Commit graph

175 commits

Author SHA1 Message Date
Tim-Philipp Müller
9d9e59622f Bump GLib requirement to >= 2.62
Can't require 2.64 yet because of
https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
2022-06-10 06:01:41 +00:00
Marc Leeman
8bdf7e8ad8 fix trivial distination -> destination
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2573>
2022-06-08 14:40:09 +02:00
Sebastian Dröge
47aab6c832 flvdemux: Make use of the streams API if used in a streams-aware bin
This allows adding audio/video streams after 6s.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2559>
2022-06-07 10:52:46 +00:00
Jan Alexander Steffens (heftig)
637406cdb1 aacparse: Avoid mismatch between src_caps and output_header_type
If our downstream caps didn't intersect, we attempted to convert between
raw and ADTS stream formats, if possible. If the caps still did not
intersect, we then used the modified `src_caps` but left the
`output_header_type` unmodified.

This caused a mismatch between caps and actual stream format.

Avoid this by first copying the `src_caps` to `convcaps` for the
additional intersection tests, replacing `src_caps` if we succeed.

While we're here, clean up the code a bit and remove the `codec_data`
field from outgoing ADTS caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2550>
2022-06-06 15:09:09 +00:00
Sebastian Dröge
e5f9bb973f flvdemux: Actually make use of the debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2552>
2022-06-06 14:36:41 +00:00
Jan Schmidt
a8f18aef18 rtpptdemux: Don't GST_FLOW_ERROR when ignoring invalid packets
https://bugzilla.gnome.org/show_bug.cgi?id=741398 changed
rtpptdemux in 2014 to not post a GST_ELEMENT_ERROR on the
bus when dropping an invalid (non-RTP) packet, but still
returned GST_FLOW_ERROR upstream - so the pipeline still
stops, but now without a useful bus error.

Return GST_FLOW_OK instead, so the pipeline keeps
running. Some old telephony equipment can send invalid
packets before the real RTP traffic starts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2520>
2022-05-29 20:27:38 +10:00
Piotrek Brzeziński
5490189b9b cutter: Include running/stream-time in messages
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2490>
2022-05-25 12:27:10 +00:00
Sebastian Dröge
7273024ae5 qtdemux: Add support for ONVIF XML Timed MetaData
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2453>
2022-05-20 13:01:44 +00:00
Sebastian Dröge
365a9af9c5 qtdemux: Add parsing/dumping of nmhd / metx boxes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2453>
2022-05-20 13:01:44 +00:00
Sebastian Dröge
04f6258863 qtdemux: Parse styp box for informational purposes
And include some more details in the debug logs for the ftyp box too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2453>
2022-05-20 13:01:44 +00:00
Jan Schmidt
7322a6d004 splitmuxsrc: Re-queue sticky events after probing.
When processing the first event after probing the
file and being activated, requeue sticky events
as there's no requirement that demuxers send tag
and other events again after a seek - that's
why they're sticky.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2432>
2022-05-17 11:55:40 +00:00
Jan Alexander Steffens (heftig)
d0fdfa76ae deinterlace: Clean up error handling in chain and _push_history
- Consistently unref the chained buffer at the end of the chain
  function, if we're not handing it off to `gst_pad_push`. This avoids a
  few buffer leaks in the error paths in `_chain` and `_push_history`.
- When mapping the video frame fails, return a flow error instead of
  crashing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2428>
2022-05-17 10:56:23 +00:00
Jan Alexander Steffens (heftig)
718d31fe63 splitmuxsink: When flushing, exit handle_mq_input quickly
If we just break the loop, we might run into the `gop != NULL` assert
that follows it. Rather, exit immediately with flushing flow.

Also use this flushing mechanism when we release a pad. This avoids
having an extra flag.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1030>
2022-05-17 09:24:10 +00:00
Jan Alexander Steffens (heftig)
fd27ee1537 splitmuxsink: Avoid deadlock on release, harder
Unlock after broadcasting and wait for the pad to be free before
relocking the muxer, giving the input probe a chance to react to our
broadcast.

Improves the fix from
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/838.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1030>
2022-05-17 09:24:10 +00:00
Shingo Kitagawa
92c0a462ae wavparse: fix typo in debug message
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2425>
2022-05-16 19:31:18 +09:00
Thibault Saunier
1cb4c050d0 rtpbin: Avoid holding lock GST_RTP_BIN_LOCK when emitting pad-added
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2411>
2022-05-13 06:25:03 +00:00
Sebastian Dröge
1223324246 qtdemux: Don't use tfdt for parsing subsequent trun boxes
The timestamp in the tfdt refers to the first trun box and if there are
multiple trun boxes then the distance between the first timestamps will
grow.

At some point this distance reaches a threshold and triggers the
resetting of the first sample's timestamp of this trun box to be reset
to the tfdt.

This threshold is implemented for files where there is a jump in the
timeline between fragments and where this can be detected via a jump
between the end timestamp of the previous fragment and the tfdt of the
next. This behaviour is preserved.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2409>
2022-05-13 04:19:36 +00:00
Sebastian Dröge
d2c6f21fc1 mp4mux: Disable aggregator's default negotiation
mp4mux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.

By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
2022-05-05 17:41:58 +00:00
Sebastian Dröge
841cba4182 flvmux: Disable aggregator's default negotiation
flvmux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.

By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
2022-05-05 17:41:58 +00:00
Matthew Waters
f4f342aa78 wavparse: ensure that any pending segment is sent before an EOS event is sent
Specifically fixes seqnum handling when an aggregator-based element
(audiomixer et al) is downstream and a seek is performed that
immediately causes an EOS from wavparse.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2356>
2022-05-04 08:00:02 +00:00
Sebastian Dröge
7466444b63 rtpjitterbuffer: Free CNAME/SSRC mappings on finalize and PAUSED->READY
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2336>
2022-04-29 23:33:47 +03:00
Sebastian Dröge
2c405da921 rtpmanager: Refactor RTCP packet loops to fix control flow
Mixing C loops with switch statements is a bad idea as break has a
different meaning in both. Breaking inside the switch statements wrongly
caused further loop iterations.

Instead use goto to get out of the loop and continue to do another loop
iteration, and never ever use break except for the end of a case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2336>
2022-04-29 23:13:15 +03:00
Seungha Yang
6619f1611f rtpjitterbuffer: Initialize variables
Avoid use of uninitialized variable
Fixing MSVC warning
gstrtpjitterbuffer.c(4733) : warning C4700: uninitialized local variable 'have_sdes' used

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2315>
2022-04-28 12:37:13 +00:00
dongil.park
5b11e6a3d0 wavparse: Unset DISCONT buffer flag for divided into multiple buffers in push mode
In push mode (streaming), if the received chunk buffer size from _chain is bigger
than output buffer size, the flags of the divided-buffers are propagated to the
DISCONT flag from first received chunk buffer. This unexpected buffers contained DISCONT
flags are abnormally transformed when changing the sampling rate by audioresample element.
So unset unnecessary DISCONT flag before pad_push().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2305>
2022-04-27 14:29:10 +00:00
Sebastian Dröge
9d5179ad3f rtpjitterbuffer: add the reference timestamp meta in more situations
Previously, we only added it when actually performing synchronization
based on the NTP time.

The information can be useful downstream in other situations too, and
we can compute a NTP time as soon as we get a sender report with the
relevant information.

Co-authored-by: Mathieu Duponchelle <mathieu@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2252>
2022-04-27 12:35:21 +00:00
Sebastian Dröge
ed425e2785 rtpgstpay: Don't push packets before the first input buffer is received
It's not possible to create a valid RTP timestamp for them, which would
cause a potentially very big RTP timestamp discontinuity between those
first packets (created from initial events) and the packet based on the
first input buffer.

As a side-effect, also simplify the packet aggregation code a bit and
work with only a single level of buffer lists.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2250>
2022-04-27 11:55:17 +00:00
Havard Graff
390ec99f1b rtptwcc: don't map the buffer twice
...and use the pt extracted rather than the one from RTPPacketInfo
when logging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2271>
2022-04-26 10:27:25 +00:00
Thibault Saunier
d673a90aea rtpsession: Emit "notify::stats" when we update stats from RR or SR
Sensibily optimizing caching the pspecs and using them directly

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2266>
2022-04-26 08:49:42 +00:00
Mathieu Duponchelle
3391a7d499 rtpredenc: quieten warning about ignoring header extensions
Turn it into a FIXME, and only log once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2279>
2022-04-23 01:04:54 +00:00
Havard Graff
b7b71e6974 rtprtxsend: mark RTX buffers with GST_RTP_BUFFER_FLAG_RETRANSMISSION
It is useful for elements downstream from rtxsend to know if the RTP
buffer they are dealing with is an RTX buffer or not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2272>
2022-04-22 19:27:45 +00:00
Tristan Matthews
27dea62304 mp4mux: fix spelling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2241>
2022-04-22 14:07:57 +00:00
Jonas Bonn
2f6ad787b2 multiudpsink: allow binding to IPv6 address
When the sink is configured to create sockets with an explicit bind
address, then the created socket gets set to the udp_socket field
irregardless of whether the bind address indicated that the socket
family should be IPv4 or IPv6.  When binding to an IPv6 address, this
results in the following error:

gstmultiudpsink.c:1285:gst_multiudpsink_configure_client:<rtcpsink>
error: Invalid address family (got 10)

This patch adds a check of the address family being bound to and sets
the created socket to used_socket or used_socket_v6, accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1551>
2022-04-22 10:43:13 +00:00
Sebastian Dröge
02115a5efc rtpmanager: Move some duplicated constant and helper function to a single place
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
c7e12974ba rtpbin/rtpjitterbuffer: Don't parse RTCP SRs twice unless needed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
82169aa140 rtpjitterbuffer: Add property to throttle handling of RTCP SR / NTP-64 syncing
This proxies the "rtcp-sync-interval" property of rtpbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
ce38614e1a rtpsession: Handle RTCP-SR-REQ (RFC6051) RTCP feedback message
This causes an RTCP SR to be sent at the earliest possible time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
0c819d2f31 rtpbin/rtpjitterbuffer: Allow syncing to an SR without CNAME if the CNAME is already known
The RTCP SR packet might be without SDES in case of a reduced-size RTCP
packet. For syncing purposes the CNAME is needed but it might be known
already from an earlier RTCP packet or out of band, via the SDP for
example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
cbaac3cdba rtpbin/jitterbuffer: Use inband 64-bit NTP timestamps according to RFC6051 for faster synchronization
When signalled via the caps that the header extension is used, it will
be read and used in the same way as the RTP/NTP time mapping from RTCP
SRs.

If the CNAME of the stream's SSRC is provided out of band via e.g. the
SDP then this allows streams to be synchronized immediately on the first
packet instead of having to wait for the first RTCP SR to arrive.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/383

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
7c796b3c05 rtpsession: Only add send latency to the running time if it is actually known
Otherwise we can't know the running time yet if rtcp-sync-send-time is
set, and have to wait until the latency is known later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
7ffc830959 rtpsession: Update 64-bit NTP header extensions with the actual NTP time in senders
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
8980c35efe rtpmanager: Add header extension implementation for the 64-bit RFC6051 NTP header extension
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Xavier Claessens
b99ecc78ca Replace gst-i18n-*.h with gi18n-lib.h
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00
Havard Graff
71891e5647 qtdemux: fix leak of channel_mapping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2179>
2022-04-14 19:41:36 +09:00
Robert Rosengren
e4a6521ac7 rtpbin: Fix division by zero when using ts-offset-smoothing-factor
avg_ts_offset may cause division by zero when calculating potential
overflow protection. This fix will avoid the division.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2151>
2022-04-11 15:29:49 +02:00
Tristan Matthews
86f0f8b67f rtpopusdepay: assume 2 channels if sprop-stereo is missing
Fixes #1064

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2125>
2022-04-08 13:11:25 +00:00
Sebastian Dröge
0813efc821 rtpstats: Remove non-existing twcc field docs from RTPPacketInfo and add missing field docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2121>
2022-04-06 10:15:13 +03:00
Sebastian Dröge
46d7763879 rtpsession: Remove unused twcc fields from the struct
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2121>
2022-04-06 10:15:13 +03:00
Xavier Claessens
b004464ac6 Remove glib and gobject dependencies everywhere
They are part of gst_dep already and we have to make sure to always have
gst_dep. The order in dependencies matters, because it is also the order
in which Meson will set -I args. We want gstreamer's config.h to take
precedence over glib's private config.h when it's a subproject.

While at it, remove useless fallback args for gmodule/gio dependencies,
only gstreamer core needs it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
2022-04-01 16:32:17 +00:00
Thibault Saunier
b358897a3b navigation: Rename parse_state to parse_modifier_state
`parse_state` sounds a bit weird and `parse_modifier_state` is clearer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2087>
2022-04-01 06:38:43 +00:00
Matthew Waters
8cdbfec5ed deinterlace: silence unused-but-set werror from imported code
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2042>
2022-03-28 03:00:58 +00:00