gst_structure_get() is declared with G_GNUC_NULL_TERMINATED, ie
__attribute__((__sentinel__)), which means gcc will generate a
warning if the last parameter passed to the function is not NULL
(where a valid NULL in this context is defined as zero with any
pointer type).
The C code callers to gst_structure_get() within gst-plugins-good
use the C NULL definition (ie ((void*)0)), which is a valid sentinel.
However gstid3v2mux.cc uses the C++ NULL definition (ie 0L), which
is not a valid sentinel without an explicit cast to a pointer type.
Upstream-Status: Pending
Signed-off-by: Andre McCurdy <armccurdy@gmail.com>
Note from Edward Hervey: Patch from git.yoctoproject.org
musl libc generates warnings if <sys/poll.h> is included directly.
Upstream-Status: Pending
Signed-off-by: Andre McCurdy <armccurdy@gmail.com>
The code before copied GstStructure twice. The first time inside
gst_value_set_structure and the second time in g_value_array_append.
Optimized version does no copies, just transfers ownership to
GValueArray. It takes advantage of the fact that array has already
enough elements preallocated and the memory is zero initialized.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.
The fix is to grab the ref to the RTPSource object in
rtp_session_process_feedback.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
These are the sources we send from, so there is no reason to
report receive statistics for them (as we do not receive on them,
and the remote side has no knowledge of them).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
Whenever got new moov or new stream-start,
demux will try to expose new pad by following rule.
Comparing stream-id in the current moov with previous one, then
* If matched stream-id is found from previous one,
reuse existing pad (most common case)
* Otherwise, expose new pad with new stream-start
* No more used stream will be freed
https://bugzilla.gnome.org/show_bug.cgi?id=684790
Whenever demux got moov, demux will create new stream. Only exception is
duplicated track-id in a moov box. In that case the first stream
will be accepted. This patch is pre-work for rework of moov handling.
https://bugzilla.gnome.org/show_bug.cgi?id=684790
Currently, enable_v4l2_probe is hard-coded to "yes" on linux, platforms
arm and aarch64. This even overrides the --disable-v4l2-probe argument.
As a result, it is impossible to disable v4l2_probe. It becomes a problem
for use-cases, when startup time is critical, because the v4l2_probe
feature increases the initialization time.
This commit makes the v4l2_probe feature configurable.
On linux, platforms arm and aarch64, the default value is still "yes".
But now it can be disabled by the --disable-v4l2-probe argument.
https://bugzilla.gnome.org/show_bug.cgi?id=795200
gst_v4l2_dup() will now take care of setting
v4l2capture->no_initial_format and keep_aspect instead of doing it
manually.
Fix a typo as keep_aspect was set twice on v4l2output but never on
v4l2capture.
https://bugzilla.gnome.org/show_bug.cgi?id=795028
Supports CEA 608 and CEA 708 CC streams
Also supports usage in "Robust Prefill" mode if the incoming caption
stream is constant (i.e. there is one incoming CC buffer for each
video frame).
https://bugzilla.gnome.org/show_bug.cgi?id=606643
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
