Use atomic ops on pending flags. Rename the segment_pending to
new_segment_pending. Set new_segment_pending not when we received seek, but
when we received the first upstream new_segment.
When we don't have specific {audio|video|text}-sink properties, don't
set them on playsink when reconfiguring.
If we do that, we end up setting the previous configured sink to
GST_STATE_NULL resulting in any potentially pending push being returned
with GST_FLOW_WRONG_STATE which will cause the upstream elements to
silently stop.
https://bugzilla.gnome.org/show_bug.cgi?id=655279
When we have a multi-stream (i.e. audio and video) input and the demuxer
adds/removes pads for a new stream (common in a mpeg-ts stream when the
program stream mapping is updated), the algorithm for EOS handling was
previously wrong (it would only drop the EOS of the *last* pad but would
let the EOS on the other pads go through).
The logic has only been changed a tiny bit for EOS handling resulting in:
* If there is no next group, let the EOS go through
* If there is a next group, but not all pads are drained in the active
group, drop the EOS event
* If there is a next group and all pads are drained, then the ghostpads
will be removed and the EOS event will be dropped automatically.
This allows us to make parsers accept both parsed and unparsed input
without decodebin plugging them in a loop until things blow up, ie.
without affecting applications that still use the old playbin or the
old decodebin.
(Making parsers accept parsed input is useful for later when we want
to use parsers to convert the stream-format into something the decoder
can handle. It's also much more convenient for application authors
who can plug parsers unconditionally in transcoding pipelines, for
example).
Add a flags property and two flags to allow one to disable the
conversion elements within encodebin. Doing so insists that the
uncompressed input to encodebin for the appropriate stream type is
sufficient to meet the caps requirements of the encoders, muxers and
encodebin target.
This is mostly beneficial to bypass slow caps negotiations in the
conversion elements.
Caps returned from gst_pad_peer_get_caps_reffed () may not be writable.
If they are not is should cause an assertion in gst_caps_merge (),
however, sometimes assertions are disabled in binary builds of -base and
it's safer to just be sure the caps are writable. Also, check that the
reffed caps pointer is not NULL.
The length check isn't sufficient, an source might
report the correct length, but then still fail to
read the requested number of bytes for some reason.
https://bugzilla.gnome.org/show_bug.cgi?id=652642
This is especially needed when switching between a non-sparse and sparse
video stream, see bug #537382. It also lowers the time needed for switching
between streams a bit.
Previously we checked mute_csource to determine wheter we need to premultiply
volumes and mute values. That fails as we unrefs mute_csource and set it to
NULL after. Use an extra flag instead.
make_lossless_changes() returns the same structure that we're passing (probably
to enable chaining). Instead of reusing s and making it point to s2 as well,
keep using s2. Drop the assignment which in the 2nd case is a dead one anyway.
Autoplug formatters for streams if a formatter with secondary or
higher rank is found. Formatters are autoplugged when there is no
muxer or when the muxer doesn't implement the tagsetter interface.
Currently only the first formatter found is plugged, this might
help in lots of cases, but it doesn't solve the
'lamemp3 ! xingmux ! id3mux'
case.
https://bugzilla.gnome.org/show_bug.cgi?id=649841
In particular, in audio only cases whose (estimated) metadata provides bitrate
information, the buffer-size based on such bitrate (and buffer-duration)
will be much more reasonable than queue2 default buffer-size.
For streams at low bitrates we need to set a limit in time because the limit
in bytes might not reached too late, sometimes more than 30 seconds.
This limit can only be set if upstream is seekable (see #584104)
Closes#647769
These reconfigure based on the caps and plugin in converters if
necessary. This also makes switching between compressed and raw
streams work flawlessly without loosing the states of any element
somewhere or having running time problems.
Before playbin2 would use different selectors for raw audio and
compressed audio (and the same for video) and used different
pads from playsink. This made the involved logic much more
complex and was not implemented completely in playsink, which
made it impossible to support files with a compressed and
uncompressed stream that is support by the sink.
playbin2 handles raw/non-raw streams the same now and the
decision is left to playsink, which now can also handle
caps changes from raw to non-raw and the other way around.
Fixes bug #632788.
Fixes#648548. Orc generates bad code for
gst_videoscale_orc_resample_merge_bilinear_u32, so we'll use the
slightly slower two-stage process. I'd fix Orc, but it's hard to
get excited about fixing a feature that I'm planning to deprecate
and replace.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
We should keep playlist/m3u8 available for normal m3u8 playlists,
which we we'll likely support some day. Also, we probably don't
want this handled like other playlists, so application/* seems
more appropriate in this case, even if it's really just a playlist.
In addition to ensuring that an element we want to select in
autoplug-select can enter the READY state, we also now check if it can
accept the caps we wish to plug it for. This is handy for sinks that
need to perform a probe to figure out whether they can actually handle a
given format.
When fixating caps, from_par should always be initialized
with a fixed value.
In case the fixation is from src to sink pad it was setting
the from par (srcpad par) to a fraction range, this patch initializes
it to 1/1, based on the assumption that missing PAR is 1/1.
https://bugzilla.gnome.org/show_bug.cgi?id=641952
Post better error messages in case typefind/decodebin2 are missing or
could not be loaded for some reason (e.g. because they inadvertently
got blacklisted).
https://bugzilla.gnome.org/show_bug.cgi?id=644892
In NULL/READY, we should be able to switch profiles on encodebin,
this patch makes it tear down old profiles when new ones are set
if in NULL/READY states
https://bugzilla.gnome.org/show_bug.cgi?id=644416
Clients are usually disconnected in the streaming thread if their inactivity
is bigger than the timeout. If no new buffers are to be rendered in the sink,
these clients will never be disconnected and for that reason it should be
handled in the select() loop too.
Parsers are the only element class that are not changing the data and
could lead to an infinite loop. Other element classes like demuxers,
e.g. id3demux, can be used multiple times in a row and sometimes are.
Previously we only checked against the raw caps but we should also
check against the return value of autoplug-continue. Additionally fix
a thread-safety issue with accessing the raw caps.
Add "source-setup" signal for convenience and discoverability. No need
to figure out "notify::source", look up the notify callback signature,
then do an g_object_get() to get the source element..
https://bugzilla.gnome.org/show_bug.cgi?id=626152
As a result, pipelines that contain multiple instances of audiotestsrc
with the 'wave' property set to 'white-noise', 'pink-noise', or
'gaussian-noise' will run much faster, since they won't be competing
for access to the global, lock-protected instance of GRand.
Fixes bug #642720.
...instead of copying the array. Returning NULL will result
in the original factories array to be used and prevents a useless
array copy in most use cases.
...instead of copying the array. Returning NULL will result
in the original factories array to be used and prevents a useless
array copy in most use cases.
Add notes about the behaviour if multiple signal handlers are connected.
For most autoplug-* signals only the first signal handler will ever
be invoked.
Also add to the autoplug-sort docs that the signal handler can return NULL
to specify that the order should change and other handlers get the chance
to sort the array.
This lock is taken when activating a group, which could result in
calling the autoplug-continue callback, which also needs this lock
to access the sinks.
See bug #642174.
Don't build merge the caps of all sinks but check them one-by-one
until one supports the caps. Also get reffed caps from the sinkpads
instead of a writable copy and add debug output if a sink claims to
support ANY caps.
The outgoing buffer timestamp is calculated by scaling an output buffer
count by the src pad frame rate caps. If these caps change, we need to
reset the count and work from a new base timestamp. The new output
buffer timestamp is then the count scaled by the new caps values added
onto the base timestamp.
with i686-apple-darwin10-gcc-4.2.1:
encoding-profile.h:134: warning: type qualifiers ignored on function return type
encoding-profile.c:240: warning: type qualifiers ignored on function return type
gstencodebin.c: In function 'next_unused_stream_profile':
gstencodebin.c:454: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
gstencodebin.c:464: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
Since we calculate timestamps by:
timestamp = t0 + (out samples) / (out rate)
and durations by:
duration = ((out samples) + (processed samples)) / (out rate) - timestamp
if t0 is nonzero, this would simplify to
duration = t0 + (processed samples) / (out rate).
This duration is too large by the amount t0. We should have done:
duration = t0 + ((out samples) + (processed samples)) / (out rate) - timestamp
so that
duration = (processed samples) / (out rate).
Frame size is given in words; it is already multiplied by two where
needed, so the left shift is superfluous. This extra multiplication
caused the code to inspect the third packet instead of the second,
which would fail for files where the second packet has a size
different from the first.
Some things aren't quite right yet and cause problems (0-sized buffers
with PREROLL flag set cause crashes in elements that don't expect those;
getting pipeline back to preroll/playing again when audio/video streams
have different lengths and a seek past the end of one of the stream
happens doesn't always work, etc.). Needs further investigation in the
next cycle.
https://bugzilla.gnome.org/show_bug.cgi?id=633700https://bugzilla.gnome.org/show_bug.cgi?id=634699
Fix conversions to IYU1, they allocated infinite amounts of memory before
because no conversion to IYU1 was actually implemented and it was running
into an infinite loop trying to find suitable intermediate formats.
Also fix the stride and sizes used for IYU1.
Fix a bug when reconfiguring the playsink where the subpicture
stream is broken by attempting to connect it through
streamsynchroniser and second time.
Going over integer arithmetic will lead to minimal rounding errors,
leading to +/-1 changes for volume==1.0. Implement the controlled
processing with floating point arithmetic, which was already done
for the C versions anyway.
