The videoframe-audiolevel element acts like a synchronized audio/video "level"
element. For each video frame, it posts a level-style message containing the
RMS value of the corresponding audio frames. This element needs both video and
audio to pass through it. Furthermore, it needs a queue after its video
source.
https://bugzilla.gnome.org/show_bug.cgi?id=748259
0x04 signifies a MPEG elementary stream but according to RP2008, 0x10 should
be used for a h264 byte-stream. This also fixes compatibility of our files
with ffmpeg.
If packet->payload_unit_start_indicator is true and pointer 0, there is no
discontinuity check. Therefore there could be a previous section not complete
that need to be cleared.
https://bugzilla.gnome.org/show_bug.cgi?id=758010
The values of channel_mapping are copied by gst_codec_utils_opus_create_caps ()
but it doesn't free or take ownership of the g_new0 allocated memory. This
needs to be freed before going out of scope.
CID 1338692
buf surely isn't NULL inside the block conditional to a buffer size bigger
than (G_MAXUINT16 - 3). Plus gst_buffer_unref() checks if the buffer is
NULL and does nothing if it is.
CID 1338693
If tsdemux never receives data for a stream, the corresponding pad will never
be added and stream->active will remain FALSE. When the stream is removed, the
pad will not be unreffed and will be leaked.
https://bugzilla.gnome.org/show_bug.cgi?id=757873
The current implementation for detecting the resolution changes
on key frames is based on vp8 bitstream alignment. Avoid this
width and height parsing for vp9 bitstream, which requires proper
frame header parsing inorder to detect the resolution change (Fixme).
https://bugzilla.gnome.org/show_bug.cgi?id=757825
It is up to the element handling the seek to send flush events
downstream, otherwise we end up with a situation where upstream
would get unexpected GST_FLOW_FLUSHING
The Onvif Streaming Specification specifies that the NTP timestamps
in the Onvif extension header indicaes the absolute UTC time associated
with the access unit. But by using running time we can not achieve that,
since a frame's running time depends on the played interval, whether a
non-flushing is done, etc. Instead we have to use the stream time.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
It is now possible to update the currently used ntp-offset with a
custom serialized downstream event. The element will read the ntp-offset
property when doing the state transition from READY to PAUSED and
use that offset until it receives a "GstNtpOffset" event, which also
has a "ntp-offset" attribute in that it's structure. In case the
property is not set and no event has been received, the element will
guess the npt-offset with help of the clock. If no clock can be
retrieved, the element will error out and stop the data flow.
The same event is also used for updating the D/E-bits in the RTP
extension header. The discont flag in a buffer can be set whenver a
live/network source looses a frame, but that is not the type of
discontinuity that the onvif extension header should reflect. The
header is mainly used for playback of a track concept, in which
gaps can be present, and it's those kind of gaps that should be
highlighted with the D- and E-bits.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
If a buffer or a buffer list is cached, no events serialized with the
data stream should get through. The cached buffers and events should
be purged when we stop flushing.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
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
String parameters are expected to be passed as (f0r_param_string *),
which actually map to char**. In the filters this is evaluated as
(*(char**)param) which currently lead to crash when passing char*.
Remove the special case for string, all types, including char* as
passed as a reference.
https://phabricator.freedesktop.org/T83
Check if downstream is seekable via a SEEKING query and output a
BYTE segment if we want to seek back to fix up the headers later,
but if we're streaming send a TIME segment instead (which goes
down better with e.g. asfmux ! rtpasfpay).
https://bugzilla.gnome.org/show_bug.cgi?id=719553
Some video bitstreams report a too restrictive set of profiles. If a video
decoder was to strictly follow the indicated profile, it wouldn't support that
stream, whereas it could in theory and in practice. So we should relax the
profile restriction for allowing the decoder to get connected with parser.
https://bugzilla.gnome.org/show_bug.cgi?id=747613
In the case where you have a source giving the GstAggregator smaller
buffers than it uses, when it reaches a timeout, it will consume the
first buffer, then try to read another buffer for the pad. If the
previous element is not fast enough, it may get the next buffer even
though it may be queued just before. To prevent that race, the easiest
solution is to move the queue inside the GstAggregatorPad itself. It
also means that there is no need for strange code cause by increasing
the min latency without increasing the max latency proportionally.
This also means queuing the synchronized events and possibly acting
on them on the src task.
https://bugzilla.gnome.org/show_bug.cgi?id=745768
VPS is not mandatory, and need not check for its presence before setting
the caps. Because of the check, in streams which don't have VPS,
sticky event mishandling happens.
https://bugzilla.gnome.org/show_bug.cgi?id=752807
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
