Commit graph

51 commits

Author SHA1 Message Date
Mathieu Duponchelle
bc39c0f54b rtspsrc: expose property for forcing usage of non-compliant URLs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7133>
2024-08-12 20:10:45 +00:00
Mathieu Duponchelle
a20ef245a0 rtspsrc: fix invalid seqnum assertions
Upon fatal errors the loop function will first post an error message
then push out an EOS event.

An application may react immediately to the error message by setting the
state of the pipeline to NULL, meaning by the time we push out the EOS
event PAUSED_TO_READY may have reset the seek seqnum to -1.

While this is harmless, the assertion when setting an invalid seqnum
isn't tidy, fix this by simply not resetting to INVALID as it serves no
practical purpose and the next READY_TO_PAUSED will select a new seqnum
anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7032>
2024-06-14 11:28:06 +02:00
Sebastian Dröge
9b60b32cf8 rtspsrc: Only update from the Content-Base header in the initial OPTION / DESCRIBE response
Some servers send a new content base in the SETUP response, which is
just the non-aggregate control URL of the individual streams.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3563

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6926>
2024-06-01 11:30:44 +00:00
Sebastian Dröge
e65344afac rtspsrc: Handle the case of * as session-wide control URL from the SDP
Just like the comment above says this is supposed to indicate that the
same URL should be used as for the connection so far. If encountering
this case simply do nothing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6926>
2024-06-01 11:30:44 +00:00
Sebastian Dröge
e73e34fd6f rtspsrc: Also handle rtsps:// and similar URLs as absolute in other places
Previously a direct comparison with `rtsp://` was performed, which
didn't catch cases like `rtsps://`.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3563

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6926>
2024-06-01 11:30:44 +00:00
Sebastian Dröge
966c39b92e rtspsrc: Don't try the SETUP workaround for broken servers with absolute control URIs
Previously only control URIs that started with "rtsp://" were ignored
but it makes more sense to ignore all absolute URIs.

gst_uri_is_valid() conveniently checks for exactly that.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3563

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6926>
2024-06-01 11:30:43 +00:00
Sebastian Dröge
df8c29e340 rtpjitterbuffer: Set max-rtcp-rtp-sync-time to -1 (disabled)
There is generally no requirement to ignore RTCP SR if the RTP time of
the SR differs a lot from the last received RTP packet. The mapping
between RTP and NTP time stays valid until there was a stream reset, in
which case we wouldn't use that information anyway.

When using rtcp-sync-send-time=false the default of 1s difference can
easily be exceeded, e.g. if encoding of the stream after capture adds
more than 1s of latency.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
4a4eb56fc2 rtspsrc: Optionally timestamp RTP packets with their receive times in TCP/HTTP mode
Until https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509
this was accidentally done inside rtpjitterbuffer for many years, and
doing so potentially solves problems on some streams while introducing
problems on others.

Make this configurable on rtspsrc and default to not set timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6529>
2024-04-08 08:34:38 +00:00
Alexander Slobodeniuk
650534c940 rtspsrc: remove 'deprecated' flag from the 'push-backchannel-sample' signal
It seems that it was added by accident when copying from push-backchannel-buffer

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6355>
2024-03-13 19:32:46 +00:00
Nirbheek Chauhan
3bed35c342 rtspsrc: Don't invoke close when stopping if we've started cleanup
When we're doing a state change from PLAYING to NULL, first we invoke
gst_rtspsrc_loop_send_cmd_and_wait (..., CMD_CLOSE, ...) during
PAUSED_TO_READY which will schedule a TEARDOWN to happen async on the
task thread.

The task thread will call gst_rtspsrc_close(), which will send the
TEARDOWN and once it's complete, it will call gst_rtspsrc_cleanup()
without taking any locks, which frees src->streams.

At the same time however, the state change in the app thread will
progress further and in READY_TO_NULL it will call gst_rtspsrc_stop()
which calls gst_rtspsrc_close() a second time, which accesses
src->streams (without a lock again), which leads to simultaneous
access of src->streams, and a segfault.

So the state change and the cleanup are racing, but they almost always
complete sequentially. Either the cleanup sets src->streams to NULL or
_stop() completes first. Very rarely, _stop() can start while
src->streams is being freed in a for loop. That causes the segfault.

This is unlocked access is unfixable with more locking, it just leads
to deadlocks. This pattern has been observed in rtspsrc a lot: state
changes and cleanup in the element are unfixably racy, and that
foundational issue is being addressed separately via a rewrite.

The bandage fix here is to prevent gst_rtspsrc_stop() from accessing
src->streams after it has already been freed by setting src->state to
INVALID.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6302>
2024-03-11 09:15:50 +00:00
Jan Schmidt
f53dbb28b2 rtspsrc: Parse Speed/Scale before Range in responses
Parse the speed and scale in the server's response
*before* the range, so that the range start/stop
are swapped (or not swapped) correctly based
on the server's actual chosen values. Otherwise,
the old rate from the segment is used - what the
last seek asked for, but not necessarily what
the server chooses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
57013e1a7c rtspsrc: Handle queries and events with no manager
When doing direct output with no session manager, we still
want to respond to queries and events from downstream, so
install the handlers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
4d2f000125 rtspsrc: return NO_PREROLL on PLAYING->PAUSED too
When transitioning back to PAUSED and rtspsrc is live, return
NO_PREROLL so the pipeline knows to skip preroll here too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Tim-Philipp Müller
4db25f1500 rtspsrc: Consider 503 Service Not Available when handling broken control urls
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6213>
2024-03-05 17:45:18 +00:00
Nirbheek Chauhan
cf2238a522 rtspsrc: Increase rank to PRIMARY for autoplug purposes
This affects autoplug by gst_element_make_from_uri() in, for example,
uridecodebin. The element should've already been PRIMARY rank, but it
was NONE because gst_element_make_from_uri() doesn't ignore NONE rank
elements when searching for element factories, unlike decodebin.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/502

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6226>
2024-02-27 11:36:01 +00:00
Jan Schmidt
f7e494f348 rtspsrc: Reset combined flows after a seek before restarting
After a flushing seek, rtspsrc doesn't reset the last_ret value for
streams, so might immediately shut down again when it resumes pushing
buffers to pads due to a cached `GST_FLOW_FLUSHING` result

Prevent a stored flushing value from immediately stopping
playback again by resetting pad flows before (re)starting
playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6137>
2024-02-21 01:50:13 +00:00
Daniel Morin
0a55c86e6a rtspsrc: update rtsp url on redirect
- If a redirect took place on a GET when rtsp is tunneled we update the
  rtsp url too.
- log source and final destination on redirect

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5222>
2024-01-31 11:43:45 +00:00
Dan Searles
1d02d7eda0 rtspsrc: fix ttl setting for udpsink[1]
Fix ttl setting being incorrectly applied to udpsink[0] rather
than to udpsink[1].

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5955>
2024-01-23 13:54:51 +00:00
Dan Searles
da55b953a1 rtspsrc: set multicast-iface on udpsinks
Copy rtspsrc property multicast-iface to its udpsinks to
allow messages over those sinks back to the server to work (and
prevent 'Network unreachable' warnings).

Closes: #3239
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5955>
2024-01-23 13:54:51 +00:00
Daniel Moberg
0e6cd64232 rtspsrc: Property for adding custom http request headers
This commit adds a property which enables adding custom http request headers to
the rtspsrc element. Added headers will be appended to http requests
made during http tunneling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5268>
2023-09-26 06:35:43 +00:00
Tim-Philipp Müller
5bbd8c2d71 rtspsrc: use version template in user-agent property
Avoids documentation churn when the version changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5090>
2023-08-07 10:41:16 +00:00
Seungha Yang
794cde703c rtspsrc: Fix crash when is-live=false
The pad's parent (i.e., rtspsrc) can be nullptr since we add pads
later.

Co-authored-by: Jan Schmidt <jan@centricular.com>

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2751
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4965>
2023-07-05 06:48:37 +00:00
Jochen Henneberg
fd1d208446 rtspsrc: Cleanup code for next pending command
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4792>
2023-06-07 20:30:36 +00:00
Jochen Henneberg
4790a8d2be rtspsrc: Do not try send dropped get/set parameter
If the set_get_param_q has been emptied we have to reset the cached
pending command to CMD_LOOP as we will not have the request parameters
anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4792>
2023-06-07 20:30:36 +00:00
Sebastian Dröge
ef89bac181 rtspsrc: Fix handling of * control path
Regression introduced by 7f9d689572.
Thanks to Tristan Matthews for reporting this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4497>
2023-04-27 13:47:56 +00:00
Sebastian Dröge
43e4db9fc9 rtspsrc: Skip PTs with caps incompatible to the global caps
Otherwise empty caps are created while all following code assumes that
the caps will have exactly one structure, and then run into assertions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4339>
2023-04-04 22:13:59 +00:00
Tim-Philipp Müller
0fc568c6b1 gst-plugins-good: re-indent with GNU indent 2.2.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4182>
2023-03-17 03:18:54 +00:00
Matt Feury
224030ff0c rtspsrc: Consider "451: Parameter Not Understood" when handling broken control urls
similar to https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3854

it seems that some implementations return this when
the server does not implement URL handling correctly

this fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2334

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4123>
2023-03-07 10:32:32 -05:00
Sebastian Dröge
269915a51e rtspsrc: Use the correct vfunc for the push-backchannel-sample action signal
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/446

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4050>
2023-02-23 09:22:23 +00:00
Sebastian Dröge
3ca85189fd rtspsrc: Also consider "Method Not Valid In This State" error in broken control URL handling workaround
Some servers send a 455 error instead of any reasonable error when using
a correctly constructed control URL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3854>
2023-02-01 07:55:24 +00:00
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
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
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
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