ULP FEC, as defined in RFC 5109, has the protected and protection
packets sharing the same ssrc, and a different payload type, and
implies rewriting the seqnums of the protected stream when encoding
the protection packets. This has the unfortunate drawback of not
being able to tell whether a lost packet was a protection packet.
rtpbasedepayload relies on gaps in the seqnums to set the DISCONT
flag on buffers it outputs. Before that commit, this created two
problems:
* The protection packets don't make it as far as the depayloader,
which means it will mark buffers as DISCONT every time the previous
packets were protected
* While we could work around the previous issue by looking at
the protection packets ignored and dropped in rtpptdemux, we
would still mark buffers as DISCONT when a FEC packet was lost,
as we cannot know that it was indeed a FEC packet, even though
this should have no impact on the decoding of the stream
With this commit, we consider that when using ULPFEC, gaps in
the seqnums are not a reliable indicator of whether buffers should
be marked as DISCONT or not, and thus rewrite the seqnums on
the decoding side as well to form a perfect sequence, this
obviously doesn't prevent the jitterbuffer from doing its job
as the ulpfec decoder is downstream from it.
https://bugzilla.gnome.org/show_bug.cgi?id=794909
This reverts commit af273b4de9.
While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
the opposite, just like the ONVIF standard.
Let's follow those RFCs as we're doing RTSP here, and add a property at
a later time if needed to switch to the SDP RFC behaviour.
https://bugzilla.gnome.org/show_bug.cgi?id=793964
After a CAPS event, in theory a new stream can start and it might start
with the FLAC headers again. We can't detect FLAC headers in the middle
of the stream, so we drain the parser to be able to detect either FLAC
headers after the CAPS event or the continuation of the previous stream.
This fixes for example
gst-launch-1.0 audiotestsrc num-buffers=200 ! flacenc ! c. \
audiotestsrc num-buffers=200 freq=880 ! flacenc ! c. \
concat name=c ! rtpgstpay ! udpsink host=127.0.0.1 port=5000
gst-launch-1.0 udpsrc multicast-group=127.0.0.1 port=5000 \
caps=application/x-rtp,media=application,clock-rate=90000,encoding-name=X-GST ! \
rtpgstdepay ! flacparse ! flacdec ! audioconvert ! pulsesin
Tested on linux with X11/wayland and semi-tested on Windows.
Windows crashes on item destruction however this is better than nothing.
Fix up some win32 build issues on the way with mismatched {} and
G_STMT_{START,END}
Looping the test 500 times to only execute the test once every
33 times means we inited and deinited gstreamer 467 times
for no reason at all, which was annoying when running the test
with valgrind.
gst_qt_mux_can_renegotiate () gets called everywhere following
that pattern:
return gst_qt_mux_can_renegotiate (ref(self));
This means the reference must be released both in the success
and failure cases, it was only done in the success case.
After investigating, we do dispose of the TLS connections
appropriately in the souphttpsrc test, which in turn
calls gnutls_deinit, but certificates get leaked anyway.
libv4l2 does not prevent exporting DMABuf even when emulated formats are
in use. As a side effect, userspace ends up with buffers of the original
formats which will cause issues.
https://bugzilla.gnome.org/show_bug.cgi?id=795097
BT2020_12 is not represented in V4L2, so drivers providing full colority
for BT2020 will set V4L2_XFER_FUNC_709 transfer function. To fix the
issue, we bump this to BT2020_12 if the resoltion is 4K, but we should
only do that if the colorspace is BT2020 to start with, otherwise it's
not possible to use normal BT709 for 4K 8bit formats.
So far we were only setting colorimetry for OUTPUT devices (v4l2sink or
m2m sink pad). This prevented selecting through caps negotiation the
colorimetry for CAPTURE devices (v4l2src or m2m src pad). This is rarely
selectable, but trying is harmless.
It can happen during teardown that the reference context becomes NULL.
In that case, trying to send the fragment-opened-closed message would
lead to a crash.
Corrupted files could potentially have multiple cdat/cdt2 atoms in
a sample entry, which is unclear how to handle.
Ignore repeated ones.
CID #1434162
CID #1434159
The code responsible for creating retransmitted buffers
assumed the stored buffer had been created with
rtp_buffer_new_allocate when copying the extension data,
which isn't necessarily the case, for example when
the rtp buffers come from a udpsrc.
https://bugzilla.gnome.org/show_bug.cgi?id=794958
Similar to the get-session and get-internal-session signals,
we expose a get-storage signal in addition to the
get-internal-storage signal to give access to the actual
element for applications that need to set properties on the
element, in particular "size-time"
https://bugzilla.gnome.org/show_bug.cgi?id=794910
With this the muxer is not set to NULL after each segment but instead
only flush events are sent to it to reset the EOS state.
As a result, the muxer will keep stream state and e.g. mpegtsmux will
keep the packet continuity counter continuous between segments as needed
by hlssink2.
https://bugzilla.gnome.org/show_bug.cgi?id=794816
The value stored in cenc_aux_sample_count wasn't in sync with the
parsing code that followed which checks whether all entries are
valid and present.
Only write the actual sample count when we know for sure.
CID #1427087
This exposes a new property, mtu, which is used to determine the
initial size of buffers from the buffer pool. If received data
exceeds this, the element gracefully handles that in a manner similar
to what we had previously: a large memory gets filled and reallocated
at the next call to "fill".
The default size is set to 1500, which should cover most use cases.
With contributions from Mathieu Duponchelle <mathieu@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=772841
Optimize GstUdpSrc for cache performance.
Move the hot properties, which are used by the read function, to the top:
@used_socket, @addr, @cancellable, @skip_first_bytes, @timeout,
@retrieve_sender_address.
Remove the unused property @ttl.
Where needed reorder so that holes are avoided (the 64-bit @timeout)
https://bugzilla.gnome.org/show_bug.cgi?id=772841
The samples table is sorted by DTS, not PTS. As such we can only get the
correct result when using a binary search on it, if we search for the
DTS.
Also if we only ever search for the frame, where the following frame is
the first one with a PTS after the search position, we will generally
stop searching too early if frames are reordered.
In forwards playback this is not really a problem (after the decoder
reordered the frames, clipping is happening), in reverse playback
it means that we can output one or more frames too few as we stop too
early and the decoder would never receive it.
https://bugzilla.gnome.org/show_bug.cgi?id=782118
76e458a119 changed the conditions from
"queued > threshold" to "queued >= threshold", which broke hlssink2 and
resulting in too small fragments being created although keyframes would
be at *exactly* the configured threshold.
https://bugzilla.gnome.org/show_bug.cgi?id=794440
Fix compilation with MSVC. We still assume that attribute
is supported by all other relevant compilers, which seems
to be the case since we haven't had any complaints about
similar code in rtpsbcpay.
