Commit graph

10195 commits

Author SHA1 Message Date
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