| Commit message (Collapse) | Author | Age | Files | Lines |
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This reverts commit e9cbc89d92049ed3fdca4fdb732c90b3430583d7.
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The hash table priv->pipelined_requests is not unref:ed in the
finalize funktion. Make sure it is.
https://bugzilla.gnome.org/show_bug.cgi?id=788704
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CID 1418985
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If we have more than one dynamic payloader in the pipeline, we need
to wait until the *last* one emits 'no-more-pads' before switching
to PREPARED.
Failure to do so would result in a race where some of the streams
wouldn't properly be prepared
https://bugzilla.gnome.org/show_bug.cgi?id=769521
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The docs already mentioned this, but we actually stored it in the hash
table with key==NULL and leaked its reference forever.
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This is a simple port of:
* a722f6e8329032c6eda4865d6a07f4ba5981d7ea
* c438545dc9e2f14f657bc0ef261fff726449867b
* cd17c71dcea5c9310d21f1347c7520983e5869ac
in gst-plugins-good.
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https://bugzilla.gnome.org/show_bug.cgi?id=786457
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From 48a5d85 to dd9d403
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Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
multiudpsink's last-sample always comes from the payloader. Which
is wrong if auxiliary streams are multiplexed in the same stream.
So check the buffer's ssrc against the caps'ssrc before to use its
seqnum. If not the same ssrc just use the payloader as done prior
the commit above or when there is no last-sample yet.
https://bugzilla.gnome.org/show_bug.cgi?id=784094
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From 60aeef6 to 48a5d85
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From 39ac2f5 to 60aeef6
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Just the build dir is not going to work for srcdir!=builddir.
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Ship meson build files in tarballs, so people who use tarballs
in their builds can start playing with meson already.
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Add an example line of how to deliver a stream to the
RTSP RECORD example
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keep-alive
If there is no Content-Length header, no body would be allocated and the
'\0' would also not be appended to the body.
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While they logically have 0 bytes length, GstRTSPConnection is appending
a '\0' to everything making the size be 1 instead.
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when disabling
gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
affected.
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This bug was accidentally introduced while fixing a segfault
in _get_server_port() function.
https://bugzilla.gnome.org/show_bug.cgi?id=776345
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Calling function gst_rtsp_stream_get_server_port() results in
segmenation fault in the RTP/RTSP/TCP case.
Port that the server will use to receive RTCP makes only
sense in the UDP case, however the function should handle
the TCP case in a nicer way.
https://bugzilla.gnome.org/show_bug.cgi?id=776345
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https://bugzilla.gnome.org/show_bug.cgi?id=777037
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Generating those files is useful for users building the GStreamer stack
using meson and having to link it to another project which is still
using the autotools.
https://bugzilla.gnome.org/show_bug.cgi?id=776810
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pcfiledir was never defined so the paths were wrong.
https://bugzilla.gnome.org/show_bug.cgi?id=776867
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Make sure that the appsink element is actually added to
the bin before trying to link it with the elements in it.
https://bugzilla.gnome.org/show_bug.cgi?id=776343
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Likely extremely bitrotten, and we should not ship this anyway.
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From f980fd9 to 39ac2f5
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Since decryption is handled within rtpbin, all outcoming stream
caps will be application/x-rtp (i.e. regular rtp)
Fixes RECORD with SRTP streams
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The function called immediately afterwards (collect_streams()) will
need it to work properly
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connected
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We're going to put a pipeline into a pipeline otherwise, which is not
exactly ideal.
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https://bugzilla.gnome.org/show_bug.cgi?id=755329
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Fix a warning on shutdown - don't keep a pointer to an
alread-unreffed object.
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https://bugzilla.gnome.org/show_bug.cgi?id=775110
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85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
detection of the srtp rollover counter to add to the SDP.
Unfortunately, it was incomplete for live pipelines where the logic
blocks the source bin before creating the SDP and thus would never have
the necessary informaiton to create a correct SDP with srtp encryption.
Move the pad blocks to rtpbin's output pads instead so that the
necessary information can be created before we need the information for
the SDP.
https://bugzilla.gnome.org/show_bug.cgi?id=770239
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The RTSP server will not timeout an idle RTSP connection
(note this is different from doing timeout on a RTSP
session).
At least for Apache this is a problem when running RTSP over
HTTPS since it uses one of the threads (there is a rather
limited number) that are available for handling requests.
https://bugzilla.gnome.org/show_bug.cgi?id=771830
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