Commit graph

10311 commits

Author SHA1 Message Date
Jan Schmidt
2623404744 rtph264depay: Warn when max SPS/PPS are collected in AVC mode.
The AVC codec_data has a flaw that it can only accomodate
31 SPS headers, even though H.264 can have 32, and 255 PPS,
when there can be 256 in H.264. When streaming RTP some
clients like to cycle through SPS/PPS ids when changing
configuration and can eventually accumulate a full set.

In that case, we have no choice but to discard one (oldest)
entry, or else the count written into the codec_data is wrong
and downstream decoding failures ensue.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
2020-10-29 14:09:21 +00:00
Havard Graff
63c7a9ae43 rtpjitterbuffer: don't send multiple instant RTX for the same packet
Due to us not properly acknowleding the time when the last RTX was sent
when scheduling a new one, it can easily happen that due to the packet
you are requesting have a PTS that is slightly old (but not too old when
adding the latency of the jitterbuffer), both its calculated second and
third (etc.) timeout could already have passed. This would lead to a burst
of RTX requests, which acts completely against its purpose, potentially
spending a lot more bandwidth than needed.

This has been properly reproduced in the test:
test_rtx_not_bursting_requests

The good news is that slightly re-thinking the logic concerning
re-requesting RTX, made it a lot simpler to understand, and allows us
to remove two members of the RtpTimer which no longer serves any purpose
due to the refactoring. If desirable the whole "delay" concept can actually
be removed completely from the timers, and simply just added to the timeout
by the caller of the API. But that can be a change for a another time.

The only external change (other than the improved behavior around bursting
RTX) is that the "delay" field now stricly represents the delay between
the PTS of the RTX-requested packet and the time it is requested on,
whereas before this calculation was more about the theoretical calculated
delay. This is visible in three other RTX-tests where the delay had
to be adjusted slightly. I am confident however that this change is
correct.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/789>
2020-10-28 01:22:24 +01:00
Jan Schmidt
0e84f42055 matroska-mux: Fix sparse stream crash
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656
introduced an invalid memory access when debug is enabled, by casting
the wrong pointer to a GstCollectPad. Fixing that showed the original
change was incorrect and leads to an infinite loop in the
testsuite. This patch fixes both problems.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/788>
2020-10-28 01:55:06 +11:00
Arun Raghavan
b4a713ff2d rtputils: Count metas with an empty tag list for copying/keeping
The GstMetaInfos registered in core do not set their tags to NULL, but
instead use an empty list (non-NULL list with a single NULL value).
Let's check explicitly for that so as to not miss some metas.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/779>
2020-10-22 09:19:53 -04:00
Nicolas Dufresne
b113516241 rtpbin: Add clear-ssrc action
This action signal will delegate to clear-ssrc onto the rtpssrcdemux element
associated with the session. This allow rtpbin users to clear pads and
elements for a specific ssrc that is known to no longer be in use. This
happens when a pad is reused in rtpsrc or ristsrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/736>
2020-10-16 16:45:56 +00:00
John-Mark Bell
3348c5ceae rtpvp8pay: payload temporally scaled bitstreams.
Co-Authored-By: Vincent Sanders <vince@pexip.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
2020-10-16 09:25:10 +00:00
Stian Selnes
29d5936749 rtpvp8pay: Add picture-id-offset property
Add property to set the initial value for picture-id. RFC7741 says
that picture-id MAY be initialized to a random value, thus it's also
valid to simply set it to a fixed initial value. A fixed value is very
useful for testing.

Default behavior is not changed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
2020-10-16 09:25:10 +00:00
Mikhail Fludkov
543b7e5024 rtpvp8pay: move duplicate code to separate functions
Two new functions to modify picture id:
gst_rtp_vp8_pay_picture_id_reset - picks random picture id of
appropriate bitsize
gst_rtp_vp8_pay_picture_id_increment - increments picture id taking
care of wrapping

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
2020-10-16 09:25:10 +00:00
Stéphane Cerveau
0429c24637 meson: update glib minimum version to 2.56
In order to support the symbol g_enum_to_string in various
project using GStreamer ( gst-validate etc.), the glib minimum
version should be 2.56.0.

Remove compat code as glib requirement
is now > 2.56

Version used by Ubuntu 18.04 LTS

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/774>
2020-10-15 18:21:54 +02:00
Mathieu Duponchelle
5fb5abc8a8 rtpst2022-1-fecenc: fix input seqnum check
We need to cast the incremented last seqnum to guint16 for
consistent checks on wraparound

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/770>
2020-10-14 14:30:34 +02:00
Jan Alexander Steffens (heftig)
a73ede42cf flvmux: Correct time types
- last_dts is in milliseconds, not nanoseconds as expected for
  GstClockTime. Make it a generic guint64.
- Use GstClockTime for the fields that actually contain nanoseconds.
  None of them should become negative.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/766>
2020-10-09 07:10:47 +00:00
Sebastian Dröge
6a84dc4146 rtpst2022-1-fecenc: Don't unconditionally use GLib 2.60 APIs
g_queue_clear_full() in this case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/768>
2020-10-09 09:31:27 +03:00
Mathieu Duponchelle
ed2b5e6cfc rtpulpfec: fix potential alignment issue in xor function
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753#note_646453
for context

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
2020-10-08 22:22:18 +00:00
Mathieu Duponchelle
591af0f38a rtpmanager: implement SMPTE 2022-1 FEC encoder
+ improve integration of FEC encoders in rtpbin

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
2020-10-08 22:22:18 +00:00
Mathieu Duponchelle
cff42d4c26 rtpmanager: implement SMPTE 2022-1 FEC decoder
+ improve integration of FEC decoders in rtpbin

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
2020-10-08 22:22:18 +00:00
Olivier Crête
7c9a5e86fe rtpfunnel: Also forward custom sticky event
This is useful to track metadata about each group of packets

Also include a unit test

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/666>
2020-10-06 20:57:49 +00:00
Thibault Saunier
6eef0967b9 isomp4: Rename GstQTMux to GstBaseQTMux to avoid breaking API
Since 52b63de19a the qtmux GType was
renamed GstQTMuxElement which breaks presets, revert that change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/755>
2020-09-30 09:18:13 -03:00
Sebastian Dröge
f95dde512c rtp: Fix allocations to support source-info property
Use gst_rtp_base_payload_allocate_output_buffer() instead of
gst_rtp_buffer_new_allocate() in order to allocate RTP buffer with
correct number of CSRCs according to the meta.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
2020-09-28 15:27:17 +00:00
Stian Selnes
d494be9916 rtpvp8pay: Fix allocation to support source-info property
Use gst_rtp_base_payload_allocate_output_buffer() in order to allocate
RTP buffer with correct number of CSRCs according to the meta.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/314

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
2020-09-28 15:27:17 +00:00
Matthew Waters
7736a21659 qtmux: output the correct limits in error messages
Having the current bytes being less than the limit was confusing!

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
2020-09-28 15:37:12 +10:00
Matthew Waters
e81ce6f2d7 qtmux: properly support initial caps nego failure
Scenario:
- gap event causes h264parse to push made up caps that may fail checks
  inside qtmux (e.g missing codec_data).
- the caps event has already been marked as received and is sticky on
  the sink pad
- gst_qt_mux_pad_can_renegotiate() will retrieve the failed caps event
  using gst_pad_get_current_caps() and reject the correct updated caps
  with codec_data.
- Failure!

Keep track of the configured caps ourselves instead of relying on the
sticky event on the pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
2020-09-28 15:37:12 +10:00
Matthew Waters
b27dc540d0 qtmux: support non-seekable downstream mode
Write an mdat per buffer in that case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
2020-09-28 15:37:12 +10:00
Nicolas Dufresne
345f74b09d rtpbin: Remove the rtpjitterbuffer with the stream
Since !348, the jitterbuffer was only removed with the session. This restores
the original behaviour and removes the jitterbuffer when the stream is
removed. This avoid accumulating jitterbuffer objects into the bin when a
session is reused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
2020-09-24 09:54:05 -04:00
Nicolas Dufresne
ecc110ca8b rtpbin: Cleanup dead code
The rtpjitterbuffer is now part of the session elements, we no longer need
to do the ref_sink dance when signalling it. It is already owned by the bin
when signalled. Also, the code that handles generic session elements already
handle the ref_sink() calls since:

03dc22951b

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
2020-09-23 15:48:24 -04:00
Matthew Waters
ea61714c70 rtph26*depay: drop FU's without a corresponding start bit
If we have not received a FU with a start bit set, any subsequent FU
data is not useful at all and would result in an invalid stream.

This case is constructed from multiple requirements in
RFC 3984 Section 5.8 and RFC 7798 Section 4.4.3.  Following are excerpts
from RFC 3984 but RFC 7798 contains similar language.

The FU in a single FU case is forbidden:

   A fragmented NAL unit MUST NOT be transmitted in one FU; i.e., the
   Start bit and End bit MUST NOT both be set to one in the same FU
   header.

and dropping is possible:

   If a fragmentation unit is lost, the receiver SHOULD discard all
   following fragmentation units in transmission order corresponding to
   the same fragmented NAL unit.

The jump in seqnum case is supported by this from the specification
instead of implementing the forbidden_zero_bit mangling:

   If a fragmentation unit is lost, the receiver SHOULD discard all
   following fragmentation units in transmission order corresponding to
   the same fragmented NAL unit.

   A receiver in an endpoint or in a MANE MAY aggregate the first n-1
   fragments of a NAL unit to an (incomplete) NAL unit, even if fragment
   n of that NAL unit is not received.  In this case, the
   forbidden_zero_bit of the NAL unit MUST be set to one to indicate a
   syntax violation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/730>
2020-09-21 08:08:38 +00:00
Seungha Yang
027940a416 imagefreeze: Response caps query from srcpad
... and chain up to default query handler for unhandled query types.
Unhandled query shouldn't be returned with FALSE if there's no special needs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/731>
2020-09-21 10:28:01 +03:00
Matthew Waters
e64227f585 qtmux: make documentation happy
introduce a base qtmux class that we can install documentation snippets
on instead of duplicating across alll the isomp4 elements

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
2020-09-21 12:09:09 +10:00
Matthew Waters
52b63de19a isomp4/mux: add a fragment mode for initial moov with data
Used by some proprietary software for their fragmented files.

Adds some support for multi-stream fragmented files

Flow is as follows.
1. The first 'fragment' is written as a self-contained fragmented
   mdat+moov complete with an edit list and durations, tags, etc.
2. Subsequent fragments are written with a mdat+moof and each stream is
   interleaved as data arrives (currently ignoring the interleave-*
   properties).  data-offsets in both the traf and the trun ensure
   data is read from the correct place on demuxing.  Data/chunk offsets
   are also kept for writing out the final moov.
3. On finalisation, the initial moov is invalidated to a hoov and the
   size of the first mdat is extended to cover the entire file contents.
   Then a moov is written as regularly would in moov-at-end mode (the
   default).

This results in a file that is playable throughout while leaving a
finalised file on completion for players that do not understand
fragmented mp4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
2020-09-21 12:08:14 +10:00
Matthew Waters
97e932d500 qtdemux: increase some logging on streams and sample parsing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
2020-09-21 12:08:14 +10:00
Matthew Waters
37f0119f49 qtdemux: bail out when encountering an atom with a size of 0
A size 0 atom means the atom extends to the end of the file.  No further
valid atoms will ever follow.  Avoids a subsequent scan for an atom from
one byte earlier after encountering a size 0 atom.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
2020-09-21 12:08:14 +10:00
Matthew Waters
868149ca5a qtdemux: fix subsequent moof parsing after moov with valid samples
reset the moof_offset back to its original value like is done in the
error case just before.

Fixes subsequent parsing of a moof following a moov that contains valid
samples in a non-streaming fragmented mp4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
2020-09-21 12:08:14 +10:00
Matthew Waters
2b9c465643 qtdemux: extend edit list when fragmented
When we are fragmented, the edit list may only refer to the portion of
the media that is in the moov.  Extend the edit list stop time when we
if there is only one qt segment and we are reading a fragmented file.

Fixes playback of some fragmented mp4 files generated by proprietary
programs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
2020-09-21 12:08:14 +10:00
Olivier Crête
c79a520946 splitmuxsrc: Implement segment query
Fixes #239

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/713>
2020-09-18 10:54:23 -04:00
Sebastian Dröge
c90af726ab rtpmp4gdepay: Allow lower-case "aac-hbr" instead of correct "AAC-hbr"
Various live555 based products are using the wrong "mode" string or
seem to assume case-insensitive matching, which is wrong.

Examples for this are the Yuan SC6C0N1 mini and the Kiloview E2.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/727>
2020-09-18 10:02:44 +03:00
Stefan Brüns
ee3ea2a94d qtdemux: Add support for AAX encrypted audio streams
This is modelled after the DASH Common Encryption scheme, but is somewhat
simpler as more parts are fixed, i.e. just one encryption scheme.

The output caps are fixed to 'application/x-aavd'. All information
required for decryption are part of the 'adrm' atom, which is passed
on as a property. The property is attached to the buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
2020-09-16 00:59:34 +00:00
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
He Junyan
15fac84f63 deinterlace: Add the missing ORC_RESTRICT define.
ORC_RESTRICT may not be defined in yadif.c and cause build error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/637>
2020-06-21 20:11:06 +08:00
Seungha Yang
9122bfdfb7 meson: deinterlace: Check host cpu type for asm build
Add host cpu type check as we would enable asm only for x86_64

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/636>
2020-06-19 20:28:14 +09:00
Jan Schmidt
5c68e06b00 qtdemux: Split tag reading functions out
Move some code out of the enormous qtdemux.c into a separate
qtdemux_tags helper, and make some structs available via qtdemux.h
to accommodate that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/634>
2020-06-18 14:41:27 +00:00
Jan Schmidt
0ddfc5020f qtdemux: Move some tree parsing files out to a separate file.
Reduce a tiny bit of the bulk of qtdemux.c by moving some
agnostic helper functions out.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/634>
2020-06-18 14:41:27 +00:00
Jan Schmidt
e2d75939bb qtdemux: Factor out svmi parsing. Fix bounds checking.
Move the SVMI stereoscopic atom parsing out to a helper
function to shrink qtdemux_parse_trak a bit.

Add a bounds check that the received atom is large enough
before parsing it.

Add a note to the atom parser that svmi comes from the
MPEG-A spec 23000-11.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/634>
2020-06-18 14:41:27 +00:00
Seungha Yang
8b4f18d53b rtspsrc: Don't return TRUE for unhandled query
Expected return value for unhandled query is FALSE

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/629>
2020-06-16 19:35:30 +09:00
Vivia Nikolaidou
536ff4776f deinterlace: Add yadif ASM optimisations
Measured to be about 3.4x faster than C

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/621>
2020-06-16 12:53:25 +03:00
Vivia Nikolaidou
ef78014d15 deinterlace: Fix invalid read in yadif
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/621>
2020-06-12 13:21:02 +03:00
Sebastian Dröge
556e7ab210 flvdemux: Change a GST_ERROR_OBJECT() back to GST_DEBUG_OBJECT()
It was accidentally changed in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/436

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/624>
2020-06-12 09:52:56 +03:00
Jordan Petridis
3e2420361a Use gst_element_class_set_metadata when passing dynamic strings
gst_element_class_set_metadata is meant to only be used with
static or inlined strings, which isn't the case for the 2 elements
here resulting in use-after-free later on.

https://gstreamer.freedesktop.org/documentation/gstreamer/gstelement.html?gi-language=c#gst_element_class_set_static_metadata

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/622>
2020-06-11 20:39:33 +03:00
Sebastian Dröge
f8196e06d5 Revert "rtpjitterbuffer: Avoid deadlock on flush"
This reverts commit 54810bf44f

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/620>
2020-06-10 16:31:06 +00:00
U. Artie Eoff
bf0842aa0c rtpjitterbuffer: g_queue_clear_full introduced in glib 2.60
Define g_queue_clear_full if glib < 2.60.

Fixes #747

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/619>
2020-06-09 13:09:20 -07:00
Thibault Saunier
29a661d4a4 rtpsession: Make internal-ssrc as show default for doc 2020-06-09 11:45:13 -04:00
Nicolas Dufresne
5b2ad31583 rtptimerqueue: Fix leak on timer collision
While the caller should make sure this does not happen, make sure timer
collision are not silently ignored and leaked.

Fixes #726

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/616>
2020-06-08 17:54:53 -04:00
Nicolas Dufresne
b4f421e9aa rtpjitterbuffer: Keep JBUF lock while processing timers
Until now, do_expected_timeout() was shortly dropping the JBUF_LOCK in order
to push RTX event event without causing deadlock. As a side effect, some
CPU hung would happen as the timerqueue would get filled while looping over
the due timers. To mitigate this, we were processing the lost timer first and
placing into a queue the remainign to be processed later.

In the gap caused by an unlock, we could endup receiving one of the seqnum
present in the pending timers. In that case, the timer would not be found and
a new one was created. When we then update the expected timer, the seqnum
would already exist and the updated timer would be lost.

In this patch we remove the unlock from do_expected_timeout() and place all
pending RTX event into a queue (instead of pending timer). Then, as soon as
we have selected a timer to wait (or if there is no timer to wait for) we send
all the upstream RTX events. As we no longer unlock, we no longer need to pop
more then one timer from the queue, and we do so with the lock held, which
blocks any new colliding timers from being created.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/616>
2020-06-08 17:54:53 -04:00
Edward Hervey
54810bf44f rtpjitterbuffer: Avoid deadlock on flush
When a GST_EVENT_FLUSH_START reaches the jitterbuffer, there is a chance that
our task is currently blocking waiting for a timer.

There was two problems:
* That wait wasn't checking for flushing situations
* The flushing handling wasn't waking up that conditional (to check whether it
should abort)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/608>
2020-06-08 13:34:26 +02:00
Mathieu Duponchelle
f63299ff2f plugins: uddate gst_type_mark_as_plugin_api() calls 2020-06-06 00:42:25 +02:00
Sebastian Dröge
e527eb3e4c rtpbin: Initialize uninitialized variable correctly
`last_out` would be used uninitialized if the element has no `set-active`
signal. Initialize it to -1 as that's what the "default" value is
further below.

CID 1455443

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/727

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/613>
2020-06-05 11:49:17 +03:00
Thibault Saunier
6f0f41fef0 doc: Fix wrong link to GString in rtpjitterbuffer 2020-06-03 22:44:09 -04:00
Mathieu Duponchelle
37c619f995 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types 2020-06-03 22:44:09 -04:00
Sebastian Dröge
b94b9988fa rtspsrc: Use the correct type for storing the max-rtcp-rtp-time-diff property
It's an integer property and rtpbin also expects an integer. Passing it
as a GstClockTime (guint64) to g_object_set() will cause problems, and
on big endian MIPS apparently causes crashes.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/737

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/605>
2020-05-27 22:33:31 +03:00
Thibault Saunier
3fdae346ca rtspsrc: Error out when failling to receive message response
And let it rety twice.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/717

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/601>
2020-05-25 20:13:06 -04:00
Sebastian Dröge
2c278bb2ab flvdemux: Send gap events if one of the streams falls behind the other by more than 3s
Same mechanism and threshold as in other demuxers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/597>
2020-05-20 18:46:41 +00:00
Sebastian Dröge
0bb9880b36 flvdemux: Remove unused audio_linked/video_linked booleans
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/597>
2020-05-20 18:46:41 +00:00
Edward Hervey
5dd3643d94 flvdemux: Answer bitrate queries from upstream
If upstream (such as queue2 in urisourcebin) asks for our bitrate, check if we
have stored audio/video bitrates, and use them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/596>
2020-05-20 16:51:47 +03:00
Edward Hervey
e8282661b6 flvdemux: Handle empty metadata strings
g_utf8_validate() errors out on empty string. But empty strings are valid,
so only check if they're not

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/596>
2020-05-20 10:48:06 +02:00
Edward Hervey
9f5f906515 flvdemux: Set ACCEPT_TEMPLATE flag on sinkpad
A demuxer can accept any caps matching its sinkpad template caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/596>
2020-05-20 10:48:06 +02:00
Jan Schmidt
d8f0deadc3 deinterlace: Split out NULL checks in yadif
Separate out explicit NULL checks for fields we depend on so
that coverity can hopefully verify dependencies better.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/585>
2020-05-09 03:09:03 +10:00
Jan Schmidt
1106eb16b6 deinterlace: Handle NV12/NV21 for the greedyl mode.
Don't fall back on the default interpolate_scanline function, which
blindly tries to copy from the next field, which can be NULL in
mixed progressive/interlaced streams

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/585>
2020-05-09 03:07:33 +10:00
Vivia Nikolaidou
82dc670f1f deinterlace: Support packed formats for YADIF
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>
2020-05-06 17:08:06 +00:00
Vivia Nikolaidou
5fce46f5ef deinterlace: Call the planar functions for the Y plane of nv12/nv21
In some algorithms (like yadif), the Y plane has to be handled different
than the UV plane. Therefore, the planar_y functions are now called for
the Y plane, and the nv12/nv21 functions are handling only the UV/VU
planes respectively.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>
2020-05-06 17:08:06 +00:00
Jan Schmidt
e9ee7ab0af deinterlace: Add C implementation of YADIF
Import the YADIF deinterlacer from ffmpeg and modify
it to match the simple deinterlace scanlines structure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>
2020-05-06 17:08:06 +00:00
Jan Schmidt
1c1bc56a3b deinterlace: Allow for 5 fields for interpolation
Add an extra field to the simple deinterlace implementation,
so that methods can potentially use 5 fields - the current
field, and 2 before and 2 after.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/444>
2020-05-06 17:08:06 +00:00
Jan Schmidt
5468988223 deinterlace: Force renegotiation when changing mode
Switching the deinterlacing mode on-the-fly from disabled to
auto used to work, but was broken by commit #1f21747c some
years ago.

Force re-negotiation with downstream when the mode or
fields properties are changed, otherwise deinterlace
never switches out of the passthrough mode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/584>
2020-05-07 01:31:59 +10:00
Sebastian Dröge
e5feaa76ed imagefreeze: Handle flushing correctly
First of all get rid of the atomic seeking boolean, which was only ever
set and never read. Replace it with a flushing boolean that is used in
the loop function to distinguish no buffer because of flushing and no
buffer because of an error as otherwise we could end up in a
GST_FLOW_ERROR case during flushing.

Also only reset the state of imagefreeze in flush-stop when all
processing is stopped instead of doing it as part of flush-start.

And last, get a reference to the imagefreeze buffer in the loop function
in the very beginning and work from that as otherwise it could in theory
be replaced or set to NULL in the meantime as we release and re-take the
mutex a couple of times during the loop function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/580>
2020-05-06 08:06:33 +00:00
Edward Hervey
756f390f56 videbox: Use MIN instead of CLAMP for uint
an unsigned int is always positive.

CID #206207
CID #206208
CID #206209
CID #206210
CID #206211

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/583>
2020-05-06 06:49:09 +00:00
Edward Hervey
619457ae26 avidemux: Avoid potential double-free
stream->name was being freed (without being NULL-ed) before we were certain it
would be set again.

CID #1456071

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/582>
2020-05-06 04:36:46 +00:00
Edward Hervey
518d192dc5 deinterlace: Don't leak frame in error case
CID #1455494

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/581>
2020-05-05 17:30:48 +02:00
Edward Hervey
cfb9a5d53a slitmuxsrc: Properly stop the loop if not part reader is present
Previously this would end up in a refcounting loop of hell.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/578>
2020-05-05 15:32:58 +02:00
Vivia Nikolaidou
6a38961561 flvmux: Add skip-backwards-streams property
Backwards timestamps confuse librtmp, even if they're only backwards
relative to the other stream. If the timestamp of a stream is going
backwards related to the other stream, this property allows the muxer to
skip a few buffers until it reaches the timestamp of the other stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/572>
2020-04-27 16:18:34 +03:00
Vivia Nikolaidou
b0855113c6 flvmux: Allow requesting streamable pads after header is written
Allows us to request pads after writing header for streamable flv's.

For non-streamable it doesn't make sense to request a new pad after
writing the header, because the headers have been written already and we
can't add the new stream. But for streamable, any clients that connect
after the new pad has been added will be able to see both streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/572>
2020-04-27 14:11:10 +03:00
Olivier Crête
3ae1bae2a3 qtdemux: Add 'mp3 ' fourcc that VLC seems to produce now
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/574>
2020-04-22 15:32:31 -04:00
Sebastian Dröge
7b22397cf5 rtpjitterbuffer: Properly free internal packets queue in finalize()
As we override the GLib item with our own structure, we cannot use any
function from GList or GQueue that would try to free the RTPJitterBufferItem.
In this patch, we move away from g_queue_new() which forces using
g_queue_free(). This this function could use g_slice_free() if there is any items
left in the queue. Passing the wrong size to GSLice may cause data corruption
and crash.

A better approach would be to use a proper intrusive linked list
implementation but that's left as an exercise for the next person
running into crashes caused by this.

Be ware that this regression was introduced 6 years ago in the following
commit [0], the call to flush() looked useless, as there was a g_queue_free()
afterward.

Signed-off-by: Nicolas Dufresne <nicolas.dufresne@collabora.com>

[0] 479c7642fd

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/573>
2020-04-22 10:28:30 -04:00
Seungha Yang
ca48f5265e splitmuxsink: Enhancement for timecode based split
The calculated threshold for timecode might be varying depending on
"max-size-timecode" and framerate.
For instance, with framerate 29.97 (30000/1001) and
"max-size-timecode=00:02:00;02", every fragment will have identical
number of frames 3598. However, when "max-size-timecode=00:02:00;00",
calculated next keyframe via gst_video_time_code_add_interval()
can be different per fragment, but this is the nature of timecode.
To compensate such timecode drift, we should keep track of expected
timecode of next fragment based on observed timecode.
2020-04-20 21:39:49 +09:00
Seungha Yang
fe73c3b0f3 splitmuxsink: Post error when requested timecode interval is invalid
In case we cannot rely on max-size-timecode for split decision,
post error instead of crashing
2020-04-19 20:23:32 +09:00
Havard Graff
981d0c02de rtpjitterbuffer: don't use RTX packets in rate-calc and reset-logic
The problem was this:

Due to the highly irregular arrival of RTX-packet the max-misorder variable
could be pushed very low. (-10).

If you then at some point get a big in the sequence-numbers (62 in the
test) you end up sending RTX-requests for some of those packets, and then
if the sender answers those requests, you are going to get a bunch of
RTX-packets arriving. (-13 and then 5 more packets in the test)

Now, if max-misorder is pushed very low at this point, these RTX-packets
will trigger the handle_big_gap_buffer() logic, and because they arriving
so neatly in order, (as they would, since they have been requested like
that), the gst_rtp_jitter_buffer_reset() will be called, and two things
will happen:
1. priv->next_seqnum will be set to the first RTX packet
2. the 5 RTX-packet will be pushed into the chain() function

However, at this point, these RTX-packets are no longer valid, the
jitterbuffer has already pushed lost-events for these, so they will now
be dropped on the floor, and never make it to the waiting loop-function.

And, since we now have a priv->next_seqnum that will never arrive
in the loop-function, the jitterbuffer is now stalled forever, and will
not push out another buffer.

The proposed fixes:
1. Don't use RTX in calculation of the packet-rate.
2. Don't use RTX in large-gap logic, as they are likely to be dropped.
2020-04-16 17:06:31 +02:00
Sebastian Dröge
d75ea5b340 splitmuxsink: Do split-at-running-time splitting based on the time of the start of the GOP
If the start of the GOP is >= the requested running time, put it into a
new fragment. That is, split-at-running-time would always ensure that a
split happens as early as possible after the given running time.

Previously it was comparing against the current incoming timestamp,
which does not tell us what we actually want to know as it has no direct
relation to the GOP start/end.
2020-04-15 17:52:41 +03:00
Sebastian Dröge
0ab0f92cac splitmuxsink: Fix off-by-one in running time comparison for split-at-running-time
If we get a keyframe exactly at the requested running time we would only
split on the next keyframe afterwards due to wrong usage of > vs. >=.
2020-04-15 13:33:17 +03:00
Thibault Saunier
fd7ecac793 rtspsrc: Properly set segments seqnums after seeks 2020-04-09 14:03:04 -04:00
Vivia Nikolaidou
9189cdcb1d flvdemux: Don't write an empty string as a tag
To stop warnings like:

GStreamer-WARNING **: 19:47:48.186: Trying to set empty string on
taglist field 'encoder'. Please file a bug.
2020-04-08 20:22:51 +03:00
Thibault Saunier
00539e1277 rtspsrc: Avoid stack overflow recursing waiting for response
Instead of recursing, simply implement a loop with gotos, the same
way it was done before 8121752887

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/710
2020-04-08 09:49:49 -04:00
Sebastian Dröge
cf3fbf57bf qtmux: Add property for enforcing the creation of chunks in single-stream files
This is disabled by default as it unnecessarily creates bigger headers
but it is something that is required by some applications and most
notably the Apple ProRes spec.
2020-04-06 16:25:59 +03:00
Jan Schmidt
a3933ea53d flvmux: Fix invalid padlist accesses.
Request pads can released at any time, so make sure to hold
the object lock when iterating the element sinkpads list where
that's safe, or to use other safe pad iteration patterns in
other places.

When choosing a best pad, return a reference to the pad to make sure it
stays alive for output in the aggregator srcpad task.

Should fix a spurious valgrind error in the CI flvmux tests and some
other potential problems if the request sink pads are released while
the element is running..

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/714
2020-04-05 11:50:43 +00:00
Vivia Nikolaidou
5817c659e6 qtmux: Add option to create a timecode trak in non-mov flavors
Even if timecode trak is officially unsupported in non-mov flavors,
some software still supports it, e.g. Final Cut Pro X:

https://developer.apple.com/library/archive/technotes/tn2174/_index.html

The user might still expect to see the timecode information in the
non-mov file despite it being officially unsupported , because other
software e.g. QuickTime will create a timecode trak even in mp4 files.
Furthermore, software that supports timecode trak in non-mov flavors
will also display the file duration in "timecode units" instead of real
clock time, which is not necessarily the same for 29.97 fps and friends.
This might confuse users, who see a different duration for the same
framerate and amount of frames depending on whether the container is mp4
or mov.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/512
2020-04-03 18:19:38 +00:00
Sebastian Dröge
db69f02dd8 rtpLXXdepay: Set the UNPOSITIONED flag on the audio-info when configuring an unpositioned layout
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/688
2020-04-03 17:57:23 +00:00
Kristofer Björkström
586fc57e55 rtpjpeg: Use gst_memory_map() instead of gst_buffer_map()
gst_buffer_map () results in memcopying when a GstBuffer contains
more than one GstMemory.
This has quite an impact on performance on systems with limited amount
of resources. With this patch the whole GstBuffer will not be mapped at
once, instead each individual GstMemory will be iterated and mapped
separately.
2020-04-03 17:01:24 +02:00
Kristofer Björkström
54b6ee0c55 buffermemory: keep track of buffer size and current offset
Added the possibility to get current offset and the total size of the
buffer.
2020-04-03 17:01:24 +02:00
Havard Graff
d9aaa15a30 rtpopuspay: make depay ! pay work
There is a use-case for a server to re-payload opus going through it.

Problem was that the payloader requires channels in the caps, but
this is not something the depayloader can parse out of the stream, meaning
caps-negotiation would fail.

Removing the requirement of channels in the template-caps fixes this.
2020-04-03 09:04:32 +00:00
Seungha Yang
599066726f splitmuxsink: Don't send too many force key unit event
splitmuxsink should requst keyframe depending on configured
threshold and previously requested time in order to avoid too many
keyframe request.
2020-04-03 15:00:37 +09:00
Jan Schmidt
78eaa7c6ed matroska: Check the return value of gst_segment_do_seek()
gst_segment_do_seek() can fail.
2020-04-02 05:23:17 +00:00
Sebastian Dröge
f757fbe0f7 qtdemux: Send instant-rate-change event if requested in the SEEK event
Handle an instant rate change seek immediately by reflecting
it downstream as an instant-rate-change event, and do no
further seek handling.
2020-04-02 05:23:17 +00:00
Sebastian Dröge
5d0657d4ae matroska-demux: Send instant-rate-change event if requested in the SEEK event
Short-circuit instant rate change events by generating
a downstream instant-rate-change event and doing no further
seek processing.
2020-04-02 05:23:17 +00:00
Seungha Yang
cb8c83e799 matroska: Update for video-hdr struct change
See the change of -base https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/594
2020-04-01 05:19:24 +00:00
Aaron Boxer
3b65663846 rtpbin: make warning messages more meaningful 2020-03-31 15:51:27 -04:00
Nicolas Pernas Maradei
ce0fb9bd29 rtpsession: rename RTCP thread
RTP session starts a new thread for RTCP and names it
"rtpsession-rtcp-thread" which happens to be longer than the maximum 16B
allowed by pthread_setname_np and causes the naming to fail.
See docs for more details.

This commit simply shortens the thread's name so it can actually be set.
2020-03-31 13:34:07 +02:00