The loop in zoomFilterSetResolution is meant to change the values in the
zf->firedec[] array. Each iteration writes the value of decc onto the arrya,
but no conditions that change the value of decc are ever met and the array is
filled with zero for each element. Which is the initial state of the
array before the loop begins.
The loop does nothing.
https://bugzilla.gnome.org/show_bug.cgi?id=728353
We never initialize clock_rate explicitly, therefore it is 0 by default. The
parameter is a uint32 and the only caller ensure that it is >0, therefore it
won't become -1 ever.
In order to have a full mapping between channel positions in the audio
stream and loudspeaker positions, the channel-mask alone is not enough:
the channels must be interleaved following some Default Channel Ordering
as mentioned in the WAVEFORMATEXTENSIBLE[1] specification.
As a Default Channel Ordering use the one implied by
GstAudioChannelPosition which follows the ordering defined in SMPTE
2036-2-2008[2].
NOTE that the relative order in the Top Layer is not exactly the same as
the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users
using so may channels are already aware of such discrepancies.
[1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx
[2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf
Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127
Otherwise the CAPS event will be dropped and we never configure any caps at
all, leading to weird behaviour in many situations. Especially header
rewriting is not going to work if a capsfilter is after wavenc.
https://bugzilla.gnome.org/show_bug.cgi?id=737735
This is about converting the format, not about converting any widths and
heights. Subclasses are expected to handler different resolutions themselves,
like the videomixers already do properly.
gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
'GstRTSPResult' [-Werror,-Wenum-conversion]
res = gst_sdp_message_new (&sdp);
~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
'GstRTSPResult' [-Werror,-Wenum-conversion]
res = gst_sdp_message_parse_uri (uri, sdp);
~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Remove pads from flow combiner and reset last
flow return to FLOW_OK by resetting the flow combiner.
This prevents FLOW_FLUSHING when trying to re-use the
demuxer after setting it back to NULL/READY state.
https://bugzilla.gnome.org/show_bug.cgi?id=737359
DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later.
This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap)
https://bugzilla.gnome.org/show_bug.cgi?id=737095
In 1.0, we pass the complete caps to transform_caps to allow for better
optimizations. Make this function actually work on non-simple caps
instead of just ignoring the configured filter caps.
We have to skip 12 bytes of data for the chunk, and the data size
passed to the sub-chunk parsing functions should have 4 bytes less
than the data size.
Also when parsing the sub-chunks, check if we actually have enough
data to read instead of just crashing.
https://bugzilla.gnome.org/show_bug.cgi?id=736266
Drop use of g_socket_get_available_bytes() which is
not useful on all systems (where it returns the size
of the entire buffer not that of the next pending
packet), and is yet another syscall and apparently
very inefficient on Windows in the UDP case.
Instead, when reading UDP packets, use the more featureful
g_socket_receive_message() call that allows to read into
scattered memory, and allocate one memory chunk which is
likely to be large enough for a packet, while also providing
a larger allocated memory chunk just in case the packet
is larger than expected. If the received data fits into the
first chunk, we'll just add that to the buffer we return
and re-use the fallback buffer for next time, otherwise we
add both chunks to the buffer.
This reduces memory waste more reliably on systems where
get_available_bytes() doesn't work properly.
In a multimedia streaming scenario, incoming UDP packets
are almost never fragmented and thus almost always smaller
than the MTU size, which is also why we don't try to do
something smarter with more fallback memory chunks of
different sizes. The fallback scenario is just for when
someone built a broken sender pipeline (not using a
payloader or somesuch)
https://bugzilla.gnome.org/show_bug.cgi?id=610364
This makes sure that also properties like the pixel-aspect-ratio are the same
between both streams and that the output caps contain all fields necessary for
complete video caps.
https://bugzilla.gnome.org/show_bug.cgi?id=735804
gst_buffer_ref and gst_buffer_writable is being used to create a writable copy of source buffer.
replacing the same with gst_buffer_copy as the functionality is same.
https://bugzilla.gnome.org/show_bug.cgi?id=735880
