Otherwise those symbols can conflict with external libraries when
linking everything statically for mobile targets.
Use the gst_gm_ prefix, short for gst geometric math.
https://bugzilla.gnome.org/show_bug.cgi?id=756882
Store and copy input state fields when setting the
output state of the decoder. Avoids problems like
the framerate set by an upstream element being ignored
Related to:
https://bugzilla.gnome.org/show_bug.cgi?id=756563
New subclass with a similar behaviour as the old liveadder, but
a slightly different API as the latency is in nanoseconds, not
milliseconds. Also, the new liveadder has a effective latency that
is latency + output-buffer-duration. In practice, just setting a non-zero
latency with the new audiomixer gives you the right behavior in 99% of the
cases.
Build error due to wrong argument type in debug message
aagg->priv->offset and next_offset are of type int64, but uint64
formatter is being used in logs. Changing all those to int64
https://bugzilla.gnome.org/show_bug.cgi?id=756065
When g_option_context_parse fails, context and error variables are not getting free'd
which results in memory leaks. Free'ing the same.
And replacing g_error_free with g_clear_error, which checks if the error being passed
is not NULL and sets the variable to NULL on free'ing.
https://bugzilla.gnome.org/show_bug.cgi?id=753854
The buffer timestamps in the collect function will already be
running time, don't try to convert them again to running time,
this would yield CLOCK_TIME_NONE now that the segment is shifted
to account for negative dts.
This fixes x264enc ! mpegtsmux ! hlssink, which was broken
because mpegtsmux would send a downstream key unit event with
running time NONE and then hlssink would immediately send
another one upstream and it would just be a flood of force
keyframe events in both directions after the first one. This
would then break hlssink because it uses multifilesink in
next-file=key-unit-event mode, and starting a new file after
every few kB does not work well for HLS.
The alsamidisrc element allows to get input event from ALSA MIDI
sequencer devices, and possibly convert them to sound using some
downstream element like fluiddec.
https://bugzilla.gnome.org/show_bug.cgi?id=738687
An IDX file or codec_data normally contains the original frame size of
the video. Allow upstream to provide this information by sending a
custom event, which will allow scaling the overlay correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=663750
Instead of only supporting writing SPU data directly to YUV frames,
render the SPU data to an intermediate AYUV overlay buffer. The overlay
data is then attached to the video frame if downstream supports overlay
composition, otherwise the AYUV overlay is blended to the video frame.
For the PGS format, the overlay buffer size is set to the size of the
Composition Window, and its position in the overlay composition is set
to the window position. The objects to render are now cropped when the
cropping flag is set.
For the Vobsub format, the overlay buffer size is set to the size of the
Display Area.
Once rendered, the overlay composition rectangle is now moved and scaled
to fit the video output size, to avoid clipping.
https://bugzilla.gnome.org/show_bug.cgi?id=663750
We might've queued up a GAP buffer that is only partially inside the current
output buffer (i.e. we received it too late!). In that case we should only
skip the part of the GAP buffer that is inside the current output buffer, not
also the remaining part. Otherwise we forward this pad too far into the future
and break synchronization.
Derive from GstVideoSink so that preroll frames will automatically
get rendered too, unless the show-preroll-frame property is set to
FALSE. Fixes intervideosrc only picking up frames if intervideosink
is in PLAYING state.
https://bugzilla.gnome.org/show_bug.cgi?id=755049
Fix the negotiation of GstVideoOverlayComposition by checking
intersection with the peer caps, rather than just accept-caps,
which might only check the pad template.
https://bugzilla.gnome.org/show_bug.cgi?id=755113
We have to queue buffers based on their running time, not based on
the segment position.
Also return running time from GstAggregator::get_next_time() instead of
a segment position, as required by the API.
Also only update the segment position after we pushed a buffer, otherwise
we're going to push down a segment event with the next position already.
https://bugzilla.gnome.org/show_bug.cgi?id=753196
x/y/w/h are signed integers. As can be seen in GstCompositorPad.
The prototype for clamp_rectangle was wrong. This commit reverts the change
and fixes the prototype.
This reverts commit bca444ea4a.
CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
number since it is an unsigned integer. Removing that check and only checking if
it is bigger than max by using MIN().
CID 1320707
As it's recursive, gst_pad_get_allowed_caps() may also return
empty for anything incompatible downstream. EMPTY is not valid caps
value for gst_caps_fixate(). This lead to assertion and then crash.
Ideally, the negotiate function should be re-factored to have a return
value, and we could make the negotiation fails earlier.
https://bugzilla.gnome.org/show_bug.cgi?id=754122
The obscured check in compositor was using the dimensions of the pad to clamp
the h/w of the pad instead of the output resolution, and was doing an incorrect
calculation to do so. Fix that by simplifying the whole calculation by using
corner coordinates. Also add a test for this bug which fell through the cracks,
and just skip all the obscured tests if the pad's alpha is 0.0.
https://bugzilla.gnome.org/show_bug.cgi?id=754107
The tsdemux latency should always be added to the minimum
latency (which is always a valid clock time value). The
"cleanup" in commit a1f709c2 made it so that it would not
be added if upstream reported 0 as minimum latency (as
e.g. udpsrc would). This broke playback of live mpeg-ts
streaming in some cases, leading to playback stutter due
to a too-small configured latency for the pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=751508
In gst_live_adder_chain() function, calls to gst_buffer_copy_region() can lead
to assertion as 'offset + size <= bufsize' is not respected.
Indeed 'offset' and 'size' parameters are calculated through calling gst_live_adder_length_from_duration(),
and thus gst_util_uint64_scale_int_round().
Depending on the nearest integers, rounded values 'offset' and 'size' can then trigger the assertion.
This case mainly occurs when 'skip' value is > 0 in chain function process.
https://bugzilla.gnome.org/show_bug.cgi?id=753759
It is faster than doing a query that propagates downstream and
should be enough
Elements: faac, gsmenc, opusenc, sbcenc, voamrwbenc, adpcmenc, sirenenc
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
A race condition in the state change function may cause buffers to be
unreffed while they are still used by the streaming thread in
gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the
parent class first in the state change function to make sure streaming
has stopped and only then free those buffers.
https://bugzilla.gnome.org/show_bug.cgi?id=741381
The SPS struct might be filled out by a call to
gst_h264_parser_parse_subset_sps, which fills out
dynamically allocated data and requires a call
to gst_h264_sps_clear() to free it. Also make sure
to clear out any allocated SPS data when returning
an error.
https://bugzilla.gnome.org/show_bug.cgi?id=753306
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
When playing mts files with embedded subtitles, the buffer is mapped,
but not unmapped at the end resulting in a memory leak.
Also unref event in handle_dvd_event as it takes ownership of the event.
https://bugzilla.gnome.org/show_bug.cgi?id=753539
When playing mts files with embedded subtitles, there are few event leaks.
Events are supposed to be transfer full. So if not forwarding the event,
they need to be freed.
https://bugzilla.gnome.org/show_bug.cgi?id=753539
h264parse and gstrtph264depay do the same, let's keep the behaviour
consistent. As we now include the codec_data inside the stream, this causes
less caps renegotiation.
https://bugzilla.gnome.org/show_bug.cgi?id=753228
rtph264depay does the same and this fixes decoding of some streams with 32
SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255),
but the field in the codec_data for the number of SPS or PPS is only 5
(or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
This looks like a mistake in the part of the spect about the codec_data.
This causes an assertion and would lead to getting a NULL instead
of a buffer. Without proper checking this would easily lead to a
segfault.
Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199
