GST_VIDEO_CONVERTER_OPT_ALPHA_MODE, GST_VIDEO_CONVERTER_OPT_CHROMA_MODE,
GST_VIDEO_CONVERTER_OPT_MATRIX_MODE, GST_VIDEO_CONVERTER_OPT_GAMMA_MODE and
GST_VIDEO_CONVERTER_OPT_PRIMARIES_MODE were G_TYPE_STRING with only a few valid
options. Changed those to real enums.
https://bugzilla.gnome.org/show_bug.cgi?id=749104
Upstream might want to use it to properly map timestamps to running/stream
times, if we just override it with 0 synchronization will be just wrong.
For this we remove some old 0.10 code related to segment accumulation, and
remove some more code that is useless now, and accumulate the group start time
(aka segment.base offset) manually now.
https://bugzilla.gnome.org/show_bug.cgi?id=635701
Embedded systems often have limited charset conversion
functionality, so don't rely on g_convert() (i.e. iconv)
for UTF-16 to UTF-8 conversions, we can easily enough do
that ourselves by converting to native endianness and
then using GLib's helper functions.
This is a fixup for b2db18cda2
audioconvert: avoid float calculations when mixing integer-formatted channels
The int matrix was using gint and gint32 synonymously, which can theoretically
cause problems if gint and gint32 are actually different types.
https://bugzilla.gnome.org/show_bug.cgi?id=747005
Try harder to look for gvfs backend changes in the right
place, to make sure the plugin gets reloaded when backends
are removed or installed. We watch the gvfs mounts directory
because the files there contain absolute paths to the
backend executables, and those may not be in the usual gio
path.
https://bugzilla.gnome.org/show_bug.cgi?id=747841
We don't expect clients to send us any data, but if they do, just
ignore it. Web browsers might send us an HTTP request for example,
but some will still be happy if we just send them data without
a proper HTTP response.
There was a bug in the reading code path. We only have a small
read buffer and would provoke an EWOULDBLOCK trying to read
because we don't bail out of the loop early enough.
https://bugzilla.gnome.org/show_bug.cgi?id=743834
When shutting down the chain, we can get a deadlock when removing
a pad, if that chain was being busy streaming but blocked (eg, while
waiting for a queue to have free space).
https://bugzilla.gnome.org/show_bug.cgi?id=746480
In case upstream does not provide videorate with framerate information,
it will detect the current framerate from the buffer it received,
but if downstream forces the use of variable framerate (most probably
through the use of a caps filter with framerate = 0 / 1), videorate will
respect that.
And add some unit tests
https://bugzilla.gnome.org/show_bug.cgi?id=734424
In the case the framerate is variable (represented by framerate=0/1),
we currently end up loop pushing the first buffer and then recompute
diff1 and diff2 without updating the videorate->next_ts at all
leading to infinitely looping pushing that first buffer.
In the case of variable framerate, we should just compute the next_ts
as previous_pts + previous_duration.
https://bugzilla.gnome.org/show_bug.cgi?id=734424
The patch calculates a second channel mixing matrix from the current one. The
matrix contains the original values * (2^10) as integers. This matrix is used
when integer-formatted channels are mixed.
On a ARM Cortex-A8, single core, 800MHz this improves performance in a
testcase from 29s to 9s for downmixing 6 channels to stereo.
https://bugzilla.gnome.org/show_bug.cgi?id=747005
If a new pad is added after playbin has been put to READY/NULL it
should ignore new pads as it is shutting down.
This can happen when the pipeline fails to preroll (is still in READY)
and the user gives up on waiting or an error that doesn't reach
the demuxer occurs (on some event handling) and it will continue to
work and exposing pads while playbin has been put to NULL.
Without this check an input-selector is created and set to PAUSED
state, preventing playbin from properly shutting down in case it
has data blocked inside it.
audio_convert_convert unpacks to default format (signed) before calling
quantize, and the unsigned variants were equivalent to signed anyway,
so we just get rid of them.
Since range size is always 2^n, we can simply use modulo (implemented
with a bitmask).
The previous implementation used 64-bit integer division, which is
done in software on ARMv7. Although the divisor was constant, the
division could not be transformed into "multiplication by magic number"
since the dividend was 64-bit.
The now-unused and not-so-fast gst_fast_random_(u)int32_range functions
were removed.
Also, implementing bug fixes:
1) ADD_DITHER_TPDF_HF_I no longer discards bias.
2) We change TPDF's noise range to be the same as RPDF's. Previously,
RPDF's noise ranged:
{ bias - dither, bias + dither }
while TPDF's noise ranged:
{ bias/2 - dither/2, bias/2 + dither/2 - 1 } +
{ bias/2 - dither/2, bias/2 + dither/2 - 1 } =
{ bias - dither, bias + dither - 2 }
Now, both range:
{ bias - dither, bias + dither - 1 }
https://bugzilla.gnome.org/show_bug.cgi?id=746661
This fixes a race where the use-buffering property on a multiqueue was
set before the queue depth was changed from it's high preroll limits to
lower playback limits. This resulted in buffering messages being emitted
by the multiqueue in the short window between use-buffering being
set and the queue depth being reset.
https://bugzilla.gnome.org/show_bug.cgi?id=744308
The variables could have changed when the lock was released
to push a gap event. Streamsynchronizer needs to check them
again before going to sleep.
Bonus: fix a comment typo
multisocketsink now understands the new GstNetControlMessageMeta to allow
sending control messages (ancillary data) with data when writing to Unix
domain sockets.
Thanks to glib's `GSocketControlMessage` abstraction the code introduced
in this commit is entirely portable and doesn't introduce and additional
dependencies or conditionally compiled code, even if it is unlikely to be
of much use on non-UNIX systems.
multisocketsink now understands the new GstNetControlMessageMeta to allow
sending control messages (ancillary data) with data when writing to Unix
domain sockets.
A later commit will introduce a new socketsrc element which will similarly
understand `GstNetControlMessageMeta`. This, when used with a
`GSocketControlMessage` of type `GUnixFDMessage` will allow GStreamer to
send and receive file-descriptions in ancillary data, the first step to
using memfds to implement zero-copy video IPC.
Thanks to glib's `GSocketControlMessage` abstraction the code introduced
in this commit is entirely portable and doesn't introduce and additional
dependencies or conditionally compiled code, even if it is unlikely to be
of much use on non-UNIX systems.
This provides notification that the socket in use was closed by the peer
and gives an opportunity to replace it with a new one which is not
closed, allowing reading from many sockets in order.
I use this in pulsevideo to implement reconnection logic to handle the
pulsevideo service dieing, such that is can be restarted without
disrupting downstream.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=739546
* Don't bother polling, just do a blocking read, the `GCancellable` will
take care of unlocking. This should also be faster on MS Windows where
the GIO documentation for `g_socket_get_available_bytes` states: "Note
that on Windows, this function is rather inefficient in the UDP case".
* Implement `GstPushSrc.fill` rather than `GstPushSrc.create`. This means
that we will be using the downstream allocator which may be more
efficient. It also means that socketsrc is likely to respect its
"blocksize" property (assuming that there is enough data available).
See https://bugzilla.gnome.org/show_bug.cgi?id=739546
`socketsrc` can be considered a source counterpart to `multisocketsink`.
It can be considered a generalization of `tcpclientsrc` and
`tcpserversrc`: it contains all the logic required to communicate over
the socket but none of the logic for creating the sockets/establishing
the connection in the first place, allowing the user to accomplish this
externally in whatever manner they wish making it applicable to other
types of sockets besides TCP.
This commit essentially copies the implementation directly from
tcpserversrc. Later patches will tidy the implementation up and
re-implement `tcpclientsrc` and `tcpserversrc` in terms of `socketsrc`.
See https://bugzilla.gnome.org/show_bug.cgi?id=739546
If a buffer is made up of non-contiguous `GstMemory`s `gst_buffer_map`
has to copy all the data into a new `GstMemory` which is contiguous. By
mapping all the `GstMemory`s individually and then using scatter-gather
IO we avoid this situation.
This is a preparatory step for adding support to multisocketsink for
sending file descriptors, where a GstBuffer may be made up of several
`GstMemory`s, some of which are backed by a memfd or file, but I think this
patch is valid and useful on its own.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=746150
When we modify a GList (via g_list_delete_link), always reassign the
new head to the original GList. Otherwise we end up with
filtered_errors being corrupt (the head might have been the element
removed)
This function is static, and only ever called with the expose lock
taken. It thus has no reason to take this lock itself.
This was introduced by one of my locking fixes from 741355.
https://bugzilla.gnome.org/show_bug.cgi?id=741355
Check if dbin->decode_chain is NULL before running drain_and_switch_chains()
because if it is, we shouldn't run that function or it will segfault.
CID #1271074
Otherwise if there are multiple parsers we would most likely break negotiation
of the stream-format/alignment wanted by the decoders as parsers generally
support all possible stream-formats and alignments.
