This makes gstconfig.h completely arch-independent. Should cover all
compilers that gstreamer is known to build on, and all architectures
that I could find information on. People are encouraged to file bugs if
their platform/arch is missing.
In ringbuffer mode we need to make sure we post buffering messages *before*
blocking to wait for data to be drained.
Without this, we would end up in situations like this:
* pipeline is pre-rolling
* Downstream demuxer/decoder has pushed data to all sinks, and demuxer thread
is blocking downstream (i.e. not pulling from upstream/queue2).
* Therefore pipeline has pre-rolled ...
* ... but queue2 hasn't filled up yet, therefore the application waits for
the buffering 100% messages before setting the pipeline to PLAYING
* But queue2 can't post that message, since the 100% message will be posted
*after* there is room available for that last buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=769802
A new event which precedes EOS in situations where we
need downstream to unblock any pads waiting on a stream
before we can send EOS. E.g, decodebin draining a chain
so it can switch pads.
https://bugzilla.gnome.org/show_bug.cgi?id=768995
Redirection messages are already used in fragmented sources and in
uridecodebin, so it makes sense to introduce these as an official message
type.
https://bugzilla.gnome.org/show_bug.cgi?id=631673
Other pads that are waiting for the stream on the selected
pad to advance before they finish waiting themselves
should be given the chance to do so when the selected pad
goes EOS. Fixes problems where input streams can end up
waiting forever if the active stream goes EOS earlier than
their own end time.
In some corner cases, the error 'code' part passed to
GST_ELEMENT_ERROR() is a valid define as well, in which
case it won't survive two levels of macro expansion, but
only one. Fixes:
oss4-sink.c: In function ‘gst_oss4_sink_open’:
error: ‘GST_RESOURCE_ERROR_0x00000002’ undeclared (first use in this function)
GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__,
which is from GST_ELEMENT_ERROR(el,RESOURCE,OPEN_WRITE,..)
and OPEN_WRITE happens to be defined to 2 here.
https://bugzilla.gnome.org/show_bug.cgi?id=756806https://bugzilla.gnome.org/show_bug.cgi?id=769117
gst_structure_id_get() returns a new reference so the returned object is
actually (transfer full).
The unit tests was already unreffing the objects.
https://bugzilla.gnome.org/show_bug.cgi?id=768776
gst_structure_id_get() returns a new reference so the returned device is
actually (transfer full).
The code using this API was already correct but the code example in
comments was not.
https://bugzilla.gnome.org/show_bug.cgi?id=768776
If segment.stop was given, and the subclass provides a size that might be
smaller than segment.stop and also smaller than the actual size, we would
already stop there.
Instead try reading up to segment.stop, the goal is to ignore the (possibly
inaccurate) size the subclass gives and finish until segment.stop or when the
subclass tells us to stop.
When dealing with small-ish input data coming into queue2, such as
adaptivedemux fragments, we would never take into account the last
<200ms of data coming in.
The problem is that usually on TCP connection the download rate
gradually increases (i.e. the rate is lower at the beginning of a
download than it is later on). Combined with small download time (less
than a second) we would end up with a computed average input rate
which was sometimes up to 30-50% off from the *actual* average input
rate for that fragment.
In order to fix this, force the average input rate calculation when
we receive an EOS so that we take into account that final window
of data.
https://bugzilla.gnome.org/show_bug.cgi?id=768649
We don't free this from gst_deinit() but from gst_task_cleanup_all(),
so more GStreamer API may be called. In particular makes unit tests
work again with CK_FORK=no.
https://bugzilla.gnome.org/show_bug.cgi?id=768577
This ensures that all async operations (started from gst_element_call_async())
have been completed and so there is no extra thread running.
Fix races when checking for leaks on unit tests as some of those
operations were still running when the leaks tracer was checking for
leaked objects.
https://bugzilla.gnome.org/show_bug.cgi?id=768577
Waiting before posting calculated bitrates seems to be the
intent of the code, so avoid adding them to the tag list
pushed with the first frame.
When the threshold is reached, gst_base_parse_update_bitrates
sets tags_changed, so this posts the calculated ones right
that moment.
This prevents an insane average calculated from just the
first (key) frame from getting posted.
https://bugzilla.gnome.org/show_bug.cgi?id=768439
There must be a SEGMENT event before the GAP event, and SEGMENT events must
come after any CAPS event. We however did not produce any CAPS yet, so we need
to ensure to insert the CAPS event before the SEGMENT event into the pending
events list.
https://bugzilla.gnome.org/show_bug.cgi?id=766970
gcc 6 has problems detecting and avoiding throwing
a warning for tautological compares in macros (they
should only trigger for compares outside macros).
Avoid them with a nasty cast of one parameter to void *
https://bugzilla.gnome.org/show_bug.cgi?id=764526
This is an update on c9b6848885
multiqueue: Fix not-linked pad handling at EOS
While that commit did fix the behaviour if upstream sent a GST_EVENT_EOS,
it would break the same issue when *downstream* returns GST_FLOW_EOS
(which can happen for example when downstream decoders receive data
from after the segment stop).
GST_PAD_IS_EOS() is only TRUE when a GST_EVENT_EOS has flown through it
and not when a GST_EVENT_EOS has gone through it.
In order to handle both cases, also take into account the last flow
return.
https://bugzilla.gnome.org/show_bug.cgi?id=763770
When syncing by running time, multiqueue will throttle unlinked streams
based on a global "high-time" and the pending "next_time" of a stream.
The idea is that we don't want unlinked streams to be "behind" the global
running time of linked streams, so that if/when they get linked (like when
switching tracks) decoding/playback can resume from the same position as
the other streams.
The problem is that it assumes elements downstream will have a more or less
equal buffering/latency ... which isn't the case for streams of different
type. Video decoders tend to have higher latency (and therefore consume more
from upstream to output a given decoded frame) compared to audio ones, resulting
in the computed "high_time" being at the position of the video stream,
much further than the audio streams.
This means the unlinked audio streams end up being quite a bit after the linked
audio streams, resulting in gaps when switching streams.
In order to mitigate this issue, this patch adds a new "group-id" pad property
which allows users to "group" streams together. Calculating the high-time will
now be done not only globally, but also per group. This ensures that within
a given group unlinked streams will be throttled by that group's high-time
instead.
This fixes gaps when switching downstream elements (like switching audio tracks).
Especially if multiple threads are waiting for buffers to be available again,
the current code was wrong. Fix this and document clearly how the GstPoll is
supposed to be used.
Also fix some potential races with reading from the GstPoll before writing
actually happened.
https://bugzilla.gnome.org/show_bug.cgi?id=767979
