Commit graph

115178 commits

Author SHA1 Message Date
Nirbheek Chauhan 90b742651d meson: Update flex, bison, and nasm
Latest flex is 2.6.4, bison is 3.8.2, nasm is 2.15.04

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3209>
2022-10-18 02:21:07 +05:30
Sam Van Den Berge 07d8e53aac examples/webrtc/signalling: Fix compatibility with Python 3.10
Fix asyncio throwing a deprecation warning when using
asyncio.get_event_loop().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3196>
2022-10-17 11:46:51 +02:00
Arun Raghavan 34ca46c786 rtmp2sink: Correctly return GST_FLOW_ERROR on error
If there is an error while connecting, the streaming task will be stopped, and
is_running() will be false, causing a GST_FLOW_FLUSHING to be returned. Instead,
we perform the error check (!self->connection) first, to return an error if
that's what occured.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3189>
2022-10-15 08:21:40 +00:00
Edward Hervey ece84d69a2 gst-play: Don't leak the stream collection
We are given a reference to the collection when parsing it from the
message. Just store it (instead of referencing it again).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3186>
2022-10-15 03:04:06 +00:00
Mathieu Duponchelle b10e0efd3a webrtc/nice: fix small leak of split strings
g_strfreev previously stopped at our manual NULL-termination. Fix by
restoring the pointer after joining.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3188>
2022-10-14 20:15:00 +00:00
Nirbheek Chauhan 48e097c315 gst-docs: Fix typo in hotdoc kwarg
The hotdoc module passes unknown keyword arguments as arguments to
hotdoc, and the fatal warnings argument is --fatal-warnings.

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972#note_1586361

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3178>
2022-10-14 16:04:00 +00:00
Devin Anderson 31b244271e wavparse: Avoid occasional crash due to referencing freed buffer.
We've seen occasional crashes in the `wavparse` module associated with
referencing a buffer in `gst_wavparse_chain` that's already been freed.  The
reference is stolen when the buffer is transferred to the adapter with
`gst_adapter_push` and, IIUC, assuming the source doesn't hold a reference to
the buffer, the buffer could be freed during interaction with the adapter in
`gst_wavparse_stream_headers`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3179>
2022-10-14 07:54:03 +00:00
Edward Hervey ae8a5e110c rtsp-client: Remove duplicate documentation
Confuses the documentation builder, since it's documented twice it complains
about a missing "Since:" marker whereas it's present in the documentation
comment further down

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3180>
2022-10-14 08:54:17 +02:00
Julian Bouzas 9197235539 riff: Mark jpeg as parsed
This is needed so that autoplugging works with avidemux and JPEG decoders that
need parsed sink caps (eg rockchip 'mppjpegdec' decoder). It also works fine
with 'jpegdec' decoder regardless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3175>
2022-10-13 13:53:29 -04:00
Devin Anderson 4e03c5f885 wavparse: Fix crash that occurs in push mode when header chunks are corrupted
in certain ways.

In the case that a test is provided for, the size of the `fmt ` chunk is
changed from 16 bytes to 18 bytes (bytes 17 - 20 below):
```
$ hexdump -C corruptheadertestsrc.wav
00000000  52 49 46 46 e4 fd 00 00  57 41 56 45 66 6d 74 20  |RIFF....WAVEfmt |
00000010  12 00 00 00 01 00 01 00  80 3e 00 00 00 7d 00 00  |.........>...}..|
00000020  02 00 10 00 64 61 74 61                           |....data|
00000028
```

(Note that the original file is much larger.  This was the smallest sub-file
I could find that would generate the crash.)

Note that, while the same issue doesn't cause a crash in pull mode, there's a
different issue in that the file is processed successfully as if it was a .wav
file with zero samples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3173>
2022-10-13 08:56:49 +00:00
He Junyan 11436be268 vp9parse: The show_existing_frame buffer should not be decode only.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>
2022-10-13 06:41:06 +00:00
He Junyan eac9c33cc1 vp9parse: Correct the pts for frames inside a super frame.
When the alignment is "FRAME" and the parse is likely connecting to
a decoder, the current PTS setting for VP9 frames inside a super
frame is not very correct.

For example, the super frame may begin with non-displayed frames and
end with a displayed frame. The current way will assign the PTS to
the first non-displayed frame, which is a decode-only frame and the
PTS will be discarded in the video decoder. While the last displayed
frame has invalid PTS, and so the video decoder needs to guess its
PTS based on the frame rate and previous frame's PTS. This is not a
decent and robust way. And more important, when the previous frames
provide DTS, the video decoder will also guess the PTS based on the
previous frames' DTS and trigger the warning like:

  gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
  <vavp9dec0> decreasing timestame

It sets the reordered_output and makes the decoder in free run mode.

We should correct the PTS for a super frame, let the non-displayed
frames have no PTS while set the correct PTS to the displayed one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>
2022-10-13 06:41:06 +00:00
Piotr Brzeziński 1e70525bc9 avfvideosrc: Allow specifying crop coordinates during screen capture
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3153>
2022-10-12 21:19:31 +00:00
Linus Svensson f5451f7ff2 rtsp-server: Free client if no connection could be created
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3164>
2022-10-12 11:09:41 +00:00
Edward Hervey 3215136d67 mxfdemux: Add support for Canon XF-HEVC
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1495

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3163>
2022-10-12 09:15:57 +00:00
Edward Hervey 9ca306d22c mxfdemux: Don't leak index table segments on failures
The segment was freed ... but not the contents :)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3163>
2022-10-12 09:15:57 +00:00
Peter Stensson 11982bcaba rtsp-server: Add since marker for adjust_error_code
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3157>
2022-10-12 08:08:27 +00:00
Xavier Claessens 9ebc3f2316 meson: Update libsoup.wrap
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3162>
2022-10-11 14:53:40 -04:00
Thibault Saunier e99393520e videorate: Do not close segment when getting a same segment twice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>
2022-10-11 11:48:09 -03:00
Thibault Saunier 11b83fb2fc videorate: Handle closing segment on EOS right after caps event
The scenario is what we try in the tests:
- we have a segment with .stop set
- some frame(s) flow
- we get a CAPS event
- we get an EOS (before getting buffers after the CAPS event)

in that case, without that patch, the segment is not properly closed
which is not correct. In this patch we keep track of previous caps until
a new buffer arrives, this way in that situation we set previous caps
again, and close the segment with the previous buffer.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1352
in this specific case

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>
2022-10-11 11:48:09 -03:00
Thibault Saunier 55dd7ff4b4 validate: Plug some leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>
2022-10-11 11:48:09 -03:00
Edward Hervey c37182b6b9 oss4: Fix debug category initialization
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1456

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3158>
2022-10-11 13:02:54 +00:00
Sebastian Dröge 430ec0d860 webrtc: Move GST_WEBRTC_ERROR_TYPE_ERROR at the end of the enum to keep ABI compatibility
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3154>
2022-10-11 11:24:19 +00:00
Sangchul Lee 0f4cf19fb9 tests/webrtc: Add test for 'add-turn-server' action signal
It just checks return value of the action signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3131>
2022-10-11 10:23:00 +00:00
Johan Sternerup 44eea7bd8a sctpenc: Prohibit sending of interleaved message parts
Apparently we cannot start sending messages from another datachannel
before the previous message was completely sent. usrsctplib will
complain about being locked on another stream id and set
errno=EINVAL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2454>
2022-10-11 09:36:13 +00:00
Nicolas Dufresne 82f63b0d64 opengl: Fix usage of eglCreate/DestroyImage
The implementation was inconsistent between create and destroy. EGLImage
creation and destruction is requires for EGL 1.5 and up, while
otherwise the KHR version is only available if EGL_KHR_image_base
feature is set. Not doing these check may lead to getting a function
pointer to a stub, which is notably the case when using apitrace.

Fixes #1389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2925>
2022-10-11 08:23:45 +00:00
Peter Stensson ec605e7b52 rtsp-server: Add support for adjusting request response on pipeline errors
The idea is to give the application the possibility to adjust the error
code when responding to a request. For that purpose the pipeline's bus
messages are emitted to subscribers through a signal handle-message.
The subscribers can then check those messages for errors and adjust
the response error code by overriding the virtual method
adjust_error_code().

Fixes #1294

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
2022-10-11 07:42:28 +02:00
Mathieu Duponchelle cddb0e951f splitmuxsrc: don't queue data on unlinked pads
Once a pad has returned NOT_LINKED, the part reader shouldn't let its
corresponding data queue run full and eventually (after 20 seconds)
stall playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3145>
2022-10-10 18:11:12 +00:00
Sebastian Dröge bd5a4d321b rtpsource: Don't do probation for RTX sources
Disable probation for RTX sources as packets will arrive very
irregularly and waiting for a second packet usually exceeds the deadline
of the retransmission.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:18 +00:00
Sebastian Dröge 5568cb33f7 rtp: examples: client-rtpaux: Provide correct caps by payload type and RTX pt map by session
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:18 +00:00
George Kiagiadakis 8dd512fd9f tests/check/rtpsession: extend test_internal_sources_timeout
to verify that rtx SSRCs do not BYE after timeout

See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:18 +00:00
Sebastian Dröge 72b6dabd32 rtpsession: Remember the corresponding media SSRC for RTX sources
This allows timing out the RTX source and sending BYE for it when the
actual media source belonging to it is timed out.

This change only applies to sending sources from this session.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:17 +00:00
Sebastian Dröge d5c072fadd rtpsource: Rename rtp_source_update_caps to rtp_source_update_send_caps
To make it clear that this is only used for sending RTP sources.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:17 +00:00
Sebastian Dröge 97a47341a7 rtpsession: Rename gst_rtp_session_sink_setcaps to gst_rtp_session_setcaps_recv_rtp
to make it clearer that this is for setting receiver caps and to make it
more consistent with gst_rtp_session_setcaps_send_rtp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:17 +00:00
Aleksandr Slobodeniuk 89fc20931b decodebin3: allow to call "dispose" multiple times
https://docs.gtk.org/gobject/concepts.html#reference-counts-and-cycles

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3135>
2022-10-10 10:24:52 +00:00
Wojciech Kapsa b618ff3369 decklink: reset calculation of gst_decklink_video_src_update_time_mapping on no_signal. When the HDMI cable was disconnected for a long time, the calculation took too much time. SDI cable works fine.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3132>
2022-10-10 08:13:30 +00:00
Nirbheek Chauhan 2d838a9b3d ci: Fix website regen on push
Don't make the integrate stage manual, we need it to regen the website

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3134>
2022-10-10 07:02:58 +00:00
Guillaume Desmottes 2df2dfce55 aggregator: fix input buffering
We need to be able to buffer at least the aggregator latency +
upstream latency, which is the value used to compute the aggregator
deadline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3138>
2022-10-07 16:27:51 +02:00
Xavier Claessens 56eb44c502 Meson: Fix libxml2 fallback
The variable xml2lib_dep does not exist. The correct name is already in
the wrap file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3136>
2022-10-07 07:56:21 -04:00
Sebastian Dröge bacd92274d rtspsrc: Retry SETUP with non-compliant URL resolution on "Bad Request" and "Not found"
Various RTSP servers/cameras assume base and control URL to be simply
appended instead of being resolved according to the relative URL
resolution algorithm as mandated by the RTSP specification.

To work around this, try using such a non-compliant control URL if the
server didn't like the URL used in the first SETUP request.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1447
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/922

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3127>
2022-10-07 09:12:00 +00:00
Sangchul Lee 93b896eb4e webrtcbin: Fix pointer dereference before null check
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3129>
2022-10-06 16:46:33 +00:00
Wojciech Kapsa 505f48f237 decklink: Add new persistent-id property and sort devices by persistent ID
The order of the devices iterator from the SDK is undefined and can
randomly change.

Keep the device-number property for backwards compatibility and
simplicity but prefer the persistent-id property and also use it for the
device provider implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3078>
2022-10-06 15:50:11 +00:00
Aleksandr Slobodeniuk 67caa45a4c decodebin3: fix mutex leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3126>
2022-10-06 14:25:27 +00:00
Thibault Saunier bf964f896f transcodebin: Implement support for upstream stream selection
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3128>
2022-10-06 09:14:41 -03:00
Thibault Saunier 31acfcd875 decodebin3: Do not try to plug a decoder on raw formats
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3123>
2022-10-06 08:41:49 +00:00
Seungha Yang c9f12b5086 videosink: Don't return unknown end-time from get_times()
... in case of reverse playback. Otherwise basesink will not
wait for clock

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3121>
2022-10-06 06:36:42 +00:00
Mathieu Duponchelle 235be306fd avauddec: address regression with WMA files ..
By outputting lead-in samples that FFmpeg now would like us to ignore,
and discarding trailing samples that it would now like us to output.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1474

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1348
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3117>
2022-10-06 01:03:10 +00:00
Thibault Saunier 9abceda343 validate:launcher: Cleanup test uuid when copying it
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
2022-10-05 20:29:22 +00:00
Thibault Saunier f3c162cc85 validate: launcher: Add a argument to avoid rereuning flaky tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
2022-10-05 20:29:22 +00:00
Thibault Saunier 1577911d75 validate: launcher: Keep variable framerate from input when possible
But disable it if forcing a framerate for some reason

Fixing our support for variable framerate in the encoding profile
serialization format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
2022-10-05 20:29:22 +00:00