Commit graph

9812 commits

Author SHA1 Message Date
Mathieu Duponchelle
f554369ed5 doc: remove xml from comments 2019-05-29 22:20:40 +02:00
Sebastian Dröge
cced65ee21 matroskamux: Add new property to offset all streams to start at zero
This takes the timestamp of the earliest stream and offsets it so that
it starts at 0. Some software (VLC, ffmpeg-based) does not properly
handle Matroska files that start at timestamps much bigger than zero.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/449
2019-05-29 11:53:02 +00:00
Tim-Philipp Müller
b47f3c9c50 rtpmp4gdepay: don't spam debug log for broken ADTS-in-RTP AAC
Print warning only once.
2019-05-28 19:28:05 +00:00
Sebastian Dröge
32c465a537 splitmuxsink: Only set running time on finalizing sink element when in async-finalize mode
There is only a single sink element in async-finalize mode, and we would
keep the running time from previous fragments set in that case. As we
don't ever set the running time for the very last fragment on EOS, this
would mean that the closing time reported for the very last fragment is
the same as the closing time of the previous fragment.
2019-05-28 17:21:06 +03:00
Nicolas Dufresne
301a46bd2d rtspsrc: Remove uneeded keep-alive hack
The rtsp connection code has been fixed now.

https://bugzilla.gnome.org/show_bug.cgi?id=744209
2019-05-27 16:04:23 +02:00
Vivia Nikolaidou
987230a759 rtpjitterbuffer: Print GstClockTimeDiff as GST_STIME_FORMAT 2019-05-26 17:46:06 +03:00
Mathieu Duponchelle
81dd2db06b videomixer: the documentation for GstVideoMixer2Pad is not exposed 2019-05-25 17:25:02 +02:00
Mathieu Duponchelle
d704790519 doc: fix element section documentations
Element sections were not rendered anymore after the hotdoc
port, fixing this revealed a few incorrect links.
2019-05-25 16:57:31 +02:00
Nicolas Dufresne
4e0bdca3f0 rtpbin: Improve RTPStorage action signal documentation
This is a tiny clarification as the storage was loosely named "storage".
This change clarify that the storage is specificaly used for received RTP
packets. This is unlike the storage found in rtprtxsend that stores a
backlog of sent RTP packets.
2019-05-25 13:44:00 +02:00
Seungha Yang
1ae4814a74 matroska: Add BT2020_10, PQ and HLG transfer functions
The direct use of newly added transfer functions
2019-05-24 16:32:38 +09:00
Seungha Yang
d2cac61113 multifilesink: Fix documentation of max-file-duration property
The max-file-duration property works with max-duration mode
2019-05-22 11:03:34 +09:00
Nicolas Dufresne
947a37f3c8 rtpsession: Always keep at least one NACK on early RTCP
We recently added code to remove outdate NACK to avoid using bandwidth
for packet that have no chance of arriving on time. Though, this had a
side effect, which is that it was to get an early RTCP packet with no
feedback into it. This was pretty useless but also had a side effect,
which is that the RTX RTT value would never be updated. So we we stared
having late RTX request due to high RTT, we'd never manage to recover.

This fixes the regression by making sure we keep at least one NACK in
this situation. This is really light on the bandwidth and allow for
quick recover after the RTT have spiked higher then the jitterbuffer
capacity.
2019-05-17 19:13:22 +00:00
Thibault Saunier
38c5ba90b3 doc: Fix some docstrings 2019-05-13 17:00:00 -04:00
Thibault Saunier
af01988534 doc: Port documentation to hotdoc 2019-05-13 11:34:56 -04:00
Thibault Saunier
232e3682ea Mark some properties as DOC_SHOW_DEFAULT 2019-05-13 10:24:40 -04:00
Thibault Saunier
0a6a62aa76 docs: Port all docstring to gtk-doc markdown 2019-05-13 10:24:40 -04:00
Thiago Santos
135e12565b rtspsrc: do not try to send EOS with invalid seqnum
The second udpsrc (rtcp) might not have seen the segment event if it was
not enabled or if rtcp is not available on the server. So if the
application tries to send an EOS event it will try to set an invalid
seqnum to the event.
2019-05-02 22:14:35 -07:00
Nicolas Dufresne
a6e7f258ac rtpsource: Add more information to probation warning 2019-05-02 14:44:58 -04:00
Nicolas Dufresne
84c102b6fe rtpsession: Call on-new-ssrc earlier
Right now, we may call on-new-ssrc after we have processed the first
RTP packet. This prevents properly configuring the source as some
property like "probation" are copied internally for use as a
decreasing counter. For this specific property, it prevents the
application from disabling probation on auxiliary sparse stream.

Probation is harmful on sparse streams since the probation algorithm
assume frequent and contiguous RTP packets.
2019-05-02 14:44:58 -04:00
Seungha Yang
74e409590a matroskamux: Write MasteringMetadata and Max{CLL,FALL}
Enable muxing with HDR meta data if upstream provided it
2019-05-01 14:28:36 +00:00
Seungha Yang
61f9a2a415 matroskademux: Add support parsing HDR metadata
Set SMPTE ST 2086 mastering-display-metadata and
content-light-level to caps, if any
2019-05-01 14:28:36 +00:00
Seungha Yang
53fedc43ae matroska: Remove white space 2019-05-01 14:28:36 +00:00
Sebastian Dröge
c4608b410c rtprawdepay: Don't get rid of the buffer pool on FLUSH_STOP
We expect there to be a pool as long as the caps are known and
FLUSH_STOP is not resetting the caps. Getting rid of the pool would
cause assertions.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/584
2019-05-01 10:00:51 +03:00
Danny Smith
037d70c01b rtpbin: Free storage when freeing session 2019-04-29 10:57:38 +02:00
Sebastian Dröge
0c7c31d197 matroskamux: Fix typo in error message 2019-04-25 21:52:42 +03:00
Sebastian Dröge
4881ea95b0 imagefreeze: Only set the DISCONT flag on the first buffer after segment start 2019-04-25 08:20:14 +00:00
Philippe Normand
aadfa5f20f scaletempo: Advertise interleaved layout in caps templates
Scaletempo doesn't support non-interleaved layout. Not explicitely stating this
would trigger critical warnings and a caps negotiation failure when scaletempo
is used as playbin audio-filter.

Patch suggested by George Kiagiadakis <george.kiagiadakis@collabora.com>.

Fixes #591
2019-04-23 13:39:20 +00:00
Seungha Yang
7fb8abf8bb meson: matroska: Ensure header dependency not only library
Library existence does not guarantee header.
2019-04-22 20:40:50 +09:00
Robert Rosengren
2476e9e4ae multidupsink: Use gst_net_utils_set_socket_tos for QoS DSCP
Util function in net library exists for setting QoS DSCP on socket, hence
use it to simplify code.
2019-04-22 09:16:20 +00:00
Tim-Philipp Müller
c6c3bed095 rtpulpfecdec,enc: unbreak plugin gtk-doc build in autotools
Fix doc chunks to not use that syntax for links that have the
url as description, it will be put verbatim into the xml/*.xml
file and then the expat parser will throw a syntax error like:

  File "../../common/mangle-db.py", line 71, in <module>
    main()
  File "../../common/mangle-db.py", line 69, in main
    patch (details.replace("-details", ""), os.path.basename(details))
  File "../../common/mangle-db.py", line 20, in patch
    doc = xml.dom.minidom.parse(related)
  File "/usr/lib/python2.7/xml/dom/minidom.py", line 1918, in parse
    return expatbuilder.parse(file)
  File "/usr/lib/python2.7/xml/dom/expatbuilder.py", line 924, in parse
    result = builder.parseFile(fp)
  File "/usr/lib/python2.7/xml/dom/expatbuilder.py", line 207, in parseFile
    parser.Parse(buffer, 0)
xml.parsers.expat.ExpatError: not well-formed (invalid token): line 84, column 7
2019-04-09 23:58:30 +01:00
Antonio Ospite
61c1385c42 rtpvrawpay: preserve GST_BUFFER_FLAG_DISCONT on the first outputted buffer
If the incoming frame buffer has GST_BUFFER_FLAG_DISCONT set this should
be preserved and set for the first output buffer too, like other
payloaders do.

Spotted with gst-validate-1.0 when adding integration tests for
rtpsession, a minimal test to reproduce the issue is:

$ gst-validate-1.0 videotestsrc num-buffers=1 ! rtpvrawpay ! identity ! fakesink
Starting pipeline
Pipeline started
   warning : Buffer didn't have expected DISCONT flag333 speed: 1.000000 />
             Detected on <identity0:sink>
             Detected on <identity0:src>
             Detected on <fakesink0:sink>
             Description : Buffers after SEGMENT and FLUSH must have a DISCONT flag

Issues found: 1

=======> Test PASSED (Return value: 0)
2019-04-09 09:32:43 +00:00
Olivier Crête
92138dc3d6 rtpulpfec*: Replace github URIs with gitlab.fdo ones 2019-04-09 08:17:28 +00:00
Olivier Crête
1bd81d3d33 rtpred*: Add example pipelines 2019-04-09 08:17:28 +00:00
Olivier Crête
11f3018170 rtpulpfec*: Improve documentation 2019-04-09 08:17:28 +00:00
Olivier Crête
070eacdd4f rtpstorage + rtpulpfecdec: Get the storage using a query as fallback
This allows it to be used using gst-launch for easier testing.
2019-04-09 08:17:28 +00:00
Nicolas Dufresne
ec06268ed8 rtpsession: Allow overriding NACK packet creation
This introduce a new signal on RTSession, on-sending-nacks is emited
right before the list of seqnums to be nacked are processed and
transformed into FB Nack. This allow implementing custom nacks
handling through another mechanism with APP feedback.
2019-04-05 18:36:36 -04:00
Mathieu Duponchelle
280d86a841 rtpsession: Add disable-sr-timestamp property
The Onvif Streaming Spec, in section 6.11, mandates that when
Rate-Control is disabled potential RTCP packets shall have
their timestamps set to 0.

<https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf>
2019-04-05 20:23:08 +02:00
Nicolas Dufresne
6bb53e75fb rtpsession: Send as many nack seqnum as possible
In order to do that, we now split the nacks registration from the actual
FB nack packet construction. We then try and add as many FB Nacks as
possible into the active packets and leave the remaining seqnums in the
RTPSource. In order to avoid sending outdated NACK later on, we save the
seqnum calculated deadline and cleanup the outdated seqnums before the
next RTCP send.

Fixes #583
2019-04-05 14:53:09 +00:00
John Bassett
74a74bfc99 rtpsession: Fix race when sending PLI, FIR and NACK packets
Calling rtp_session_send_rtcp before marking the source as requiring a
pli/fir/nack meant the rtcp_thread could be scheduled and start running
before the source was updated. This meant the request would not be sent
early but instead was transmitted with the next regular RTCP packet.

Add test for nack generation.
2019-04-05 14:53:09 +00:00
Nicolas Dufresne
6b50d142f3 rtpsession: Fix early rtcp time comparision
If the current time is equal to the early rtcp time deadline, there is
no need to schedule a timer. This ensure that immediate feedback is
really immediate and simplify implementing unit tests with the test
clock, which stops perfectly on the timeout time.

This fix has been extracted from Pexip feature patch called
  "rtpsession: Allow instant transmission of RTCP packets"
2019-04-05 14:53:09 +00:00
Mathieu Duponchelle
74e3eb1f1d rtpgstpay: Set DELTA_UNIT flag when appropriate
When used in combination with a rtponviftimestamp element
downstream, forwarding this flag ensures it gets correctly
serialized in the ONVIF header extension.
2019-04-04 19:08:23 +02:00
Antonio Ospite
435f67debf docs: fix typo s/abonormally/abnormally/ 2019-04-03 16:42:26 +02:00
Antonio Ospite
d6939c4031 docs: fix typo s/incomming/incoming/ 2019-04-03 16:38:56 +02:00
Antonio Ospite
f7c8317668 rtp: fix indentation after G_DEFINE_TYPE
A missing colon after G_DEFINE_TYPE declaration was confusing gst-indent
and causing problem in the pre-commit hook.

Add the missing colon and fix the following function declaration to
follow the normal GStreamer style.
2019-04-03 16:37:34 +02:00
Antonio Ospite
114de8cc96 rtpsession: fix comment to refer to buffers instead of groups
One comments in gst_rtp_session_chain_send_rtp_common() is referring to
groups in a buffer list, however this concept of "group" comes from
GStreamer 0.10 and does not exist anymore in GStreamer 1.0, so update the
comment to refer to buffers instead.
2019-04-02 13:03:56 +02:00
Antonio Ospite
e98b0ca8da rtpsource: add comment to explain why probation queue is not always cleared 2019-04-02 13:03:56 +02:00
Antonio Ospite
0fae88b5fd rtpsource: fix stats about received packets
The update_receiver_stats() function is called also when sending packets
in rtp_source_send_rtp(), and sending packets may happen using a buffer
list rather than individual buffers.

So update the stats using the actual number of packets sent.

NOTE: this is fine for the receive path too (rtp_process_send_rtp)
because the receive path does not support buffer lists and
pinfo->packets would always be equal to 1 in this case.
2019-04-02 09:26:03 +02:00
Olivier Crête
915a9c99bb rtpstorage: Limit the queue size
Limit to the queue size in case there is no arrival time or in case there is
a huge flood of packets.
2019-03-29 22:51:54 +00:00
Olivier Crête
0ecc52c2ee rtpbin: Request the FEC decoder even if ignore-pt is set 2019-03-28 16:24:17 -04:00
Olivier Crête
c2dd263562 rtpbin: Factor out the code that exposes the src pad 2019-03-28 16:24:12 -04:00