Commit graph

417 commits

Author SHA1 Message Date
Olivier Crête
c51ae6112d rtpopus: Put MULTIOPUS in all caps
The RTP payload encoding-name are always in caps in GStreamer.
In SDP, they are not case-sensitive, but since caps are, we need to pick
a caps, and we picked upper-case along time ago.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Seungha Yang
6540c4e89c rtspsrc: Fix string leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>
2022-12-28 04:39:18 +09:00
Seungha Yang
9b305df1cc rtptimerqueue: Fix memory leak
Should chain up to parent's finalize

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>
2022-12-27 19:31:16 +00:00
Patricia Muscalu
d752bf1b46 qtmux: Fix buffer leak in fragment_buffers
When pushing buffers from one of the sink pads fail,
make sure that all buffers added to fragment_buffers on other pads
are freed as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3624>
2022-12-22 14:11:10 +00:00
Mathieu Duponchelle
194dcd91e0 qtmux: For video with N/1001 framerates use N as timescale instead of centiframes
This is recommended by various specifications for such framerates, while
for integer framerates we continue using centiframes to allow for some
more accuracy.

Using N means that no rounding error accumulates, eventually leading to
outputting a packet with a different duration.

Some tools such as MediaInfo determine that a stream is variable
framerate if any packet has a different duration than the others, and
there is no reason I can see for not using the full 4 bytes of
resolution that the mp4 timescale offers.

Example problematic pipeline:

```
videotestsrc num-buffers=5001 ! video/x-raw,framerate=60000/1001,width=320,height=240 ! \
videoconvert ! x264enc bitrate=80000 speed-preset=1 tune=zerolatency ! h264parse ! \
video/x-h264,profile=high-10 ! mp4mux ! filesink location="result2.mp4"
```

This results in a media file that MediaInfo detects as variable
framerate because the 5000th packet has duration 99 instead of 100.

With this patch, the timescale is 60000 and all packets have duration
1001.

Related issue for context: https://bugzilla.gnome.org/show_bug.cgi?id=769041

Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3049>
2022-12-22 12:31:06 +02:00
Sebastian Dröge
066558cba1 qtdemux: Always use tfdt if available in BYTE segments
This reverts the decision from
  https://bugzilla.gnome.org/show_bug.cgi?id=754230
where it was decided that we rather play safe and only use the `tfdt` if
it is "significantly different" to the sum of sample durations.

As the specification says

    If the time expressed in the track fragment decode time (‘tfdt’) box
    exceeds the sum of the durations of the samples in the preceding
    movie and movie fragments, then the duration of the last sample
    preceding this track fragment is extended such that the sum now
    equals the time given in this box.

we have to use the `tfdt` in general to allow for it to signal gaps in
the stream.

A muxer producing fragments might not yet know the full duration of the
last sample of a previous fragment if the next fragment starts with a
gap, and knowing the actual start of the next fragment would potentially
require to violate latency requirements.

Additionally, the existence of `tfdt` allows to avoid accumulating
rounding errors from summing up the durations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3586>
2022-12-17 19:26:19 +02:00
Xabier Rodriguez Calvar
87ae60176b qtdemux: Clear protection events when we get new ones
If we keep the old events they can be end up being passed to the app, that could
discard the protection information because it has been seen before.

Drive by improvement: use g_queue_clear_full instead of foreach+clear for
protection events.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3547>
2022-12-14 11:01:23 +01:00
Mathieu Duponchelle
fa71217502 rtpvp9depay: expose keyframe-related properties
This simply brings in the wait-for-keyframe and request-keyframe
properties from rtpvp8depay.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/909>
2022-12-10 13:28:07 +00:00
Jacek Skiba
61c17c5665 qtdemux: exit when protection caps are not defined during PIFF parsing
Reproduction testcase (uses PlayReady):
https://developers.canal-plus.com/rx-player/upc/?appTileLocation=[object%20Object]

In test streams we are using PIFF box, but caps did not had
present GST_PROTECTION_SYSTEM_ID_CAPS_FIELD. In consequence, invalid
system_id was returned which caused SIGSEGV crash.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3535>
2022-12-07 18:35:37 +00:00
Philippe Normand
b9011f3541 flacparse: Fix handling of headers advertising 32bps
According to the flac bitstream format specification, the sample size in bits
corresponding to `111` is 32 bits per sample.

https://xiph.org/flac/format.html#frame_header

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3517>
2022-12-04 11:47:57 +00:00
Aleksandr Slobodeniuk
38f6a0ba2e rtspsrc: fix seek event leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3500>
2022-12-01 23:52:40 +00:00
Matt Crane
ca7f66f9b5 rtpsession: Support disabling late adjustment of ntp-64 header ext
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.

This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
2022-11-24 08:23:03 +00:00
Johan Sternerup
9794c9bfd0 Use the correct SSRC(s) when routing a RTCB FB FIR
Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>
2022-11-23 11:31:23 +00:00
Jan Schmidt
cb225b3682 rtpsource: Track the seqnum for senders
RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.

Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>
2022-11-23 10:26:29 +00:00
Jan Alexander Steffens (heftig)
1d7c936db0 rtspsrc: Don't replace 404 errors with "no auth protocol found"
When getting a "404 Not Found" response from the DESCRIBE request, the
source produced a "No supported authentication protocol was found" error
instead of passing on the 404, which was confusing.

Only produce this error message when we're handling a response of "401
Unauthorized" without a compatible WWW-Authenticate header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3414>
2022-11-22 13:07:17 +00:00
Edward Hervey
f3c2f612ce rtspsrc: Don't leak sticky events
We have incremented the reference 2 lines above, and
gst_pad_store_sticky_event() does not take a reference, therefore release it

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>
2022-11-21 19:02:44 +00:00
Vivia Nikolaidou
f29c19be58 splitmuxsink: Avoid assertion when WAITING_GOP_COLLECT on reference context
I have seen a backtrace out in the wild where this happened. Maybe after
receiving EOS and stream-start on the reference context.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3005>
2022-11-18 15:52:03 +00:00
Edward Hervey
845dcf7ec5 imagesequencesrc: Don't leak caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3428>
2022-11-18 07:22:23 +00:00
Matthew Waters
8e355d23a1 qtmux: use trun with multiple entries in more cases
The only case where we definitely need to write a new trun is when the
data_offset value does not match the end of the list of entries.
Needing multiple trun atoms is required when interleaving multiple
streams together.

All other cases can be covered by adding more entries to the existing
trun atom.

Fixes playback of fragemented mp4 in ffplay and chrome.

Using e.g. mp4mux fragment-duration=1000 fragment-mode=dash-or-mss
and
mp4mux fragment-duration=1000 fragment-mode=first-moov-then-finalise

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3426>
2022-11-17 21:04:57 +11:00
Nirbheek Chauhan
13723198a1 rtspsrc: Fix regression when using hostname in the location property
When the address can't be parsed as an IP address, it should just be
treated as a hostname and used as-is.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1576

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3420>
2022-11-16 11:30:26 +00:00
Sebastian Dröge
3d79402344 rtpjitterbuffer: Reschedule timers when updating their offset
As EXPECTED timers are skipped the order of the timers relative to each
other can change if there are EXPECTED timers and rescheduling needs to
happen.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1422

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3416>
2022-11-16 08:26:41 +00:00
Sanchayan Maity
02fd7fb777 wavparse: Do not run all typefinders for all output
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, wavparse calls the typefinder helper
except that means it runs all typefinders.

Since it only cares about checking for DTS, we should only run the
audio/x-dts typefinder (if present). Commit 858e516 did not really
fix things.

Use the new type helper with the caps to fix this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3417>
2022-11-16 10:32:25 +05:30
Sebastian Dröge
424e208170 rtspsrc: Consistently set seqnums on events
Set udpsrc seqnums on all events sent to the udpsrc's, and before
forwarding events out of rtspsrc set the latest seek seqnum on them if
any.

Also produce a consistent seqnum in rtspsrc from the very beginning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
e6efd288c2 rtspsrc: Make segment event writable before overriding the seqnum and use the proper API to do so
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
4099fd064b rtspsrc: Intercept and handle events when using no manager too
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
e6a2e41c06 rtspsrc: Don't blindly copy over sticky events from manager pad to external source pad
This would get around the code that modifies some events when they go
through the ghost pad's proxypad. Instead go via the event function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
a4674a1e17 rtspsrc: Don't make udpsrc segment events writable just to retrieve their seqnum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
b181686211 rtspsrc: Reset EOS flag also on FLUSH_STOP and not only on ssrc-active
Also don't bother not sending EOS if EOS was sent already:
gst_pad_push_event() takes care of that for us already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Edward Hervey
30886fa9ea rtpjitterbuffer: Unlock timer waits on flushing
If there is a pending EOS wait for example, we would never unblock on flushing

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3401>
2022-11-15 18:30:43 +00:00
Víctor Manuel Jáquez Leal
64cb38685b matroskademux: Handle element's duration query.
This is small regression from commit f7abd81a.

When calling `gst_element_query()` no pad is associated with that query, but the
current code always forwards the query to the associated pad, which is NULL in
previous case. This patch checks for the pad before forwarding the query.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3404>
2022-11-14 15:10:44 +00:00
Colin Kinloch
99fc124f25 videocrop, videobox: Simplify navigation event handling and support touch events
Signed-off-by: Colin Kinloch <colin.kinloch@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3053>
2022-11-11 06:45:49 +00:00
Colin Kinloch
d7aba91518 videoflip: Use gst_video_orientation_from_tag to parse orientation
Signed-off-by: Colin Kinloch <colin.kinloch@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3053>
2022-11-11 06:45:48 +00:00
Christian Wick
2498457b2f rtspsrc: Introduce new action signal push-backchannel-sample with correct ownership semantics
Signals are not supposed to take ownership of their arguments but only
borrow them for the scope of the signal emission.

The old action signal `push-backchannel-buffer` is now marked deprecated.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3363>
2022-11-10 13:04:04 +02:00
Justin Chadwell
fd96fc23c5 qtdemux: use unsigned int types to store result of QT_UINT32
In a few cases throughout qtdemux, the results of QT_UINT32 were being
stored in a signed integer, which could cause subtle bugs in the case of
an integer overflow, even allowing the the result to equal a negative
number!

This patch prevents this by simply storing the results of this function
call properly in an unsigned integer type. Additionally, we fix up the
length checking with stsd parsing to prevent cases of child atoms
exceeding their parent atom sizes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3344>
2022-11-06 12:00:31 +00:00
Sebastian Dröge
b368a5fcd2 qtmux: Add durations to raw audio buffers from the raw audio adapter in prefill mode
This ensures that a duration can also be calculated and stored for the
last buffer at EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3321>
2022-11-04 19:02:22 +00:00
Sebastian Dröge
7b60e48c8c qtmux: Release object lock before posting an error message
GST_ELEMENT_ERROR() also takes the object lock and this would then
deadlock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3321>
2022-11-04 19:02:22 +00:00
Edward Hervey
97bfb8b6cb imagesequencesrc; Fix leaks
* The path was leaked
* The custom buffer was never freed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Edward Hervey
6ffae88a9f qtdemux: Fix cenc-related leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Edward Hervey
aa61662632 deinterlace: Don't leak metas
There is no correlation between the frame being NULL and the metas not being
present.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Sanchayan Maity
858e516383 wavparse: Speed up type finding for DTS
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, right now we call the typefinder helper
which runs all typefinders.

Speed up this type finding process by specifying the extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3294>
2022-10-28 19:01:26 +05:30
Matthew Waters
e2081ce31e mp4mux: enable muxing VP9 streams
As specified in https://www.webmproject.org/vp9/mp4/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3260>
2022-10-28 00:06:07 +00:00
Matthew Waters
5bed545113 qtmux: add support for writing vpcC box for VP9
Increases compatibility for VP9 in .mov in at least VLC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3260>
2022-10-28 00:06:07 +00:00
Thibault Saunier
f7abd81a45 matroskademux: Let upstream handle seeking/duration query in time if possible
So proper response are given for dash streams

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
2022-10-27 19:45:44 +00:00
Thibault Saunier
8c7579e129 matroskademux: Start support for upstream segments in TIME format
So we can use matroskademux for dash webm dash streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
2022-10-27 19:45:44 +00:00
Tim-Philipp Müller
d132592423 xingmux: move from gst-plugins-ugly to gst-plugins-good
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/415

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3251>
2022-10-25 12:40:20 +00:00
Sebastian Dröge
e392d9c597 rtspsrc: Only EOS on timeout if all streams are timed out/EOS
Otherwise a stream that is just temporarily inactive might time out and
then can never become active again because the EOS event was sent
already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3238>
2022-10-24 09:19:12 +00:00
Matthew Waters
093e9c8c9d rtpulpfecdec: add property for passthrough
Support for enabling and disabling decoding of FEC data decoding on
packet loss events and unconditional seqnum rewriting of packets.

See
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/581
for background.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3212>
2022-10-23 23:44:07 +00:00
Devin Anderson
31b244271e wavparse: Avoid occasional crash due to referencing freed buffer.
We've seen occasional crashes in the `wavparse` module associated with
referencing a buffer in `gst_wavparse_chain` that's already been freed.  The
reference is stolen when the buffer is transferred to the adapter with
`gst_adapter_push` and, IIUC, assuming the source doesn't hold a reference to
the buffer, the buffer could be freed during interaction with the adapter in
`gst_wavparse_stream_headers`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3179>
2022-10-14 07:54:03 +00:00
Devin Anderson
4e03c5f885 wavparse: Fix crash that occurs in push mode when header chunks are corrupted
in certain ways.

In the case that a test is provided for, the size of the `fmt ` chunk is
changed from 16 bytes to 18 bytes (bytes 17 - 20 below):
```
$ hexdump -C corruptheadertestsrc.wav
00000000  52 49 46 46 e4 fd 00 00  57 41 56 45 66 6d 74 20  |RIFF....WAVEfmt |
00000010  12 00 00 00 01 00 01 00  80 3e 00 00 00 7d 00 00  |.........>...}..|
00000020  02 00 10 00 64 61 74 61                           |....data|
00000028
```

(Note that the original file is much larger.  This was the smallest sub-file
I could find that would generate the crash.)

Note that, while the same issue doesn't cause a crash in pull mode, there's a
different issue in that the file is processed successfully as if it was a .wav
file with zero samples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3173>
2022-10-13 08:56:49 +00:00
Mathieu Duponchelle
cddb0e951f splitmuxsrc: don't queue data on unlinked pads
Once a pad has returned NOT_LINKED, the part reader shouldn't let its
corresponding data queue run full and eventually (after 20 seconds)
stall playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3145>
2022-10-10 18:11:12 +00:00
Sebastian Dröge
bd5a4d321b rtpsource: Don't do probation for RTX sources
Disable probation for RTX sources as packets will arrive very
irregularly and waiting for a second packet usually exceeds the deadline
of the retransmission.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:18 +00:00
Sebastian Dröge
72b6dabd32 rtpsession: Remember the corresponding media SSRC for RTX sources
This allows timing out the RTX source and sending BYE for it when the
actual media source belonging to it is timed out.

This change only applies to sending sources from this session.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:17 +00:00
Sebastian Dröge
d5c072fadd rtpsource: Rename rtp_source_update_caps to rtp_source_update_send_caps
To make it clear that this is only used for sending RTP sources.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:17 +00:00
Sebastian Dröge
97a47341a7 rtpsession: Rename gst_rtp_session_sink_setcaps to gst_rtp_session_setcaps_recv_rtp
to make it clearer that this is for setting receiver caps and to make it
more consistent with gst_rtp_session_setcaps_send_rtp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:17 +00:00
Sebastian Dröge
bacd92274d rtspsrc: Retry SETUP with non-compliant URL resolution on "Bad Request" and "Not found"
Various RTSP servers/cameras assume base and control URL to be simply
appended instead of being resolved according to the relative URL
resolution algorithm as mandated by the RTSP specification.

To work around this, try using such a non-compliant control URL if the
server didn't like the URL used in the first SETUP request.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1447
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/922

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3127>
2022-10-07 09:12:00 +00:00
Edward Hervey
f2a1769236 qtdemux: Don't stop task when resetting
This is a regression that was introduced in
cca2f555d1 (yes, 9 years ago).

The only place where a demuxer streaming thread should be stopped is when the
sinkpad is deactivated from pull mode (i.e. PAUSED->READY).

Attempting to stop the task in this function would cause this to happen when a
FLUSH_STOP or STREAM_START event is received... which can cause deadlocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3109>
2022-10-03 14:41:18 +02:00
Mathieu Duponchelle
f8d8d67b8b splitmuxsrc: don't consider unlinked pads when deactivating part
If splitmuxsrc exposes multiple pads, but only one is linked, part pads
will never see an EOS event. This shouldn't prevent the part from being
eventually deactivated.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3099>
2022-10-01 02:33:08 +00:00
Nirbheek Chauhan
0aa9d8ade6 rtspsrc: Fix usage of IPv6 connections in SETUP
If the SETUP request returns an IPv6 server address in the Transport
field, we would generate an incorrect URI, and multiudpsink would fail
to initialize:

```
     rtspsrc gstrtspsrc.c:9780:dump_key_value:<source>    key: 'Transport', value: 'RTP/AVP;unicast;source=fe80::dc27:25ff:fe5e:bd13:8080;client_port=62696-62697;server_port=4000-4001'
...
     rtspsrc gstrtspsrc.c:4595:gst_rtspsrc_stream_configure_udp_sinks:<source> configure RTP UDP sink for fe80::dc27:25ff:fe5e:bd13:8080:4000
...
multiudpsink gstmultiudpsink.c:1229:gst_multiudpsink_configure_client:<udpsink0> error: Invalid address family (got 23)
```

We can't look at stream->is_ipv6 because we can't rely on the server
returning the right value there. In the issue reported about this,
server reported itself as `KuP RTSP Server/0.1`, and the SDP was:

```
c=IN IP4
m=video 54608 RTP/AVP 96
a=rtpmap:96 H264/90000
```

So we need to parse the string value and figure out the family
ourselves.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1058

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1819>
2022-09-27 18:59:59 +00:00
Tim-Philipp Müller
02a8f9973b qtdemux: guard against timestamp calculation overflow in gap event loop
Could possibly cause an endless loop.

Fixes #1400.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3084>
2022-09-27 13:07:15 +00:00
Matt Crane
e64a5b9a85 rtpjitterbuffer: Fix calculation of reference timestamp metadata
Add support for RTCP SRs that contain RTP timestamps later than the
current timestamps in the RTP stream packet buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3019>
2022-09-12 20:17:08 +00:00
Sebastian Dröge
648b8f3362 rtpjitterbuffer: Make it more explicit that update_rtx_timers() takes ownership of the passed in timer
It is not valid anymore afterwards and must not be used, otherwise an
already freed pointer might be used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2973>
2022-09-03 09:26:24 +00:00
Sebastian Dröge
e66f5e2423 rtpjitterbuffer: Don't shadow variable
While this didn't cause any problems in this context it is simply
confusing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2973>
2022-09-03 09:26:24 +00:00
Sebastian Dröge
0b19c457ca rtpjitterbuffer: Change RTX timer availability checks to assertions
It's impossible to end up in the corresponding code without a timer for
RTX packets because otherwise it would be an unsolicited RTX packet and
we would've already returned early.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2973>
2022-09-03 09:26:24 +00:00
Sebastian Dröge
2ca849499e rtpjitterbuffer: Only unschedule timers for late packets if they're not RTX packets and only once
Timers for RTX packets are dealt with later in update_rtx_timers(), and
timers for non-RTX packets would potentially also be unscheduled a
second time from there so avoid that.

Also don't shadow the timer variable from the outer scope but instead
make use of it directly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2973>
2022-09-03 09:26:24 +00:00
Patricia Muscalu
3c9e4f4886 rtph265: keep delta unit flag
Without this patch all buffers that pass the payloader
are marked as non-delta-unit buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2969>
2022-09-02 08:56:13 +00:00
Thibault Saunier
6a4425e46a meson: Call pkgconfig.generate in the loop where we declare plugins dependencies
Removing some copy pasted code

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
2022-09-01 21:17:35 +00:00
Raul Tambre
e1d3612321 rtpjitterbuffer: remove lost timer for out of order packets
When receiving old packets remove the running lost timer if present.
This fixes incorrect reporting of a lost packet even if it arrived in time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2922>
2022-09-01 09:01:31 +00:00
Sebastian Dröge
cbc6761199 rtpvp8depay: If configured to wait for keyframes after packet loss, also do that if incomplete frames are detected
This can happen if the data inside the packets is incomplete without the
seqnums being discontinuous because of ULPFEC being used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2947>
2022-08-31 08:58:03 +00:00
Mathieu Duponchelle
8756f523d1 playback: add onvif metadata caps to raw caps
+ remove encoding from x-onvif-metadata caps output by qtdemux

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2889>
2022-08-24 12:21:18 +03:00
zhiyuan.liu
ffebd52e46 isoff: Fix earliest pts field parse issue
earliest pts will be covered by first_offset field on version 0 case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2927>
2022-08-23 10:59:56 +00:00
Jan Schmidt
4a6c2e6720 splitmuxsrc: Stop pad task before cleanup
When stopping the element, make sure the pad task
is stopped before destroying the part readers.

Closes a race where the pad task might access
a freed pointer.

Also add a guard against this sort of thing
by holding a ref to the reader in the pad loop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2901>
2022-08-17 09:42:50 +10:00
Jan Schmidt
c2fa0b50ce qtdemux: Avoid crash on reconfiguring.
When reconfiguring a stream that never created
an output pad, don't access a NULL GstPad pointer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2869>
2022-08-16 19:01:28 +00:00
Sebastian Dröge
a3037eb453 qtdemux: Set parsed=true on ONVIF Timed Metadata caps
Inside MP4 the metadata must be properly parsed into frames and in
order.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2897>
2022-08-16 18:11:53 +00:00
Sebastian Dröge
8e77c8155c rtspsrc: Consider the actual control base URI also in case the connection URI contains a query string
That is, get rid of unnecessary and wrong special-casing.

This could always use gst_rtsp_url_get_request_uri_with_control() but as
we only have the control base URI as string it is easier to just call
gst_uri_join_strings().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2868>
2022-08-12 18:52:29 +00:00
Sebastian Dröge
b0533d1ea0 qtdemux: Add reference timestamp meta with UTC times based on the ONVIF Export File Format CorrectStartTime box to outgoing buffers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2525>
2022-08-12 16:13:50 +00:00
Nirbheek Chauhan
d8c4ebccab rtpst2022-1-fecenc: Drain column packets on EOS
Otherwise we won't send the protection packets for the last few
packets when a stream ends.

Also send EOS on the FEC src row pad immediately, and on the FEC src
column pad after draining is complete. This makes it so that the FEC
src pads on rtpbin behave the same way as the RTCP src pads on rtpbin
when EOS is received on the send_rtp_sink pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2863>
2022-08-12 12:59:19 +00:00
Edward Hervey
63dcee34fb qtdemux: Don't use invalid values from failed trex parsing
If parsing the fragment default values (`trex` atom) failed, don't try to
compute a bogus sample_description_id value.

Fixes #1369

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2860>
2022-08-11 08:50:34 +02:00
Piotr Brzeziński
c883a9f54b videoflip: Add support for 10/12bit planar formats
Implements support for I420, I422 and Y444 in 10/12 bit LE/BE variants.
I422 is handled separately from the rest, as it needs to consider
the endianness of the current format during most transforms.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2788>
2022-08-10 10:52:27 +00:00
Haihua Hu
82025897c4 alpha: fix stride issue when out buffer has padding on right
if outbuf has padding on right, need jump to next line use stride,
otherwise downstream element will show a wrong picture when use the
same stride

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2842>
2022-08-09 13:04:11 +08:00
Nirbheek Chauhan
5da9f62313 rtsp+rtmp: Forward warning added to tls-validation-flags to our users
With the 2.72 release, glib-networking developers have decided that
TLS certificate validation cannot be implemented correctly by them, so
they've deprecated it.

In a nutshell: a cert can have several validation errors, but there
are no guarantees that the TLS backend will return all those errors,
and things are made even more complicated by the fact that the list of
errors might refer to certs that are added for backwards-compat and
won't actually be used by the TLS library.

Our best option is to ignore the deprecation and pass the warning onto
users so they can make an appropriate security decision regarding
this.

We can't deprecate the tls-validation-flags property because it is
very useful when connecting to RTSP cameras that will never get
updates to fix certificate errors.

Relevant upstream merge requests / issues:

https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214

https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179

https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
2022-07-30 11:27:12 +00:00
Mark Nauwelaerts
b5707e2371 videobox: avoid dropping caps fields for passthrough caps transform
Fixes potential negotiation failure in case downstream element
is a bit picky regarding the fields in question.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2786>
2022-07-29 18:44:13 +00:00
Adrian Fiergolski
8e6872a36e videoflip: Fix caps negotiation when method is selected
The caps negotiation should respect the selected method to the test pipeline below works properly.
gst-launch-1.0 videotestsrc ! video/x-raw,width=320,height=600 ! videoflip method=clockwise ! video/x-raw,width=600,height=320 ! fakesink

Signed-off-by: Adrian Fiergolski <adrian.fiergolski@fastree3d.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2803>
2022-07-28 00:00:47 +00:00
Jan Schmidt
ab459f0528 splitmuxsink: Fix memory leak
Fix a leak of the buffer info struct when reaching
EOS without data on the reference input.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2751>
2022-07-12 11:22:33 +00:00
Sebastian Dröge
eb0746ba97 rtpjitterbuffer: Fix calculation of RFC7273 RTP time period start
This has to be based directly on the current estimated clock time and
has to allow for negative period starts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2655>
2022-07-11 15:33:42 +00:00
Seungha Yang
b233df3537 splitmuxsink: Don't crash on EOS without buffer
Fix a case where upstream pushed EOS without buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2174>
2022-07-05 11:33:35 +00:00
Thibault Saunier
339f950e79 rtprtx: Fix copying extension headers
There was a typo leading to reading memory from the buffer we were
writing to.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2696>
2022-07-04 19:20:57 +00:00
Marc Leeman
db5a4b490d rtpsession: properly initialise favor-new property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2625>
2022-06-17 13:05:18 +00:00
Sebastian Dröge
cf887f1b8e matroskademux: Avoid integer-overflow resulting in heap corruption in WavPack header handling code
blocksize + WAVPACK4_HEADER_SIZE might overflow gsize, which then
results in allocating a very small buffer. Into that buffer blocksize
data is memcpy'd later which then causes out of bound writes and can
potentially lead to anything from crashes to remote code execution.

Thanks to Adam Doupe for analyzing and reporting the issue.

CVE: CVE-2022-1920

https://gstreamer.freedesktop.org/security/sa-2022-0004.html

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1226

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2612>
2022-06-15 18:35:12 +00:00
Sebastian Dröge
14d306da6d qtdemux: Fix integer overflows in zlib decompression code
Various variables were of smaller types than needed and there were no
checks for any overflows when doing additions on the sizes. This is all
checked now.

In addition the size of the decompressed data is limited to 200MB now as
any larger sizes are likely pathological and we can avoid out of memory
situations in many cases like this.

Also fix a bug where the available output size on the next iteration in
the zlib decompression code was provided too large and could
potentially lead to out of bound writes.

Thanks to Adam Doupe for analyzing and reporting the issue.

CVE: tbd

https://gstreamer.freedesktop.org/security/sa-2022-0003.html

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1225

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2610>
2022-06-15 17:50:55 +00:00
Sebastian Dröge
ad6012159a matroskademux: Fix integer overflows in zlib/bz2/etc decompression code
Various variables were of smaller types than needed and there were no
checks for any overflows when doing additions on the sizes. This is all
checked now.

In addition the size of the decompressed data is limited to 120MB now as
any larger sizes are likely pathological and we can avoid out of memory
situations in many cases like this.

Also fix a bug where the available output size on the next iteration in
the zlib/bz2 decompression code was provided too large and could
potentially lead to out of bound writes.

Thanks to Adam Doupe for analyzing and reporting the issue.

CVE: CVE-2022-1922, CVE-2022-1923, CVE-2022-1924, CVE-2022-1925

https://gstreamer.freedesktop.org/security/sa-2022-0002.html

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1225

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2610>
2022-06-15 17:50:55 +00:00
Sebastian Dröge
f503caad67 avidemux: Fix integer overflow resulting in heap corruption in DIB buffer inversion code
Check that width*bpp/8 doesn't overflow a guint and also that
height*stride fits into the provided buffer without overflowing.

Thanks to Adam Doupe for analyzing and reporting the issue.

CVE: CVE-2022-1921

See https://gstreamer.freedesktop.org/security/sa-2022-0001.html

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1224

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2608>
2022-06-15 16:40:48 +00:00
Adam Doupe
be11a6e26b smpte: Fix integer overflow with possible heap corruption in GstMask creation.
Check that width*height*sizeof(guint32) doesn't overflow when
allocated user_data for mask, potential for heap overwrite when
inverting.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1231

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2603>
2022-06-15 14:53:50 +00:00
Tim-Philipp Müller
9d9e59622f Bump GLib requirement to >= 2.62
Can't require 2.64 yet because of
https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
2022-06-10 06:01:41 +00:00
Marc Leeman
8bdf7e8ad8 fix trivial distination -> destination
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2573>
2022-06-08 14:40:09 +02:00
Sebastian Dröge
47aab6c832 flvdemux: Make use of the streams API if used in a streams-aware bin
This allows adding audio/video streams after 6s.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2559>
2022-06-07 10:52:46 +00:00
Jan Alexander Steffens (heftig)
637406cdb1 aacparse: Avoid mismatch between src_caps and output_header_type
If our downstream caps didn't intersect, we attempted to convert between
raw and ADTS stream formats, if possible. If the caps still did not
intersect, we then used the modified `src_caps` but left the
`output_header_type` unmodified.

This caused a mismatch between caps and actual stream format.

Avoid this by first copying the `src_caps` to `convcaps` for the
additional intersection tests, replacing `src_caps` if we succeed.

While we're here, clean up the code a bit and remove the `codec_data`
field from outgoing ADTS caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2550>
2022-06-06 15:09:09 +00:00
Sebastian Dröge
e5f9bb973f flvdemux: Actually make use of the debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2552>
2022-06-06 14:36:41 +00:00
Jan Schmidt
a8f18aef18 rtpptdemux: Don't GST_FLOW_ERROR when ignoring invalid packets
https://bugzilla.gnome.org/show_bug.cgi?id=741398 changed
rtpptdemux in 2014 to not post a GST_ELEMENT_ERROR on the
bus when dropping an invalid (non-RTP) packet, but still
returned GST_FLOW_ERROR upstream - so the pipeline still
stops, but now without a useful bus error.

Return GST_FLOW_OK instead, so the pipeline keeps
running. Some old telephony equipment can send invalid
packets before the real RTP traffic starts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2520>
2022-05-29 20:27:38 +10:00
Piotrek Brzeziński
5490189b9b cutter: Include running/stream-time in messages
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2490>
2022-05-25 12:27:10 +00:00
Sebastian Dröge
7273024ae5 qtdemux: Add support for ONVIF XML Timed MetaData
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2453>
2022-05-20 13:01:44 +00:00
Sebastian Dröge
365a9af9c5 qtdemux: Add parsing/dumping of nmhd / metx boxes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2453>
2022-05-20 13:01:44 +00:00
Sebastian Dröge
04f6258863 qtdemux: Parse styp box for informational purposes
And include some more details in the debug logs for the ftyp box too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2453>
2022-05-20 13:01:44 +00:00
Jan Schmidt
7322a6d004 splitmuxsrc: Re-queue sticky events after probing.
When processing the first event after probing the
file and being activated, requeue sticky events
as there's no requirement that demuxers send tag
and other events again after a seek - that's
why they're sticky.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2432>
2022-05-17 11:55:40 +00:00
Jan Alexander Steffens (heftig)
d0fdfa76ae deinterlace: Clean up error handling in chain and _push_history
- Consistently unref the chained buffer at the end of the chain
  function, if we're not handing it off to `gst_pad_push`. This avoids a
  few buffer leaks in the error paths in `_chain` and `_push_history`.
- When mapping the video frame fails, return a flow error instead of
  crashing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2428>
2022-05-17 10:56:23 +00:00
Jan Alexander Steffens (heftig)
718d31fe63 splitmuxsink: When flushing, exit handle_mq_input quickly
If we just break the loop, we might run into the `gop != NULL` assert
that follows it. Rather, exit immediately with flushing flow.

Also use this flushing mechanism when we release a pad. This avoids
having an extra flag.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1030>
2022-05-17 09:24:10 +00:00
Jan Alexander Steffens (heftig)
fd27ee1537 splitmuxsink: Avoid deadlock on release, harder
Unlock after broadcasting and wait for the pad to be free before
relocking the muxer, giving the input probe a chance to react to our
broadcast.

Improves the fix from
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/838.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1030>
2022-05-17 09:24:10 +00:00
Shingo Kitagawa
92c0a462ae wavparse: fix typo in debug message
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2425>
2022-05-16 19:31:18 +09:00
Thibault Saunier
1cb4c050d0 rtpbin: Avoid holding lock GST_RTP_BIN_LOCK when emitting pad-added
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2411>
2022-05-13 06:25:03 +00:00
Sebastian Dröge
1223324246 qtdemux: Don't use tfdt for parsing subsequent trun boxes
The timestamp in the tfdt refers to the first trun box and if there are
multiple trun boxes then the distance between the first timestamps will
grow.

At some point this distance reaches a threshold and triggers the
resetting of the first sample's timestamp of this trun box to be reset
to the tfdt.

This threshold is implemented for files where there is a jump in the
timeline between fragments and where this can be detected via a jump
between the end timestamp of the previous fragment and the tfdt of the
next. This behaviour is preserved.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2409>
2022-05-13 04:19:36 +00:00
Sebastian Dröge
d2c6f21fc1 mp4mux: Disable aggregator's default negotiation
mp4mux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.

By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
2022-05-05 17:41:58 +00:00
Sebastian Dröge
841cba4182 flvmux: Disable aggregator's default negotiation
flvmux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.

By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
2022-05-05 17:41:58 +00:00
Matthew Waters
f4f342aa78 wavparse: ensure that any pending segment is sent before an EOS event is sent
Specifically fixes seqnum handling when an aggregator-based element
(audiomixer et al) is downstream and a seek is performed that
immediately causes an EOS from wavparse.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2356>
2022-05-04 08:00:02 +00:00
Sebastian Dröge
7466444b63 rtpjitterbuffer: Free CNAME/SSRC mappings on finalize and PAUSED->READY
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2336>
2022-04-29 23:33:47 +03:00
Sebastian Dröge
2c405da921 rtpmanager: Refactor RTCP packet loops to fix control flow
Mixing C loops with switch statements is a bad idea as break has a
different meaning in both. Breaking inside the switch statements wrongly
caused further loop iterations.

Instead use goto to get out of the loop and continue to do another loop
iteration, and never ever use break except for the end of a case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2336>
2022-04-29 23:13:15 +03:00
Seungha Yang
6619f1611f rtpjitterbuffer: Initialize variables
Avoid use of uninitialized variable
Fixing MSVC warning
gstrtpjitterbuffer.c(4733) : warning C4700: uninitialized local variable 'have_sdes' used

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2315>
2022-04-28 12:37:13 +00:00
dongil.park
5b11e6a3d0 wavparse: Unset DISCONT buffer flag for divided into multiple buffers in push mode
In push mode (streaming), if the received chunk buffer size from _chain is bigger
than output buffer size, the flags of the divided-buffers are propagated to the
DISCONT flag from first received chunk buffer. This unexpected buffers contained DISCONT
flags are abnormally transformed when changing the sampling rate by audioresample element.
So unset unnecessary DISCONT flag before pad_push().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2305>
2022-04-27 14:29:10 +00:00
Sebastian Dröge
9d5179ad3f rtpjitterbuffer: add the reference timestamp meta in more situations
Previously, we only added it when actually performing synchronization
based on the NTP time.

The information can be useful downstream in other situations too, and
we can compute a NTP time as soon as we get a sender report with the
relevant information.

Co-authored-by: Mathieu Duponchelle <mathieu@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2252>
2022-04-27 12:35:21 +00:00
Sebastian Dröge
ed425e2785 rtpgstpay: Don't push packets before the first input buffer is received
It's not possible to create a valid RTP timestamp for them, which would
cause a potentially very big RTP timestamp discontinuity between those
first packets (created from initial events) and the packet based on the
first input buffer.

As a side-effect, also simplify the packet aggregation code a bit and
work with only a single level of buffer lists.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2250>
2022-04-27 11:55:17 +00:00
Havard Graff
390ec99f1b rtptwcc: don't map the buffer twice
...and use the pt extracted rather than the one from RTPPacketInfo
when logging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2271>
2022-04-26 10:27:25 +00:00
Thibault Saunier
d673a90aea rtpsession: Emit "notify::stats" when we update stats from RR or SR
Sensibily optimizing caching the pspecs and using them directly

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2266>
2022-04-26 08:49:42 +00:00
Mathieu Duponchelle
3391a7d499 rtpredenc: quieten warning about ignoring header extensions
Turn it into a FIXME, and only log once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2279>
2022-04-23 01:04:54 +00:00
Havard Graff
b7b71e6974 rtprtxsend: mark RTX buffers with GST_RTP_BUFFER_FLAG_RETRANSMISSION
It is useful for elements downstream from rtxsend to know if the RTP
buffer they are dealing with is an RTX buffer or not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2272>
2022-04-22 19:27:45 +00:00
Tristan Matthews
27dea62304 mp4mux: fix spelling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2241>
2022-04-22 14:07:57 +00:00
Jonas Bonn
2f6ad787b2 multiudpsink: allow binding to IPv6 address
When the sink is configured to create sockets with an explicit bind
address, then the created socket gets set to the udp_socket field
irregardless of whether the bind address indicated that the socket
family should be IPv4 or IPv6.  When binding to an IPv6 address, this
results in the following error:

gstmultiudpsink.c:1285:gst_multiudpsink_configure_client:<rtcpsink>
error: Invalid address family (got 10)

This patch adds a check of the address family being bound to and sets
the created socket to used_socket or used_socket_v6, accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1551>
2022-04-22 10:43:13 +00:00
Sebastian Dröge
02115a5efc rtpmanager: Move some duplicated constant and helper function to a single place
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
c7e12974ba rtpbin/rtpjitterbuffer: Don't parse RTCP SRs twice unless needed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
82169aa140 rtpjitterbuffer: Add property to throttle handling of RTCP SR / NTP-64 syncing
This proxies the "rtcp-sync-interval" property of rtpbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
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