When the framesize is not specified, we try and calculate a size from
the strides and offset information. This was done with the sum of
offsets + the size of the last frame. That is just wrong method. We also
need to account for video meta that may be flipping two planes. An
example is if you convert I420 to YV12 by flipping the two last offsets.
https://bugzilla.gnome.org/show_bug.cgi?id=760270
To make parser work with image having non-standard strides, plane
offsets or with padding between images.
For now, since element doesn't check for videometa, we can't directly
push buffers when these properties are set so it convert the frame
in the pre_push_buffer method to remove any custom padding.
https://bugzilla.gnome.org/show_bug.cgi?id=760270
Allows the subclass to completely override the chosen src caps.
This is needed as videoaggregator generally has no idea exactly
what operation is being performed.
- Adds a fixate_caps vfunc for fixation
- Merges gst_video_aggregator_update_converters() into
gst_videoaggregator_update_src_caps() as we need some of its info
for proper caps handling.
- Pass the downstream caps to the update_caps vfunc
https://bugzilla.gnome.org/show_bug.cgi?id=756207
When sps data is NULL, the buffer allocated and mapped is not being freed.
In this scenario there is no need to allocate the buffer as we are supposed to return NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=761070
It's useful enough already to be used in other elements for audio aggregation,
let's give people the opportunity to use it and give it some API testing.
https://bugzilla.gnome.org/show_bug.cgi?id=760733
It's not enough to have timeout or event based VPS/SPS/PPS information
sent in RTP packets. There are some scenarios when key frames may appear
more frequently than once a second, in which case the minimum timeout
for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough.
It might also be desirable in general to make sure the VPS/SPS/PPS is
available with every keyframe (packet loss aside), so receivers can
actually pick up decoding immediately from the first keyframe if
VPS/SPS/PPS is not signaled out of band.
This commit adds the possibility to send VPS/SPS/PPS with every key frame.
This mode can be enabled by setting "config-interval" property to -1. In
this case the payloader will add VPS, SPS and PPS before every key (IDR)
frame.
https://bugzilla.gnome.org/show_bug.cgi?id=757892
The MPEG standard (ISO-13880-1) says the reserve bits need to be set
to one (2.1.64). This is causing transport streams to fail validation
on some systems.
https://bugzilla.gnome.org/show_bug.cgi?id=760127
This works usually in this place, unless the compiler optimizes things in
interesting ways in which case it causes stack corruption and crashes later.
The compiler in question here is clang with -O1, which seems to pack the stack
a bit more and causes writing to the guint as pointer to overwrite map.memory,
which then later crashes during unmapping of the memory.
While encoding the frame in ASCII mode, per component four bytes are needed
and after every 20 bytes, a \n will be added. So the calculation should be
size = size * (4 + 1 / 20). This should exclude the header being written.
Since header is also being included in the calculations, memory mishandlings
are happening.
https://bugzilla.gnome.org/show_bug.cgi?id=759520
rename gst-launch --> gst-launch-1.0
replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
fix caps in examples
https://bugzilla.gnome.org/show_bug.cgi?id=759432
The edit rate is only supposed to be the same in a source package, but there
might be multiple source packages with the same essence container. As such
just comparing the body/index SID is not sufficient.
This was completely broken before and could only work on a very constrained
set of files. After these changes it should work except for situations where
PTS != DTS, which is not handled at all in mxfdemux currently.
https://bugzilla.gnome.org/show_bug.cgi?id=759118
According to S377-1-2009c 9.2 the local tags must not be resolved from the
primer pack, which as a result means that there can't be any other tags than
statically assigned ones.
This is to support byte-stream decoder that does not remember the
PPS/SPS after a flush. This is not needed by all decoders, but is
harmless for those that do remember.
https://bugzilla.gnome.org/show_bug.cgi?id=758405
The order in which program switch must happen is:
1) drain all data on old pads (but don't push EOS)
2) add new pads (but don't push any data on them)
3) Push EOS and remove old pads
4) Start pushing data on new pads
There was one caveat in this implementation, which is that when
we activate a sparse pad (step 2) we would push a GAP event. The problem
is that, while being an event, it is actually *data*.
We therefore need to make sure pushing those GAP event is done at the step
we start pushing data.
https://bugzilla.gnome.org/show_bug.cgi?id=750402
Before we add any streams, make sure we drain all streams. This ensures
there's consistency that only "new" data will be pushed on buffers once
the new pads are added
https://bugzilla.gnome.org/show_bug.cgi?id=750402
When changing programs, the order of events needs to be the following:
* add pads from new program
* send EOS on old pads
* remove old pads
* emit 'no-more-pads'
Previously tsdemux was not doing that, and was first deactivating and
removing old pads before adding new ones.
We fix this by allowing subclasses of mpegtsbase to be able to handle
themselves the deactivation of programs. In this case tsdemux will
properly deactivate it once it has activated the new program.
https://bugzilla.gnome.org/show_bug.cgi?id=750402
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
