... so subclasses can release a frame all the way (also from frame list)
without having to pass through _finish_frame or _drop_frame.
The latter may not be applicable, or may or may not have already
been called for the frame in question.
See https://bugzilla.gnome.org/show_bug.cgi?id=693772
allows configuration of whether GstVideoGLTextureUploadMeta is
added to buffers resulting from a buffer pool. This is sperate
to the caps feature in that an element may want to add the upload
meta itself rather than allowing the buffer pool to.
https://bugzilla.gnome.org/show_bug.cgi?id=712798
Raise an error in case no frames are decoded before EOS and we
have input, meaning that data was received but it was somehow invalid.
Based on the videodecoder change, merged here for consistency.
https://bugzilla.gnome.org/show_bug.cgi?id=711094
Allows using -1 to make audiodecoder never post an error message
after decoding errors.
Based on the videodecoder change, merged here for consistency.
https://bugzilla.gnome.org/show_bug.cgi?id=711094
We could have allocation query before caps event and even without caps inside
the query. In such cases , the downstream can return a bufferpool object with
out actually configuring it. This feature is helpful to negotiate the bufferpool
with out knowing the output video format. For eg: some hardware accelerated
decoders can interpret the o/p video format only after it finishes the decoding
of one buffer at least.
https://bugzilla.gnome.org/show_bug.cgi?id=687183
Accumulate buffers in an adapter instead of appending them because append causes
a lot of memcpys.
Keep track of the last tagsize and accumulate enough data before attempting to
parse more data.
This patch implements a minimal amount of changes in order to not change the
behaviour. We should really rewrite the tag handling and trimming using
the adapter API instead of merging and trimming into a buffer.
Added new functions gst_rtsp_connection_set_tls_validation_flags() to
allow setting the TLS certificate validation flags when establishing a
TLS connection.
A getter is also available, gst_rtsp_connection_get_tls_validation_flags().
https://bugzilla.gnome.org/show_bug.cgi?id=711231
gst_audio_ring_buffer_set_channel_positions() checks whether the given
positions are identical with the current setup and returns
immediately if so. But it also clears need_reorder flag before this
comparison, thus this flag might be wrongly cleared if the function is
called twice with the same channel positions.
Move the flag clearance after the check.
https://bugzilla.gnome.org/show_bug.cgi?id=709754
We're checking the caps to see if we got more caps details after a parser got
plugged. This will also have a flipped 'parsed' field. If the field was already
present before the parse the match will fail. Add a function that will do the
check while excluding this field.
Creating a GSource and not attaching it to a context will cause
a leak of it's child sources. That is why we create writesrc right
before attaching it to a context.
https://bugzilla.gnome.org/show_bug.cgi?id=708667
Makes it easier to track how many users there are
Also make it possible to create a dmabuf struct on systems without mmap,
it just won't be possible to map it.
https://bugzilla.gnome.org/show_bug.cgi?id=707793
The payload type can't be between 72 and 76 because with the marker bit set,
this could be mistaken for an RTCP packet then. We do a relaxed check and
only refuse 72-76 when the marker bit is set. The effect is that when
we try to map an RTCP packet as an RTP packet, we will certainly fail.
