Commit graph

24506 commits

Author SHA1 Message Date
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
Matthew Waters
ebb9c3c298 tests/webrtc: factor out sdp validation into a single function 2019-05-30 21:26:46 +10:00
Matthew Waters
eb79f95bf8 tests/webrtc: validate number of sdp media using validate_sdp 2019-05-30 21:26:46 +10:00
Matthew Waters
7e1cdbfd4d tests/webrtc: allow multiple validation functions 2019-05-30 21:26:46 +10:00
Matthew Waters
120a40cf25 tests/webrtc: test that duplicate negotiations succeed 2019-05-30 21:26:46 +10:00
Philippe Normand
9595a7a721 webrtcbin: Expose current and pending local/remote description properties
They are already handled in the property getter and setter functions but were
not formally declared in the GObject class.
2019-05-30 10:35:58 +01:00
Mathieu Duponchelle
da6afdec9c doc: remove xml from comments 2019-05-29 22:58:08 +02:00
Adam Duskett
43eaf5ac4a ext/hls/meson.build: fix dependency logic
Currently, if one was to set -Dhls-crypto to either libgcrypt or openssl
instead of auto, the following lines would fail because hls_crypto_dep is not
yet set:

if not hls_crypto_dep.found() and ['auto', 'libgcrypt'].contains(hls_crypto)
if not hls_crypto_dep.found() and ['auto', 'openssl'].contains(hls_crypto)

Instead, change "if not hls_crypto_dep.found()" to "if not have_hls_crypto"
which fixes the error.
2019-05-29 18:33:02 +00:00
Tim-Philipp Müller
1b774e2da8 docs: remove stale plugins from docs plugin cache
gtk plugin has moved to -good, schroedinger has been removed.
2019-05-29 12:48:06 +01:00
Nicolas Dufresne
f14206f2b3 waylandsink: Workaround gnome-shell bug
Use a timeout to limit that amount of time we wait after the compositor
for the initial configure event. Compositor are support to emit a
configure event before any wl_buffer can be attached. The problem is
that Weston strongly enforce this, while gnome-shell simply does not
emit such an event.
2019-05-26 17:49:29 +02:00
Nicolas Dufresne
112baf404e kmssink: Fixup all errno tracing
All DRM ioctl uses errno to report the error and simply returns -1
when some error occured. This patch fixes all usage of the return
value instead of errno to trace the error type and moves to g_strerror
instead of string.h strerror in order to be consistent with the rest
of GStreamer.
2019-05-26 12:17:29 +02:00
Mathieu Duponchelle
85c389df24 doc: update plugin cache 2019-05-25 19:48:25 +02:00
Nicolas Dufresne
bb646787d0 docs: Updated plugins cache file
This was done so that the duplicated rist element would go way.
2019-05-25 17:36:31 +02:00
Mathieu Duponchelle
102b1346e7 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 16:58:13 +02:00
Tim-Philipp Müller
0d59589935 codecparsers: fix debug category initialisation
Make thread-safe.
2019-05-25 15:29:25 +02:00
Sebastian Dröge
1c712ca555 avwait: Protect properties and some other code with the mutex
These variables are all accessed from multiple threads.

Also fix some minor leaks in error code paths.
2019-05-24 10:41:35 +00:00
Sebastian Dröge
d55dda6252 avwait: Insert some empty lines to give the code some space to breath 2019-05-24 10:41:35 +00:00
Sebastian Dröge
c8876a37ba avwait: Allow setting start timecode after end timecode and the other way around
This might be necessary temporarily for changing the previous settings.
Make it an actual error if the settings are like this while processing a
buffer.
2019-05-24 10:41:35 +00:00
Haihua Hu
9d0ba0f27a wayland/wlbuffer: just return if used_by_compositor is true when attach
When buffer is used by compositor, we don't need attach it and hold one
more reference. Just check used_by_compositor, just return if it is true.
Assert error log is not need, this is normal behavior.
2019-05-22 09:14:03 +00:00
Sebastian Dröge
ab9d42cc7f proxy: Forward queries/events sent directly to the element correctly 2019-05-22 07:48:33 +00:00
Sebastian Dröge
70b08bdbfa proxy: Set SOURCE flag on the source and SINK flag on the sink
So that they are properly recognized as such.
2019-05-22 07:48:33 +00:00
Haihao Xiang
7820109b88 ivfparse: Check the data size against IVF_FRAME_HEADER_SIZE
It is parsing frame data and so should check the data size against the
frame header size instead of the file header size. If don't, it is
possible to drop the last frame because IVF_FILE_HEADER_SIZE is greater
than IVF_FRAME_HEADER_SIZE
2019-05-22 12:37:29 +08:00
Nicolas Dufresne
abe339fb31 doc: Add rist plugin to hotdoc 2019-05-21 18:49:18 +00:00
Nicolas Dufresne
98acb3260d rist: Add combined bonding-method support
This patchs add support for configuring the bonding method used. There is
two method specified

 - redundant: All the RTP packets are replicated
 - combined: RTP packet are evenly distributed over each links

Additionally, an application can set the "dispatcher" property in order
to implement custom dispatching method. Whenever the "dispatcher"
property is set, "bonding-method" property will be ignored.
2019-05-21 18:49:17 +00:00
Nicolas Dufresne
9a443c04bc ristsrc: Implement per session stats
As we can now have multiple sessions, stats need to be implemented per
session. This follow RTPSession model with sources. The stats are now:

    dropped: 0
    received: 0
    recovered: 0
    permanently-lost: 0
    duplicates: 0
    retransmission-requests-sent: 0
    rtx-roundtrip-time: 0
    session-stats:
        session-id=0
            rtp-from=""
            rtcp-from=""
            dropped=0
            received=0
        session-id=1
            rtp-from=""
            rtcp-from=""
            dropped=0
            received=0
        . . .

session-stats is a GValueArray as there is no better alternatives.
2019-05-21 18:49:17 +00:00
Nicolas Dufresne
0c26aaa614 ristsrc: Cleanup unused include 2019-05-21 18:49:17 +00:00
Nicolas Dufresne
73edff67c7 ristsink: Implement per session stats
As we can now have multiple sessions, stats need to be implemented per
session. This follow RTPSession model with sources. The stats are now:

  sent-original-packets: 0
  sent-retransmitted-packets: 0
       session-stats:
            session-id=0
              sent-original-packets=0
              sent-retransmitted-packets=0
              round-trip-time=0
            session-id=1
              sent-original-packets=0
              sent-retransmitted-packets=0
              round-trip-time=0
            . . .

session-stats is a GValueArray as there is no better alternatives.
2019-05-21 18:49:17 +00:00