There are some few but certain conditions where it is possible for the
dest_width to be smaller than x. So we check this before assigning a negative
value to src_width, which is a unsigned and would be promoted to a number that
can segfault videoblend.
https://bugzilla.gnome.org/show_bug.cgi?id=738242
This was never reset when going from PAUSED->READY and resulted
in encoders being not reusable after EOS. They just rejected any
buffer because they received EOS in their previous life.
The flag wasn't used anywhere except for rejecting buffers after
EOS, and this is now handled by GstPad directly.
The spec mentions a version of the MPEG-2 frame with a base frame and
extension frame. I don't have IEC 13818-3 to figure out what that is,
and don't see any references in search results, so it's a FIXME for now.
https://bugzilla.gnome.org/show_bug.cgi?id=736797
Move the conversion code used in videoconvert to the video library
and expose a simple but generic API to do arbitrary conversion. It can
currently do colorspace conversion but the plan is to add videoscale to
it as well.
See https://bugzilla.gnome.org/show_bug.cgi?id=732415
When playing chained data the audio ringbuffer is released and
then acquired again. This makes it reset the segbase/segdone
variables, but the next sample will be scheduled to play in
the next position (right after the sample from the previous media)
and, as the segdone is at 0, the audiosink will wait the duration
of this previous media before it can write and play the new data.
What happens is this:
pointer at 0, write to 698-1564, diff 698, segtotal 20, segsize 1764, base 0
it will have to wait the length of 698 samples before being able to write.
In a regular sample playback it looks like:
pointer at 677, write to 696-1052, diff 19, segtotal 20, segsize 1764, base 0
In this case it will write to the next available position and it
doesn't need to wait or fill with silence.
This solution is borrowed from pulsesink that resets the clock to
start again from 0, which makes it reset the time_offset to the time
of the last played sample. This is used to correct the place of
writing in the ringbuffer to the new start (0 again)
https://bugzilla.gnome.org/show_bug.cgi?id=737055
Move the assert to the error handling block at the end of the function so the
the logging is still triggered. Reword the logging slightly and add another
comment to hint what went wrong.
Fixes#737138
Issue:
During a PAUSED->PLAYING transition when we are rendering an audio buffer in AudioBaseSink
we make adjustments to the sink's provided clock i.e. fix clock calibration using the external
pipeline clock, within "gst_audio_base_sink_sync_latency function inside gstaudiobasesink.c".
For the calibration adjustment we need to get the sink clock time using "gst_audio_clock_get_time".
But before calling "gst_audio_clock_get_time" we acquire the Object Lock on the Sink. If sink is
a pulsesink, "gst_audio_clock_get_time" internally calls "gst_pulsesink_get_time" which needs to
acquire Pulse Audio Main Loop Lock before querying Pulse Audio for its stream time using
"pa_stream_get_time". Please see "gst_pulsesink_get_time in pulsesink.c".
So the situation here is we have acquired the Object lock on Sink and need PA Main Loop Lock.
Now Pulse Audio Main Thread itself might be in the process of posting a stream status
message after Paused to Playing transition which in turn acquires the PA Main loop lock and
needs the Object Lock on Pulse Sink. This causes a deadlock with the earlier render thread.
Fix:
Do not acquire the object Lock on Sink before querying the time on PulseSink clock. This is
similar to the way we have used get_time at other places in the code. Acquire it after the
get_time call. This way PA Main loop will be able to post its stream status message by
acquiring the Sink Object lock and will eventually release its Main Loop lock needed for
gst_pulsesink_get_time to continue.
https://bugzilla.gnome.org/show_bug.cgi?id=736071
The timeout parameter is only allowed in a session response header
but some clients, like Honeywell VMS applications, send it as part
of the session request header. Ignore everything from the semicolon
to the end of the line when parsing session id.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736267
Fixes a crash when controlsrc, readsrc or writesrc are modified from
gst_rtsp_source_dispatch_read/write and gst_rtsp_watch_reset at the
same time.
https://bugzilla.gnome.org/show_bug.cgi?id=735569
This avoids a race where we would get new tag but we are already
prerolled and analyzing results.
It is the way it is supposed to be handled as stated in comment:
"If preroll is complete, drop these tags - the collected information is
possibly already being processed and adding more tags would be racy"
Reset last_timestamp_out when applying the output segment
change, to avoid decoder confusion over new timestamp timelines when
a seamless segment change happens.
Move some locks/unlocks to later when they're actually needed.
https://bugzilla.gnome.org/show_bug.cgi?id=734617
As was done for the base video decoder in commit 695675, don't
flush out the decoder on a new SEGMENT event. Segment events
may be a new segment, but are also often segment updates for
the current segment where the old data should be kept. For new
segments, a STREAM_START event will already trigger a drain, but
make sure to flush any remaining partial data then as well.
https://bugzilla.gnome.org/show_bug.cgi?id=734666
