Commit graph

1052 commits

Author SHA1 Message Date
Jonas K Danielsson
b0becfa46b splitmuxsrc: Use natural ordering to find files
Today when using the `splitmuxsrc` on a collection of files named as:

```
item0.mkv
item1.mkv
item2.mkv
[...]
item10.mkv
item11.mkv
[...]
```

You will get a continuous stream made in the order of:

```
item0.mkv -> item1.mkv -> item10.mkv -> item11.mkv -> [...]
```

You can fix this by having smarter names of the items:

```
item000.mkv
item001.mkv
item002.mkv
[...]
item010.mkv
item011.mkv
[...]
```

Will get you:
```
item000.mkv -> item001.mkv -> item003.mkv -> item004.mkv -> [...]
```

But, we could also "fix" the former case by using natural ordering when
comparing the files in gstsplitutils.c.

Fixes #2523

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4491>
2024-01-24 20:15:19 +00:00
Dan Searles
1d02d7eda0 rtspsrc: fix ttl setting for udpsink[1]
Fix ttl setting being incorrectly applied to udpsink[0] rather
than to udpsink[1].

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5955>
2024-01-23 13:54:51 +00:00
Dan Searles
da55b953a1 rtspsrc: set multicast-iface on udpsinks
Copy rtspsrc property multicast-iface to its udpsinks to
allow messages over those sinks back to the server to work (and
prevent 'Network unreachable' warnings).

Closes: #3239
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5955>
2024-01-23 13:54:51 +00:00
Guillaume Desmottes
fae6fbaa6b flvdemux: don't re-use segment from one stream if the other has buffer earlier
Fix first audio buffers being out of segment because the audio stream
is starting earlier than the video one which was the first demuxed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5940>
2024-01-19 11:05:05 +01:00
Guillaume Desmottes
632ee523fb flvdemux: factor out ensure_new_segment()
- Use the pad instead of the element for logs, so it's clearer on which
  pad this segment will be pushed.
- One copy was checking for invalid seq num, the other was not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5940>
2024-01-19 11:05:01 +01:00
Hou Qi
2539bb0b1d rtpjitterbuffer: Fix build warning in rtp_jitter_buffer_append_query()
This is to fix build warnings when using [-Wmaybe-uninitialized]
../gst/rtpmanager/rtpjitterbuffer.c:1237:10: warning: 'head' may be used uninitialized [-Wmaybe-uninitialized]
 1237 |   return head;

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5907>
2024-01-13 15:00:19 +00:00
Philippe Normand
8a99589d2c vpxdec: Use appropriate domain and code for decoding errors
STREAM domain and DECODE error is commonly used in other decoders. ENCODE is for
encoders.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5916>
2024-01-12 14:10:36 +00:00
Olivier Crête
814f21557f soup: Avoid using GUri before GLib 2.66
Let's use gpointer for now

Fixes: #3169
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5906>
2024-01-11 18:06:59 +00:00
Sebastian Dröge
6fa41f78bb rtpsession: Remove some unused fields
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5899>
2024-01-08 12:57:04 +02:00
Sanchayan Maity
00bbac6541 rtphdrext-clientaudiolevel: Fix level value being written by the extension
When level value is greater than 127, it was being clamped but this clamped
value was not the one being actually used. For level values greater than 127
this resulted in an incorrect value being used. As an example, a level value
of 187, after and'ed with 0x7F, it would result in 0x3B being reported as the
level value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5893>
2024-01-07 16:00:18 +05:30
Tim-Philipp Müller
bf4755331a vpx: fix plugin description
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5864>
2023-12-30 11:33:52 +00:00
Sebastian Dröge
c292da7044 rtpsession: Only warn once if configured latency needs to be known but isn't yet
Otherwise we would warn about this once for every single packet until
the LATENCY event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5854>
2023-12-27 11:00:44 +00:00
Tim-Philipp Müller
d415816cb1 rtpvrawdepay: only announce supported formats in sink template
For most video formats we currently just assume that they
have a depth of 8 bits, whilst advertising that we can
handle 8/10/12/16 bit depth.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5866>
2023-12-25 19:00:18 +01:00
Sebastian Dröge
c9c26eab26 rtpvp8pay: Also set partition IDs in the packets if meta exists but without temporal_scalability
Encoders will add the meta to every single buffer, but we only cannot set
partition IDs properly when the meta has temporal_scalability set

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5814>
2023-12-21 11:26:49 +00:00
Sebastian Dröge
2e86fb691a video-format: Fix format order once again
RGBA should be before RBGA. Both the Python script and the gstreamer-rs
tests agree on that, but somehow this is not caught by the CI.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5837>
2023-12-20 05:33:43 +00:00
Chao Guo
2e75b8c8e9 v4l2object: clear old fds in poll when closing v4l2object
When reopening a v4l2 device, the v4l2object->poll will include some old fds,
which was assigned to this device before. If the pipeline opens multiple v4l2
devices, the old fd may been assigned to other v4l2 devices when reopening
devices.

This will cause the timing of the pipeline become confusing when polling devices,
leading functional abnormalities.

Therefore, when closing v4l2object, remove the old fds in poll to ensure that the
pipeline timing is normal.

Signed-off-by: Chao Guo <chao.guo@nxp.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5820>
2023-12-19 15:23:23 +00:00
Arun Raghavan
ee903a5afd rtp: Fix incorrect RTP channel order lookup by name
The g_ascii_strcasecmp() logic is inverted, since it returns 0 on equality.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5815>
2023-12-15 15:21:20 -05:00
Víctor Manuel Jáquez Leal
4f27b50c2e gtkglsink: template caps to only 2D & rectangle texture targets
Apparently external-oes is not supported by the plugin as texture target,
while DMABuf uploading prefers it because it's zero copy.

This patch enables DMABuf uploading and rendering by using either 2D or
rectangle texture targets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5795>
2023-12-11 13:17:48 +01:00
Olivier Crête
e8d7604a6a adaptivedemux2: Parse cookies in downloadhelper
We need to parse any cookie headers, otherwise we end up
sending back attributes likes "Secure" and "httponly" which break
some servers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5776>
2023-12-09 18:30:30 +00:00
Sebastian Dröge
14b94ea00b rtpvp9pay: Don't include unused dboolhuff.h header
It's only used by the VP8 payloader.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5784>
2023-12-09 11:17:15 +00:00
Xavier Claessens
b80f4a1fa4 v4l2src: Consider framerate during caps selection
This simplifies the way it picks the closest caps to preference and take into
consideration the framerate to avoid picking high resolution at 5fps or so.
Simply calculate a "distance" of caps A and B from the preference and put
closest first, sorting by framerate first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5777>
2023-12-08 21:05:46 +00:00
Guillaume Desmottes
a56923d5e6 qtdemux: fix bug report URL
Using PACKAGE_BUGREPORT as in other modules.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5762>
2023-12-05 09:25:22 +01:00
Thibault Saunier
14c7d3f4e9 qtdemux: Do not update demux->offset when droping data on EOS
The offset is updated right after and we were breaking it by updating it
twice.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5724>
2023-12-02 08:08:26 +00:00
Thibault Saunier
b1b29de0fb qtdemux: Do not mark stream as EOS only if all streams are EOS
The `GstFlowCombiner` is responsible for tracking the flow of each
stream and handle the overal flow return value. Without that, we
can end up with the following scenario:

- Audio+video stream
- Only the video stream is linked downstream
- The audio stream goes EOS, video doesn't yet
  -> We update the Flow in the combiner with OK as all streams are not EOS
- Video goes EOS because downstream returned EOS
-> `qtdemux` returns `FLOW_OK` forever because the unlinked audio pad
  has `last_flowret==FLOW_OK`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5724>
2023-12-02 08:08:26 +00:00
Thibault Saunier
8295b2ae5c qtdemux: Determine EOS based on the stream segment
Depending on the stream segment might vary (because of edts for example)
leading to EOS being sent at the wrong time (too early for example).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5724>
2023-12-02 08:08:26 +00:00
Hosang Lee
7bf646e5ba qtdemux: Don't overflow sample index
Don't reduce sample index if it is already at 0.
Assigning -1 to a guint32 variable causes unexpected behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5743>
2023-12-01 13:34:12 +00:00
Hosang Lee
041e0c6cab qtdemux: Fix reverse playback for pcm audio stream
Some raw lpcm or adpcm may have larger sample sizes than the max
buffer size value set.
Trimming the buffer causes bogus size error on reverse playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5742>
2023-12-01 15:11:04 +09:00
Seungha Yang
5cbd062856 video: Add RBGA format
This new format is intended to be used by hardware decoders
where VUYA is only supported 4:4:4 decoding surface but
stream is encoded with GBR color space, HEVC and VP9 GBR streams
for example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5703>
2023-11-29 16:54:16 +00:00
Philippe Normand
ee1b905ff3 dashdemux2: Fix a couple leaks and a use-after-move
The tags and caps were leaked for unknown streams, I'm not sure they'd be valid
in that case, but better safe than sorry.

The tags ownership is transfered when calling `gst_adaptive_demux_track_new()`
so unreffing those afterwards was a mistake.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5714>
2023-11-24 17:01:33 +00:00
Robin Gustavsson
38a8411bdf rtpklvdepay: Recover after invalid fragmented KLV unit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4816>
2023-11-17 09:01:10 +00:00
Sebastian Dröge
db77deef00 rtpjitterbuffer: Add new "rfc7273-reference-timestamp-meta-only" property
If this property is enabled then the jitterbuffer will do the normal PTS
calculations according to the configured mode instead of making use of
the RFC7273 media clock.

The timestamp calculated from the RFC7273 media clock will only be
stored in the reference timestamp meta, if addition of that meta is enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
eae3ef7461 rtpjitterbuffer: Add new rfc7273-use-system-clock property
When this property is used, it is assumed that the system clock is
synced close enough to the media clock used by an RFC7273 stream.

As long as both clocks are at most a few seconds from each other this
will give the correct results and avoids having to create an actual
network clock that has to sync first.

If the system clock is actually synchronized to the media clock then
everything will behave exactly the same, otherwise the reference
timestamp meta will be correct but the buffer timestamps will be off by
the difference between the two clocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
2956ba48fc rtpjitterbuffer: Improve handling of media clocks
Do more checks for clock equality than just checking pointers. The same
NTP/PTP clock might be used as pipeline clock but a new instance, so
instead also check what clock they are synced to.

Also handling setting / resetting of the media clock and pipeline clock
correctly by resetting the media clock's state accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Piotr Brzeziński
4037334143 qtdemux: Ignore raw audio streams when adjusting seek
Because we treat raw audio chunks/samples as keyframes, they were interfering
with seek time adjustment.
Became apparent when the accompanying video stream was I-frame only,
for example ProRes.
Since raw audio streams can be seeked freely, it's fine to just ignore them here,
giving priority to the real keyframes in the video stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4946>
2023-11-15 07:55:27 +00:00
Dongyun Seo
8db184085a dcaparse: keep upstream buffer meta
Some audio decoders cannot decode DTS stream if there is no
valid timestamp. So, keep upstream buffer meta.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5655>
2023-11-14 16:51:44 +09:00
Olivier Crête
c2a357c867 rtpopusdepay: set resync flag
- Set re-sync flag on output buffer when rtp had the marker flag set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5529>
2023-11-10 21:45:13 +00:00
Philippe Normand
1fc2bd8032 adaptivedemux2-stream: Use gst_clear_object when releasing collection
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5606>
2023-11-08 09:16:55 +00:00
Johan Adam Nilsson
808c27b4cc wavparse: fix buffer leak with adtl tag
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3020
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5595>
2023-11-03 19:38:38 +00:00
robert
e3e8147a74 ximagesrc: fix xnavigation linking issue
Fixes #3083

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5589>
2023-11-03 17:36:58 +00:00
Seungha Yang
5e147ed3b8 meson: Fix MSVC build with GST_DISABLE_GST_DEBUG
MSVC does not understand Wno-unused

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5585>
2023-11-03 13:31:03 +00:00
Sebastian Dröge
2dd65d8715 mpg123audiodec: Update rank from MARGINAL to PRIMARY
This is our primary MP3 decoder after mad got removed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5590>
2023-11-02 14:17:06 +00:00
robert
737c32b9b6 ximagesrc: fix compile-time warning and XInitThreads()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5493>
2023-11-01 09:17:24 +00:00
Tim-Philipp Müller
f6c40bb15c pngenc: mark output frames as I-frames
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5546>
2023-10-27 05:47:37 +00:00
Tim-Philipp Müller
d69885e0f7 pngenc: output one frame only in snapshot mode
In snapshot mode pngenc should output exactly one frame
and then return FLOW_EOS to upstream. If upstream sends
more input frames before shutting down, it should keep
returning FLOW_EOS but not output any more encoded frames.

After a flushing seek it should output frames again though.

Fixes #3069.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5546>
2023-10-27 05:47:37 +00:00
Shengqi Yu
25c00b5ba2 v4l2object: scale the encoded sizeimage based on maximum resolution
The default 2MB ENCODED_BUFFER_SIZE can't support some 4K video playback. We now
detect the driver reported maximum resolution and choose an appropriate
default bitstream size accordingly. For 4K video these results in around 4MB
buffer instead of 2MB.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4549>
2023-10-23 14:10:56 +00:00
Matthias Fuchs
2bbc2a4c52 qml6glsrc: sync on the streaming thread
After rendering a QML scene the qml6glsrc element copies the contents of
the scene to a GStreamer buffer. This happens on the Qt render thread.
Then it attaches a sync point to the destination buffer. This sync point
must be awaited by other threads which use the buffer later on. The
current implementation relies on the downstream elements to wait for the
sync point. However, there are situation where this does not work. The
GstBaseTransform e.g. copies the buffer metadata (which overwrites the
sync point without waiting for it) *before* waiting for the sync point.

This commit waits for the sync point inside the qml6glsrc element before
sending it downstream. The wait command is issued on the streaming
thread with the pipeline OpenGL context, i.e. it will synchronize with
the GStreamer OpenGL thread.

This is a port of the original fix for the qmlglsrc element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5519>
2023-10-23 08:43:16 +00:00
Tim-Philipp Müller
654f3370a0 meson: Bump GLib requirement to >= 2.64
This includes fixes to make GstBus watches non-racy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2126>
2023-10-22 10:48:12 +01:00
Tim-Philipp Müller
136c82d735 flacenc: signal in output caps that the output is framed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5524>
2023-10-22 00:25:50 +00:00
Tim-Philipp Müller
bce1d121ba rtpac3depay: should output audio/x-ac3 not audio/ac3
audio/x-ac3 is the canonical media format in GStreamer.
audio/ac3 is sometimes accepted as input (e.g. in rtpac3pay
or ac3parse), but shouldn't be output.

Fixes #3038.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5472>
2023-10-19 13:27:58 +00:00
Matthias Fuchs
24ae3de107 qmlglsrc: sync on the streaming thread
After rendering a QML scene the qmlglsrc element copies the contents of
the scene to a GStreamer buffer. This happens on the Qt render thread.
Then it attaches a sync point to the destination buffer. This sync point
must be awaited by other threads which use the buffer later on. The
current implementation relies on the downstream elements to wait for the
sync point. However, there are situation where this does not work. The
GstBaseTransform e.g. copies the buffer metadata (which overwrites the
sync point without waiting for it) *before* waiting for the sync point.

This commit waits for the sync point inside the qmlglsrc element before
sending it downstream. The wait command is issued on the streaming
thread with the pipeline OpenGL context, i.e. it will synchronize with
the GStreamer OpenGL thread.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5506>
2023-10-19 08:19:05 +00:00