Commit graph

24382 commits

Author SHA1 Message Date
Matthew Waters
32d217a9df vulkan: add a command pool object for tracking 2019-06-04 09:03:44 +00:00
Matthew Waters
2d0cab20c0 vkswapper: don't output duplicate formats 2019-06-04 09:03:44 +00:00
Matthew Waters
25dd3b32e5 vulkan: make a gstvulkan library out of the existing API 2019-06-04 09:03:44 +00:00
Matthew Waters
8def3b3743 vkbufferpool: Fix multiplaner allocations
Use the plane width/height and the sizes required by vulkan

Fixes allocation of:
videotestsrc ! video/x-raw,format=I420 ! vulkanupload ! fakesink
2019-06-04 09:03:44 +00:00
Matthew Waters
268dfcaad9 vk*memory: explicitly error out for driver NPOT alignment 2019-06-04 09:03:44 +00:00
Matthew Waters
0526310a95 vulkan/image: initialize the requirements struct before using it 2019-06-04 09:03:44 +00:00
Matthew Waters
7ee28e2e4b vulkan: don't require every element to have a display
Only sink elements really care about a valid display
2019-06-04 09:03:44 +00:00
Matthew Waters
eb0f7f3279 vulkan: remove unused X11 window system references
We use XCB instead
2019-06-04 09:03:44 +00:00
Matthew Waters
873add374a vulkan: remove unused layer enablement
This is possible now via the vulkan loader
2019-06-04 09:03:44 +00:00
Ali Yousuf
69e06ced7d webrtc: Fix log when adding stun server 2019-06-04 07:54:25 +00:00
Matthew Waters
95488812b2 webrtc: fix the location of signalling-state change notification
1. The spec indicates that the notification should occur near the end of
   'setting the description' processing
2. The current location with the drop of the lock could cause the 'check
   if negotiation is needed' logic to execute and become confused about
   the state of the webrtcbin's current local descriptions.
   In the bad case, the following assertions could be hit:
   g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_local_description->sdp));
   g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_remote_description->sdp));

Moving the signalling state change later in the set description task
means that checking for a renegotiation will early abort as the
signalling state is not STABLE before the session description and
transceivers have been updated.
2019-06-04 05:43:43 +00:00
Nicolas Dufresne
2667081654 make: rtp: Remove spurious header file
This header file no longer exist.
2019-06-03 20:29:18 -04:00
Marc Leeman
3ef737605a rtpmanagerbad: add RTP streaming elements
This is a re-implementation of the RTP elements that are submitted in
2013 to handle RTP streams. The elements handle a correct connection
for the bi-directional use of the RTCP sockets.

https://bugzilla.gnome.org/show_bug.cgi?id=703111

The rtpsink and rtpsrc elements add an URI interface so that streams
can be decoded with decodebin using the rtp:// interface.

The code can be used as follows

```
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234

gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink

gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay ! avdec_mpeg4 ! videoconvert ! xvimagesink
```

rtpmanagerbad: add pkg-config
rtpmanagerbad: Rtp should be uppercase
rtpmanagerbad: add G_OS_WIN32 for shielding unix headers
rtpmanagerbad: remove Since from documentation
rtpmanagerbad: rename lib name from nrtp to rtpmanagerbad
rtpmanagerbad: sync meson.build with other modules
rtpmanagerbad: add Makefile.am
rtpmanagerbad: use GstElement to count pads
rtpmanagerbad: use gst_bin_set_suppressed_flags
rtpmanagerbad: check element creation
rtpmanagerbad: post message when trying to access missing rtpbin
rtpmanagerbad: return FALSE with g_return tests
rtpmanagerbad: use gsocket multicast check
rtpmanagerbad: use gst_caps_new_empty_simple iso gst_caps_from_string
rtpmanagerbad: sync with gstrtppayloads.h
rtpmanagerbad: correct media type X-GST
rtpmanagerbad: test if a compatible pad was found
rtpmanagerbad: remove evil copy of GstRTPPayloadInfo
rtpmanagerbad: add gio_dep to meson
rtpmanagerbad: revert to old glib boilerplate

GStreamer 1.16 does not yet support the newer GLib templates, so revert.

rtpmanagerbad: return GST_STATE_CHANGE_NO_PREROLL for live sources

for live sources, NO_PREROLL should be returned for PLAYING->PAUSED and
READY->PAUSED transitions.

rtpmanagerbad: use GstElement pad counting
rtpmanagerbad: just use template name to request pad
rtpmanagerbad: remove commented code
rtpmanagerbad: use funnel to send multiple streams on one socket
rtpmanagerbad: avoid beaches

beaches should only be used during the summer, so rewrite the code to
return explicitly and avoid beaches during the winter.

rtpmanagerbad: add copyright to test code
rtpmanagerbad: g_free is NULL safe
rtpmanagerbad: do not trace rtpbin
rtpmanagerbad: return NULL explitly
rtpmanagerbad: warn when data port is not even

According to RFC 3550, RTP data should be sent on even ports, while RTCP
is sent on the following odd port.

rtpmanagerbad: document port allocation in rtpsink/src
rtpmanagerbad: improve uri description
rtpmanagerbad: add comment re-use socket
rtpmanagerbad: rename gst_object_set_properties_from_uri_query
rtpmanagerbad: loan prop/val setter from rist
rtpmanagerbad: rtpsrc: fix unitialised pointer
rtpmanagerbad: fix silly typo
rtpmanagerbad: test for empty key/value
rtpmanagerbad: rtpsrc: deprecate ssrc collision to INFO
rtpmanagerbad: sync debug with rist
rtpmanagerbad: small strings allocated on stack
rtpmanagerbad: correct rename
rtpmanagerbad: add locking on prop setters/getters

Locking is added because the URI allows to access the properties too.

rtpmanagerbad: allow for RTCP through NAT
rtpmanagerbad: move gio to header file
rtpmanagerbad: free small strings too
rtpmanagerbad: ttl_mc for ttl on dynudpsink
rtpmanagerbad: add comments on the URI registered
rtpmanagerbad: correct macro after file rename
rtpmanagerbad: code style
rtpmanagerbad: handle wrong URIs in setter
rtpmanagerbad: nit URI notation correction

In an URI, the first key/value pair should not have an ampersand, the
parser did not die though.
2019-06-03 20:08:23 +00:00
Niels De Graef
da085a3713 meson: Bump minimal GLib version to 2.44
This means we can use some newer features and get rid of some
boilerplate code using the G_DECLARE_* macros.

As discussed on IRC, 2.44 is old enough by now to start depending on it.
2019-06-02 21:25:24 +02:00
Alex Ashley
015566daec tests/dash_mpd: take account of Period start in expected timestamps
The start of each segment is relative to the Period start, minus
the presentation time offset.

As specified in section 5.3.9.6 of the MPEG DASH specification:
	The value of the @t attribute minus the value of the
	@presentationTimeOffset specifies the MPD start time of
	the first Segment in the series.

Several tests use a Period@start value of 10 seconds, which either
needs to be taken into account when calculating expected timestamps
or have that attribute removed.

This commit uses a mix of updating the timestamps and removing the
start attribute, so that both the case of its presence and absence
is tested.
2019-06-01 21:25:33 +00:00
Alex Ashley
a11f7ed924 dashdemux: include both Period start and presentationTimeOffset in segment start
The start of each segment is relative to the Period start, minus
the presentation time offset.

As specified in section 5.3.9.6 of the MPEG DASH specification:
    The value of the @t attribute minus the value of the
    @presentationTimeOffset specifies the MPD start time of
    the first Segment in the series.

dashdemux was not taking account of presentationTimeOffset and in
some methods was not taking into account the Period start time.
This commit modifies the segment->start value to always be
relative to the MPD start time (zero for VOD,
availabilityStartTime for live streams). This makes all uses of
the segment list consistent.

Fixes #841
2019-06-01 21:25:33 +00:00
Mathieu Duponchelle
51ea6ec6b7 docs: document gstreamer-bad-audio
And unprefix subproject paths, making a special case for
webrtc, to not conflict with the webrtc plugin
2019-06-01 02:58:09 +00:00
Mathieu Duponchelle
7e5ae06ffe libs: build a gir file for gstreamer-bad-audio 2019-06-01 02:58:09 +00:00
Vivia Nikolaidou
50075616f2 avwait: Don't print warnings for every buffer passed 2019-05-31 18:47:03 +03:00
Haihao Xiang
1ec231b85d msdk: return a right pointer for VUYA format
The first channel in memory is V for VUYA format, note
GST_VIDEO_FORMAT_VUYA is mapped to MFX_FOURCC_AYUV in this plugin
2019-05-31 14:51:35 +08:00
Tim-Philipp Müller
bc0c99a3ab docs: update plugin doc cache and add more plugins 2019-05-30 20:50:07 +02:00
Tim-Philipp Müller
7853700b50 meson: add more plugins to plugins list
Makes sure their path gets added to the uninstalled environment
and makes sure they get included in the docs.
2019-05-30 20:41:57 +02:00
Mathieu Duponchelle
f5495700fb basetsmux: don't reset pad on flush_stop
This was mistakenly added when porting to aggregator, this
restores the old behaviour, by only resetting them when the
muxer itself is reset
2019-05-30 17:20:49 +02:00
Mathieu Duponchelle
f9c0367619 mpegtssection: events don't necessarily have a structure 2019-05-30 17:20:12 +02:00
Mathieu Duponchelle
1e72aa6e85 basetsmux: fix send_event by chaining up 2019-05-30 17:20:12 +02:00
Mathieu Duponchelle
02ded087a4 mpegtsmux: add SECTION comment
We include an example for injecting sections in the transport
stream in the documentation
2019-05-30 13:53:05 +00:00
Mathieu Duponchelle
1d90a0afc5 tests: add example for injecting MPEG-TS sections 2019-05-30 13:53:05 +00:00
Mathieu Duponchelle
76c3d98962 basetsmux: preserve user-specified sections across resets
As sections can be provided by the user through send_event
when the element state is NULL, their lifetime is expected
to match that of the muxer, and they must be preserved when
the state changes
2019-05-30 13:53:05 +00:00
Mathieu Duponchelle
fdfd4600c1 atscmux: send empty RRT / MGT / STT tables
These are mandated by A/65, their absence gets flagged by
stream analyzers. Users can of course provide filled up
versions through the send_event API.
2019-05-30 13:53:05 +00:00
Mathieu Duponchelle
5d41740ff6 tsmux: maintain packet counters in a global array
We can have multiple TsMuxPacketInfo objects for the same PID
with user-provided sections, for example ATSC requires multiple
tables with the same PID.
2019-05-30 13:53:05 +00:00
Mathieu Duponchelle
09749192d8 mpegts: extend support for ATSC tables
Adds constructors for the following sections:

STT: System Time Table
MGT: Master Guide Table
RRT: Rating Region Table

Also adds parsing code for RRT
2019-05-30 13:53:05 +00:00
Matthew Waters
f8911deccf webrtc: only set sctp ports if they are different
SCTPassociation will complain if we do that while running and resetting
is not something we support at the moment
2019-05-30 21:33:09 +10:00
Matthew Waters
62cc5e51d1 tests/webrtc: wait until the SDP has been set before continuing
If we renegotiate, then it is currently possible for an added stream to
be added to webrtcbin before the SDP is complete.  This causes an
internal inconsistency as there is a 'pending sink transceiver' without
a corresponding media section in the sdp.  It also does not have an
associated transport stream and will fail in _connect_input_stream().
2019-05-30 21:33:09 +10:00
Matthew Waters
979daea7f2 tests/webrtc: fix racy test with a prenegotiated data channel
If both data channels become ready simultaneously, then the two integer
read-add-update cycles can execute concurrently and only ever increment
once instead of the required twice.  Use an atomic add instead.
2019-05-30 21:33:09 +10:00
Matthew Waters
be011d2086 webrtc/dc: move some code from webrtcbin into the datachannel 2019-05-30 21:33:09 +10:00
Matthew Waters
a51db86ac4 webrtc: hold onto any unknown ICE candidates until the next SDP set
It is very possible for badly behaving signalling or peers to send
us ICE candidates before we receive an SDP.  While we had consideration
for that on the first set SDP, subsequent SDP's could result in
misconfigured ICE transports.  Expand the previous code to also take
into account reconfigurations.
2019-05-30 21:33:09 +10:00
Matthew Waters
177aa22bcd webrtc: Initial support for stream addition/removal
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
  will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
2019-05-30 21:33:09 +10:00
Matthew Waters
015cb75f66 tests/webrtc: a couple of debug/error string fixes 2019-05-30 21:33:09 +10:00
Matthew Waters
be35735989 tests/webrtc: rewrite bundle checks for separate validate_sdp passes
Improves reusability
2019-05-30 21:33:09 +10:00
Matthew Waters
2bb1fde47c tests/webrtc: add helper for getting the offer/answer element 2019-05-30 21:33:09 +10:00
Matthew Waters
b48e2947bf tests/webrtc: only check audio/video for direction attributes 2019-05-30 21:33:09 +10:00
Matthew Waters
033e55695f webrtcbin: expose the transceiver as a pad property 2019-05-30 21:33:09 +10:00
Matthew Waters
c3c4b07ad3 webrtc/transceiver: add a set_direction function
Matches the setDirection() from the W3C spec and allows changing the
transceiver direction at the next negotiation cycle.
2019-05-30 21:33:09 +10:00
Matthew Waters
6ad0edbe92 webrtc: track and log more rtpbin state
like bye's timeouts, validation, activation, etc
2019-05-30 21:33:09 +10:00
Matthew Waters
2df7da85fe webrtc: add support for intersecting inactive transceiver directions 2019-05-30 21:33:09 +10:00
Matthew Waters
5ea7031bd0 webrtc: mark remote/local-description as readonly 2019-05-30 21:32:06 +10:00
Matthew Waters
19b3d744d8 webrtc: don't reuse stopped transceivers at all 2019-05-30 21:26:46 +10:00
Matthew Waters
4d34fe7617 webrtc: also check for a null mid to signify an unassociated transceiver
We always give our transceivers an mline on creation so that check is
not useful by itself
2019-05-30 21:26:46 +10:00
Matthew Waters
00977f263a webrtc: only check sink pads for a 'sink pads have caps' check 2019-05-30 21:26:46 +10:00
Matthew Waters
bd92b2f7c4 webrtc: fix answer creation with multiple streams and similar caps 2019-05-30 21:26:46 +10:00