Commit graph

128 commits

Author SHA1 Message Date
Xavier Claessens
b004464ac6 Remove glib and gobject dependencies everywhere
They are part of gst_dep already and we have to make sure to always have
gst_dep. The order in dependencies matters, because it is also the order
in which Meson will set -I args. We want gstreamer's config.h to take
precedence over glib's private config.h when it's a subproject.

While at it, remove useless fallback args for gmodule/gio dependencies,
only gstreamer core needs it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
2022-04-01 16:32:17 +00:00
Thibault Saunier
b358897a3b navigation: Rename parse_state to parse_modifier_state
`parse_state` sounds a bit weird and `parse_modifier_state` is clearer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2087>
2022-04-01 06:38:43 +00:00
Matthew Waters
8cdbfec5ed deinterlace: silence unused-but-set werror from imported code
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2042>
2022-03-28 03:00:58 +00:00
Thibault Saunier
2db3ddaa9d navigationtest: Add some support for modifiers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2010>
2022-03-25 15:16:03 +00:00
Matthew Waters
c1a3f958e7 rtpptdemux: fix leak of caps when ignoring a pt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2025>
2022-03-25 05:44:36 +00:00
Vivienne Watermeier
97bc8f193f navigationtest: Display touchscreen events, log all events
Represents touchscreen events as a trail of black squares, one for each
reported position. Additionally, this adds the `display-mouse` and
`display-touch` properties to toggle visibility of mouse/touchscreen
events, since touchscreens often emulate mouse events, as well as
logging for all received navigation events.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
2022-03-23 13:14:52 +00:00
Vivienne Watermeier
6c2f6c3bd4 all: Use new navigation interface and API
Use and implement the new navigation interface in all relevant sink elements,
and use API functions everywhere instead of directy accessing the event structure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
2022-03-23 13:14:52 +00:00
Stéphane Cerveau
1170ab3c29 wavparse: handle query in any parse state
In order to create the stream_id, we need to
pass the query to the default query handler.

If the parse state is different from GST_WAVPARSE_DATA
the query should be passed to the default query
handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1987>
2022-03-22 16:25:35 +00:00
Jan Alexander Steffens (heftig)
074f7c2e4e flvmux: Clean up aggregate's control flow
This unifies exits to go through a single out label. It mostly
simplifies how EOS is handled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1035>
2022-03-22 15:28:57 +00:00
Matthew Waters
206021e4d4 rtpmanager/rtx: implement initial support for reading/writing rid extensions
Two RTP Header extensions are very relevant for rtprtxsend/receive.
1. "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id": will always be removed
2. "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id": will be written
    instead of the "rtp-stream-id" header extension.

Currently it's only a simple replacement of one header extension for
another however a future change would only add the relevant extension
based on some heuristics (like, video frames only on one of the rtp key
frame buffers, or only until the rtx ssrc has been validated by the peer)
in order to reduce the required bandwidth.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1759>
2022-03-21 03:18:18 +00:00
Matthew Waters
1e55e2d654 rtpmanager: add support for RFC8852 (rid) RTP header extensions
Both for regular RID and for adding on a repaired (RTX) etc stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1759>
2022-03-21 03:18:18 +00:00
Matthew Waters
ecd9cce3b1 rtpmanager: add support for writing RFC8843 (BUNDLE mid) RTP header extension
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1759>
2022-03-21 03:18:18 +00:00
Sebastian Dröge
3de245ed17 videocrop: Add support for v210
Like UYVY and similar formats this is rounding down to the start of the
previous macro-pixel to not mix up the different components.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1988>
2022-03-19 01:25:07 +00:00
Sebastian Dröge
49ec82b209 videocrop: Use GST_ROUND_DOWN_2 instead of re-defining a local version
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1988>
2022-03-19 01:25:07 +00:00
Sebastian Dröge
cd86181d54 videocrop: Rename PACKED_COMPLEX to PACKED_YVYU
It's not handling any kind of complex packed format, only formats that
are like YVYU.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1988>
2022-03-19 01:25:07 +00:00
Sangchul Lee
7691c6776a rtpjitterbuffer: Fix invalid memory access in rtp_jitter_buffer_pop()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1973>
2022-03-17 12:46:14 +00:00
Tim-Philipp Müller
7895bf38ad rtspsrc: proxy new "add-reference-timestamp-meta" property from rtpjitterbuffer
When syncing to an RFC7273 clock this will add the original
reconstructed reference clock timestamp to buffers in form
of a GstReferenceTimestampMeta.

This is useful when we want to process or analyse data based
on the original timestamps untainted by any local adjustments,
for example reconstruct AES67 audio streams with sample accuracy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1964>
2022-03-16 09:52:58 +00:00
Tim-Philipp Müller
c29d741c0e rtpbin: proxy new "add-reference-timestamp-meta" property from rtpjitterbuffer
When syncing to an RFC7273 clock this will add the original
reconstructed reference clock timestamp to buffers in form
of a GstReferenceTimestampMeta.

This is useful when we want to process or analyse data based
on the original timestamps untainted by any local adjustments,
for example reconstruct AES67 audio streams with sample accuracy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1964>
2022-03-16 09:52:58 +00:00
Tim-Philipp Müller
c88bfc0b3e rtpjitterbuffer: add "add-reference-timestamp-meta" property
When syncing to an RFC7273 clock this will add the original
reconstructed reference clock timestamp to buffers in form
of a GstReferenceTimestampMeta.

This is useful when we want to process or analyse data based
on the original timestamps untainted by any local adjustments,
for example reconstruct AES67 audio streams with sample accuracy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1964>
2022-03-16 09:52:58 +00:00
Sebastian Dröge
5ca39060f4 rtpjitterbuffer: Improve accuracy of RFC7273 clock time calculations
Previously the result of the calculations included inaccuracies caused
by the NTP clock estimation, which caused the timestamps to jitter
+/- 1/clockrate.

By reorganizing the calculations it is possible to get rid of this
inaccuracy and calculate deterministic and exact packet timestamps based
on the actual NTP clock as long as the estimation is not off by more
than 2**31 clockrate units.

The only remaining inaccuracy that is introduced now is caused by the
conversion from the NTP clock to the pipeline clock.

Also split up debug output, demote many messages to the trace debug
level and output more intermediate results.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1955>
2022-03-15 23:33:37 +00:00
Nirbheek Chauhan
8c2ef0f025 twcc: Add some logging to debug TWCC feedback
This should allow people to debug when TWCC feedback is not enabled
because they haven't set the extmap in the caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1952>
2022-03-15 22:32:07 +00:00
Nirbheek Chauhan
a6bb63dcd7 twcc: Note that packet-loss-pct can count reordering as loss
This is difficult to encounter in ordinary networks, but is
encountered when using tc-netem to add random delays to packets, and
also when your UDP stream is bonded over multiple links with varying
characteristics.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1952>
2022-03-15 22:32:07 +00:00
Havard Graff
e5bd9839c4 rtprtxsend: don't require clock-rate in caps
For multiplexing, the rtpstreams you are multiplexing might not use
the same clock-rate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1881>
2022-03-15 19:05:00 +00:00
Havard Graff
4d31641302 rtprtxsend: don't start the task unless we are doing rtx
The rtxsend element can do pass-through when not enabled (no pt-map set)
and in those cases there is no point in starting an additional task
that does absolutely nothing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1880>
2022-03-15 12:03:27 +00:00
Havard Graff
6f57199958 rtprtxreceive: add ssrc-map property
Mirroring the rtxsend, this allows the application to "pre-map" the
retransmission-ssrcs to the "real" ssrc, if this information is known.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1878>
2022-03-14 09:14:10 +00:00
Carlos Rafael Giani
671c89c392 mpg123: Add gapless playback support
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
2022-03-14 10:32:15 +02:00
Carlos Rafael Giani
0431a0845c mpegaudioparse: Support gapless playback
Gapless playback is handled by adjusting buffer timestamps & durations
and by adding GstAudioClippingMeta.

Support for "Frankenstein" streams (= poorly stitched together streams)
is also added, so that gapless playback support doesn't prevent those
from being properly played.

Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
2022-03-14 10:32:15 +02:00
Jan Alexander Steffens (heftig)
2db283499e deinterlace: scalerbob: Reduce latency to 0
We only need the current field, just like `linear`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1926>
2022-03-12 22:48:39 +00:00
Vivia Nikolaidou
8c648384f2 yadif: Fix CHECK macro for YUY2 format
Used to make comb artifacts for videotestsrc pattern=ball for YUY2
format only (not AYUV).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1938>
2022-03-12 17:18:47 +00:00
Sangchul Lee
67df5815a9 rtpvp8depay: Fix crash when making 'GstRTPPacketLost' custom event
This patch fixes a seg.fault in gst_structure_new() with warnings as below.

GLib-GObject-WARNING **:
 ../gobject/gtype.c:4330: type id '0' is invalid
GLib-GObject-WARNING **:
 can't peek value table for type '<invalid>' which is not currently referenced

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1918>
2022-03-10 19:37:49 +00:00
Tomasz Andrzejak
e74435008f rtpbin: allow FEC elements with Always pads
This patch enable picking up FEC decoder or enocder that have
static repair packets pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1860>
2022-03-10 08:33:27 +00:00
Edward Hervey
568b918971 qtdemux: Propagate stick events downstream when creating pads
If upstream provided a stream collection event before any pads were created,
make sure it's propagated downstream when pads are created.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1891>
2022-03-09 16:09:31 +00:00
Havard Graff
a2c25ccd09 rtprtxsend: if no rtx is present, don't expose a rtx-ssrc in caps
The point here is that rtpsession will create a new rtpsource when
the field "rtx-ssrc" is present, and when not doing rtx, that means
a random ssrc will create a new rtpsource that will be included in RTCP
messages for the current session.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1882>
2022-03-09 15:30:37 +00:00
Havard Graff
2a8fa45ba8 rtprtxsend: don't process or warn if no map is set
This makes it more gentle when doing "pass-through"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1879>
2022-03-09 12:01:22 +05:30
Mikhail Fludkov
815d279f2e rtprtxreceive: fix crash when RTX payload has zero length
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1875>
2022-03-08 09:07:41 +00:00
Havard Graff
86c7231dae rtprtxreceive: allow passthrough and non-rtp buffers
To avoid mapping rtp buffers when RTX is not in use, and to not
do a full error on receiving a non-rtp buffer, since you have no control
of what a rouge sender might send you.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1874>
2022-03-07 23:43:49 +00:00
Havard Graff
a475c93346 rtprtx: don't access type-system per buffer
When doing only a single stream of audio/video this hardly matters,
but when doing many at the same time, the fact that you have to get
a hold of the glib global type-system lock every time you process a buffer,
means that there is a limit to how many streams you can process in
parallel.

Luckily the fix is very simple, by doing a cast rather than a full
type-check.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1873>
2022-03-07 22:01:03 +00:00
Hou Qi
b11084f729 flvmux: Add protection when unref GstFlvMuxPad
This is to avoid gst_object_unref: assertion 'object != NULL' failed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1843>
2022-03-07 13:03:16 +00:00
Nicolas Dufresne
0f15580853 matroska: Fix AV1 alignment to TU
Matroska stores AV1 in temporal unit, so that all OBU sharing the same
timestamp are put together. This was previously just assumed, which isn't
safe now that we have more alignments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1837>
2022-03-04 21:58:15 +00:00
Nicolas Dufresne
f6c070fbff isomp4: Fix AV1 default alignment
ISOMP4 store TU (temporal units) worth of AV1. Expose this in the
caps to reduce overhead in the parser, and in the muxer to avoid
storing frames split in the wrong way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1837>
2022-03-04 21:58:15 +00:00
Tristan Matthews
9d0d001d19 matroskamux: allow width+height caps changes for VP8/9
For VP8 and VP9, width+height changes are signalled inband.

Refs https://github.com/Kurento/bugtracker/issues/535 and
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047/diffs?commit

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1657>
2022-03-04 14:17:20 -05:00
Tristan Matthews
c6ba57eb8e matroskamux: allow width + height changes for avc3|hev1
For avc3 and hev1, the intent was to allow more flexibility for caps changes
(see https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047/diffs?commit_id=9bd8d608d5bae27ec5ff09e733f76ca32b17420c)
however width and resolution were previously omitted.

avc3 and hev1 specifically support changing stream-parameters on the fly, whereas avc1/hvc1 disallow in-band SPS.

This commit allows for changes to width and height for these which is in line with matroskamux's behaviour prior to 1.14.0.

Practically speaking, one use case where this is commonly seen is when capturing a WebRTC stream, as the browser will adapt the resolution live.

Suggested-by: Mathieu Duponchelle "<mathieu@centricular.com>"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1657>
2022-03-04 14:17:20 -05:00
Jan Alexander Steffens (heftig)
ce503d0645 deinterlace: Prevent race between _set_method and latency query
It's possible that the method is being manipulated while downstream
queries our latency, leading to crashes.

Prevent that from happening.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1854>
2022-03-04 16:14:46 +00:00
Sebastian Dröge
9f798776e5 matroska-mux: Handle pixel-aspect-ratio caps field correctly when checking caps equality
Not having this field is equivalent with it being 1/1 so consider
it like that. The generic caps functions are not aware of these
semantics and would consider the caps different, causing a negotiation
failure when caps are changing from caps with to caps without or the
other way around.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1826>
2022-03-02 10:27:47 +00:00
Sebastian Dröge
1b851ae23f matroska-mux: Handle multiview-mode/flags caps fields correctly when checking caps equality
Not having these fields is equivalent with them being mono/0 so consider
them like that. The generic caps functions are not aware of these
semantics and would consider the caps different, causing a negotiation
failure when caps are changing from caps with to caps without or the
other way around.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1826>
2022-03-02 10:27:47 +00:00
Jan Schmidt
cebf769725 matroska-mux: If a stream has a TITLE tag, use it for the name.
If a title tag is pushed to a pad, store it as the Track name.
This means that players will use it as the human readable
description of the track, instead of something generic like 'Video'
or 'Subtitle'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1798>
2022-03-01 13:17:40 +00:00
Jan Schmidt
7efdc9c7f5 matroskademux: Don't parse Tracks element twice
If the tracks element was parsed from the SeekEntry, don't
parse it a second time and recreate tracks, as this
loses any tags that were read using the seek table.

If a genuinely new Tracks element is found, do read that
as it is needed for MSE support.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1798>
2022-03-01 13:17:40 +00:00
Vivia Nikolaidou
b699feefee yadif.asm: Fix improper usage of LOAD macro
LOAD macro relies in m7 being zero for interleaving purposes. Using LOAD
on the m7 register makes it interleave with its new content instead of
with 0.

The effect of this bug was bobbing on some static lines that appeared
over fast-moving content.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1816>
2022-03-01 07:22:10 +00:00
Vivia Nikolaidou
d499342f0d yadif.asm: Typo fixes in comments
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1816>
2022-03-01 07:22:10 +00:00
Vivia Nikolaidou
087ca88213 yadif: Fix bug in C implementation of CHECK
It was different compared to the corresponding part in both ffmpeg and
the asm implementation. Fixing this makes videotestsrc pattern=spokes
not jump at all when not using the asm optimisations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1816>
2022-03-01 07:22:10 +00:00