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>
Updated API usage appropriately, and now we have a versioned package to
track breaking vs. non-breaking updates.
Deprecates a number of properties (and we have to plug in our own values
for related enums which are now gone):
* echo-suprression-level
* experimental-agc
* extended-filter
* delay-agnostic
* voice-detection-frame-size-ms
* voice-detection-likelihood
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2943>
The `switch (n_rear)` supports up to 5 rear channels, but our channel
set only had space for 3. Size the set properly to fix this.
This didn't actually cause any memory unsafety as `PUSH_CHAN` would stop
incrementing `n_rear` if the channel set is already full.
Thanks to @alatiera for noticing this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4712>
New vulkan formats don't match the number of planes with the number of memories
attached to the buffer. This patch changes the pattern of using planes for
traverse the memories with the number of attached memories.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
When transitioning from state PAUSED to READY, the sctpenc element
could previously be stuck in an endless loop trying to resend data
in case the underlying sctp stream was in the process of
resetting. usrsctp_sendv() would repeatedly return EAGAIN with the
result that 0 bytes were sent and then sctpenc would retry forever.
To bring sctpenc out of the resend loop we just need to inform the
sink pad that it is flushing, which is already done for the associated
data queue, but we also need to set the bools associated with the
sinkpads that are used as the loop criterion.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4601>
Depending on the exact output format, 0x00 may be a better default for
padding than 0x80. 0x00 is the recommended padding value when used in
CDP (and cc_data) but is not when used in s334-1a. See CTA-708-E 4.3.5
amd SMPTE 334-1-2007 5.3.2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4578>
A blocking pad probe is added on new sink pads, it's usually removed after the
caps have been negotiated or the signaling state switched to stable, but if that
never happens and the pad is released we kept the pad probe active, leaving the
pad blocked, preventing clean disposal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4529>
Current implementation can in some cases detect
that all data is sent but in reality it is not,
leading to a push to an unlinked pad.
This is a race between the probe used to track data sent and a
call to close.
This patch sends an EOS before starting the close procedure
and then waits for the EOS event to come through to the
src pad before commencing with tear down.
This ensures that any queued data before EOS is flushed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4462>
`webrtc->signaling_state` (from) and `new_signaling_state` (to) had the
same value when printed in a trace log. This commit adds a
`old_signaling_state` variable to hold the previous value, so that the
print statement works as intented.
Spotted by: Mustafa Asım REYHAN
Fixes#1802
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4362>
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>
g_string_free(.., FALSE) gives us ownership of the string
already, no need to duplicate that again with g_strdup(),
and doing so will leak the string returned by g_string_free()
here. Caught by compiler warnings in newer GLib versions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4273>
Fix compiler warnings about not using the return value when
freeing the GString segment with g_string_free(.., FALSE):
ignoring return value of ‘g_string_free_and_steal’ declared with attribute ‘warn_unused_result’
which we get with newer GLib versions. These were all harmless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4273>
In webrtc_data_channel_send functions, both data and string,
an early return on a non-open datachannel caused it to leak
the buffer used for pushing to appsrc, meaning any buffer
sent after leaving the open state was leaked in full.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4191>
