last_segment is only being updated in dequeue_buffer(),
which is only called from _pull_sample(). _pull_preroll()
simply re-uses an old or dummy segment while the actual
one sits and waits in the queue.
https://bugzilla.gnome.org/show_bug.cgi?id=751147
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.
API: gst_video_decoder_sink_query_default
https://bugzilla.gnome.org/show_bug.cgi?id=753623
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.
API: gst_audio_decoder_sink_query_default
https://bugzilla.gnome.org/show_bug.cgi?id=753623
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 audio tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
There was a confusion, six depayloaders where passing through the
timestamp while the base class was re-timestamping to running
time. This inconstancy has been unnoticed has in most use cases
the incoming segment is [0, inifnity] in which case timestamps are
the same as running time. With DTS/PTS shifting added (to avoid
negative values) and pcapparse sending a different segment this
started being an issue.
https://bugzilla.gnome.org/show_bug.cgi?id=753037
gstvideoencoder.h:228: Warning: GstVideo: "@transform_meta"
parameter unexpected at this location:
* @transform_meta: Optional. Transform the metadata on ...
Push all pending events before pushing the gap. This ensures the
segment is pushed before the gap so it can be properly translated
to the running time
Includes unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=753360
When there is no clock_base provided, the start position is
set to 0 instead of the original segment start value. This
would break synchronization if start was not 0.
https://bugzilla.gnome.org/show_bug.cgi?id=752228
Add logging categories for most video objects.
Remove some useless debug lines in video-info and videotestsrc.
Add a performance debug line in the video scaler.
GST_TYPE_RTSP_LOWER_TRANS used to be defined in there, not
including the generated file makes older gst-p-good fail to build,
so it constitues an API break.
Add process_rtp_packet() vfunc that works just like the
existing process() vfunc only that it takes the GstRTPBuffer
that the base class has already mapped (with MAP_READ),
which means that the subclass doesn't have to map it again,
which allows more performant processing of input buffers
for most RTP depayloaders.
https://bugzilla.gnome.org/show_bug.cgi?id=750235
This minor update should workaround a build system bug. While the
makefile has been updated to generate more enum type, there is nothing
that updates the header and would lead to the generated code to be
produced again. This minor doc fix should ensure no one get a build with
missing symbols.
E.g.
video-scaler.c: In function 'gst_video_scaler_horizontal':
video-scaler.c:1332:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized]
func (scale, src, dest, dest_offset, width, n_elems);
^
video-scaler.c: In function 'gst_video_scaler_vertical':
video-scaler.c:1373:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized]
func (scale, src_lines, dest, dest_offset, width, n_elems);
^
GCC's analyses seem to be correct, for the simple fact that if you pass
get_functions a known format, but no hscale or vscale, it'll return
True without having done anything.
Some callers check for the scale values to be not NULL, but then
hscale->resampler.max_taps could return 0.
A different approach to the one presented in this patch is to check
for those max_taps, too, before calling get_functions.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=752051
The padding (if any) is included in the length of the last packet, see
RFC 3550.
Section 6.4.1:
padding (P): 1 bit
If the padding bit is set, this individual RTCP packet contains
some additional padding octets at the end which are not part of
the control information but are included in the length field. The
last octet of the padding is a count of how many padding octets
should be ignored, including itself (it will be a multiple of
four).
Section A.2:
* The padding bit (P) should be zero for the first packet of a
compound RTCP packet because padding should only be applied, if it
is needed, to the last packet.
* The length fields of the individual RTCP packets must add up to
the overall length of the compound RTCP packet as received.
https://bugzilla.gnome.org/show_bug.cgi?id=751883
It's needed to check if pixel-aspect-ratio exists before fixating.
It does not exist if input caps is not set yet and allowed caps
does not contain pixel-aspect-ratio (e.g. when using GST_VIDEO_CAPS_MAKE)
https://bugzilla.gnome.org/show_bug.cgi?id=751932
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
The default implementation copies all metadata without tags, and metadata
with only the video tag. Same behaviour as in GstVideoFilter.
This currently does not work if the ::parse() vfunc is implemented as all
metas are getting lost inside GstAdapter.
https://bugzilla.gnome.org/show_bug.cgi?id=742385
CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
number since it is a division of an unsigned integer (i). Removing that check
and only checking if it is bigger than max and setting it appropriately.
CID #1308950
For alaw/mulaw we should also try to initialize the channel positions in the
ringbuffer's audio info. This allow pulsesink to directly use the channel
positions instead of using the default zero-initialized ones, which doesn't
work well.
https://bugzilla.gnome.org/show_bug.cgi?id=751144
Add a utility function that, given a video size and a
packed stereoscopic mode, attempts to guess if the video
is packed at half resolution per view or not, since
very few videos provide the information.
We need to scale groups of 4 bytes for YUY2 formats so round up to 4.
It's possible that there is no Y byte for the last pixel so make sure
we clamp correctly.
The API does not follow the type naming convention. Re-enable
only if one take the time to box and rename (see (rename-to SYMBOL)
annotation) all types.
When copying info from the reference input state, duplicate
all the fields of the video info. The sub-class will have the
chance to override them later.
Add flags and enums to support multiview signalling in
GstVideoInfo and GstVideoFrame, and the caps serialisation and
deserialisation.
videoencoder: Copy multiview settings from reference input state
Add gst_video_multiview_* support API and GstVideoMultiviewMeta meta
https://bugzilla.gnome.org/show_bug.cgi?id=611157
This new clock slaving method allows for installing a callback that is
invoked during playback. Inside this callback, a custom slaving
mechanism can be used (for example, a control loop adjusting a PLL or an
asynchronous resampler). Upon request, it can skew the playout pointer
just like the "skew" method. This is useful if the clocks drifted apart
too much, and a quick reset is necessary.
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
https://bugzilla.gnome.org/show_bug.cgi?id=708362
Otherwise ssrc changes via rtpsession's (deprecated!) internal-ssrc property
are not possible anymore. rtpsession was now patched to only suggest an ssrc
if it makes sense to do so.
In 2.0 we should get rid of all the properties that are also negotiated via
caps, the code and behaviour is too confusing otherwise.
https://bugzilla.gnome.org/show_bug.cgi?id=749581
According to this section of the rfc.
https://tools.ietf.org/html/rfc5506#section-3.4.2
The validation should be updated to accept more types of RTCP
packages, with this mask change feedback packages will be also
accepted.
Change-Id: If5ead59e03c7c60bbe45a9b09f3ff680e7fa4868
Micro-optimisation: if the buffer consist of just one memory, we
know we have already mapped that memory to read the headers, so
no need to map it another time to get to the payload data, we
can just set up the payload data details right there and then
and avoid another map call in gst_rtp_buffer_get_payload().
Adds up when receiving RTP-payloaded raw video which can easily
be thousands of packets per frame.
Implement a chain_list function, which avoids lots of locking
compared to the default fallback implementation in GstPad.
We may also want to do some more sophisticated timestamp
tracking here at some point, but for now leave it up to the
jitterbuffer and/or subclasses (in case buffers in the
buffer list have no timestamp set on them, there may only
be a timestamp for the whole list on the first buffer).
This provides the exact same behaviour as the default
fallback implementation.
Summary:
So that the user can easily use the same encoding profile to render
with/without audio/video stream.
API:
gst_encoding_profile_is_disabled
gst_encoding_pofile_set_enabled
https://bugzilla.gnome.org/show_bug.cgi?id=749056
Instead of returning the first video meta found on a buffer, return the
one with the lowest id (which is usually the same thing, except on
multi-view buffers)
Use g_utf16_to_utf8() instead of the more generic g_convert(), so
that we can extract text in UTF-16 format even on embedded systems
with crippled iconv support.
This code path is exercised by the id3demux test_unsync_v23
check in gst-plugins-good.
https://bugzilla.gnome.org/show_bug.cgi?id=741144
From the API documentation: "Note that it is generally not
a good idea to reuse an existing cancellable for more
operations after it has been cancelled once, as this
function might tempt you to do. The recommended practice
is to drop the reference to a cancellable after cancelling
it, and let it die with the outstanding async operations.
You should create a fresh cancellable for further async
operations."
https://bugzilla.gnome.org/show_bug.cgi?id=739132
[API] gst_discoverer_info_to_variant
[API] gst_discoverer_info_from_variant
[API] GstDiscovererSerializeFlags
+ Serializes as a GVariant
+ Adds a test
+ Does not serialize potential GstToc (s)
https://bugzilla.gnome.org/show_bug.cgi?id=748814
This affects the pt, ssrc, seqnum-offset and timestamp-offset properties. If
they were set from a property, or we configured caps before, we try to use
that value for them. Even if the first structure of the downstream caps
specifies a different value, we check if the value is supported by other
structures.
Only if all this fails, we use the values given by downstream in the first
structure, i.e. if no properties were set and these are the first caps we
negotiate or downstream does not support our values.
By doing this we ensure that we don't spuriously change ssrcs or other fields
in the middle of the stream (and also consider property values more). Ssrc
changes would currently happen after sending an RTX packet (thus creating a
new internal source inside the rtpsession), and then renegotiating the
payloader (which then gets the RTX ssrc from rtpsession).
https://bugzilla.gnome.org/show_bug.cgi?id=749581
All those where super straight forward from the warnings gtkdoc prints. It kind
of makes sense to apply them before the list of warnings is >100 and people
complain that gtkdoc is noisy.
GST_VIDEO_CONVERTER_OPT_ALPHA_MODE, GST_VIDEO_CONVERTER_OPT_CHROMA_MODE,
GST_VIDEO_CONVERTER_OPT_MATRIX_MODE, GST_VIDEO_CONVERTER_OPT_GAMMA_MODE and
GST_VIDEO_CONVERTER_OPT_PRIMARIES_MODE were G_TYPE_STRING with only a few valid
options. Changed those to real enums.
https://bugzilla.gnome.org/show_bug.cgi?id=749104
2 second frame duration is rather unlikely... but if we don't clip
away buffers that far before the segment we can cause the pipeline to
lockup. This can happen if audio is properly clipped, and thus the
audio sink does not preroll yet but the video sink prerolls because
we already outputted a buffer here... and then queues run full.
In the worst case we will clip one buffer too many here now if no
framerate is given, no buffer duration is given and the actual
framerate is less than 0.5fps.
Fixes seeking on HLS/DASH streams, when seeking into the middle of
fragments and having no framerate/buffer duration.
