rounding errors.
Packets of a given PID are meant to have sequential continuity counters
(modulo 16). If there are not sequential, this is the sign of a broken
stream, which we then consider as a discontinuity.
But if that new packet is a frame start (PUSI is true), then we can resume
from that packet without any damage.
Allows for "low latency" mpeg-ts mode which is not standard, but somewhat common.
For this to work the sender has to put timestamps at a higher frequency than the spec requires.
PES packets with size 0 are unbounded, and
could therefore overflow the 32-bit size
accumulator.
Add a 32MB limit, which is larger than
any PES packet should ever get. If one does,
then output a 32MB chunk and continue.
Don't signal a pipeline error when processing incomplete
j2pk PES packets that are too small. That can happen normally
during a DISCONT and shouldn't shut down the whole pipeline
Remove some custom and incomplete seek calculation
logic in favour of gst_segment_do_seek(), and
short-circuit any actual seeking or recalculation
if the position didn't change and just send an updated
segment directly.
This removes the custom seeking logic in favour of
using standard core seek handling.
Unless we only have sparse streams. In this case we will consider them.
It fixes a bug happening when first observed timestamp comes from a
sparse stream and other streams don't have a valid timestamp, yet. Thus
leading the timestamp from sparse stream to be the start of the
following segment. In this case, if the timestamp is really bigger than
non-sparse stream (audio/video), it will lead the pipeline to clip
samples from the non-parse stream.
https://bugzilla.gnome.org/show_bug.cgi?id=744469
* Avoid copying the pending data and instead create a buffer directly from
that data with the appropriate offset.
* Locate the jp2k magic to determine the exact location of the (first) frame
data instead of assuming that the header is of an expected size
https://bugzilla.gnome.org/show_bug.cgi?id=786111
The jp2k specification (ITU-T T.800) specifies that the 'brat' box
has two fields and the second one (AUF2) can be set to 0 for progressive
streams.
The problem is that the mpeg-ts specification (ITU-T H.222.0 06/2012)
says that the AUF2 field is only present if the stream is interlaced
In order to cope with both situation, accept those next 32bit if the
stream is marked as progressive and those bits contain 0
https://bugzilla.gnome.org/show_bug.cgi?id=786111
For each MpegTSBaseStream, we have a GstStream object which
subclasses can extend with information.
For each program a GstStreamCollection is created with all
GstStream from each stream.
EAC3 bit streams shall be identified with a stream_type value of 0x87 when
transmitted as PES streams conforming to ATSC-published standards. It is specified
in ATSC Standard A/52.
https://bugzilla.gnome.org/show_bug.cgi?id=770528
When draining a program, we might send a newsegment event on the pads
that are going to be removed (and then the pending data).
In order to do that, calculate_and_push_newsegment() needs to know
what list of streams it should take into account (instead of blindly
using the current one).
All callers to calculate_and_push_newsegment() and push_pending_data()
can now specify the program on which to act (or NULL for the default
one).
A simple fix for the problem of creating new pads with duplicate
names when switching program, easier than the alternative of
trying to work out which pads might persist and manage that.
See https://bugzilla.gnome.org/show_bug.cgi?id=758454
When the sub-class claims a program for later freeing, make
sure it's not left in the hash table, or it can cause crashes on shutdown.
Make sure tsdemux frees any program it has kept around at shutdown
if it wasn't freed already.
https://bugzilla.gnome.org/show_bug.cgi?id=763503
The order in which program switch must happen is:
1) drain all data on old pads (but don't push EOS)
2) add new pads (but don't push any data on them)
3) Push EOS and remove old pads
4) Start pushing data on new pads
There was one caveat in this implementation, which is that when
we activate a sparse pad (step 2) we would push a GAP event. The problem
is that, while being an event, it is actually *data*.
We therefore need to make sure pushing those GAP event is done at the step
we start pushing data.
https://bugzilla.gnome.org/show_bug.cgi?id=750402
Before we add any streams, make sure we drain all streams. This ensures
there's consistency that only "new" data will be pushed on buffers once
the new pads are added
https://bugzilla.gnome.org/show_bug.cgi?id=750402
When changing programs, the order of events needs to be the following:
* add pads from new program
* send EOS on old pads
* remove old pads
* emit 'no-more-pads'
Previously tsdemux was not doing that, and was first deactivating and
removing old pads before adding new ones.
We fix this by allowing subclasses of mpegtsbase to be able to handle
themselves the deactivation of programs. In this case tsdemux will
properly deactivate it once it has activated the new program.
https://bugzilla.gnome.org/show_bug.cgi?id=750402
The values of channel_mapping are copied by gst_codec_utils_opus_create_caps ()
but it doesn't free or take ownership of the g_new0 allocated memory. This
needs to be freed before going out of scope.
CID 1338692
If tsdemux never receives data for a stream, the corresponding pad will never
be added and stream->active will remain FALSE. When the stream is removed, the
pad will not be unreffed and will be leaked.
https://bugzilla.gnome.org/show_bug.cgi?id=757873
The tsdemux latency should always be added to the minimum
latency (which is always a valid clock time value). The
"cleanup" in commit a1f709c2 made it so that it would not
be added if upstream reported 0 as minimum latency (as
e.g. udpsrc would). This broke playback of live mpeg-ts
streaming in some cases, leading to playback stutter due
to a too-small configured latency for the pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=751508
