Some h264 payloaders are unfortunately buggy and don't correctly set the
E bit in FU-A NAL when they have ended. Work around this by assuming
such a fragmentation unit has ended when there was no packet loss and a
new NAL is started
When pushing out buffers over S/PDIF or HDMI, IEC 61937 payloading
requires each buffer to contain 6 blocks from each substream. This adds
code to collect all the frames needed to meet this requirement before
pushing out a buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=650313
Using the current RTCP interval to timeout SSRC collision can lead to
collisions being timed out immediately if a BYE packet is sent because
it is sent immediately, so the interval is 0. This is not what we
want. So just set a static 10 times the default RTCP interval, it
should be enough
https://bugzilla.gnome.org/show_bug.cgi?id=648642
... which is particularly needed when merging NAL units, where not resetting
would lead to output of an older (pre-flush) AU (with unintended timestamp).
Current matroska demux calculates the pixel aspect ratio only if both
DisplayHeight and DisplayWidth are set, but it is legal to use only
one variable if the other is equal to PixelWidth or PixelHeight, at
least the mkclean utility is doing that. So this makse mkcleaned
files play correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=654744
A missing sys/param.h include results in:
/usr/include/sys/proc.h:64: error: 'MAXLOGNAME' undeclared here (not in a
function)
/usr/include/sys/proc.h:285: error: 'MAXCOMLEN' undeclared here (not in a
function)
when compiling goom on openbsd/ppc. We can just remove the two sys/ includes
here, they are not needed for anything.
https://bugzilla.gnome.org/show_bug.cgi?id=654749
... introduced when shuffling around code for the async implementation
by setting state of source (and udp sources) in _play before downstream
flushing is undone.
The gst_base_parse_set_frame_rate call was predicated on a change to
sample rate, duration or profile. However, the block count per frame can
also change between packets, which would result in incorrect buffer
durations.
Pretending to handle 8-bit signed causes distorted audio when
actually given such audio, which you will get if passing 8-bit
unsigned through audioconvert ! audioresample, as audioresample
only handles 8-bit signed. Fixes#605834.
Signed-off-by: David Schleef <ds@schleef.org>
Some video frames, for example alt-ref frame in VP8, will be
never displayed. This is why it has duration=0.
This patch allow to use this duration.
Bug: 654175
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the
deferred call to be run before returning. This causes a race when
READY->NULL is executed shortly after, which stops the mainloop. This
leaks the element reference which is passed as userdata for the callback
(introduced in commit 7cf996, bug #614765).
The correct fix is to wait in READY->NULL for all outstanding calls to
be fired (since libpulse doesn't provide a DestroyNotify for the
userdata). We get rid of the reference passing from 7cf996 altogether,
since finalization from the callback would anyways lead to a deadlock.
Re-fixes bug #614765.
So that pulsesrc/pulsesink get chosen over other possible PRIMARY
src/sinks by autoaudiosink. Presumably, if pulse is available, it
is always preferred over another src/sink.
Fixes: #647540.
For some reason, in code dating to 2001, encoded jpeg buffers were
rounded up to multiples of 4 bytes. With the added bonus that the
extra bytes are unwritten, causing valgrind issues. Oops. I can't
think of any reason why JPEG buffers need to be multiples of 4 bytes,
so I removed the padding. There might be some code somewhere that
depends on this behavior, so if this needs to be reverted, please fix
the valgrind issues.
This drops support fof PulseAudio versions prior to 0.9.16, which was
released about 1.5 years ago. Testing with very old versions is not
feasible and we don't want to maintain 2 independent code-paths.
