Commit graph

18139 commits

Author SHA1 Message Date
Olivier Crête
9b0a373eac rtpstorage: Add debug funcptr to chain function 2019-03-26 18:08:57 -04:00
Julian Bouzas
2ebdd70c21 flac: report latency in flacenc and flacdec
The FLAC specification states that the data is processed in blocks, regardless of the number of channels. Thus, The latency can be calculated using the blocksize and rate. For example a 1 second block sampled at 44.1KHz has a blocksize of 44100
2019-03-25 15:14:32 +01:00
Tim-Philipp Müller
d682c74c1e examples: rtsp: fix compiler warning
"control reaches end of non-void function"
2019-03-22 23:37:09 +00:00
Nicolas Dufresne
79fd0af152 gstrtpsession: Remove set but not use running-time 2019-03-22 20:01:52 +00:00
Olivier Crête
7ecbd7271d rtpmanager: Register chain functions to debug 2019-03-22 16:44:41 +00:00
Nicolas Dufresne
2ff7519d73 rtpbin: Allow reusing the sender AUX bin
This is needed for the case you don't know in advance all the sessions
you will be using, but would like to place all the related AUX element
in the same GstBin. As per current implementation, each time an sender
AUX bin is requested and returned, RTPBin will walk the src pads and
create sessions for these pads.

In the current implementation, if a src pad already have a sessions, it
returns an error and stops. As a side effect, if an AUX bin is reused in
a following AUX bin request, it can only work if the pads are created on
the last request.

This change simply relax the restriction in order to keep walking, and
just ensure that all newly created pads have a sessions.
2019-03-21 21:10:43 +00:00
Philipp Zabel
16e5b32bc1 v4l2videoenc: set GstVideoCodecFrame sync point flag
The V4L2 elements already set the delta unit buffer flag when dequeueing
the buffer, but gst_video_encoder_finish_frame overwrites it from the
passed codec frame's sync point flag. Set the flag correctly.
2019-03-21 18:05:51 +00:00
George Kiagiadakis
d5ce10240a gstrtpsession: improve stats about rtx requests 2019-03-21 13:40:31 -04:00
George Kiagiadakis
db647ee55b rtprtxsend: Improve looging of not found RTX packet
When an RTX packet is not found, display a message that say if the
packet have not arrived yet or if it was already removed from the RTX
packet queue.
2019-03-21 13:19:52 -04:00
Nicolas Dufresne
0aff8a7d30 rtpsession: Remove unused rtp_session_create_source 2019-03-21 13:19:52 -04:00
Tim-Philipp Müller
e8583cebe7 meson: add -Wno-unused also to C++ args when gst debug system is disabled
And check if argument is supported instead of just passing it blindly,
and make meson code slightly cleaner, centralising the argument setting
in one place.
2019-03-21 16:45:03 +00:00
Piotr Drąg
a0474578a0 Update LINGUAS 2019-03-21 15:59:31 +00:00
Seungha Yang
63bb1e3a4d qtdemux: Don't pass zero to denominator for framerate
Need to respect return of gst_video_guess_framerate() to ensure
non-zero denominator.

This patch is to fix below error with an abnormal (but has valid frame) file.
(gst-play-1.0:17940): GStreamer-CRITICAL **: passed '0' as denominator for `GstFraction'
2019-03-19 12:35:08 +09:00
Philippe Normand
75f26bc954 v4l2: Set Hardware classifier on encoders 2019-03-18 10:51:15 +00:00
Philippe Normand
3296a4b7e2 v4l2: Set Hardware classifier on video decoders 2019-03-18 10:51:07 +00:00
Philipp Zabel
cdf15e9032 v4l2transform: don't segfault if flushed without pools
The v4l2output and v4l2capture v4l2objects can have pool == NULL if they
have been stopped before.
2019-03-17 13:17:21 +00:00
Charlie Turner
39d32b2394 qtdemux: Find mp4a esds atoms in protected streams sample description tables.
This problem was found in Test. 2 of the YouTube 2018 EME
tests[1]. The code was accidentally not finding an mp4a's esds atom in
the sample description table when the stream was encrypted. It assumed
that if the stream is protected, then only an enca atom will be found
here. What happens with YouTube is they often provide protected
content with a few seconds of clear content, and then switch to the
encrypted stream.

The failure case here was an incorrect codec_data field being sent
into aacparse. The advertisement of stereo audio @ 44.1kHz for the
mp4a (unprotected) stream was incorrect. As usual, the esds contained
the real values here which were mono at 22050 Hz.

Here's what the MP4 tree looks like for these types of files,
demonstrating why the code was making a wrong assumption (or maybe
YouTube is being unusual),

[ftyp] size=8+16
...
[moov] size=8+1571
...
  [trak] size=8+559
...
          [stsd] size=12+234
            entry-count = 2
            [enca] size=8+147
              channel_count = 2
              sample_size = 16
              sample_rate = 44100
              [esds] size=12+27
                ...
            ...
            [mp4a] size=8+67
              channel_count = 2
              sample_size = 16
              sample_rate = 44100
              [esds] size=12+27
                ...

In addition to fixing this, the checks for esds atoms in mp4a and mp4v
have been made symmetrical. While I haven't seen a test case for video
with the same problem, it seemed better to make the same checks. This
also fixes a crash reported from another user[2], they also noted the
asymmetry with mp4v and mp4a.

[1] https://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2018.html?test_type=encryptedmedia-test
[2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/398
2019-03-15 12:41:33 +00:00
Andreas Frisch
3160713abf flvmux: Fix scale of time values in warning message 2019-03-15 09:55:32 +00:00
Sebastian Dröge
a676c17259 rtspsrc: Don't remove udpsrc/sink from rtspsrc if they were not added to it
This can happen in various error cases that could happen between the
creation of the element in question and the adding to the rtspsrc.

It causes an ugly critical warning right now but is otherwise harmless.
2019-03-15 08:21:11 +00:00
Antonio Ospite
2513edf229 test: imagefreeze: add test for the num-buffers property 2019-03-14 09:12:28 +01:00
Antonio Ospite
8c26e33f20 imagefreeze: add a num-buffers property
The imagefreeze element can be handy for benchmarking downstream
elements because it re-uses the same buffer memory and introduces less
overhead compared to always creating new frames with videotestsrc.

However it's not possible to make imagefreeze send EOS when using
gst-launch-1.0.

Add a num-buffers property to make it look more like a source in the
above scenario.
2019-03-14 09:12:28 +01:00
Guillaume Desmottes
fcd568dd56 matroskamux: add support for new color primaries 2019-03-12 16:52:45 +01:00
Philipp Zabel
8b068fb78b v4l2sink: fix pool-less allocation query handling
This fixes a critical warning if the last-sample property is enabled:

  (gst-launch-1.0:391): GStreamer-CRITICAL **: 01:12:57.428: gst_object_unref: assertion 'object != NULL' failed

If the allocation query does not contain any allocation pools,
gst_query_parse_nth_allocation_pool will leave the local pool,
min, and max variables undefined, so check the array length first.
If pool is NULL, do not call gst_object_unref.
2019-03-08 22:01:14 +01:00
Seungha Yang
3f9170bd02 meson: Build v4l2 example only if v4l2 plugin was built
Otherwise v4l2 example will be built with MSVC
2019-03-08 11:06:32 +09:00
Antonio Ospite
2dfe228740 docs: fix typos s/recieve/receive/ 2019-03-07 12:41:40 +01:00
Antonio Ospite
30db93e3a4 rtpsource: fix documentation of rtp_source_send_rtp parameters
In commit 28e5f9098 (rtpbin: use PacketInfo for the sender, 2013-09-13)
the rtp_source_send_rtp signature changed but the documentation was not
adjusted to match the new one.

Update the documentation to match the function signature.
2019-03-07 12:41:40 +01:00
Antonio Ospite
38285e5bcf rtpsession: fix typo in a comment, s/SESSION_LOCK/RTP_SESSION_LOCK/
Fix a typo in a comment, mainly to avoid confusing autocompletion in
text editors.
2019-03-07 12:41:40 +01:00
Antonio Ospite
43e4226844 rtpsession: fix typos and update parameters names in comments
Some functions now accept a generic 'gpointer data' parameter because
they can work either on a single buffer or a buffer list.

However the comments were still referring to the old 'GstBuffer *buffer'
parameter, so update the comments to match the actual functions
signature.
2019-03-07 12:41:40 +01:00
Antonio Ospite
b2b60c4d8f rtpstats: fix some fields names in the RTPSourceStats documentation
Fix documentation of RTPSourceStats to use the actual fields names.
2019-03-07 10:36:11 +01:00
Mathieu Duponchelle
0da8f111e6 rtpulpfdecdec: only put recovered packet back into storage if not recovered from there 2019-03-06 19:40:10 +00:00
Mathieu Duponchelle
f9b49aef09 rtpulpfecdec: fix buffer leak when packet is recovered from storage
Exposed by rtpulpfecdec_recovered_from_storage test.
2019-03-06 19:40:10 +00:00
Tim-Philipp Müller
a2d01b3a8b tests: rtpulpfec: fix buffer leak in unit test
This freed wrapped memory instead of the GstMemory or buffer.
2019-03-06 19:40:10 +00:00
Tim-Philipp Müller
c79cf179cc rtph264depay: fix caps leak
Exposed by rtp_h264depay_bytestream() unit test.
2019-03-06 18:21:20 +00:00
Tim-Philipp Müller
081da67444 tests: rtpjitterbuffer: fix leaks in new test_push_eos() test 2019-03-06 17:28:57 +00:00
Tim-Philipp Müller
55d43dbbde tests: states: blacklist gtk sinks for state change test
gtk_init() throws GLib-GIO-WARNING **: unknown schema extension 'd'
unrelated to our test environment.
2019-03-06 17:27:32 +00:00
Tim-Philipp Müller
6b68b73341 tests: .gitignore more test and example binaries 2019-03-06 17:26:03 +00:00
Matthew Waters
7f95a809e9 gtkgl: Also try retrieving an EGL context from Gdk with X11
Some embedded platforms will use EGL instead of GLX within the X11
ecosystem.
2019-03-05 15:26:45 +11:00
Tim-Philipp Müller
4f3dda36b4 Back to development 2019-03-04 09:07:30 +00:00
Tim-Philipp Müller
899d0c4b3b matroskademux: fix AV1 caps when there's no codec_data
There is no "byte-stream" format for AV1 in Matroska, this
was probably cargo-culted from H.264. codec_data / CodecPrivate
is now mandatory for AV1 in Matroska[*], but there are sample
files out there which don't have it (e.g. some Elecard ones).

[*] https://github.com/Matroska-Org/matroska-specification/blob/master/codec/av1.md#codecprivate-1
2019-03-01 17:37:55 +00:00
Tim-Philipp Müller
83b45abe74 meson: don't build icles when tests are disabled
They are manual tests, so let them be controlled
via the tests option.
2019-02-28 08:52:28 +00:00
Marc Leeman
8737e29a49 rtpsource: small spell correct 2019-02-27 16:14:22 +01:00
Tim-Philipp Müller
48b46e3f14 Release 1.15.2 2019-02-26 21:09:04 +00:00
Tim-Philipp Müller
1ef240e24d Update docs 2019-02-26 21:09:02 +00:00
Tim-Philipp Müller
6b212368cf Update translations 2019-02-26 19:25:59 +00:00
Mauro Carvalho Chehab
dc7bd48326 v4l2: accept Bayer as possible input/output for V4L2 codecs
A V4L2 transform codec may input/output data on Bayer format.

Add support for that.
2019-02-26 13:59:46 +00:00
Mauro Carvalho Chehab
55c1274dba v4l2: fix a typo on a debug message at v4l2_calls
suppored -> supported
2019-02-26 13:59:46 +00:00
Matthew Waters
0acbf40060 v4l2dec: also remove the colorimetry and chroma-site fields
If a different format is chosen, then these values are incorrect.
2019-02-26 07:04:54 +00:00
Nicolas Dufresne
e72ef633a6 rtpsession: Fix EOS forwarding
So far we assumed that if all sources are bye, this meant we needed to
send an EOS on the RTCP sink. The problem is that this case may happens
if we only had one internal source and it detected a collision.

So now we limit the EOS forwarding to when there is a send_rtp_sink pad
and that this pad has received EOS. We don'tcheck the recv_rtp_sink
since the code does not wait for the bye to be send before sending EOS
to the RTCP src pad.
2019-02-25 17:06:50 +00:00
Jan Schmidt
098f936be8 wavparse: Declare support for RF64
RF64 encode support was added to wavenc quite some time
ago, but not declared in wavparse. It seems wavparse can
decode it though, so add it to the sink pad.

The RF64 support was added in
https://bugzilla.gnome.org/show_bug.cgi?id=735627
2019-02-24 14:29:27 +00:00
Nicolas Dufresne
06c340edd4 rtp: Add property to disable RTCP reports per internal rtpsource
This is useful when implementing custom retransmission mechanism like
RIST to prevent RTCP from being produces for the retransmitted SSRC.
This would also be used in general for various purpose when customizing
an RTP base pipeline.
2019-02-13 15:07:39 -05:00