The parser can accept input that is not completely specified. Use the
ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
check for intersection only. This allows us to proxy downstream
constraints while still allowing non-subset caps as input.
We can then also remove the appended template caps workaround.
Make a unit-test to check the new feature.
This reverts commit 26040ee38c
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024
In SmoothStreaming fragmented scenario, the timestamps are calculated
starting from the fragment buffer timestamp. When there is a not-linked
return from downstream, qtdemux will return upstream and will keep the
non-pushed data into its adapter.
On a new fragment buffer pushed to qtdemux, the new buffer timestamp
would overwrite the previous one that should be used on the still
to be pushed buffers. Because of this, this patch will also
update the fragment_start timestamp from the adapter last pts
to make sure the moof and timestamps are in sync and will result
in correct timestamps for all fragments.
In the scenario of "mdat | moov (with fragmented artifacts)" qtdemux
could read the moov again after the mdat because it was considering the
media as a fragmented one.
To avoid this loop this patch makes it store
the last processed moov_offset to avoid parsing it again.
And it also checks if there are any samples to play before
resturning to the mdat, so that it knows there is new data to be played.
https://bugzilla.gnome.org/show_bug.cgi?id=691570
When parsing a trak only free streams on failures if those streams
were created locally. They could have been created from a previous
fragment, in this case we they have valid info from the other fragment.
Including pads.
https://bugzilla.gnome.org/show_bug.cgi?id=691570
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
As for text subtitles and as suggested in #712643, throw
away the 2 byte terminator packets that some encoders insert.
This will make things better when remuxing and causes generation
of gap events.
Otherwise there were race conditions where we would send tags
on a flushing srcpad.
We have a test for that in GES, but this should be tested
systematically with harness in the future as I believe it
is useful for exactly that kind of cases.
https://bugzilla.gnome.org/show_bug.cgi?id=708165
Clean up the handling of mp4s streams. Use the generic esds
descriptor function to extract the palette, instead of hard coding
a wrong magic offset.
Add some more size safety checks when parsing ES descriptors, and
replace magic numbers with the descriptive constants that are already
defined.
Enhance dump output for stsd atoms.
Streams from both bug 712643 and historic bug 568278 now both work
correctly.
Fixes: #712643
Remove bogus reconfigure event on collision, we don't want to send the event on
the receiving RTP pad and the collision event is now handling this
case.
See https://bugzilla.gnome.org/show_bug.cgi?id=711560
The problem here was that the jitterbuffer lock was unlocked to push
the event, but that caused another thread to remove the timer currently
being processed, probably because the amount of rtx events
(and therefore timers) was getting too high. The solution is to
unlock and push the event only after timer processing has finished.
fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131
Restore the behavior of the element to the state before commit
db29522a43. A non-empty header is
generated and when the EOS event is received the header is generated
again, this time with the correct size.
https://bugzilla.gnome.org/show_bug.cgi?id=711699
An internal sender in a session is also a receiver of its own packets so update
the receiver stats. Other senders in the session will use this info to generate
correct RB blocks in their SR reports.
Assume a file with atoms in the following order: moov, mdat, moof,
mdat, moof ...
The first moov usually doesn't contain any sample entries atoms (or
they are all set to 0 length), because the real samples are signaled
at the moofs. In push mode, qtdemux parses the moov and then finds the mdat,
but then it has 0 entries and assumes it is EOS.
This patch makes it continue parsing in case it is a fragmented file so that
it might find the moofs and play the media.
https://bugzilla.gnome.org/show_bug.cgi?id=710623
In push mode, when qtdemux can't use a seek to skip the mdat buffer it has
to buffer it for later use.
The issue is that after parsing the next moov/moof, there might be some
trailing bytes from the next atom in the file. This data was being discarded
along with the already parsed moov/moof and playback would fail to continue
after the contents of this moov/moof are played.
This is particularly bad on fragmented files that have the mdat before the
corresponding moof. So you'd get:
mdat|moof|mdat|moof ...
When a moof was received, it usually came with some extra bytes that would
belong to the next mdat (because upstream doesn't care about atoms alignment).
So those bytes were being discarded and playback would fail.
This patch makes qtdemux store those extra bytes to reuse them later after the
mdat is emptied.
https://bugzilla.gnome.org/show_bug.cgi?id=710623
Add a new timestamp mode that assumes the local and remote clock are
synchronized. It takes the first timestamp as a base time and then uses the RTP
timestamps for the output PTS.
matroska-demux.c: In function 'gst_matroska_demux_add_stream':
matroska-demux.c:1379:7: error: format '%u' expects argument of type 'unsigned int', but argument 4 has type 'guint64' [-Werror=format=]
"%03u", context->uid);
^
WebM has a couple of specific requirements we need to handle.
Idea is to set this flag once and just rely on mux->is_webm
at run time instead of repeatedly figuring this out from
GST_MATROSKA_DOCTYPE_WEBM (which requires a strcmp()).
WebM spec states SegmentUID is Unsupported. Files produced
with gstreamer without this change will spit an error like
this when passed to mkvalidator:
ERR201: Invalid 'SegmentUID' for profile 'webm' in Info at 192
When flush-stop arrives before we process the result of the _push() in the
loop function, we might pause even though we are not flushing anymore. Fix this
race by waiting for the srcpad loop function to completely pause after doing the
flush-start.
If we were not waiting for the missing seqnum when we insert the lost packet
event in the jitterbuffer, we end up not updating the next_seqnum and wait
forever for the lost packets to arrive. Instead, keep track of the amount of
packets contained by the jitterbuffer item and update the next expected
seqnum only after pushing the buffer/event. This makes sure we correctly handle
GAPS in the sequence numbers.
Doing so would be a regression over 1.0 and breaks the unit test.
However the result will be most likely unusable, so let's post
a warning message on the bus.
Use g_date_time seconds manipulation to allow to cover the quicktime
spec for creation_time. It uses seconds since 1904.
Both paths could be done using the generic approach of seconds since
1904 with GDateTime handling, but the first path using seconds from
1970 should be more commonly found and avoids a few objects creation and
ref/unref, so keep it there for performance.
Additionally, the code for handling seconds since 1970 changed from >
to >= because having 0 seconds since 1970 is also a valid case for that
path to handle.
https://bugzilla.gnome.org/show_bug.cgi?id=707975
Always prepare a lost event in the jitterbuffer, it is to wake up and make the
pushing thread continue. We drop the event when we are not supposed to push lost
events downstream.
Schedule the lost event by placing it inside the jitterbuffer with the seqnum
that was lost so that the pushing thread can interleave and push it properly.
Make the jitterbuffer operate on a structure containing all the packet
information. This avoids mapping the buffer multiple times just to get the RTP
information. It will also make it possible to store other miniobjects such as
events later.
Improve the order of the timeout events, if there are timers with the same
timeout, we want to trigger the lowest seqnum first. For this we need to loop
over the complete array of timers to find the best one before triggering the
timeout.
First send the lost event, then update the next_seqnum counter and then
send the signal to the pushing thread that it can retry to push a buffer. This
avoids pushing out buffers before the lost event is pushed.
There is no need to unschedule the timer in flush-start, flush-stop will remove
the timers and unschedule.
Unschedule the current timer before attempting to join the timer thread.
Use the more correct POFFSET macro to get the offset of a component in its
plane. The offset macro gives the offset of the component relative to the start
of the frame.
clang does not want or need a clobber list for emms:
error: clobbers must be last on the x87 stack
Patch taken from the FreeBSD ports, provided by
Dan McGregor <dan.mcgregor@usask.ca>
The streamable property only make sense for fragmented formats.
For regular MP4, when downstream is not seekable we can't rewrite
the headers, so qtmux can only work with fast-start=TRUE, where
the headers are written finishing the file.
For fragmented MP4, when streamable is not seekable and the streamable
property is FALSE, we must enforce streamable=TRUE warning the user
about this change
https://bugzilla.gnome.org/show_bug.cgi?id=707242
The most common use case for fragmented MP4 (Dash and Smooth Streaming)
is producing streamable content (even for VOD). streamable=FALSE would only
be used to generate fragmented MP4 with and index of MOOF's that could
be reproduced without a playlist/manifest
https://bugzilla.gnome.org/show_bug.cgi?id=707242
Don't assume planar formats have just one memory block with the data but use the
macros to access the right memory block where a component can be found.
Keep a separate delay in the timer so that we still know the original timestamp
of the packet that this timer refers to. We can then place the correct
running-time in the Retransmission event.
Instead of pushing the lost event from the chain function, schedule a timeout
that will push the lost event from the timer thread. This avoid blocking the
upstream thread while we push and sync the event.
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
Keep track of the current time in the timeout loop.
Loop over all timers and trigger all the expired ones, we can do this in the
same loop that selects the new best timer.
Also update the timers when retransmission is disabled. We need to
do this because when we added LOST timers when we detected missing packets and
we need to remove those timers when the packet finally arrives.
