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
The segment should start at first PTS, and the vairable name lower_pts
state so correctly. Though we where using the first DTS instead. This
could lead to small desynchronization of video stream.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
Chinese broadcaster encapsulate AVS video codec into MPEG2-TS. They
use the stream_id 0x42 to identify AVS video streams. It should be noted
that this id is currently within the ISO reserved range, hence it's
utilisation is unofficial.
https://bugzilla.gnome.org/show_bug.cgi?id=727731
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.
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
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
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)
Co-Authored by: Thibault Saunier <tsaunier@gnome.org>
From a high level perspective, the new process for seeking h264
streams is as follows:
1) Rewind the stream until we find the first I-slice of a frame,
and mark its offset in the stream.
2) Rewind the stream until we find SPS and PPS informations,
to make sure the subsequent parser is up to date.
3) Accumulate optionnal SEI NAL units on the way.
4) Push the SPS, PPS and SEI units before the new keyframe.
https://bugzilla.gnome.org/show_bug.cgi?id=675132
Since all the other timestamp tracking now gets reset on a discont,
it makes sense to wait for a PCR and timestamp buffers like when
playback first starts
Due to mpegts streaming nature some pads are created but are only added
later to the element. This can cause a scenario where the first stream
doesn't have an available decoder (while the next ones still pending
would have) and tsdemux will fail with not-linked as the first stream
added wouldn't be linked.
To avoid this tsdemux needs to add pads to the flowcombiner
when they are created instead of only when adding them to the
element.
gst_ts_demux_push_pending_data() will check if it now can activate the
stream and add the pad, we don't have to check that ourselves.
Fixes playback of very short MPEG TS files.
Apart from just adding detection of the proper stream type, we also need to only
output the first substream (0x71) which contains the core substream.
While this does not provide *full* DTS-HD support (since it will miss the complementary
substreams), it will still work in the way legacy (non-DTS-HD) bluray players would work.
https://bugzilla.gnome.org/show_bug.cgi?id=725563
Keep a list of current global tags around and push them
whenever a new stream is started. Also convert all stream
specific tags to global as they are stream specific for
the container, so they are global for the streams from
within that container.
https://bugzilla.gnome.org/show_bug.cgi?id=644395
It is quite possible that we might get PTS/DTS before the first
PCR/Offset observation.
In order to end up with valid timestamp we wait until at least one
stream was able to get a proper running-time for any PTS/DTS.
Until then, we queue up the pending buffers to push out.
Once we see a first valid timestamp, we re-evaluate the amount of
running-time elapsed (based on returned inital running-time and amount
of data/DTS queued up) for any given stream.
Taking the biggest amount of elapsed time, we set that on the packetizer
as the initial offset and recalculate all pending buffers running-time
PTS/DTS.
Note: The buffer queueing system can also be used later on for the
dvb fast start proposal (where we queue up all stream packets before
seeing PAT/PMT and then push them once we know if they belong to the
chosen program).
ATSC ac3 streams are always guaranteed to be AC3 if EAC3 descriptor
is not present
If stream registration id is 'AC-3' then it's also guaranteed to be AC3.
Finally if AC3 descriptor is present it's guaranteed to be AC3.
Only silences a warning, but still.
The new seek handling re-creates the segment time information once it
has enough information after a seek.
The problem was that we'd completely ignore the requested rate. So store
that and use it in the newly created segment.
https://bugzilla.gnome.org/show_bug.cgi?id=694369
The program_number attribute was overloaded, trying to indicate both
the currently playing program, and the program requested via the
"program-number" property. The end result was that setting the
property didn't work (see #690934).
I added a new requested_program_number field rather than reviving the
current_program_number field because it seemed this would result in
fewer changes overall and be less confusing. It breaks symmetry with
the "program-number" property, but it retains parallels with the likes
of program->program_number.
Because gst_ts_demux_reset is called after the properties have been
parsed, requested_program_number is initialised in gst_ts_demux_init.
Whether this is exactly the right place, I don't know.
Setting the program-number property does not affect which program
is actually being demuxed.
Moving the initialization of the program_number from
gst_ts_demux_reset to gst_ts_demux_init seems to fix this issue.
https://bugzilla.gnome.org/show_bug.cgi?id=690934
* Avoids handling twice the same seek (can happen with playbin and files
with subtitles)
* Set the sequence number of the segment event to the sequence number of
the seek event that generated it (-1 for the initial one).
The seeking start time is approximated from the seek offset in bytes
using the accumulated PCR observations, so on a VBR stream there might
be a big difference between the actual PCR and the estimated one after
the seek. This might result in a long wait to skip all out of segments
packets.
Instead we just recalculate the new segment to start at the first PTS
after the seek, so that playback starts immediatly.
Until now we simply ignored those streams (since we couldn't do anything
with it anyway). Now that we have the mpegts library and we offload the
section handling to the application side we can properly identify and
extract them.
By default it is disabled for tsparse and enabled for tsdemux, but there is
a property to change that.
This should open the way to properly handle all private section streams,
including:
* DSM-CC
* MHEG
* Carousel data
* Metadata streams (though I haven't seen any of those in the wild)
* ... And all other specs/protocols making use of those
Partially fixes#560631
We still have some other stream types which haven't been ported, but
we will do so once we have defined the enums in the mpegts library.
Also add some FIXMEs regarding items discovered during analysis
* Only mpeg-ts section packetization remains.
* Improve code to detect duplicated sections as early as possible
* Add FIXME for various issues that need fixing (but are not regressions)
https://bugzilla.gnome.org/show_bug.cgi?id=702724
Since there is a conflict between the DCII stream type and BluRay
stream types, moved the processing of BluRay-specific stream types
to the beginning of the function. Only if a BluRay stream type
IS NOT found do we proceed to check the rest of the stream type
identifiers
Previous code was also "sort-of" handling a similar conflict between
BluRay AC3 audio and standard AC3 audio. Moved the special case BluRay
AC3 handling in the main switch statement to the new BluRay-specific
switch.
https://bugzilla.gnome.org/show_bug.cgi?id=697892
And if we detect a discontinuity there (like... when losing packets
or having MPEGTS over raw UDP with out-of-order packets) we just
drop the corresponding packet.
A future version could try to implement a re-ordering algorithm based
on that, similar to what rtpjitterbuffer does.
