For VP8 it's possible to signal width or height to be 0, but it does
not make sense to do so. For VP9 it's impossible. Hence, we most
likely have a corrupt stream. Trying to negotiate caps downstream with
either width or height as 0 will fail with something like
gst_video_decoder_negotiate_default: assertion 'GST_VIDEO_INFO_WIDTH (&state->info) != 0' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/610>
If core is built as a subproject (e.g. as in gst-build), make sure to use
the gst-plugin-scanner from the built subproject. Without this, gstreamer
might accidentally use the gst-plugin-scanner from the install prefix if
that exists, which in turn might drag in gst library versions we didn't
mean to drag in. Those gst library versions might then be older than
what our current build needs, and might cause our newly-built plugins
to get blacklisted in the test registry because they rely on a symbol
that the wrongly-pulled in gst lib doesn't have.
This should fix running of unit tests in gst-build when invoking
meson test or ninja test from outside the devenv for the case where
there is an older or different-version gst-plugin-scanner installed
in the install prefix.
In case no gst-plugin-scanner is installed in the install prefix, this
will fix "GStreamer-WARNING: External plugin loader failed. This most
likely means that the plugin loader helper binary was not found or
could not be run. You might need to set the GST_PLUGIN_SCANNER
environment variable if your setup is unusual." warnings when running
the unit tests.
In the case where we find GStreamer core via pkg-config we use
a newly-added pkg-config var "pluginscannerdir" to get the right
directory. This has the benefit of working transparently for both
installed and uninstalled pkg-config files/setups.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/603>
The profiles and levels were applied to the common caps instead of the copy.
That had the side effect of setting profiles/level from one CODEC onto
another. Leaving to encoder not being registered or not-negotiated errors.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/599>
In some algorithms (like yadif), the Y plane has to be handled different
than the UV plane. Therefore, the planar_y functions are now called for
the Y plane, and the nv12/nv21 functions are handling only the UV/VU
planes respectively.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>
Switching the deinterlacing mode on-the-fly from disabled to
auto used to work, but was broken by commit #1f21747c some
years ago.
Force re-negotiation with downstream when the mode or
fields properties are changed, otherwise deinterlace
never switches out of the passthrough mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/584>
First of all get rid of the atomic seeking boolean, which was only ever
set and never read. Replace it with a flushing boolean that is used in
the loop function to distinguish no buffer because of flushing and no
buffer because of an error as otherwise we could end up in a
GST_FLOW_ERROR case during flushing.
Also only reset the state of imagefreeze in flush-stop when all
processing is stopped instead of doing it as part of flush-start.
And last, get a reference to the imagefreeze buffer in the loop function
in the very beginning and work from that as otherwise it could in theory
be replaced or set to NULL in the meantime as we release and re-take the
mutex a couple of times during the loop function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/580>
Backwards timestamps confuse librtmp, even if they're only backwards
relative to the other stream. If the timestamp of a stream is going
backwards related to the other stream, this property allows the muxer to
skip a few buffers until it reaches the timestamp of the other stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/572>
Allows us to request pads after writing header for streamable flv's.
For non-streamable it doesn't make sense to request a new pad after
writing the header, because the headers have been written already and we
can't add the new stream. But for streamable, any clients that connect
after the new pad has been added will be able to see both streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/572>
As we override the GLib item with our own structure, we cannot use any
function from GList or GQueue that would try to free the RTPJitterBufferItem.
In this patch, we move away from g_queue_new() which forces using
g_queue_free(). This this function could use g_slice_free() if there is any items
left in the queue. Passing the wrong size to GSLice may cause data corruption
and crash.
A better approach would be to use a proper intrusive linked list
implementation but that's left as an exercise for the next person
running into crashes caused by this.
Be ware that this regression was introduced 6 years ago in the following
commit [0], the call to flush() looked useless, as there was a g_queue_free()
afterward.
Signed-off-by: Nicolas Dufresne <nicolas.dufresne@collabora.com>
[0] 479c7642fd
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/573>
The calculated threshold for timecode might be varying depending on
"max-size-timecode" and framerate.
For instance, with framerate 29.97 (30000/1001) and
"max-size-timecode=00:02:00;02", every fragment will have identical
number of frames 3598. However, when "max-size-timecode=00:02:00;00",
calculated next keyframe via gst_video_time_code_add_interval()
can be different per fragment, but this is the nature of timecode.
To compensate such timecode drift, we should keep track of expected
timecode of next fragment based on observed timecode.
The problem was this:
Due to the highly irregular arrival of RTX-packet the max-misorder variable
could be pushed very low. (-10).
If you then at some point get a big in the sequence-numbers (62 in the
test) you end up sending RTX-requests for some of those packets, and then
if the sender answers those requests, you are going to get a bunch of
RTX-packets arriving. (-13 and then 5 more packets in the test)
Now, if max-misorder is pushed very low at this point, these RTX-packets
will trigger the handle_big_gap_buffer() logic, and because they arriving
so neatly in order, (as they would, since they have been requested like
that), the gst_rtp_jitter_buffer_reset() will be called, and two things
will happen:
1. priv->next_seqnum will be set to the first RTX packet
2. the 5 RTX-packet will be pushed into the chain() function
However, at this point, these RTX-packets are no longer valid, the
jitterbuffer has already pushed lost-events for these, so they will now
be dropped on the floor, and never make it to the waiting loop-function.
And, since we now have a priv->next_seqnum that will never arrive
in the loop-function, the jitterbuffer is now stalled forever, and will
not push out another buffer.
The proposed fixes:
1. Don't use RTX in calculation of the packet-rate.
2. Don't use RTX in large-gap logic, as they are likely to be dropped.
This patch will now set the maximum of buffers to 32, allowing to grow the
pool for drivers that supports that and will respect the minimum buffers
reported by the driver. This was made to fix a stall with the virtio CODEC
driver.
Fixes#672
If the start of the GOP is >= the requested running time, put it into a
new fragment. That is, split-at-running-time would always ensure that a
split happens as early as possible after the given running time.
Previously it was comparing against the current incoming timestamp,
which does not tell us what we actually want to know as it has no direct
relation to the GOP start/end.
