Commit graph

22 commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
Thibault Saunier
5ff769d731 Move files from gst-plugins-good into the "subprojects/gst-plugins-good/" subdir 2021-09-24 16:13:50 -03:00