Commit graph

10176 commits

Author SHA1 Message Date
Stefan Brüns
6e68873d7f qtdemux: Add 'aavd' and related fourcc codes for AAX encrypted audio
The 'aavd' box is contained in the 'stsd' sample description. The 'aavd'
box follows the layout of an 'mp4a' entry, i.e. it contains a single
standard 'esds' extension box, and the two proprietary 'adrm' and 'aabd'
extension boxes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
2020-09-16 00:59:34 +00:00
Camilo Celis Guzman
5340de5c33 rtp/vrawpay: use alloc_output_buffer from base class
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/726>
2020-09-13 23:16:10 +02:00
Ricky Tang
cfae2a37be rtspsrc: Fix push-backchannel-buffer parameter mismatch
When using python, signal parameter must match with function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/724>
2020-09-11 18:33:04 +08:00
Jan Alexander Steffens (heftig)
953ceba80d flvmux: Improve logging of gst_flv_mux_buffer_to_tag_internal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
2020-09-10 09:20:46 +00:00
Jan Alexander Steffens (heftig)
deeb3917a5 flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer
Besides looking like the correct place to put this, it allows us to drop
the entire aggregator queue. The old implementation only dropped at most
one buffer for each call of aggregate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
2020-09-10 09:20:46 +00:00
Mathieu Duponchelle
19860200ed splitmuxsink: fix sink pad release while PLAYING
- Release the split mux lock while removing the probes

- Flush the sinkpad to unblock other pads

- Turn check_completed_gop into a do while statement, when
  waking up we want to recheck whether the current GOP is
  ready for sending

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/719>
2020-09-09 19:03:12 +02:00
Sebastian Dröge
47c43b29eb gst: Update for gst_video_transfer_function_*() function renaming
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/715>
2020-09-07 12:13:18 +03:00
Jan Alexander Steffens (heftig)
2d08d16002 flvmux: Avoid crash when best pad gets flushed
The 'best' pad might receive a flush event between us picking it and us
popping the buffer. In this case, the buffer will be missing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/711>
2020-08-31 14:19:14 +00:00
Jan Alexander Steffens (heftig)
01594d19b8 flvmux: Correct breaks in gst_flv_mux_find_best_pad
The code seems to use `continue` and `break` as if both refer to the
surrounding `while` loop. But because `break` breaks out of the
`switch`, they actually have the same effect.

This may have caused the loop not to terminate when it should. E.g. when
`skip_backwards_streams` drops a buffer we should abort the aggregation
and wait for all pads to be filled again. Instead, we might have just
selected a subsequent pad as our new "best".

Replace `break` with `done = TRUE; break`, and `continue` with `break`.
Then simplify the code a bit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/710>
2020-08-31 15:14:56 +02:00
Zeid Bekli
3211c65a5e rtpL16depay: unref buffer on error
gst_rtp_L16_depay_process to unref buffer on wrong payload size or
reorder failure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/702>
2020-08-24 19:43:15 +00:00
Sebastian Dröge
85a6e95c7d rtputils: Don't call NULL GstMeta transform function
It's optional and if it does not exist then no transformation is
possible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/701>
2020-08-18 10:27:52 +03:00
Julian Bouzas
91972c91aa rtp: Do not register rtpreddec and rtpredenc twice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/699>
2020-08-13 15:27:25 -04:00
Sebastian Dröge
e4ce9887cd rtpmanager: Improve readability of "stats" docs by making the fields an actual list
Otherwise they end up all in the same line one after another.

Also add docs for the "avg-jitter" stats field of the jitterbuffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/698>
2020-08-13 07:24:17 +00:00
Vivia Nikolaidou
c95cc6a015 flvmux: Return NEED_DATA when we drop a buffer
When we are dropping a buffer in find_best_pad (e.g. waiting for a
keyframe, or skipping backwards timestamp), return
GST_AGGREGATOR_FLOW_NEED_DATA to make sure we have enough data at the
next run. Otherwise, a stream that accidentally fell behind (e.g.
relinking race, or just waiting for a keyframe) will never get the
opportunity to catch up to the other one, because the other one will
always keep advancing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>
2020-08-10 20:36:51 +03:00
Vivia Nikolaidou
75f6ca8a11 flvmux: Return NEED_DATA when no best pad is found
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>
2020-08-10 20:20:04 +03:00
Vivia Nikolaidou
59aab55e71 flvmux: Fix possible crash on GST_ITERATOR_RESYNC
Wrong pointer type

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>
2020-08-10 20:18:30 +03:00
Sebastian Dröge
e9a0307b94 rtph26[45]pay: Change default aggregate-mode to "none" for backwards compatibility
We didn't aggregate at all in previous versions and there are apparently
various RTP implementations that don't handle aggregation well at all.

As part of this also document that for RTSP it is recommended to keep it
set to "none" while for WebRTC it should be set to "zero-latency".

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/692>
2020-08-08 10:08:31 +03:00
Jan Alexander Steffens (heftig)
28a616f693 splitmuxsink: Make sure flushing doesn't block
* Trying to disconnect a stream from a running splitmuxsink by flushing
  it results in the FLUSH_START blocking in the stream queue's
  gst_pad_pause_task because the flush did not unblock
  complete_or_wait_on_out, so add a check for ctx->flushing there.

* Add a GST_SPLITMUX_BROADCAST_INPUT so check_completed_gop notices
  flushing changed and the incoming push is unblocked.

* Pass the FLUSH_STOP along to the muxer without waiting.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/687>
2020-08-04 15:15:27 +00:00
Vivia Nikolaidou
af9e66d7a5 imagefreeze: Wait until we have a clock
Otherwise it can happen that it tries to get the clock in PAUSED state
in live mode, which does not exist.

Thanks to Sebastian Dröge for helping debugging.

Fixes #775

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/691>
2020-08-04 17:28:39 +03:00
Tim-Philipp Müller
a27e171bfa qtdemux: extract bit depth from codec data for ALAC
The info in the sound sample description might not be
accurate if it's an older version atom.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/771

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/686>
2020-07-31 11:05:02 +01:00
Jordan Petridis
516db3f1d0 auparse: fix compiler warnings
GCC 10 was complaining like following. It really is complaining about default cases returning
with potentially unitialized *desval, but those cases in the switch should never be hit.

```
 ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c: In function 'gst_au_parse_chain':
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:481:37: error: 'timestamp' may be used uninitialized in this function [-Werror=maybe-uninitialized]
  481 |       GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:482:36: error: 'duration' may be used uninitialized in this function [-Werror=maybe-uninitialized]
  482 |       GST_BUFFER_DURATION (outbuf) = duration;
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:480:34: error: 'offset' may be used uninitialized in this function [-Werror=maybe-uninitialized]
  480 |       GST_BUFFER_OFFSET (outbuf) = offset;
cc1: all warnings being treated as errors
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/671>
2020-07-29 19:21:31 +03:00
George Kiagiadakis
d997a8d48b rtspsrc: drop stream-start message posted by the internal udp sink(s)
See #1368

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/685>
2020-07-29 14:06:55 +03:00
Hosang Lee
f8e686078d qtdemux: create correct pad names in encrypted streams
Refer to "original-media-type" when setting stream's subtype
for encrypted streams in mss mode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/628>
2020-07-28 11:41:51 +00:00
Thibault Saunier
18aeb5bac1 matroskamux: Do caps renegotiation when it only adds fields
Matroskamux can accept caps renegotiation if the new caps is a
superset of the old one, meaning upstream added new info to
the caps.

Same logic as a5f22f03aa in qtmux.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/678>
2020-07-28 07:35:37 +00:00
Tim-Philipp Müller
10f07e84a5 rtpfunnel: protect internal srccaps with lock
These are modified from sink pad event handlers, so
could be accessed from multiple threads at the same
time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/681>
2020-07-28 07:08:04 +00:00
Havard Graff
f5fc34ae83 rtpfunnel: copy caps before sending them in a caps-event
Reason being we don't want downstream to own a ref to our
internal caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/681>
2020-07-28 07:08:04 +00:00
Mathieu Duponchelle
aa34c29d3b rtpmanager: fix various documentation issues
Improper naming of properties, improper links, misc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/684>
2020-07-27 13:51:15 +00:00
Stéphane Cerveau
c943be8b25 qtdemux: add Dolby Vision fourcc
This identifiers are registered in the MPEG-RA and defined
to be used by the Dolby Vision AVC/HEVC streams.

This is a first step to present the stream to the decoder.
Additional box parsing of DOVIConfigurationBox is necessary
to complete the media presentation with proper Dolby Vision
enhancements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/658>
2020-07-21 15:53:52 +00:00
Luke Yelavich
1e39fe66ad imagefreeze: Copy GstCapsFeatures to caps for source pad
Allows using imagefreeze with buffers in GLMemory. The following pipeline
works.

gst-launch-1.0 filesrc location=image.jpg ! jpegdec ! glupload ! \
imagefreeze ! glcolorconvert ! glimagesinkelement

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/594>
2020-07-20 21:12:09 +00:00
Tim-Philipp Müller
913e17e19e rtpmanager: fix "redefinition of typedef RTPTWCCManager" compiler warning
G_DECLARE_FINAL_TYPE includes this typedef as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/675>
2020-07-20 18:20:59 +01:00
Olivier Crête
7effe918d1 rtp*pay: Allocate using the base class for audio codecs
This is required to add RTP header extensions from the
meta automatically.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>
2020-07-17 16:53:40 -04:00
Ognyan Tonchev
adb044c9ed rtspsrc: Fix segfault with illegal free
set_get_param_q is not a pointer so it is illegal to call g_queue_free_full().
Freeing the requests by popping them from the queue instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/669>
2020-07-15 13:19:38 +00:00
Justin Chadwell
738f32d5d0 qtdemux: fix allocation explosion with stsd entries
Previously, the user input for stsd entries is trusted completely, and
so a maliciously crafted file could choose the length of the stsd
entries arbitrarily and cause qtdemux to try to allocate up to 2GB of
memory (half of a 32 bit max int).

This patch fixes this by sanity checking the stsd input against the
size of the entire stsd atom.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/670>
2020-07-15 12:10:45 +00:00
Justin Chadwell
e6f66f4681 qtdemux: fix crashes when input stream contained no stsd entries
During trak parsing, we need to check for the existence of stsd_entries,
otherwise, we end up with a NULL pointer to them. It is entirely
possible for the stsd to exist, but for it to have no entries, which the
previous checks did not take into account.

This patch adds a simply check to ensure that all files that do not
contain a stsd entry are deemed corrupt, and adds a test case to prevent
a regression.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/670>
2020-07-15 12:10:45 +00:00
Sebastian Dröge
54bc0157b5 qtmux: Don't lock object lock twice in prefill mode
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/762

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/663>
2020-07-07 12:36:01 +03:00
Tim-Philipp Müller
31ff328727 meson: add update-orc-dist target
Add target to update backup orc -dist.[ch] files.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/662>
2020-07-04 15:04:59 +01:00
Nirbheek Chauhan
c7f8c8d4ef deinterlace: Disable nasm support on x32
The assembly assumes pointers are 64-bit, so just disable it.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/757

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/660>
2020-07-02 07:53:14 +05:30
Nirbheek Chauhan
3fe4626e3c deinterlace: Fix build on x32
Need to pass `-f elfx32` to nasm in that case.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/757

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/657>
2020-07-01 19:43:41 +00:00
Jan Schmidt
7ae40045ba matroska-mux: Wait for caps on sparse streams
Don't set sparse streams to non-waiting at the collectpads
level until after capa arrive, as we need caps on all
pads before the file headers get written, or else the
subtitle track will be silently absent in the final file.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/724

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656>
2020-07-01 19:24:49 +01:00
Jan Schmidt
ed5e935fb7 matroska-mux: Warn on late caps arrival
As well as warning when caps change after the headers
were already written, make sure to warn if the *first* caos
arrive late too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656>
2020-07-01 16:13:27 +10:00
Sebastian Dröge
e589a950c3 imagefreeze: Return TRUE from the LATENCY query handling
We always answer it successfully no matter what.

The default return value in the function is FALSE even if the code below
sets it again to FALSE.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/654>
2020-06-30 18:37:06 +03:00
Sebastian Dröge
8345caf6e0 imagefreeze: Add a live mode
Previously imagefreeze would always operate as non-live element and
output frames as fast as possible according to the configured segment
(via SEEK events) and the negotiated framerate from start to stop or the
other way around.

With the new live mode (enabled via the is-live property) it would only
output frames in PLAYING. Frames would be output according to the
negotiated framerate unless it would be too late, in which case it would
jump ahead and skip over the requirement amount of frames.

This makes it possible to actually use imagefreeze in live pipelines
without having to manually ensure somehow that it would start outputting
at the current running time and without still risking to fall behind
without recovery.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>
2020-06-29 12:07:14 +03:00
Sebastian Dröge
06b29a4aef imagefreeze: Correctly answer the LATENCY query
We never run as a live element, even if upstream is live, and never
output any buffers with latency but immediately generate buffers as
fast as we can according to the negotiated framerate.

Passing the query upstream would proxy whatever mode of operation
upstream has, which has nothing to do with how we produce buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>
2020-06-28 22:26:23 +03:00
Tim-Philipp Müller
4f555ecf8e splitmuxsink: flesh out docs for format-location* signals
Make explicit that the returned strings need to be g_free()-able.

Fixes #753

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/652>
2020-06-26 18:31:10 +00:00
Havard Graff
57eebe8b05 rtpstats: guard against division by zero
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/646>
2020-06-26 15:29:25 +00:00
Havard Graff
e45cc475bd rtptwcc: fix pruning of ack'ed twcc-packets
Fixes #750

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/645>
2020-06-26 12:53:07 +01:00
Sebastian Dröge
13331e051f splitmuxsink: Add new properties for setting muxer/sink presets
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/644>
2020-06-25 17:53:00 +03:00
Mathieu Duponchelle
8d464c8361 autodetect: mark filter-caps property as DOC_SHOW_DEFAULT
When generating the cache we inspect the base class through
an instance of one of its subclasses. We don't want potential
assignments in subclasses initialization to leak into the
base class documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/641>
2020-06-24 17:04:51 +02:00
Mathieu Duponchelle
f97430f6e9 docs: mark GstIirEqualizer as plugin API 2020-06-23 19:04:03 +02:00
Mathieu Duponchelle
c897fe2b73 docs: mark more types as plugin API 2020-06-23 10:25:55 -04:00