Commit graph

1495 commits

Author SHA1 Message Date
Guillaume Desmottes
1796f3f5e4 wavparse: fix seeking in READY state
wavparse claims to be able to support seeking in the READY state by
saving the pending seek event and actually seeking later after having parsed the
header.
Problem was that this seek event was reset on the READY to PAUSED
transition, making all this code useless. Fixing it by stop resetting
on READY to PAUSED transition as we already reset on PAUSED to READY
and when initiating the element.

Note that DTS marker detection isn't support in such scenario as
gst_type_find_helper_for_buffer() needs a buffer containing the
beginning of the stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>
2021-02-18 16:32:24 +01:00
Guillaume Desmottes
4aa39da2d3 tests: wavparse: factor out create_pipeline()
No semantic change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>
2021-02-18 10:38:18 +01:00
Jakub Adam
b105797163 tests: add rtpopus multichannel test cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
2021-02-11 07:46:04 +00:00
Guillaume Desmottes
7b7e49de31 rtp: add rtphdrextrfc6464
Header Extension for Client-to-Mixer Audio Level Indication as
defined in RFC 6464.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
2021-02-04 11:12:51 +01:00
Guillaume Desmottes
4b6c3c9a1b level: add GstRTPAudioLevelMeta on buffers
This meta can be used by a RTP payloader to send the level information
to the peer.

Part of https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/446

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
2021-02-04 11:12:47 +01:00
Matthew Waters
db15ec9286 videoflip: fix possible crash when setting the video-direction while running
A classic case of not enough locking.

One interesting thing with this is the interaction between the
rotation value and caps negotiation.  i.e. the width/height of the caps
can be swapped depending on the video-direction property.  We can't lock
the entirety of the caps negotiation for obvious reasons so we need to
do something else.  This takes the approach of trying to use a single
rotation value throughout the entirety of the negotiation and then
subsequent output frame in a kind of latching sequence.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/792
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
2021-01-04 12:10:12 +00:00
Matthew Waters
35018d67ef tests: add tests for videoflip
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
2021-01-04 12:10:12 +00:00
Jan Schmidt
2d24a45c89 splitmuxsink: Unit test - check format/opened/closed sequence
Check the sequence of format-location/fragment-opened/fragment-closed
events is respected. There should be 1 format-location call for each
fragment-opened message, and 1 fragment-closed for each.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
2020-12-12 03:28:56 +11:00
Marijn Suijten
030b1b3fa5 tests/rtp-payloading: Use new AudioFormatInfo::fill_silence function
The function is renamed to be properly associated with AudioFormatInfo
(its instance) instead of AudioFormat (an unrelated enum), see [1] for
the rename itself.

[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940
2020-11-26 10:06:25 +02:00
Havard Graff
79748dab2b rtpsession: never send on a non-internal source
This will end up as a "received" packet, due to the code in
source_push_rtp, which will think this is a packet being received.

Instead drop the packet and hope that either:
1. Something upstream responds to the GstRTPCollision event and changes
   SSRC used for sending.
2. That the application responds to the "on-ssrc-collision" signal, and
   forces the sender (payloader) to change its SSRC.
3. That the BYE sent to the existing user of this SSRC will respond to
   the BYE, and that we timeout this source, so we can continue sending
   using the chosen SSRC.

The test reproduces a scenario where we previously would have sent
on a non-internal source.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
2020-11-13 21:35:58 +01:00
Tim-Philipp Müller
f5310ce346 tests: qtdemux: fix typo in caps field
timesacle -> timescale

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
2020-11-12 23:40:13 +00:00
Tim-Philipp Müller
2ce5909f3a tests: qtdemux: fix crash on 32-bit architectures
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/803

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
2020-11-12 23:40:13 +00:00
Jan Schmidt
81ecf076e8 splitmuxsrc: Fix comment in a test
Fix a comment in the splitmuxsrc robust muxing test so it
describes the test properly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
2020-10-31 02:50:51 +00:00
Stian Selnes
95579a00c0 rtpvp9depay: Improve SVC parsing, aggregate all layers
- Fix start and end of picture to support multiple layers. Start of
  picture is the first packet of the base layer, while end of picture
  is when the marker bit is set (last packet of the enhancement
  layers).
- All "layers" (aka "frames") of a picture are pushed downstream in a
  single buffer when picture is complete.
- Forgive SID=0 for enhancement layers (invalid, but Chrome and
  Firefox sends it)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/773>
2020-10-30 17:46:30 +01:00
Stian Selnes
d77fcf251b rtpvp8depay: Send lost events when marker bit is missing
This means the previous frame was incomplete.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/796>
2020-10-30 03:43:19 +01:00
Mikhail Fludkov
346b077ae0 rtpvp*depay: possibly forward might-have-been-fec PacketLost events
This is ad adaptation of a Pexip patch for dealing with spurious
GstRTPPacketLost events caused by lost ulpfec packets: as FEC packets
under that scheme are spliced in the same sequence domain as the media
packets, it is not generally possible to determine whether a lost packet
was a FEC packet or a media packet.

When upstreaming pexip's ulpfec patches, we decided to drop all lost
events at the base depayloader level, and where the original patch
from pexip was making use of picture ids and marker bits to determine
whether a packet should be forwarded, this patch makes use of those
to determine whether they should be dropped instead (by removing their
might-have-been-fec field).

Spurious lost events coming out of the depayloader can cause the
decoder to stop decoding until the next keyframe and / or request a new
keyframe, and while this is not desirable it makes sense to forward
that information when we have other means to determine whether a lost
packet was indeed a FEC packet, as is the case with VP8 / VP9 payloads
when they carry a picture id.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>
2020-10-29 19:56:07 +01:00
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
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
John-Mark Bell
d9cedee042 vp8enc: finish support for temporally scaled encoding
- introduce two new properties:

    * temporal-scalability-layer-flags:

      Provide fine-grained control of layer encoding to the
      outside world. The flags sequence should be a multiple of
      the periodicity and is indexed by a running count of encoded
      frames modulo the sequence length.

    * temporal-scalability-layer-sync-flags:

      Specify the pattern of inter-layer synchronisation (i.e.
      which of the frames generated by the layer encoding
      specification represent an inter-layer synchronisation).
      There must be one entry per entry in
      temporal-scalability-layer-flags.

  - apply temporal scalability settings and expose as buffer
    metadata.

    This allows the codec to allocate a given frame to the correct
    internal bitrate allocator. Additionally, all the
    non-bitstream metadata needed to payload a temporally scaled
    stream is now attached to each output buffer as a
    GstVideoVP8Meta.

  - add unit test for temporally scaled encoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
2020-10-16 09:25:10 +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
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
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
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
John-Mark Bell
8f684913cf vp8enc: improve unit tests
- make test_encode_simple cope with libvpx built with
    CONFIG_REALTIME_ONLY. Sadly, there's no way to detect this at
    runtime beyond trying to set lag-in-frames to >0, pushing a
    buffer and catching the GST_FLOW_NOT_NEGOTIATED return.

  - fix bitrot in test_encode_simple_when_bitrate_set_to_zero.

  - port test_encode_simple to GstHarness and introduce a separate
    test for the lag-in-frames property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/708>
2020-09-08 22:59:29 +00:00
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
Matthew Waters
3296a03d73 build: update for gl pkg-config file split
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/680>
2020-08-07 07:58:29 +00:00
Hosang Lee
d6f6e8410e tests: qtdemux: test correct pad names are created
Test correct pad names are created in accordance to their media type
in mss mode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/628>
2020-07-28 11:41:51 +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
Tim-Philipp Müller
3b0437e58d examples: hook up rpicamsrc examples
webrtc one should probably go into gst-examples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>
2020-07-10 17:37:28 +01:00
Tim-Philipp Müller
c22b71e181 examples: fix indentation of rpicamsrc examples
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>
2020-07-10 17:37:28 +01:00
Tim-Philipp Müller
84dbf94313 Merge branch 'plugin-move-rpicamsrc'
Move rpicamsrc from https://github.com/thaytan/gst-rpicamsrc/

It's a useful little element and works well, so might as well
make sure it's widely available so people can stop piping
raspivid output into fdsrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/667>
2020-07-10 17:36:14 +01:00
Jan Schmidt
41f41f1fdd rpicamsrc: webrtc example: Add a STUN server to the configuration
To let the webrtc example work through NAT firewalls
2020-07-10 16:46:30 +01:00
Jan Schmidt
b333e32e18 rpicamsrc: webrtc example: Modify HTML to support other ports than 57778 2020-07-10 16:46:28 +01:00
Jan Schmidt
d9115ef1eb rpicamsrc: webrtc example: Remove external fmtp insertion
GStreamer 1.14.2 should contain the backport of gst-plugins-bad
commit 5c450c5 adding FEC and RTX support, and incidentally
the fmtp field in the SDP
2020-07-10 16:46:26 +01:00
Jan Schmidt
fa840da606 rpicamsrc: webrtc example: Set the locale
Make the date format in the overlay respect the current
locale
2020-07-10 16:46:24 +01:00
Jan Schmidt
39a026760d rpicamsrc: Add webrtc streaming example
Add an example for testing webrtc streaming from the rpi
camera, based on the code from
https://bugzilla.gnome.org/show_bug.cgi?id=795404

Requires GStreamer 1.14.1 or git master
2020-07-10 16:46:21 +01:00
Philippe Normand
cda483cb3c rpicamsrc: Basic orientation interface support
The (h,v)flip attributes are now supported through this interface.
It should also be possible to support (h,v)center attributes using the
ROI properties.
2020-07-10 16:45:13 +01:00
Philippe Normand
c51503fc41 rpicamsrc: add test-color-balance example
This small test will display a live video preview of the rpicam with
the balance controls being updated once a second. The controls to
update can be disabled in the source by setting the CONTROL_* macros
values to 0.
2020-07-10 16:45:02 +01:00
Jan Schmidt
acc7449d28 rpicamsrc: Add dynamic properties example
Python example of adjusting saturation on the fly
2020-07-10 16:44:41 +01:00
Sebastian Dröge
3ad86bdf30 imagefreeze: Add test for new live mode
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>
2020-06-29 12:07:14 +03:00
Nirbheek Chauhan
0fcd87e42a meson: Build Qt5 tests with -std=c++11
We already do this for the plugin.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/780#note_548179

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/642>
2020-06-25 15:20:55 +00:00
Havard Graff
cdba5952ed rtpsession: make tests more stable
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/410>
2020-06-20 19:45:33 +00:00
Tim-Philipp Müller
87d4374655 examples: qmlsink: rename qrc file to avoid naming conflicts with older meson versions
Would get "Tried to create target "qt5-qmlsink_qrc", but a
target of that name already exists." with older meson versions.

Work around that by renaming the qrc file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/633>
2020-06-18 10:58:32 +01:00
Tim-Philipp Müller
d654c6feae tests: don't pull in all -bad plugin, only allow the one we need
Set up our plugin include list for tests in such a way that
we don't pull in *all* plugins from -bad but only the one
used in the splitmuxsink unit test, i.e. the timecode plugin,
so we don't accidentally use other encoders/decoders such as
nvenc/dec for example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/617>
2020-06-09 15:23:40 +01:00
Guillaume Desmottes
0594d2f981 tests: vp9enc: enforce I420 format
Test was not enforcing a video format on videotestsrc. I420 was picked
as it was the first format in GST_VIDEO_FORMATS_ALL which will no longer
be true (gst-plugins-base!689).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/615>
2020-06-08 17:58:29 +02:00
Mikhail Fludkov
7b390a8bbd vpxenc: Add new bit-per-pixel property to select a better "default" bitrate
As part of this also change the default bitrate value to 0. The default
value was 256000 previously. In reality, if the property was not set the
bitrate value would be scaled according to the resolution which is not
very intuitive behavior. It is better to use 0 for this purpose. Now
together with newly introduced property "bits-per-pixel" 0 means to
assign the bitrate according to resolution/framerate.

The default bitrates are now
 - 1.2Mbps for VP8 720p@30fps
 - 0.8Mbps for VP9 720p@30fps
and scaled accordingly for different resolutions/framerates.

Previously the default bitrate was also not scaled according to the
framerate but only took the resolution into account.

This also fixes the side effect of setting bitrate to 0. Previously
encoder would not produce any data at all.

Addition from Sebastian Dröge <sebastian@centricular.com> to assume
30fps if no framerate is given in the caps instead of not calculating
any bitrate at all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/611>
2020-06-04 20:14:03 +00:00
Stian Selnes
44e4de43da vpxdec: Check that output width and height != 0
For VP8 it's possible to signal width or height to be 0, but it does
not make sense to do so. For VP9 it's impossible. Hence, we most
likely have a corrupt stream. Trying to negotiate caps downstream with
either width or height as 0 will fail with something like

gst_video_decoder_negotiate_default: assertion 'GST_VIDEO_INFO_WIDTH (&state->info) != 0' failed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/610>
2020-06-02 23:59:20 +03:00