Commit graph

290 commits

Author SHA1 Message Date
Sebastian Dröge
ce38614e1a rtpsession: Handle RTCP-SR-REQ (RFC6051) RTCP feedback message
This causes an RTCP SR to be sent at the earliest possible time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
0c819d2f31 rtpbin/rtpjitterbuffer: Allow syncing to an SR without CNAME if the CNAME is already known
The RTCP SR packet might be without SDES in case of a reduced-size RTCP
packet. For syncing purposes the CNAME is needed but it might be known
already from an earlier RTCP packet or out of band, via the SDP for
example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
cbaac3cdba rtpbin/jitterbuffer: Use inband 64-bit NTP timestamps according to RFC6051 for faster synchronization
When signalled via the caps that the header extension is used, it will
be read and used in the same way as the RTP/NTP time mapping from RTCP
SRs.

If the CNAME of the stream's SSRC is provided out of band via e.g. the
SDP then this allows streams to be synchronized immediately on the first
packet instead of having to wait for the first RTCP SR to arrive.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/383

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
7c796b3c05 rtpsession: Only add send latency to the running time if it is actually known
Otherwise we can't know the running time yet if rtcp-sync-send-time is
set, and have to wait until the latency is known later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
7ffc830959 rtpsession: Update 64-bit NTP header extensions with the actual NTP time in senders
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
8980c35efe rtpmanager: Add header extension implementation for the 64-bit RFC6051 NTP header extension
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Xavier Claessens
b99ecc78ca Replace gst-i18n-*.h with gi18n-lib.h
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00
Havard Graff
71891e5647 qtdemux: fix leak of channel_mapping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2179>
2022-04-14 19:41:36 +09:00
Robert Rosengren
e4a6521ac7 rtpbin: Fix division by zero when using ts-offset-smoothing-factor
avg_ts_offset may cause division by zero when calculating potential
overflow protection. This fix will avoid the division.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2151>
2022-04-11 15:29:49 +02:00
Tristan Matthews
86f0f8b67f rtpopusdepay: assume 2 channels if sprop-stereo is missing
Fixes #1064

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2125>
2022-04-08 13:11:25 +00:00
Sebastian Dröge
0813efc821 rtpstats: Remove non-existing twcc field docs from RTPPacketInfo and add missing field docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2121>
2022-04-06 10:15:13 +03:00
Sebastian Dröge
46d7763879 rtpsession: Remove unused twcc fields from the struct
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2121>
2022-04-06 10:15:13 +03:00
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
Sebastian Dröge
b0afaffc5d rtp: In payloaders map the RTP marker flag to the corresponding buffer flag
This allows downstream of a payloader to know the RTP header's marker
flag without first having to map the buffer and parse the RTP header.

Especially inside RTP header extension implementations this can be
useful to decide which packet corresponds to e.g. the last packet of a
video frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
2022-02-28 10:13:11 +00:00
Sanchayan Maity
cc3419daf6 rtp: ldac: Set frame count information in payload
The RTP payload seems to be required as it carries the frame count
information. Also, gst_rtp_base_payload_allocate_output_buffer had
the second argument incorrect.

Strangely some devices like Shanling MP4 and Sony XM3 would still
work without this while some like the Sony XM4 do not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
2022-02-26 21:09:57 +05:30
Xavier Claessens
3d8372cc50 devenv: Add some missing GStreamer specific env variables
This should make "meson devenv" closer to what "gst-env.py" sets.

- GST_VALIDATE_SCENARIOS_PATH
- GST_VALIDATE_APPS_DIR
- GST_OMX_CONFIG_DIR
- GST_ENCODING_TARGET_PATH
- GST_PRESET_PATH
- GST_PLUGIN_SCANNER
- GST_PTP_HELPER
- _GI_OVERRIDES_PATH

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1768>
2022-02-25 20:35:26 +00:00
Jan Alexander Steffens (heftig)
d6ec88c775 deinterlace: greedyh: Stop adding 2 to cur_field_idx
Just a simplification.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1790>
2022-02-25 12:06:56 +00:00
Jan Alexander Steffens (heftig)
dc1ae0aaa0 deinterlace: greedyh: Use _plane in _packed, fix planar formats
This greatly reduces code duplication. It also exposed the cause for
planar formats not being properly deinterlaced:

The planar path was missing the initial offset adjustment that the
packed path did to `L2` and `L2P` in the case of an even field, which
caused it to select the wrong weave lines every other field.

Add those offsets in `_plane`.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1047
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1790>
2022-02-25 12:06:55 +00:00
Jan Alexander Steffens (heftig)
625cbcf70a deinterlace: greedyh: Rename _planar_plane to _plane
As well as `i` to `plane`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1790>
2022-02-25 12:06:55 +00:00
Jan Alexander Steffens (heftig)
7e16955e4d deinterlace: greedyh: Move code from _planar into _planar_plane
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1790>
2022-02-25 12:06:55 +00:00
Jan Alexander Steffens (heftig)
19ca706fe8 deinterlace: greedyh: Move _planar_plane upwards
In preparation of refactoring. No functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1790>
2022-02-25 12:06:55 +00:00
Guillaume Desmottes
8bbdd9addb rtpsource: fix rtp_source_get_nack_deadlines doc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1775>
2022-02-22 09:40:35 +00:00
Matthew Waters
b0f72ed788 ulpfecenc: slightly safer dispose impl
Technically dispose can be called more than once (even if gstelement is
not actually set up to do that) so need to protect against that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1761>
2022-02-21 09:43:33 +00:00
Matthew Waters
629b427a13 ulpfecenc: fix unmatched free() call
One must always match a g_slice_new with a g_slice_free and a g_new with
a g_free.  This was not the case for the internal ctx struct.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1761>
2022-02-21 09:43:33 +00:00
Matthew Waters
acc9024039 rtpulpfecenc: add some debug logging
Like, what configuration we are using or whether a fec packet is
generated.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1761>
2022-02-21 09:43:33 +00:00
Nirbheek Chauhan
4e22ef5bd2 matroska-demux: Emit a warning when no codec data found
It is bad if an mkv file does not have codec data for the ProRes
variant, so we should emit a warning. ffmpeg does the same thing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1739>
2022-02-21 08:49:28 +00:00
Sebastian Wick
e61e069189 matroska: default prores fourcc apcn
If there is no codec private data for prores it should default to Apple
ProRes 422 Standard Definition (apcn). Can be tested with
strobe_scientist.mkv from
https://developers.google.com/media/vp9/hdr-encoding

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1734>
2022-02-18 08:38:31 +00:00
Seungha Yang
53ed876002 qtdemux: Do not send unnecessary GAP events
Each stream may have its own segment timeline
(i.g., different segment.start or segment.base)
depending on edit-list and composition-to-decode atom.

Make sure whether time position of a stream has been actually
far behind than that of current target stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1352>
2022-02-17 19:39:53 +00:00
Sebastian Dröge
8bda2ef474 qtmux: Don't post an error message if pushing a sample failed with FLUSHING
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1711>
2022-02-15 13:43:41 +02:00
Robert Rosengren
265878c4ba rtpbin: Safer ts-offset-smoothing-factor calculation
Protect the ts-offset-smoothing-factor calculation from overflow. Output
warning and fallback to ts-offset if it is detected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1409>
2022-02-08 11:11:35 +00:00
Robert Rosengren
31dd9226ce rtpbin: add ts-offset-smoothing-factor property
Add property to set the TS offset smoothing factor and set default value
to not use it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1409>
2022-02-08 11:11:35 +00:00
Danny Smith
bc964141c8 rtpbin: applied smoothing to jittery sender time-stamps
Applying a moving average filter to the timestamp offsets
for smoothing jittery and preventing aggressive skew handling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1409>
2022-02-08 11:11:34 +00:00
Danny Smith
d5e257afd1 rtpbin: added option for setting min_ts_offset in ntp-sync mode
Constantly updating the ts_offset results in audiable glitches
when streaming audio using ntp-sync=true. By requiring a minimum
offset before updating ts_offset this can be mitigated. Added a
parameter which can be used to set min_ts_offset in ntp-sync mode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1409>
2022-02-08 11:11:34 +00:00
Stéphane Cerveau
d191180061 autodetect: fix debug init category
Since the split of elements, the debug category
was default for autodetect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1590>
2022-01-28 10:35:35 +00:00
Nirbheek Chauhan
980925a6a3 rtspsrc: Fix critical while serializing timeout element message
The "cause" field wasn't registered as a GEnumValue, so do that.

Fixes this critical in gst_structure_to_string():

`gst_value_serialize: assertion 'G_IS_VALUE (value)' failed`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1573>
2022-01-26 11:07:07 +00:00
Sebastian Dröge
241a26631d splitmuxsink: Warn when calculating the next fragment time in timecode mode fails
But only if timecode mode is enabled as it will fail all the time
otherwise.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1557>
2022-01-25 12:08:20 +00:00
Aleksandar Topic
002c5ae7ef imagefreeze: Fix example launch line format issue
The currently shown example launch line will not run, because it
cannot handle png images.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1547>
2022-01-20 18:12:39 +00:00
Heinrich Kruger
6dd15acf2d rtp-hdrext-colorspace: Fix color range encoding
The color space RTP header extension encodes color range as specified in
https://www.webmproject.org/docs/container/#Range. In other words:
0: Unspecified,
1: Broadcast Range,
2: Full range,
3: Defined by matrix coefficients and transfer characteristic.

This does not match the values of GstVideoColorRange, so it is not
correct to just write the colorimetry.range value to the header
extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1482>
2021-12-30 16:31:33 +00:00
Jeongki Kim
04f6fbc237 rtpg726depay: fix endian conversion
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1469>
2021-12-24 14:52:38 +09:00
Mathieu Duponchelle
d12d45db77 reddec: implement support for the BUNDLE case
When multiple streams are bundled together, there may be more
than one red payload type to handle.

In addition, as the red decoder works by filling in gaps in
the seqnums, there needs to be one rtp_history queue per sequence
domain.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
2021-12-14 17:34:53 +00:00
Mathieu Duponchelle
a09b8ded30 rtpbin: add new request-fec-decoder-full signal for BUNDLE
When multiple streams are bundled together, the application needs
to know about the payload type in order to instantiate the appropriate
FEC decoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
2021-12-14 17:34:53 +00:00
Mathieu Duponchelle
5dc280de9f rtp/redenc|ulpfecenc: add support for TWCC
In redenc, when input buffers have a header for the TWCC extension,
we now add one to our wrapper buffers.

In ulpfecenc we add one in that case to our protection buffers.

This makes TWCC functional when UlpRed is used in webrtcbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1414>
2021-12-14 03:26:56 +00:00
Thibault Saunier
49055f1cd5 rtph264pay: Handle 'profile' field
In order to allow "level-asymmetry-allowed" we now handle a new
"profile" field, which as the same semantics as the "profile" field in
H.264 stream so that we can force payloaded stream to have the right
format when using the `gst_sdp_media_get_caps_from_media` to set caps
filter after the payloader. This allows a simple negotiation in standard
RTP negotiation based on SDPs (like webrtc) for that particular case,
closely respecting the specs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1410>
2021-12-12 10:59:00 -03:00
Célestin Marot
f509578de5 multifilesrc: fix caps leak
since `gst_caps_replace()` and `gst_pad_set_caps()` both ref the caps and neither of them takes the ownership of the caps -> it must be unreffed in `gst_multi_file_src_set_property()`

to test the leak (on Unix): `echo coucou > /tmp/file.txt && GST_TRACERS=leaks GST_DEBUG="GST_TRACER:7" gst-launch-1.0 multifilesrc location=/tmp/file.txt caps='txt' ! fakesink`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1436>
2021-12-10 18:34:57 +01:00
Mathieu Duponchelle
4412198c05 rtpfunnel: fix extmap handling on accept-caps
Follow-up on 97d83056b3, only check
for intersection with the current srccaps when checking if a sinkpad
can accept caps.

I must have been lucky in my firefox testing then, and always entered
the code path with audio getting negotiated first, thus not failing
the is_subset check when srccaps had been negotiated as
application/x-rtp, and an accept-caps query was made for the video
caps with a defined extmap.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1384>
2021-11-23 20:26:30 +00:00
Mathieu Duponchelle
97d83056b3 rtpfunnel: don't enforce twcc during upstream negotiation
A previous patch has caused rtpfunnel to output twcc-related
information downstream, however this leaked into upstream
negotiation (through funnel->srccaps), causing payloader to
negotiate twcc caps even when not prompted to do so by the user.

Fix this by only enforcing that upstream sends us application/x-rtp
caps as was the case originally.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>
2021-11-12 18:40:32 +00:00
Mathieu Duponchelle
72118b9db4 rtptwcc: complete bufferlist fix
When dealing with bufferlists, we need to store one "SentPacket"
structure per buffer, not one per buffer list!

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>
2021-11-12 18:40:32 +00:00
Sebastian Dröge
efb2b6d478 qtdemux: Log cslg_shift that was determined
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:04 +00:00
Sebastian Dröge
12e918428a qtdemux: Use a composition time offset of 0 for "no decode samples" for the time being
This needs codec-specific handling, but using 0 instead of G_MININT32 at
least gives somewhat reasonable behaviour.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/883

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Sebastian Dröge
ad412d257b qtdemux: Always check ctts for unreasonably large offsets
If this happens then ignore the whole ctts. Previously we only did this
if the PTS/DTS shift was determined from the ctts instead of the cslg.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Sebastian Dröge
93a10a4ba1 qtdemux: Dump composition time offsets in trun as signed integers
Just like we do for ctts without regard of the version of the box.
Huge offsets are interpreted as negative offsets by qtdemux so this
works.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Sebastian Dröge
a6f3391c81 qtdemux: Add a comment why only positive cslg shifts are considered
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Sebastian Dröge
a33e30cfc4 qtdemux: Only adjust segment.stop by cslg_shift if stop is not -1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Sebastian Dröge
bb5a5ae8a8 qtdemux: Handle negative composition offsets in the trun box the same way as for non-fragmented streams
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Sebastian Dröge
767e8bf668 qtdemux: Parse ctts version
Negative composition time offsets are only allowed with version 1 of the
box, however we parse it as a signed value also for version 0 boxes as
unfortunately there are such files out there and it's unlikely to have
(valid) huge composition offsets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Sebastian Dröge
284dd5443f qtdemux: Add support for version 1 cslg boxes
They use 64 bit fields instead of 32 bit.

Also parse offset as a signed integer (in both versions) and clamp it to
a positive value as negative values don't really interest us here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Sebastian Dröge
7f105a919a qtdemux: Don't free cslg data that we don't own on corrupt files
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
2021-11-12 17:51:03 +00:00
Rafał Dzięgiel
41385ab6f7 matroska: Ref index table when updating track info
Track index table array was being lost during track info update.
Ref it over to updated info, so it can be used for finding
nearest seek points.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1203>
2021-11-12 12:28:40 +00:00
Rafał Dzięgiel
478f94edc7 matroska: Use g_array_unref everywhere
Instead of using g_array_free which is not thread safe use g_array_unref instead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1203>
2021-11-12 12:28:40 +00:00
Tim-Philipp Müller
972615cf22 docs: fix unnecessary ampersand, < and > escaping in code blocks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1340>
2021-11-12 11:39:19 +00:00
Mathieu Duponchelle
792fb05cec st2022-1-fecdec: fix packet trimming
g_sequence_remove_range's end iter is exclusive, so if one
wants to remove that item as well, it should be called with
the next iter.

This could in theory fix an issue where:

* The sequence isn't entirely trimmed, with an old item lingering

* Following FEC packets are immediately discarded because they
  arrived later than corresponding media packets, long enough for
  seqnums to wrap around

* We now try to reconstruct a media packet with a completely obsolete
  FEC packet, chaos ensues.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1341>
2021-11-12 08:15:28 +00:00
Matthew Waters
71dd47516c rtpbin: separate out the two fec decoder locations
The pipeline flow for receiving looks like this:

rtpsession ! rtpssrcdemux ! session_fec_decoder ! rtpjitterbuffer ! \
  rtpptdemux ! stream_fec_decoder ! ...

There are two places where a fec decoder could be placed.
1. As requested from the 'request-fec-decoder' signal: after rtpptdemux
   for each ssrc/pt produced
2. after rtpssrcdemux but before rtpjitterbuffer: added for the
   rtpst2022-1-fecenc/dec elements,

However, there was some cross-contamination of the elements involved and
the request-fec-decoder signal was also being used to request the fec
decoder for the session_fec_decoder which would then be cached and
re-used for subsequent fec decoder requests.  This would cause the same
element to be attempted to be linked to multiple elements in different
places in the pipeline.  This would fail and cause all kinds of havoc
usually resulting in a not-linked error being returned upstream and an
error message being posted by the source.

Fix by not using the request-fec-decoder signal for requesting the
session_fec_decoder and instead solely rely on the added properties for
that case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1300>
2021-11-10 10:38:26 +00:00
Zhao, Gang
6cad2a7150 qtdemux: Fix can not demux Opus track made by qtmux
Opus stream info is read from dOps box [1]. The offset of dOps box in Opus box is different in mp4a version 1 and 0 [2]. Calculate the offset of dOps box according to mp4a version.

[1] https://opus-codec.org/docs/opus_in_isobmff.html

[2] subprojects/gst-plugins-good/gst/isomp4/atoms.c:sample_entry_mp4a_copy_data:2146

Fixed: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/918
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1329>
2021-11-09 17:57:49 +00:00
Sanchayan Maity
3f49b21b80 imagefreeze: Only set caps if they do not match current caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1311>
2021-11-05 15:34:47 +05:30
Sebastian Dröge
dac82a8932 multifilesink: Make minimum distance between keyframes in next-file=key-frame mode configurable
Previously this was hardcoded to 10s, which is not necessarily the
desired behaviour.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1292>
2021-11-02 20:21:10 +00:00
Erlend Eriksen
0805ffdce9 qtmux: Fix deadlock in gst_qt_mux_prepare_moov_recovery
Regression from 5766731bd4

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1288>
2021-11-02 19:26:57 +00:00
Mathieu Duponchelle
c414f9560a rtptwcc: don't assume all PacketInfo->data are buffers
They can also be buffer lists

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1268>
2021-10-29 03:46:25 +02:00
Jan Schmidt
f6ed40c93a qtdemux: Fix text and closed-caption handling.
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182
broke text and and closed caption extraction when introducing WebVTT
support, by making the output buffers not have timestamps any more.

Fix that by making the process functions copy buffer metadata
when generating new output buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1262>
2021-10-28 21:45:33 +11:00
Sebastian Dröge
2853c085c7 qtdemux: Add pasp box to the list of known boxes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1235>
2021-10-23 11:39:36 +00:00
Rafał Dzięgiel
b57a7c3de7 matroska: Set image/attachment structure mimetype
Set structure mimetype to fix data detection by mimetype in other plugins.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1205>
2021-10-21 21:42:00 +00:00
Rafał Dzięgiel
0ba65a00c9 matroska: Treat non-image structure as attachment
Otherwise each structure is named as GstTagImageInfo even if
it does not contain any images which is misleading.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1205>
2021-10-21 21:42:00 +00:00
Matthew Waters
8c35850f23 rtpbin: fix leak of pad when a fec encoder and aux sender a created
The ghost sink pad retrieved by rtpbin from the aux sender was not freed
when there was a previous element (fec encoder) in the chain.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1222>
2021-10-21 13:46:03 +00:00
Olivier Crête
0dbe0e21fe rtphdrext-clientaudiolevel: Rename RFC 6464 element
Multiplying elements named after RFC numbers is confusing,
so let's give them meaningful names.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1125>
2021-10-20 00:03:09 +00:00
Jan Schmidt
6cada5b064 qtdemux: Add support for wvtt (WebVTT) subtitles.
WebVTT in ISO MP4 is specified in ISO 14496-30,
and needed for DASH support. It's stored in an
mp4 specific format. To handle it compatibly,
the wvtt boxes are converted back into WebVTT text
and pushed as application/x-subtitle-vtt

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>
2021-10-19 08:56:58 +00:00
Jan Schmidt
41d6f47f2b isomp4: Split buffer process functions.
Split the different handling for closed captions, VobSub subpicture
and timed text samples into separate simplified process functions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>
2021-10-19 08:56:58 +00:00
Jan Schmidt
c47c6b4c2b isomp4: Use a function pointer for buffer splitting.
Swap the `need_process` boolean check on qtdemux streams
for a direct function pointer to the splitting function,
so we can stop adding extra cases to the single growing
`gst_qtdemux_process_buffer()` function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>
2021-10-19 08:56:58 +00:00
Thibault Saunier
20f6af7e3c qtmux: Allow more fields changes renegotiating h264
And consider interlace-mode=progress as equivalent to the field not
being specified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1062>
2021-10-18 10:10:24 -03:00
Thibault Saunier
32a88fe16f matroskamux: Ignore some fields when renegotiating
Those values are never used in the muxer so we should not fail the
negotiation if those are changing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1062>
2021-10-18 10:10:24 -03:00
Olivier Crête
c272d0bfcd rtopuspay: Set marker bit inside RTP packet too
At the end of a talk spurt, not only set the marker flag on the
GstBuffer, but also set the bit inside the RTP header as recommended
by the RFC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1124>
2021-10-12 17:18:19 -04:00
Sebastian Dröge
2d4491e195 splitmuxsink: Don't assert on the input side if no GOP is available when shutting down
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1119>
2021-10-12 19:34:16 +00:00
Sebastian Dröge
4316be6571 splitmuxsink: Keep track of the pending input GOPs in a queue
This cleans up input GOP handling and makes it possible to handle more
than 2 pending GOPs, which could happen before if keyframes are arriving
with too short of a distance between them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>
2021-10-05 12:35:19 +00:00
Sebastian Dröge
ae8ceb801c splitmuxsink: Handle frame reordering due to B frames better
Instead of assuming that the PTS of a keyframe is the lowest PTS of a
GOP, wait until the DTS has passed this PTS and take the minimum PTS up
to that point. That way the minimum PTS of a GOP can be determined, at
least for closed GOP streams. Open GOP streams still can't be handled
properly.

By knowing the minimum PTS of each GOP, keyframes can be requested at
the correct time relative to the GOP (and thus fragment) start and
fragment overflow calculations can calculate the correct durations of
the GOPs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>
2021-10-05 12:35:19 +00:00
Sebastian Dröge
f83ed50c20 splitmuxsink: Reset timecodes when resetting the sink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>
2021-10-05 12:35:19 +00:00
Sebastian Dröge
ed6823ad67 splitmuxsink: Use GST_CLOCK_STIME_NONE instead of 0 to initialize max out running time
Otherwise streams with only DTS would misbehave while it is negative.

For outputting everything immediately at EOS, use G_MAXINT64 instead
which is bigger-or-equal to any other running time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>
2021-10-05 12:35:19 +00:00
Sebastian Dröge
eed843eea0 splitmuxsink: Only update max in/out running time if it's actually bigger
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>
2021-10-05 12:35:19 +00:00
Sebastian Dröge
68683d15c4 splitmuxsink: Only count keyframes for the reference context, consistently
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>
2021-10-05 12:35:19 +00:00
Olivier Crête
5ec82c1c4e rtphdrext: Pass just the attributes to the subclass
Since the base class now does the parsing, there is no need
to reproduce that code in all the subclasses, just pass the attributes
which are the only relevant bit anyway.

Also, only store the direction if the subclass accepted the caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
2021-09-28 20:04:55 +00:00
Olivier Crête
2db9f42b36 rtphdrext-rfc6464: Use helper function to set caps field
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
2021-09-28 20:04:55 +00:00
Olivier Crête
ba328fb98d rtphdrext: Set caps without attributes as the default
Most subclasses just use the simple function, so just let the base class
do it. It makes less code in subclasses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
2021-09-28 20:04:55 +00:00
Olivier Crête
498740fe57 rtphdrext: Put simple caps generation as the base class default
Instead of having a helper function that gets called by almost every
subclass, just let the base class set the caps fields automatically.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
2021-09-28 20:04:55 +00:00
Thibault Saunier
5ff769d731 Move files from gst-plugins-good into the "subprojects/gst-plugins-good/" subdir 2021-09-24 16:13:50 -03:00