Commit graph

334 commits

Author SHA1 Message Date
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
Stéphane Cerveau
c3bb04f779 soup: fix soup debug category
Use soup debug category in souploader
for soup plugin element load.

Inititalize properly soup utils category.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2535>
2022-06-02 12:55:13 +00:00
Seungha Yang
238ce301b6 hlsdemux2: Adjust debug log level
HLS manifest might not be represented by a single common caps
when different codecs are mixed in a playlist, but it
does not seem to be a critical issue we need to warn.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2516>
2022-06-01 14:32:55 +00:00
Stéphane Cerveau
6b55f97529 adaptivedemux2: fix plugin/element init
In case of per features registration such as the
customizable gstreamer-full library, each
element should check that the soup library can be loaded to
facilitate the element registration.

Initialize the debug categories properly

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2348>
2022-05-30 13:13:48 +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
Nirbheek Chauhan
a6db1a4c81 jack: Always use jack_free as specified by the docs
Fixes a crash on Windows due to a CRT mismatch. The JACK installation
still uses MSVCRT, and we the Universal CRT for both MinGW and MSVC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2497>
2022-05-28 11:34:23 +00:00
Nirbheek Chauhan
4865f36344 jack: Add support for detecting libjack on Windows
No source code changes were necessary to get the plugin working on
Windows with MSVC.

Run QJackCtl and audiotestsrc ! jackaudiosink just works.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2497>
2022-05-28 11:34:23 +00:00
Hou Qi
7f2fb3b783 v4l2: Reset transfer in gst_v4l2_object_acquire_format()
get_colorspace() checks input caps transfer when mapping V4L2_XFER_FUNC_709
back to V4L2_COLORSPACE_BT2020 and GST_VIDEO_TRANSFER_BT2020_12. After
receiving source change event, decoder will G_FMT and S_FMT again. So need
to reset transfer when acquiring format to avoid using the old transfer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2475>
2022-05-26 17:09:28 +00:00
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
Stéphane Cerveau
39b13fcdbd soup: Fix plugin/element init
In case of per features registration such as the
customizable gstreamer-full library, each
element should check that the soup library can be loaded to
facilitate the element registration.

Initialize the debug category properly

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2349>
2022-05-25 09:33:56 +02:00
Eli Schwartz
79c3bbc2ed meson: use better zlib dependency fallback
zlib is required, and if it isn't found it is checked several ways and
then forced via subproject(). This code was added in commit
b93e37592a, to account for systems where
zlib doesn't have pkg-config files installed.

But Meson already does dependency fallback, and also, since 0.54.0, does
the in-between checks for find_library('z') and has_header('zlib.h') via
the "system" type dependency. Simplify dependency lookup by marking it
as required, which also makes sure that the console log doesn't
confusingly list "not found".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2484>
2022-05-24 19:39:29 +00:00
Nicolas Dufresne
6d6ebbbb36 v4l2videodec: Fix missing handling of resolution-change
The pool process function may poll and get the resolution-change event
whenever it is not possible to share our buffers. This typically happen
when downstream does not support GstVideoMeta.

Not handling this would cause the decoder thread to exit silently and the
pipeline to stall.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2457>
2022-05-24 18:02:12 +00:00
Nicolas Dufresne
4e441417b8 v4l2videodec: Downgrade to info resolution-change trace
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2457>
2022-05-24 18:02:12 +00:00
Tim-Philipp Müller
e64c6f0b93 shout2: fix compiler warnings and bump req to libshout >= 2.4.2
Fix compiler warnings with latest libshout version (2.4.6).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2473>
2022-05-23 20:23:38 +01:00
László Károlyi
fedb48c68f shout2send: Adding send-title-info and user-agent options
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2420>
2022-05-23 16:01:48 +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
Hou Qi
85b53bb65d v4l2: set default resolution if caps has no such information
Output may attemp to set the width and height to zero values if
caps has no such information, which will cause capture get invalid
dimensions. Then decoder reports negotiation failure.

So need to set default resolution if caps has no such information.
Real values can be set again until source change event is signaled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2400>
2022-05-20 12:23:40 +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
Guillaume Desmottes
aa3b6a11e0 vpxenc: enforce strictly increasing pts
From vpx_codec_encode() documentation:
  "The presentation time stamp (PTS) MUST be strictly increasing."

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2405>
2022-05-12 13:00:53 +02:00
Guillaume Desmottes
10b837ae5e vpxenc: conver input pts to running time
The input pts needs to be strictly increasing, see vpx_codec_encode() doc, so convert it to
running time as we don't want to reset the encoder for each segment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2405>
2022-05-12 13:00:53 +02:00
Guillaume Desmottes
1e829696e8 vpxenc: fix crash if encoder produces unmatching ts
If for some reason the encoder produces frames with a pts higher than
the input one, we were dropping all the video encoder frames and ended
up crashing when trying to access the pts of a NULL pointer returned by
gst_video_encoder_get_oldest_frame().

I hit this scenario by feeding a decreasing timestamp to vp8enc which
seem to confuse the encoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2405>
2022-05-12 13:00:53 +02:00
Nicolas Dufresne
522f19e013 v4l2videoenc: Setup crop rectangle if needed
Hantro H1 and Rockchip VEPU2 drivers will pad the width/height to a
multiple of 16. In order to obtain the right JPEG size, the image needs
to be cropped using the S_SELECTION API. This support is added as best
effort since older drivers may emulate this by looking at the capture
queue width/height.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2329>
2022-05-07 11:35:14 +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
Edward Hervey
7c9eb0335f mssdemux2: Don't expose/use streams we can't handle yet
Avoids issues further down

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2319>
2022-04-28 10:45:37 +00:00
Edward Hervey
2ec79418df mssdemux2: Ensure stream/track uniqueness
If there is more than one track of the same type (say audio), we would end up
creating several stream/types with the same name.

Instead use the MSS stream name property to make them unique

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2319>
2022-04-28 10:45:37 +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
Camilo Celis Guzman
5eadde319c rtphdrextsdes: fixup test trying to g_free a local variable
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2235>
2022-04-22 08:41:59 +00:00