Commit graph

6768 commits

Author SHA1 Message Date
Matthew Waters
02c8e66ff1 webrtc: fix an off-by-one calculating low-threshold
We were not signalling low-threshold when the previous amount was at
exactly the low-threshold mark.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>
2020-05-06 15:49:58 +10:00
Matthew Waters
18de5f8f04 webrtc: remove debugging leftover
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>
2020-05-06 15:49:58 +10:00
Matthew Waters
50644f5718 webrtc: always reply to a promise
Otherwise, we defeat the purpose of a promise.

We were not replying when the state was closed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
1f395e3ddb webrtc: name threads based on the element name
Makes debugging a busy loop possibly easier

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
d552c6556c webrtc: correctly use the pad template
GstHarness uses this for releasing request pads correctly. Fixes
numerous leaks in the webrtc unit tests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
46176fbcc7 webrtc: Fix a couple of renegotiation races
When negotiating the SDP we should only connect the streams that are
actually mentioned in the SDP.  All other streams are not relevant at
this time and would likely be part of a future SDP update.  Fixes a
couple of the renegotiation webrtc unit tests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Edward Hervey
75289d83a1 iqa: Fix all leaks in error path
CID 
CID 
CID 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1244>
2020-05-05 17:33:20 +00:00
Matthew Waters
3baf0d5dc4 sctp: enable usrsctp debug when supported
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1234>
2020-05-05 03:38:06 +00:00
Ederson de Souza
b68e47968b avtpsink: Log that AVTPDU transmission failure is due lateness
As ENOBUFS is not really clear about what is going on, let's check
socket error queue to see if packets are being dropped due being late.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
2020-05-02 17:42:15 +00:00
Ederson de Souza
32281ddd33 avtpsink: Accept buffers that fall out of segment
Proper calculate running time for buffers that are out of current
segment and try to honor them.

A typical case is for AVTP packets coming from avtpcvfpay element, as
those may have DTS that falls out of segment (which is about PTS).

By using gst_segment_to_running_time_full(), avtpsink can properly
calculate when to transmit those buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
2020-05-02 17:42:15 +00:00
Ederson de Souza
7edaeb3fae avtpcvfpay: Warn about timestamp issues on non-flushing seek
Seek events will cause new segments to be sent to avtpcvfpay, and for
flushing seeks, a pipeline running time reset. This running time
reset, which effectively changes pipeline base time, will cause
avtpcvfpay element to generate incorrect DTS for the initial set of
buffers sent after FLUSH_STOP.

This happens due the fact that base time change happens only when the
sink gets the first buffer after the FLUSH_STOP - so avtpcvfpay used
the wrong base time to do its calculations.

However, if the pipeline is paused before the seek, sink will update
base time when pipeline state goes to PLAYING again, before avtpcvfpay
gets the first buffers after the flush. Then avtpcvfpay element will be
able to normally calculate DTS for the outgoing packets.

This patch simply adds a warning message in case a flushing seek is
performed on a playing pipeline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
2020-05-02 17:42:15 +00:00
Ederson de Souza
12838af353 avtpcvfpay: Ensure NAL fragments are transmitted following stream specs
TSN streams are expected to send packets to the network in a well
defined "pace", which is arbitrarily defined for each stream. This pace
is defined by the "measurement interval" property of a stream.

When the AVTP CVF payloader element - avtpcvfpay - fragments a video
frame that is too big to be sent to the network, it currently defines
that all fragments should be transmitted at the same time (via DTS
property of GstBuffers generated, as sink will use those to time the
transmission of the AVTPDU). This doesn't comply with stream definition,
which also has a limit on how many packets can be sent on a given
measurement interval.

This patch solves that by spreading in time the DTS of the GstBuffers
containing the AVTPDUs. Two new properties, "measurement-interval" and
"max-interval-frames", added to avptcvfpay element so that it knows
stream measurement interval and how many AVTPDUs it can send on any of
them. More details on the method used to proper spread DTS/PTS according
to measurement interval can be found in a code commentary inside this patch.

Tests also added for the new property and behaviour.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
2020-05-02 17:42:15 +00:00
Matthew Waters
b266652043 webrtcbin: also mark data channel transports as active
Fixes negotiation of a bundled sdp with only a data channel.

Without marking the transport as active, we would never unblock the
transportreceivebin and thus no data would ever reach us.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1231>
2020-05-01 03:13:46 +00:00
Matthew Waters
ce9b41f5d4 webrtcbin: fix bundle none case with remote offer bundling
If the remote is bundling, but we are not and remote is offering.
we cannot put the remote media sections into a bundled transport as that
is not how we are going to respond.

This specific failure case was that the remote ICE credentials were
never set on the ice stream and so ice connectivity would fail.

Technically, this whole bunde-policy=none handling should be removed
eventually when we implement bundle-policy=balanced.  Until such time,
we have this workaround.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1231>
2020-05-01 03:13:46 +00:00
Vedang Patel
e47fa2006f avtp: Introduce the CRF Check element
This commit introduces the AVTP Clock Reference Format (CRF) Checker
element. This element re-uses the GstAvtpCrfBase class introduced along
with the CRF Synchronizer element.

This element will typically be used along with the avtpsrc element to
ensure that the AVTP timestamp (and H264 timestamp in case of CVF-H264
packets) is "aligned" with the incoming CRF stream. Here, "aligned" means
that the timestamp value should be within 25% of the period of the media
clock recovered from the CRF stream.

The user can also set an option (drop-invalid) in order to drop any packet
whose timestamp is not within the thresholds of the incoming CRF stream.
2020-04-30 23:31:25 +00:00
Vedang Patel
12ad2a4bcd avtp: Introduce the CRF Sync Element
This commit introduces the AVTP Clock Reference Format (CRF) Synchronizer
element. This element implements the AVTP CRF Listener as described in IEEE
1722-2016 Section 10.

CRF is useful in synchronizing events within different systems by
distributing a common clock. This is useful in a scenario where there are
multiple talkers who are sending data to a single listener which is
processing that data. E.g.  CCTV cameras on a network sending AVTP video
streams to a base station to display on the same screen.

It is assumed that all the systems are already time-synchronized with each
other. So, the AVTP Talker essentially adjusts the AVTP Presentation Time
so it's phase-locked with the reference clock provided by the CRF stream.

There are 2 different roles of systems which participate in CRF data
exchange.  A system can either be a CRF Talker, which samples it's own
clock and generates a stream of timestamps to transmit over the network, or
a CRF Listener, the system which receives the generated timestamps and
recovers the media clock from the timestamps. It then adjusts it's own
clock to align with recovered media clock. The timestamps generated by the
talker may not be continuous and the listener might have to interpolate
some timestamps to recover the media clock. The number of timestamps to
interpolate is mentioned in the CRF stream AVTPDU (Refer IEEE 1722-2016
Section 10.4 for AVTPDU structure). Only CRF Listener has been implemented
in this commit.

The CRF Sync element will create a separate thread to listen for the CRF
stream. This thread will calculate and store the average period of the
recovered media clock. The pipeline thread will use this stored period
along with the first timestamp of the latest CRF AVTPDU received to
calculate adjustment for timestamps in the audio/video streams. In case of
CRF AVTPDUs with single timestamp, two consecutive CRF AVTPDUs will be used
to figure out the average period of the recovered media clock.

In case of H264 streams, both AVTP timestamp and H264 timestamp will be
adjusted.

In the future commits, another "CRF Checker" element will be introduced
which will validate the timestamps on the AVTP Listener side. Which is why
a lot of code has been implemented as part of the gstcrfbase class.
2020-04-30 23:31:25 +00:00
krivoguzovVlad
b769af0c4f Update gstsrtobject.c
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/999>
2020-04-30 18:57:13 +00:00
Matthew Waters
80ede09193 webrtcbin: only start gathering on local descriptions
If we are in a state where we are answering, we would start gathering
when the offer is set which is incorrect for at least two reasons.

1. Sending ICE candidates before sending an answer is a hard error in
   all of the major browsers and will fail the negotiation.
2. If libnice ever adds the username fragment to the candidate for
   ice-restart hardening, the ice username and fragment would be
   incorrect.

JSEP also hints that the right call flow is to only start gathering when
a local description is set in 4.1.9 setLocalDescription

"This API indirectly controls the candidate gathering process."

as well as hints throughout other sections.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1226>
2020-04-30 14:47:55 +00:00
Seungha Yang
0b102d22ec webrtc: Correct symbol visibility to fix build warning on Windows
GstWebRTCDataChannel is fully internal of plugin

webrtcdatachannel.c(50): warning C4273: 'gst_webrtc_data_channel_get_type': inconsistent dll linkage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1225>
2020-04-30 10:27:47 +00:00
Thibault Saunier
a3595f7e0f lv2: Namespace global variables and explicitly make them private
And fix a LV2_PORT_GROUPS__rearLeft/LV2_PORT_GROUPS__rearRight typo

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1125>
2020-04-29 19:49:45 +00:00
Debarshi Ray
a0cd455dd0 lv2: Make it build with -fno-common
GCC 10 defaults to -fno-common. This means that global variables shared
across multiple translation units should be declared as 'extern' in
header files and defined in exactly one C file. See:
https://gcc.gnu.org/gcc-10/porting_to.html

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1125

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1125>
2020-04-29 19:49:45 +00:00
Seppo Yli-Olli
90f374dd0c openh264: memcmp return value 0 means match
Commit e2aa76db79 introduced version
check guard for OpenH264 binary. There was a boolean error in
memcmp so matching OpenH264 was erroneously rejected.
Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1219>
2020-04-27 15:40:10 +00:00
Mathieu Duponchelle
62d1a3a143 cccombiner: don't drop buffers on video timestamp discontinuities
If we receive video buffers with non-perfect timestamps, the
caption buffers' timestamps might fall in the interval between
the end of one video buffer and the start of the next one.

Make our criteria for dropping that the caption buffer has
a timestamp older than the end of the previous video buffer,
not older than the start of the new one, unless of course
this is the first video buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1207>
2020-04-24 08:47:50 +00:00
Mathieu Duponchelle
f02300eef5 cccombiner: handle gap buffers adequately
- Don't try to map them as actual CC data, that was raising
  a critical

- Consume video buffers up to the end of the gap

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1207>
2020-04-24 08:47:50 +00:00
Guillaume Desmottes
4e9030a0b6 spanplc: add 'stats' property
Allow users to retrieve the number of samples, and their duration,
generated using PLC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1210>
2020-04-23 23:15:29 +00:00
Seppo Yli-Olli
e2aa76db79 Have strict version check for OpenH264 to avoid ABI issues
This fixes  and no longer trusts soname alone

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1206>
2020-04-23 22:52:23 +00:00
Jan Alexander Steffens (heftig)
97c05d3f4b
srt: Accumulate total bytes sent/received over all connections/callers
So we don't lose them. Split gst_srt_object_open_internal for internal
reconnections that don't reset the accumulated bytes.
2020-04-15 10:42:48 +02:00
Jan Alexander Steffens (heftig)
d19b3fccb5
srt: Fix type of bytes-received-lost
The field is a uint64_t.
2020-04-15 10:42:47 +02:00
Jan Alexander Steffens (heftig)
132e3a1af9
srt: Remove use of closures for signal emission
It seems overly complicated.
2020-04-15 10:42:47 +02:00
Jan Alexander Steffens (heftig)
d2d00e07ac
srt: Clean up locking
Use GST_OBJECT_LOCK (srtobject->element) to protect only the fields
involved in property access.

Introduce a new mutex srtobject->sock_lock to go with
srtobject->sock_cond and protect the list of callers from concurrent
access.
2020-04-15 10:42:47 +02:00
Jan Alexander Steffens (heftig)
37ee389913
srt: Remove trailing whitespace 2020-04-15 10:42:47 +02:00
Philippe Normand
991bcb22d5 wpe: Add support for SHM without requiring EGLDisplay
The previous version of the SHM export support still required a valid
EGLDisplay. The upcoming WPEBackend-FDO 1.8.x aims to remove this requirement,
hence allowing wpesrc to be used without GPU.
2020-04-13 11:53:16 +00:00
J. Kim
04f3f4be4f srtobject: fix mutex lock target
GstSRTObject is a structure that has an actual GstElement
which is extended to srt{src,sink}.
2020-04-13 15:23:46 +09:00
Zeid Bekli
663cd44ef0 srtp: Added support for BYE packet
SRTCP can't get SSRC from BYE packet, this will make srtpdec element
to drop the package. Adding support to get the SSRC from BYE packets.
2020-04-09 15:11:19 +00:00
Stéphane Cerveau
d59bd5f674 dash: fix VARARGS coverity error
va_end was not called in every code path due to
g_return_val_if_fail.

API usage errors  (VARARGS)
va_end was not called for "myargs".

CID: 1461294
2020-04-08 20:02:57 +00:00
worldofpeace
f10b424418 meson: build with neon 0.31
No API/ABI changes https://github.com/notroj/neon/blob/0.31.0/NEWS#L3
2020-04-03 18:50:16 -04:00
Nirbheek Chauhan
387b6df948 meson: Don't use get_option('buildtype')
We should directly check the values of the `debug` and `optimization`
options instead.

`get_option('buildtype')` will return `'custom'` for most combinations
of `-Doptimization` and `-Ddebug`, but those two will always be set
correctly if only `-Dbuildtype` is set. So we should look at those
options directly.

For the two-way mapping between `buildtype` and `optimization`
+ `debug`, see this table:
https://mesonbuild.com/Builtin-options.html#build-type-options
2020-04-03 17:07:47 +05:30
Miguel Paris
45a1070203 srtpdec: reduce log level for replay cases
These are normal cases, so DEBUG level is enough.
2020-04-01 17:45:15 +00:00
Miguel París Díaz
ed71e262b0 srtpdec: do not warning old replay errors
Reordered packets producing decrypting errors are very normal,
so we should filter which errors are warning and which not.
2020-04-01 17:45:15 +00:00
Miguel Paris
075ff1e8b0 srtpdec: fix reseting RTP sequence number on ROC changes
Each srtp_stream_t is tied to an specific SSRC, so a
roc_changed flag should be kept per each SSRC in order to
properly reset RTP sequence number on ROC changes.
2020-04-01 16:49:44 +02:00
Seungha Yang
770a851e03 x265enc: Update for video-hdr struct change
See the change of -base https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/594
2020-04-01 05:18:11 +00:00
Matthew Waters
8da177c0bf dtls/connection: fix EOF handling with openssl 1.1.1e
openssl 1.1.1e does some stricker EOF handling and will throw an error
if the EOF is unexpected (like in the middle of a record).  As we are
streaming data into openssl here, it is entirely possible that we push
data from multiple buffers/packets into openssl separately.

From the openssl changelog:

 Changes between 1.1.1d and 1.1.1e [17 Mar 2020]
  *) Properly detect EOF while reading in libssl. Previously if we hit an EOF
     while reading in libssl then we would report an error back to the
     application (SSL_ERROR_SYSCALL) but errno would be 0. We now add
     an error to the stack (which means we instead return SSL_ERROR_SSL) and
     therefore give a hint as to what went wrong.
     [Matt Caswell]

We can relax the EOF signalling to only return TRUE when we have stopped
for any reason (EOS, error).

Will also remove a spurious EOF error from previous openssl version.
2020-03-27 11:43:53 +11:00
Matthew Waters
319a5e5779 webrtc: mark streams as active on renegotiation as well.
Otherwise when bundling, only the changed streams would be considered as
to whether the bundled transport needs to be blocked as all streams are
inactive.

Scenario is one transceiver changes direction to inactive and as that is
the only change in transciever direction, the entire bundled transport would
be blocked even if there are other active transceivers inside the same bundled
transport that are still active.

Fix by always checking the activeness of a stream regardless of if the
transceiverr has changed direction.
2020-03-25 14:46:15 +11:00
Philippe Normand
26f76dd927 wpe: Enable SHM support for new stable WPEBackend-FDO release
1.5.0 was the development version.
2020-03-23 13:08:46 +00:00
Philippe Normand
49560b4ba8 wpe: Mouse scroll events support 2020-03-23 13:08:46 +00:00
Philippe Normand
158a2b3fd1 webrtcdsp: Fix documentation markup 2020-03-15 12:44:31 +00:00
Philippe Normand
b36e36f74a openni2: Remove spurious gtk-doc markers 2020-03-15 10:47:02 +00:00
yychao
cb0e4bffea smoothstreaming: fix H264 CodecPrivateData parsing
Do not pass SPS nal_unit_type (0x67) into gst_h264_parse_sps()

Fixes 
2020-03-10 12:55:05 +00:00
Sebastian Dröge
5a2053e0af webrtcbin: Use GPtrArrays or store items inline instead of using GArrays of pointers 2020-03-09 21:38:42 +02:00
Jan Schmidt
8274fcd311 webrtcbin: Prevent ICE gathering state reaching complete early
The ICE gathering state can transition to complete prematurely if the
underlying ICE components complete their gathering while the initial
ICE gathering state task is queued and still pending.

In that situation, the ice gathering state task will report complete
while there are still ICE candidates queued for emission.

Prevent that by storing ICE candidates in an array and checking if
there are any pending before reporting a completed ICE gathering
state.
2020-03-10 05:47:40 +11:00