Otherwise we will use fields from the old caps with everything set up for the
new caps, causing crashes and worse.
Also don't do anything if the same caps are set twice.
qtdemux->streams is an array, it will never evaluate to true when comparing
to NULL. Instead we want to check the number of streams to make sure the
stream is available.
https://bugzilla.gnome.org/show_bug.cgi?id=753614
CID 1358389
The head of the queue is the oldest packet (as in lowest seqnum), the tail is
the newest packet. To calculate the fill level, we should calculate tail-head
while considering wraparounds. Not the other way around.
Other code is already doing this in the correct order.
https://bugzilla.gnome.org/show_bug.cgi?id=764889
When downstream blocks, "lost" timers are created to notify the
outgoing thread that packets are lost.
The problem is that for high packet-rate streams, we might end up with
a big list of lost timeouts (had a use-case with ~1000...).
The problem isn't so much the amount of lost timeouts to handle, but
rather the way they were handled. All timers would first be iterated,
then the one selected would be handled ... to re-iterate the list again.
All of this is being done while the jbuf lock is taken, which in some use-cases
would return in holding that lock for 10s... blocking any buffers from
being accepted in input... which would then arrive late ... which would
create plenty of lost timers ... which would cause the same issue.
In order to avoid that situation, handle the lost timers immediately when
iterating the list of pending timers. This modifies the complexity from
a quadratic to a linear complexity.
https://bugzilla.gnome.org/show_bug.cgi?id=762988
After clearing the adapter due to a DISCONT, as might happen when some packet(s)
have been lost, the depayloader was pushing data into the adapter (which had no
header due to the clear), creating a headerless frame out of it, and sending it
downstream. The downstream decoder would then usually ignore it; unless there
were lots of DISCONTs from the jitterbuffer in which case the decoder would reach
its max_errors limit and throw an element error. Now we just discard that data.
It is probaby not worth trying to salvage this data because non-progressive
jpeg does not degrade gracefully and makes the video unwatchable even with
low packet loss such as 3-5%.
The PIFF data is stored in a custom UUID box which is parsed and the
crypto_info of the element is updated accordingly. This allows
downstream decryptors to process and decrypt the protected content.
https://bugzilla.gnome.org/show_bug.cgi?id=753614
payload_buffer hasn't been assigned a value before the jumps to
switch_failed or packet_short. So the value must be NULL. No need
to unmap and unref.
CID #1316476
Free memory of current macroblock once it isn't needed so it isn't leaked
by the call of the gst_rtp_h263_pay_B_mbfinder function.
if (!(mac = gst_rtp_h263_pay_B_mbfinder (context, gob, mac, mb))) {
CID 1212156
Make sure that all data is drained out when the reference pad
goes EOS. Fixes a problem where data that arrives on other
pads after the reference pad finishes can stall forever and
never pass EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=763711
Deadlock occurs when splitting files if one stream received no buffer during
the first GOP of the next file. That can happen in that scenario for example:
1) The first GOP of video is collected, it has a duration of 10s.
max_in_running_time is set to 10s.
2) Other streams catchup and we receive the first subtitle buffer at ts=0 and
has a duration of 1min.
3) We receive the 2nd subtitle buffer with a ts=1min. in_running_time is set to
1min. That buffer is blocked in handle_mq_input() because
max_in_running_time is still 10s.
4) Since all in_running_time are now > 10s, max_out_running_time is now set to
10s. That first GOP gets recorded into the file. The muxer pop buffers out
of the mq, when it tries to pop a 2nd subtitle buffer it blocks because the
GstDataQueue is empty.
5) A 2nd GOP of video is collected and has a duration of 10s as well.
max_in_running_time is now 20s. Since subtitle's in_running_time is already
1min, that GOP is already complete.
6) But let's say we overran the max file size, we thus set state to
SPLITMUX_STATE_ENDING_FILE now. As soon as a buffer with ts > 10s (end of
previous GOP) arrives in handle_mq_output(), EOS event is sent downstream
instead. But since the subtitle queue is empty, that's never going to
happen. Pipeline is now deadlocked.
To fix this situation we have to:
- Send a dummy event through the queue to wakeup output thread.
- Update out_running_time to at least max_out_running_time so it sends EOS.
- Respect time order, so we set out_running_tim=max_in_running_time because
that's bigger than previous buffer and smaller than next.
https://bugzilla.gnome.org/show_bug.cgi?id=763711
RFC 2435 mentions in section 4.1 that U/V use table number 1, but this seems
just like an example. Some encoders are not following that and there seems to
be no reason to reject their streams.
https://bugzilla.gnome.org/show_bug.cgi?id=761345
It's not like we could accept any other caps here. The caps are decided by the
upstream caps event.
Also keep the filter order intact when filtering the results against the
filter caps.
https://bugzilla.gnome.org/show_bug.cgi?id=763326
If we don't find the index of the sample correctly in src_convert function,
we have to unref about the qtdemux before returning value.
So, I have modify it about instead pass qtdemux as a parameter into
src_convert function.
https://bugzilla.gnome.org/show_bug.cgi?id=763973
Currently, get_duration function always return the TRUE even though
it can't be set duration correctly. So, we need to add the else condition
about the fail case. Also, we already set the GST_CLOCK_TIME_NONE
in this function. So I have modify it which is related code in some
function.
https://bugzilla.gnome.org/show_bug.cgi?id=763968
In other words, gst_pad_get_current_caps should never return NULL
in a pad-added callback from the demuxer.
Added tests for the two special cases with AAC and H.264 where this
would happen every time.
https://bugzilla.gnome.org/show_bug.cgi?id=763780
Changing the input caps and not using them anymore afterwards is useless, and
it breaks negotiation in pipelines like:
gst-launch-1.0 videotestsrc ! "video/x-raw,framerate=25/1,interlace-mode=interleaved" !
deinterlace fields=all ! "video/x-raw,framerate=50/1,interlace-mode=progressive" !
fakesink
This reverts commit 4065fcb80a.
flacparse should not push tags by itself, the base class is going to do that
while properly merging in upstream tags. It just didn't because of a bug in
the base class, which was hidden by this commit.
https://bugzilla.gnome.org/show_bug.cgi?id=763553
When upstream is running in bytes in push-mode, qtdemux will
convert seeks from time to bytes and send it upstream. Upstream
element will perform a byte seek and send a byte segment to qtdemux
that will convert it to time and push it downstream.
There is, however, the pending_segment variable that stores a new
segment event to be pushed before the next data. When handling seeks
as mentioned above this variable was being ignored and, if it contained
some segment event, it would override the one resulting from the seek.
This would restore a previous segment and would cause the seek segment
to be discarded downstream.
This patch fixes this issue by unrefing any pending segment as the
seek from upstream should contain the latest one that should be
used, as requested by the application.
https://bugzilla.gnome.org/show_bug.cgi?id=763165
On Windows the socket will be bound to ANY instead of the multicast group,
as binding to a multicast group does not work. Which would mean that we
override src->addr to become ANY and won't automatically join a multicast
group anymore on Windows.
On Linux we would automatically join a multicast group, keep it consistent.
https://bugzilla.gnome.org/show_bug.cgi?id=763093
