Commit graph

114097 commits

Author SHA1 Message Date
Jan Alexander Steffens (heftig)
a4121afebb rtmp2/connection: Pass triggering GError in 'error' signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>
2022-11-15 02:38:32 +00:00
Jan Alexander Steffens (heftig)
7a80611c3a rtmp2/connection: Pass triggering GError to _emit_error
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>
2022-11-15 02:38:32 +00:00
Jan Alexander Steffens (heftig)
90f39a4c7e rtmp2/connection: Discern reasons for cancelling all commands
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>
2022-11-15 02:38:32 +00:00
Jan Alexander Steffens (heftig)
7b9fbf9d4f rtmp2/connection: Handle EOF like error
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>
2022-11-15 02:38:32 +00:00
Jan Alexander Steffens (heftig)
aea80f0529 rtmp2/client: Make sure 'salt' is not NULL
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>
2022-11-15 02:38:32 +00:00
Jan Alexander Steffens (heftig)
c802180d6b rtmp2/client: Make sure 'reason' is not NULL
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>
2022-11-15 02:38:32 +00:00
Jan Alexander Steffens (heftig)
1bc0e9e1cb rtmp2/client: Make sure 'desc' is not NULL
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>
2022-11-15 02:38:32 +00:00
Jan Alexander Steffens (heftig)
e2fa6916a9 rtmp2/client: Make sure 'code' is not NULL
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>
2022-11-15 02:38:32 +00:00
Justin Chadwell
7954f0539f qtdemux: use unsigned int types to store result of QT_UINT32
In a few cases throughout qtdemux, the results of QT_UINT32 were being
stored in a signed integer, which could cause subtle bugs in the case of
an integer overflow, even allowing the the result to equal a negative
number!

This patch prevents this by simply storing the results of this function
call properly in an unsigned integer type. Additionally, we fix up the
length checking with stsd parsing to prevent cases of child atoms
exceeding their parent atom sizes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3367>
2022-11-09 10:39:51 +00:00
Sebastian Dröge
30d894866d allocator: Switch allow-none annotations to nullable / optional
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3369>
2022-11-09 01:47:51 +00:00
Sebastian Dröge
ef8474aad5 allocator: Copy allocator name in gst_allocator_register()
The parameter is not marked as `transfer full` and stays around in the
hash table, so we will have to copy it ourselves.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3369>
2022-11-09 01:47:51 +00:00
Edward Hervey
3cc45d5a36 subparse: Fix non-closed tag handling.
Unclear what the goal was, but we could end up reading way past the next_tag.

Instead just move everything from after the end tag ('>') to the next_tag.

Fixes https://bugs.chromium.org/p/oss-fuzz/issues/detail?id=53040

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3351>
2022-11-07 12:45:29 +00:00
Seungha Yang
b6309a569a d3d11videosink: Always clear back buffer on resize
Swapchain may not need to be resized if the size of backbuffer
is equal to the previous size. Then previously rendered frame will be stay
on the screen. Do clear back buffer whenever resize() is called

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3342>
2022-11-06 03:26:31 +09:00
Tim-Philipp Müller
c2f58cf2e3 qt: initialize GError properly in gst_qt_get_gl_wrapcontext()
Spotted by Claus Stovgaard.

Fixes #1545

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3339>
2022-11-05 01:14:37 +00:00
Sebastian Dröge
4dca76396e qtmux: Add durations to raw audio buffers from the raw audio adapter in prefill mode
This ensures that a duration can also be calculated and stored for the
last buffer at EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3338>
2022-11-05 00:26:26 +00:00
Sebastian Dröge
afa15e6284 qtmux: Release object lock before posting an error message
GST_ELEMENT_ERROR() also takes the object lock and this would then
deadlock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3338>
2022-11-05 00:26:25 +00:00
Edward Hervey
5ec482bdcf gstpad: Fix non-serialized sticky event push
With non-serialized sticky events, such as GST_EVENT_INSTANT_RATE, we both want
to store the event (for later re-linking) *AND* push the event in a non-blocking
way.

We therefore must *not* propagate pending sticky events if the event is "sticky
or serialized" but only if it's "serialized"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3332>
2022-11-04 19:33:12 +00:00
Jan Alexander Steffens (heftig)
39f4d5849f srt: Remove callers for which srt_bstats fails
This keeps them from accumulating in the element and in the stats while
the sink is not being fed, as long as we at least periodically grab
stats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>
2022-11-04 18:17:06 +00:00
Jan Alexander Steffens (heftig)
253a5cc125 srt: Use simpler list operations for callers
Avoid `g_list_append` and `g_list_remove` (which have to scan the list)
and replace them with `g_list_prepend` and `g_list_delete_link`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>
2022-11-04 18:17:06 +00:00
Jan Alexander Steffens (heftig)
68cd5e1de1 srt: Clean up poll/sock lifecycle
Make sure `srtobject->poll_id` is never invalid as long as `srtobject`
exists. Only remove our caller socket from it when the socket becomes
invalid.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>
2022-11-04 18:17:06 +00:00
Jan Alexander Steffens (heftig)
debb19868f srt: Clean up error handling
- Make the srt_epoll_wait loops more uniform.

- Error only via GError when possible; let the element send the error
  message. Avoids a second error message.

- Return 0 when cancelled. Avoids an error message from the element.

- Don't send an error message from send_headers when we're a server
  sink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>
2022-11-04 18:17:06 +00:00
Jan Alexander Steffens (heftig)
7425fdf2ba srt: Simplify socket stats
Don't hide stats depending on whether we're a sending or receiving
socket. While we're here, add some more debug logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>
2022-11-04 18:17:06 +00:00
Jan Alexander Steffens (heftig)
34f9788dd6 srt: Replace stats accumulation with naive byte counting
srt_bstats cannot be used to get the stats of closed connections, so the
best we can do is keep the running count ourselves.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>
2022-11-04 18:17:06 +00:00
Sebastian Dröge
4925002d87 core/base: Only post latency messages if the latency values have actually changed
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1525

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3291>
2022-10-27 23:18:37 +01:00
Edward Hervey
74ec0d4b0c videodecoder: Only post latency message if it changed
Posting latency messages causes a full and potentially expensive latency
recalculation of the pipeline. While subclasses should check whether the latency
really changed or not before calling this function, we ensure that we do not
post such messages if it didn't change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3291>
2022-10-27 23:15:31 +01:00
Nicolas Dufresne
c29bfbe448 avdec_h265: Fix endless renegoation with alternate interlacing
The picture parameter picture->top_field_first is reused in this mode
to signal the TOP fields. As a side effect, it will change every frame
and current code assumed that if this changes then a renegotiation is
needed. Fixed this by ignoring that change whenever we are decoding one field
only.

Fixes #1523

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3289>
2022-10-27 18:02:54 +01:00
Edward Hervey
5d22503a55 concat: Properly propagate EOS seqnum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3288>
2022-10-27 16:38:02 +01:00
Nicolas Dufresne
7071c4ec2f avviddec: Avoid flushing on framerate changes
A framerate change does not require flushing the decoder and causes
issues with some specific fragmented files if the two fragments have
different framerate.

Fixes #1522

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3280>
2022-10-27 01:32:05 +01:00
Nicolas Dufresne
ea1dc750ae alphacombine: Add missing query handler for gaps
The gap handling was in place, but there was no event handler to trigger it.
Implement the alpha sink event handler for the gaps. This fixes handling of
valid streams which may not refresh the alpha frames for every video frames.
It will also allow a clean error if the stream was missing the initial
alpha frame, at least until we find a better way to handle these
invalid frames.

Related to #1518

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3279>
2022-10-26 23:42:43 +01:00
Ignacio Casal Quinteiro
049737966d avfdeviceprovider: do not leak the properties
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3278>
2022-10-26 22:57:14 +01:00
Seungha Yang
c865a2b9c3 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/3277>
2022-10-26 22:03:16 +01:00
Patrick Griffis
fef136b3d0 build: Fix building ges with tools disabled
If you configure with `tools=disabled` then ges_launch is undefined.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3274>
2022-10-26 19:55:27 +01:00
Patrick Griffis
c560ec47fb webrtc: Fix double free in webrtc-recvonly-h264 demo
The "message" signal does not transfer ownership of the GBytes passed
to it so calling g_bytes_unref() on it is incorrect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3272>
2022-10-26 18:28:35 +01:00
Sebastian Dröge
d186e19568 rtspsrc: Only EOS on timeout if all streams are timed out/EOS
Otherwise a stream that is just temporarily inactive might time out and
then can never become active again because the EOS event was sent
already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3271>
2022-10-26 17:44:57 +01:00
Patrick Griffis
98bc763f2b webrtc: Fix critical in webrtc-recvonly-h264 example
This signal only takes 2 properties yet a third was passed.
This would cause a critical in GLib.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3269>
2022-10-26 15:42:42 +01:00
Nirbheek Chauhan
d1d365373f gl/vulkan: Fix static linking on macOS
duplicate symbol '__invoke_on_main' in:
    /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstvulkan-1.0.a(cocoa_gstvkwindow_cocoa.m.o)
    /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstgl-1.0.a(cocoa_gstglwindow_cocoa.m.o)
ld: 1 duplicate symbol for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)

Also make the same change in iOS for consistency.

Continuation of https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1132

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3263>
2022-10-25 11:06:34 +01:00
Seungha Yang
8a7ae99567 padtemplate: Fix annotations
gst_caps_replace() does not take ownership of the new caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3262>
2022-10-25 10:12:21 +01:00
Nirbheek Chauhan
ccd032c569 FFmpeg.wrap: Fix to the 4.4.2 stable release
We've had a lot of changes in the meson-4.4 branch that can and have
broken builds, so let's fix it to the stable release right before
that.

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1497

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3187>
2022-10-23 23:40:02 +00:00
Tim-Philipp Müller
a2a300a0e6 gst-plugins-base: bump core req to 1.20.4
For newly-added _gst_meta_tag_memory_reference symbol.

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2928#note_1591459

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3185>
2022-10-23 17:29:40 +00:00
Matthew Waters
a679a55ecc webrtcbin: also add rtcp-fb ccm fir for video mlines by default
In addition to the 'nack pli' already added.  Both are supported by
rtpbin/rtpsession by default already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3247>
2022-10-22 11:37:16 +01:00
Matthew Waters
af92290306 build/gl: fix automatic dispmanx detection for rpi4
rpi4 doesn't contain dispmanx but still contains bcm_host.pc which
confuses the configure detection.  Add an explicit check for
EGL_DISPMANX_WINDOW_T to ensure that we only build with dispmanx when
the types are available.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/893
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/952
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3246>
2022-10-22 10:42:17 +01:00
Bunio_FH
b7ae311c75 gstminiobject: shares capacity increase
during the MSE (WebKit) tests from Apple suite:
https://hls-streaming.cdn-apple.com/hls_conformance/dist/v1.1/index.html?pretty=true&whitelist=MSE%20Suite
webkit attempts to add a single audio buffer containing ~35.5k frames.
when corresponding GstSamples are pulled buffer is being referenced
more than object capacity allows: 2^15-1. since the case could be considered
malformed a surgical patch is applied to increase the capacity.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3233>
2022-10-20 12:42:56 +00:00
Jordan Petridis
e144ec015e cdparanoia: Ignore compiler warning coming from the cdparanoia header
When trying to build the plugin, GCC starts complaining about issues
with one of the cdparanoia headers and it block us from being able
to build the plugin with Werror.

The current warning in the header look like this:

```
[1/2] Compiling C object subprojects/gst-plugins-base/ext/cdparanoia/libgstcdparanoia.so.p/gstcdparanoiasrc.c.o
In file included from ../subprojects/gst-plugins-base/ext/cdparanoia/gstcdparanoiasrc.h:37,
                 from ../subprojects/gst-plugins-base/ext/cdparanoia/gstcdparanoiasrc.c:31:
/usr/include/cdda/cdda_interface.h:164:3: warning: initialization discards ‘const’ qualifier from pointer target type [-Wdiscarded-qualifiers]
  164 |   "Success",
      |   ^~~~~~~~~
...
/usr/include/cdda/cdda_interface.h:163:14: warning: ‘strerror_tr’ defined but not used [-Wunused-variable]
  163 | static char *strerror_tr[]={
      |              ^~~~~~~~~~~
[2/2] Linking target subprojects/gst-plugins-base/ext/cdparanoia/libgstcdparanoia.so
```

Last release of cdparanoia was in 2008, so our best bet for the
time is to ignore the warnings.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3232>
2022-10-20 11:41:10 +01:00
Sam Van Den Berge
3e4f94db14 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/3201>
2022-10-17 14:15:04 +02:00
Mathieu Duponchelle
6bb564968b 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/3190>
2022-10-15 11:01:23 +00:00
Arun Raghavan
b0fa56a333 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/3192>
2022-10-15 11:00:20 +01:00
Edward Hervey
c9006504bb 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/3191>
2022-10-15 10:15:26 +01:00
Jonas Bonn
cba7eb67d0 multiudpsink: allow binding to IPv6 address
When the sink is configured to create sockets with an explicit bind
address, then the created socket gets set to the udp_socket field
irregardless of whether the bind address indicated that the socket
family should be IPv4 or IPv6.  When binding to an IPv6 address, this
results in the following error:

gstmultiudpsink.c:1285:gst_multiudpsink_configure_client:<rtcpsink>
error: Invalid address family (got 10)

This patch adds a check of the address family being bound to and sets
the created socket to used_socket or used_socket_v6, accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3183>
2022-10-14 11:57:12 +01:00
Devin Anderson
3286e0942f 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/3181>
2022-10-14 10:40:24 +01:00
Devin Anderson
80de451c06 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/3174>
2022-10-13 12:15:03 +01:00