This will prevent the converter to be picked automatically in case
someone implement dynamic converter selection support. I'd like this
to be ranked only for known device, as it's hard to be sure a device is
a converter suited for general purpose. Re-negotiation is also needed
before we can rank it.
https://bugzilla.gnome.org/show_bug.cgi?id=733607
Even though the UVC driver do a great deal of effort to prevent bad
timestamp to be sent to userspace, there still exist UVC hardware that
are so buggy that the timestamp endup nearly random. This code detect
and ignore timestamp from these drivers, making these camera usable.
This has been tested on both invalid and valid cameras, making sure it
does not trigger for valid cameras.
https://bugzilla.gnome.org/show_bug.cgi?id=732910
There is still around 18 drivers not yet ported to videobuf2. These driver
don't support freeing buffetrs through REQBUFS(0) hence for these the
memory type probing fails. In order to gain back our previous behaviour in
presence of these, we implement a workaround that assuming MMAP is
supported. Note that an allocator is only created for device with
STREAMING support in the device capabilities. In such case one of MMAP,
USERPTR and DMABUF is required. Though DMABUF came afterward, so is
not an option and in practice none of these drivers will only do USERPTR.
https://bugzilla.gnome.org/show_bug.cgi?id=735660
Also-by: Hans de Goede <hdegoede@redhat.com>
Since we can get the minimum number of buffers needed by an output
device to work, use it to set min_latency which will determine how many
buffers are queued.
https://bugzilla.gnome.org/show_bug.cgi?id=736072
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
In gst_gdk_pixbuf_dec_setup_pool(), query is being allocated using
gst_query_new_allocation(), but the same is not unreferenced
hence calling gst_query_unref() after usage of query.
https://bugzilla.gnome.org/show_bug.cgi?id=735950
They are reported properly by libvpx if the correct struct members are used.
This also fixes handling of resolution changes without input caps changes.
https://bugzilla.gnome.org/show_bug.cgi?id=719359
Adding an extra condition while calling gst_caps_unref (templ)
and replacing gst_caps_make_writable (gst_caps_ref (caps)) with
gst_caps_copy (caps) in line 177, since the functionality is same.
https://bugzilla.gnome.org/show_bug.cgi?id=735795
Most V4L2 ioctls like try_fmt will adjust input fields to match what the
hardware can do rather then returning -EINVAL. As is docmented here:
http://linuxtv.org/downloads/v4l-dvb-apis/vidioc-g-fmt.html
EINVAL is only returned if the buffer type field is invalid or not supported.
So upon requesting V4L2_FIELD_NONE devices which can only do interlaced
mode will change the field value to e.g. V4L2_FIELD_BOTTOM as only returning
half the lines is the closest they can do to progressive modes.
In essence this means that we've failed to get a (usable) progessive mode
and should fall back to interlaced mode.
This commit adds a check for having gotten a usable field value after the first
try_fmt, to force fallback to interlaced mode even if the try_fmt succeeded,
thereby fixing get_nearest_size failing on these devices.
https://bugzilla.gnome.org/show_bug.cgi?id=735660
They may have been modified by the ioctl even if it failed. This also makes
the S_FMT fallback path try progressive first, making it consistent with the
preferred TRY_FMT path.
https://bugzilla.gnome.org/show_bug.cgi?id=735660
We return EOS after the first buffer, and GstPad will make sure now that we
won't get any other buffer afterwards until a flush happens. No need to check
for it ourselves.
https://bugzilla.gnome.org/show_bug.cgi?id=735581
FLV documentation stipulates that timestamp must start at zero.
In order to respect this rule, keep the first timestamp around
and offset the timestamp from this value. This allow for longer
recording time in presence of timestamp that does not start
at 0 already.
https://bugzilla.gnome.org/show_bug.cgi?id=731352
The tags in FLV are DTS. In audio cases, and for many video format this makes
no difference, but for AVC with B-Frames, PTS need to be computed from
composition timestamp CTS, with PTS = DTS + CTS.
https://bugzilla.gnome.org/show_bug.cgi?id=731352
The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.
https://bugzilla.gnome.org/show_bug.cgi?id=734322
udpsrc gtk-doc documentation refers to sockfd and closefd properties which has
been removed. This patch replaces those references to socket and close-socket
respectively.
https://bugzilla.gnome.org/show_bug.cgi?id=734987
The old default timescale of 1 millisecond produces irrational
numbers for a lot of framerate/audio-packet-duration multiples.
1/1800 is a nicer number, as it tends to produce better fractions
and therefore slightly higher accuracy overall
Change the way the output framerate is calculated
to ignore the first sample (which is sometimes truncated
in my testing) and use the new gst_video_guess_framerate()
function to recognise common standard framerates better.
Remove the code that was sorting the first 20 sample
durations and then ignoring the result.
