The default behaviour of rtponviftimestamp is to drop buffers
outside the segment. This creates obvious problems for reverse
playback.
The ONVIF specification unfortunately doesn't describe how to handle
that specific use case, but we can expose a property to let the
user disable the dropping behaviour, and forward these buffers with
a G_MAXUINT64 ONVIF timestamp.
Also modify rtponvifparse to handle such timestamps appropriately.
We don't support negotiation with downstream but simply set caps based
on the buffers we receive. This prevents renegotiation to other formats,
and negotiation to NTSC in mode=auto in the beginning until the first
buffer is received.
As side-effect of this, also remove various other caps handling code
that was working around the behaviour of the default
BaseSrc::negotiate().
We reject caps with other framerates as it's impossible to generate
timecodes unless we actually know a constant framerate. Reflect this
also in the pad template caps.
During GstVideoInfo conversion from GstCaps, interlace-mode is
inferred to progressive so unspecified interlace-mode should not cause any
negotiation issue. Simly set GST_PAD_FLAG_ACCEPT_INTERSECT flag
on sinkpad to fix issue.
Encoded bitstream might not have valid framerate. If upstream
provided non-variable-framerate (i.e., fps_n > 0 and fps_d > 0)
use upstream framerate instead of parsed one.
Instead of using a static hardcoded PCR interval, allow the user
to configure it.
Also revert back the default to a 40 ms interval, that was changed
in recent patches for no good reason.
x265 does not allow user to configure a picture size smaller than
at least one CU size, and maxCUSize must be 16, 32, or 64.
Therefore, the CU size must be set according to the input resolution,
and the input resolution can not be less than 16.
3-byte emulation bytes can confuse the current code that skips
bits at the end of an SEI. Use a simpler method that's also
quicker because it skips all remaining bits in one go instead
of 1 bit at a time.
Encoding thread is terminated without any notification so
upstream streaming thread is locked because there is nothing
to pop from GAsyncQueue. If downstream returns error,
we need put SHUTDOWN_COOKIE to GAsyncQueue for chain function
can wakeup.
When negotiating a data channel, Chrome as recent as 75 still uses SDP
based on version 05 of the SCTP SDP draft, for example:
m=application 9 DTLS/SCTP 5000
a=sctpmap:5000 webrtc-datachannel 1024
Implement support for parsing SCTP port out of SDP message with sctpmap
attribute. Fixes data channel negotiation with Chrome browser.
WebKit's websrc depends on the main-thread for download completion
rendezvous. This exposed a number of deadlocks in adaptivedemux due to
it holding the MANIFEST_LOCK during network requests, and also needing
to hold it to change_state and resolve queries, which frequently occur
during these download windows.
Make demux->running MT-safe so that it can be accessed without using the
MANIFEST_LOCK. In case a source is downloading and requires a MT-thread
notification for completion of the fragment download, a state change
during this download window will deadlock unless we cancel the downloads
and ensure they are not restarted before we finish the state-change.
Also make demux->priv->have_manifest MT-safe. A duration query happening
in the window described above can deadlock for the same reason. Other
src queries (like SEEKING) that happen in this window also could
deadlock, but I haven't hit this scenario.
Increase granularity of API_LOCK'ing in change_state as well. We need to
cancel downloads before trying to take this lock, since sink events
(EOS) will hold it before starting a fragment download.
The row based multi threading control was introduced after 1.0.0 version
of libaom released. It adds a guard to check the relevant control
definition declared. It fixes#1025
By adding system memory support for nvdec, both en/decoder
in the nvcodec plugin are able to be usable regardless of
OpenGL dependency. Besides, the direct use of system memory
might have less overhead than OpenGL memory depending on use cases.
(e.g., transcoding using S/W encoder)
There's no point in working with invalid LTC timestamps as all future
calculations will be wrong based on this, and invalid LTC timestamps can
sometimes be read via the audio input.
False warning from MSVC, or it does not understand that
g_assert_not_reached() does not return.
...\gst-plugins-bad-1.0-1.17.0.1\sys\decklink\gstdecklink.cpp(1647) : warning C4715: 'gst_decklink_configure_duplex_mode': not all control paths return a value
