GstId3Mux sink pad is an always (static) pad. Thus releasing it
as if a request pad triggers:
(sound-juicer:11826): GStreamer-CRITICAL **:
gst_element_release_request_pad: assertion `GST_PAD_PAD_TEMPLATE (pad)
== NULL || GST_PAD_TEMPLATE_PRESENCE (GST_PAD_PAD_TEMPLATE (pad)) ==
GST_PAD_REQUEST' failed
https://bugzilla.gnome.org/show_bug.cgi?id=685110
Need to store the old running time and frame numbers when renegotiating and
start from 0 again when a new caps is set, preventing that framerate changes
cause timestamping issues.
For example, if a stream pushed 10 buffers on framerate=2/1, its
running time will be 5s. If a new framerate of 1/1 is set, it would
make the running time go to 10s as it would count those 10 buffers
as being sent on this new framerate.
Fixes camerbin unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=682973
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c: In function 'gst_audio_resample_dump_drain':
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c:729:9: warning: variable 'in_len' set but not used [-Wunused-but-set-variable]
streams with non-TIME segments will not have timestamps ...
... and therefore will never unblock the other streams.
Fixes blocking issue when using playbin suburi feature
People expect audiorate to fix things up and not make things worse
by default, so let's default to a similar tolerance as audiosinks
do. Should help with transcoding and the like, though one might
possible still want higher values then.
We can't just make a vfunc that takes a union of int
and pointer as argument, and then set up subclass-specific
action signals and signals that take int (in multifdsink's
case) or a GSocket * (in multisocketsink's case), and then
expect everything to Just Work. This blows up spectacularly
on PPC G4 for some reason.
Fixes multifdsink unit test on PPC, and fixes aborts in
multisocketunit test (now hangs in gst_pad_push - progress).
* Update outgoing segment.base with accumulated time, ensuring all
streams are synchronized.
* Only consider streams as "new" is they have a STREAM_START event
with a different seqnum.
* Use GstStream segment.base instead of separate variable to store
the past running time.
* Disable passthrough
* Switch to glib 2.32 GMutex/GCond
* Avoid getting pad parent the expensive way
* Minor other fixes
Make sure to send a CAPS event downstream when we get our
first input caps. This fixes not-negotiated errors and
adder use with downstream elements other than fakesink.
Even gst-launch-1.0 audiotestsrc ! adder ! pulsesink works now.
Also, flag the other sink pads as FIXED_CAPS when we receive
the first CAPS event on one of the sink pads (in addition to
setting those caps on the the sink pads), so that a caps query
will just return the fixed caps from now on.
There's still a race between other upstreams checking if
caps are accepted and sending a first buffer with possibly
different caps than the first caps we receive on some other
pad, but such is life.
Also need to take into account optional fields better/properly.
https://bugzilla.gnome.org/show_bug.cgi?id=679545
Fix invalid memory access caused by broken pointer arithmetic.
If we have a uint16_t *tmpbuf and add n * dest->stride to it, we
skip twice as much as we intended to because dest->stride is in
bytes and not in pixels. This made us write beyond the end of
our allocated temp buffer, and made the unit test crash.
