Commit graph

11 commits

Author SHA1 Message Date
Philippe Normand
1b01415c3b srtpenc: Fix potential leak
When attempting to process a buffer after the rtcp session was closed the output
buffer memory would remain referenced.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6029>
2024-01-31 18:46:59 +00:00
Nirbheek Chauhan
fd4828bafe meson: Add a top-level option to enable webrtc
There are a bunch of plugins that you need for webrtc support, and
it's not obvious at all to users which those are.

With this commit, srtp, sctp and dtls options will be auto-enabled if
the webrtc option is enabled.

Requires meson 1.1

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5505>
2023-10-19 06:38:45 +00:00
François Laignel
32fbad8d39 srtpdec: fix Got data flow before segment event
A race condition can occur in `srtpdec` during the READY -> NULL transition:
an RTCP buffer can make its way to `gst_srtp_dec_chain` while the element is
partially stopped, resulting in the following critical warning:

> Got data flow before segment event

The problematic sequence is the following:

1. An RTCP buffer is being handled by the chain function for the
   `rtcp_sinkpad`. Since, this is the first buffer, we try pushing the sticky
   events to `rtcp_srcpad`.
2. At the same moment, the element is being transitioned from PAUSED to READY.
3. While checking and pushing the sticky events for `rtcp_srcpad`, we reach the
   Segment event. For this, we try to get it from the "otherpad", in this case
   `rtp_srcpad`. In the problematic case, `rtp_srcpad` has already been
   deactivated so its sticky events have been cleared. We won't be pushing any
   Segment event to `rtcp_srcpad`.
4. We return to the chain function for `rtcp_sinkpad` and try pushing the
   buffer to `rtcp_srcpad` for which deactivation hasn't started yet, hence the
   "Got data flow before segment event".

This commit:

- Adds a boolean return value to `gst_srtp_dec_push_early_events`: in case the
  Segment event can't be retrieved, `gst_srtp_dec_chain` can return  an error
  instead of calling `gst_pad_push`.
- Replaces the obsolete `gst_pad_set_caps` with `gst_pad_push_event`. The
  additional preconditions checked by previous function are guaranteed here
  since we push a fixed Caps which was built in the same function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4844>
2023-06-14 11:59:33 +00:00
François Laignel
96450f4c59 srtpdec: fix assertion 'parent->numsinkpads <= 1' failed
A race condition can occur in `srtpdec` during the READY -> NULL transition:
an RTCP buffer can make its way to `gst_srtp_dec_chain` while the element is
partially stopped, resulting in the following critical warning:

> assertion 'parent->numsinkpads <= 1' failed

This can occur when the first RTCP buffer is received during the READY -> NULL
transition. If deactivation of the `rtp_srcpad` has already reached
`post_activate`, the sticky events are removed from this Pad. In this case,
`gst_srtp_dec_push_early_events` branches to the generation of a stream id
using `gst_pad_create_stream_id`. This function ensures that the element
doesn't own more than 1 sink pad. Since `srtpdec` owns two of them, the
assertion fails.

This commit uses `gst_element_decorate_stream_id` which doesn't perform this
check. The preconditions is not necessary in this particular context since the
stream id for the RTP / RTCP pads are derived from the same id.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4844>
2023-06-14 11:59:33 +00:00
Michael Olbrich
fe6b76c64e srtpdec: fix "srtp-key" check
The original code was:

if (!gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {
  goto error;
} else {
  stream->key = buf;
}

So use "srtp-key" if it is set so a non NULL buffer. The condition was
incorrectly inverted in ad7ffe64a6 to:

if (gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {
  stream->key = buf;
} ...

Fix the condition so it works as originally intended and avoid accessing
'buf' uninitialised.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4401>
2023-04-12 18:16:21 +00:00
Nirbheek Chauhan
cc3078d819 meson: Add a wrap file for libsrt2p
And allow fallback to it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3708>
2023-01-25 11:38:52 +00:00
Tim-Philipp Müller
c095a1d620 srtp: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Fabian Orccon
50c6c54675 srtp: Fix test skipping when plugin option is disabled
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3200>
2022-10-18 22:12:41 +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
Stéphane Cerveau
c77d07752a srtpdec: add counts in stats
In order to count the buffers which have been received and dropped for
decryption reason, add a stats to track it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2027>
2022-04-25 13:57:42 +00:00
Thibault Saunier
019971a3c7 Move files from gst-plugins-bad into the "subprojects/gst-plugins-bad/" subdir 2021-09-24 16:14:36 -03:00