Bringing back the check removed in the previous commit but have that check be a
g_assert. Changing the function to static void since return can never be False,
because audio format will never be unkown.
func_index is set by the sum of three ternary operators which add, 0:4, 0:2,
and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7.
The conditional checking if func_index is >= 0 and < 8 will always be true.
Removing it.
CID 1226442
We (currently?) can't really handle gaps between RTP packets if they're not
properly timestamped. The current code would go into calculations with
GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
better to error out cleanly instead.
We set to PLAYING after we have configured the caps, otherwise we
might end up calling request_key (with SRTP) while caps are still
being configured, ending in a crash.
https://bugzilla.gnome.org/show_bug.cgi?id=740505
They should always be built, while the speex elements are not.
Need to check for a smaller number of buffers then (7->4) because
speexenc will add 3 header buffers while alawenc will just output
as many buffers as it receives as input.
https://bugzilla.gnome.org/show_bug.cgi?id=742098
basesrc assumes that we don't return a buffer if
something else than OK is returned. It will just
leak any buffer we might accidentially provide
here.
This can potentially happen during flushing.
Maybe fixes https://bugzilla.gnome.org/show_bug.cgi?id=741993
Actually copy the codec data instead of copying nothing
and then bombing out because there's no data.
Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=741783
Apparently linphone sends an invalid RTP packet as very
first packet. We want to ignore that instead of erroring
out (same for any other invalid packets really).
https://bugzilla.gnome.org/show_bug.cgi?id=741398
Set positioning-mode=pixels-absolute to allow positioning with
absolute coordinates, meaning negative x/y offsets will be
interpreted as being to the left/above the video frame instead
of being interpreted as relative to the right/bottom edge of
the video frame (which is a silly default, but that's how it is).
This means we can nicely slide images into and out of the frame,
see gdkpixbufoverlay-test.
https://bugzilla.gnome.org/show_bug.cgi?id=739566
After creating the ringbuffer we have to set the device on the ringbuffer as
it defaults to kAudioDeviceUnknown. At this point it can't have changed to
anything else yet and we don't have to notify about changes to the sink/src
"device" property. It's also not a good idea because GstAudioBaseSrc has the
object lock taken while the ringbuffer is created, which might cause a
deadlock if something calls back into the element from "notify::device".
Once the base class is done with the NULL_TO_READY state change, it has opened
the device via the ringbuffer and this might have chosen a different device.
Especially if we initially used kAudioDeviceUnknown. Also notify about this
property change as initially intended by this code.
Instead of constantly querying upstream, just cache the last duration,
and in the unlikelyness we might have gone over query again before
deciding we are EOS.
Cut 15% cpu off matroskademux streaming thread (srsly...)
This is meant to be so (https://wiki.xiph.org/MatroskaOpus - while
it is marked as a draft, this part was confirmed to be correct on
IRC), and allows one to determine whether a demuxed stream is
multistream or not, and thus set the multistream caps field
accordingly. In turn, this means downstream does not have to guess.
https://bugzilla.gnome.org/show_bug.cgi?id=740744
It's like rendering a buffer list, just with one buffer.
Has the added advantage that if there are multiple clients
we can send the buffer to all the clients in one go.
We unlock and re-lock the client lock while emitting the
removed signal, which causes inconsistencies in the client
list vs. the client counts. Instead, remove the client from
the list already before emitting the signal and put it into
a temporary list of clients to be removed. That way things
look consistent to the streaming thread, but signal callbacks
can still do things like get stats from removed clients.
Add prototype for a render_list() function that can use a
sendmmsg-style g_socket_send_messages() function once it lands
in GLib. We can use this infrastructure to send multiple buffers
made up by multiple memories to multiple clients in one go, which
drastically reduces the number of syscalls made when sending
high-bitrate video streams.
https://bugzilla.gnome.org/show_bug.cgi?id=732152
Use the refcount for memory management and keep track
of the number of duplicate clients in a separate
variable. This will be useful later, and means we
don't have to hold the OBJECT_LOCK all the time.
https://bugzilla.gnome.org/show_bug.cgi?id=732866
It looks like libv4l2 support for CREATE_BUF is incomplete. That
combine with existing bugs may lead to crash in GStreamer. These
check will make it robust by:
- Checking create buf index isn't an already in used index
- Checking that the index out of QUERYBUF matches the requested
index
Right now we try to be clever by detecting if device format have
changed or not, and skip setting format in this case. This is valid
behaviour with V4L2, but it's also very error prone. The rational
for not setting these all the time is for speed, though I can't
measure any noticeable gain on any HW I own. Also, until recently,
we where doing get/set on the format for each format we where
probing, making it near to impossible that the format would match.
This also fixes bug where we where skipping frame-rate setting if
format didn't change.
https://bugzilla.gnome.org/show_bug.cgi?id=740636
Since "basetransform: Fix caps equality check" commit a7f357,
set_info() will not be called anymore if crop didn't change
the caps. This is fixed by setting "need_update" boolean when
cropping properties has been changed, and then applying these
if they where not applied before rendering the next frame. This
patch also fixed the locking, dropping un-needed custom lock,
and no holding needless lock while doing the operation as we
already hold the streaming lock.
https://bugzilla.gnome.org/show_bug.cgi?id=740787
It's unlikely that setting a channel layout will do much for AC3/DTS
streams. If we find at some point that it does make sense, we can
perform the structure copying unconditionally (i.e., the current code is
wrong, since AC3/DTS will get two structures now - one with the channel
layout, one without).
https://bugzilla.gnome.org/show_bug.cgi?id=740987
