Commit graph

115187 commits

Author SHA1 Message Date
Sebastian Dröge
75731aec53 app: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
2022-10-18 08:56:58 +00:00
Sebastian Dröge
e0b06df223 audio: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
2022-10-18 08:56:58 +00:00
Sebastian Dröge
2e5c73fff7 video: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
2022-10-18 08:56:58 +00:00
Sebastian Dröge
64c376b5b2 webrtc: Add/fix various annotations
And mark string parameters as const.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
2022-10-18 08:56:58 +00:00
Edward Hervey
0c47735c4a urisourcebin: Fix usage of raw and non-raw source provider
The computation in analyze_source was wrong, and would state that the element
has "all raw source pads" if it had at least one.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1029

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3199>
2022-10-18 07:46:06 +00:00
Sebastian Dröge
4df3da3bab rtpbuffer: Initialize extended timestamp to the first wraparound period
This allows correct handling of wrapping around backwards during the
first wraparound period and avoids the infamous "Cannot unwrap, any
wrapping took place yet" error message.

It allows makes sure that for actual timestamp jumps a valid value is
returned instead of 0, which then allows the caller to handle it
properly. Not having this can have the caller see the same timestamp (0)
for a very long time, which for example can cause rtpjitterbuffer to
output the same timestamp for a very long time.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1500

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3202>
2022-10-18 06:09:08 +00:00
Matthew Waters
d586c2cc28 examples/webrtc: don't use factory_make_full() for enums
They are not currently translated into their respective enum values and
will produce an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3210>
2022-10-18 01:30:37 +00:00
Seungha Yang
cb7958e710 wasapi2: Add support for process loopback capture
Adding loopback capture mode for specified PID.

Note that this feature requires Windows 10 build 20348
(Windows 11/Windows Server 2022 or later),
and any process loopback related properties will not be exposed
if OS does not support it.

Example launch lines:
* wasapi2src loopback-mode=include-process-tree loopback-target-pid=<PID>
 Captures audio generated by an application (specified by PID)
 and its child process
* wasapi2src loopback-mode=exclude-process-tree loopback-target-pid=<PID>
 Captures desktop audio excluding PID and its child process

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1278
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3195>
2022-10-17 23:28:48 +00:00
Nirbheek Chauhan
2c3f9d4587 ci: Run windows jobs when win-* binary subprojects are updated
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3209>
2022-10-18 02:48:54 +05:30
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