Commit graph

460 commits

Author SHA1 Message Date
Jan Schmidt 4d2f000125 rtspsrc: return NO_PREROLL on PLAYING->PAUSED too
When transitioning back to PAUSED and rtspsrc is live, return
NO_PREROLL so the pipeline knows to skip preroll here too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Tim-Philipp Müller 4db25f1500 rtspsrc: Consider 503 Service Not Available when handling broken control urls
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6213>
2024-03-05 17:45:18 +00:00
Nirbheek Chauhan cf2238a522 rtspsrc: Increase rank to PRIMARY for autoplug purposes
This affects autoplug by gst_element_make_from_uri() in, for example,
uridecodebin. The element should've already been PRIMARY rank, but it
was NONE because gst_element_make_from_uri() doesn't ignore NONE rank
elements when searching for element factories, unlike decodebin.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/502

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6226>
2024-02-27 11:36:01 +00:00
Sebastian Dröge 69e4564c87 rtphdrext-clientaudiolevel: Fix typo in documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6175>
2024-02-21 17:25:43 +00:00
Tim-Philipp Müller 0a6948ee20 rtppassthroughpay: fix critical in gst-inspect
gst_segment_to_running_time() will fail noisily
if the segment has not been initialised yet.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6151>
2024-02-21 11:25:10 +00:00
Jan Schmidt f7e494f348 rtspsrc: Reset combined flows after a seek before restarting
After a flushing seek, rtspsrc doesn't reset the last_ret value for
streams, so might immediately shut down again when it resumes pushing
buffers to pads due to a cached `GST_FLOW_FLUSHING` result

Prevent a stored flushing value from immediately stopping
playback again by resetting pad flows before (re)starting
playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6137>
2024-02-21 01:50:13 +00:00
Jochen Henneberg 6608b89977 rtpxqtdepay: Enabled header extension aggregation
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg 5d1d0cf9a5 rtpmp4gdepay: Enabled header extension aggregation
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg 75849c63c8 rtpsbcdepay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg 3fffcd021a rtpvorbisdepay: Enabled header extension aggregation
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg e1e7421982 rtpmp4vdepay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg 334ceaca21 rtptheoradepay: Enabled header extension aggregation
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg 0a4918a509 rtpsv3vdepay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg d810049f01 rtpmp4adepay: Enabled header extension aggregation
Because this depayloader may build several output buffers within one process
run we push them all into a GstBufferList and push them out at once to
make sure that each buffer gets notified about each header extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg 90b5d2eb93 rtpklvdepay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg 2c3f169ebb rtpjpegdepay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:16 +00:00
Jochen Henneberg 460813f7ee rtpj2kdepay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:16 +00:00
Jochen Henneberg ae3a00abd2 rtph263pdepay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:16 +00:00
Jochen Henneberg 4fd4c240e0 rtph263depay: Enabled header extensions aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:16 +00:00
Jochen Henneberg ae5bdaa7e1 rtph261depay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:16 +00:00
Sebastian Dröge 499474a76d Revert "rtpvp8pay: Use GstBitReader instead of dboolhuff implementation from libvpx"
This reverts commit b730e7a1b2.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3300

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6116>
2024-02-14 15:45:24 +00:00
Mathieu Duponchelle 91317aacaf webrtcbin, rtpbin: check before setting properties on jitterbuffer
In rtpbin we already systematically check for all property names
except latency, correct that.

In webrtcbin we need to check before trying to use the do-retransmission
property.

This is useful for the case where an element like identity gets passed
to rtpbin's request-jitterbuffer property, when the application wants
to use webrtcbin in an SFU situation, with no reordering and no added
latency

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6112>
2024-02-14 08:52:50 +00:00
Sebastian Dröge c726add352 rtpfunnel: Handle NTP-64 RTP header extension in caps similar to TWCC
This is another header extension that is handled by rtpsession and needs
to be preserved in the caps that are created by rtpfunnel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6109>
2024-02-14 08:05:33 +00:00
Sebastian Dröge 17e7af7181 rtpfunnel: Also write TWCC RTP header extension into buffer list buffers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6110>
2024-02-14 01:56:20 +00:00
Philippe Normand e9ecde83a7 matroska-demux: Basic support for container-specific-track-id tag
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6041>
2024-02-12 10:37:29 +00:00
Philippe Normand 30bb88a91b qtdemux: Basic support for container-specific-track-id tag
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6041>
2024-02-12 10:37:29 +00:00
Ignazio Pillai 34741e1db2 cutter: add audio-level-meta
Set GstAudioLevelMeta on buffers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5771>
2024-02-08 13:52:40 +00:00
Sebastian Dröge b730e7a1b2 rtpvp8pay: Use GstBitReader instead of dboolhuff implementation from libvpx
All compressed frame header values that are read as part of the
payloader are encoded as bits with 50:50 probability, and as such are
just the plain bits as they are.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5810>
2024-01-31 16:52:28 +00:00
Daniel Morin 0a55c86e6a rtspsrc: update rtsp url on redirect
- If a redirect took place on a GET when rtsp is tunneled we update the
  rtsp url too.
- log source and final destination on redirect

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5222>
2024-01-31 11:43:45 +00:00
Thibault Saunier e1a8ce16b4 matroskademux: Lower verbosity of some often happenning warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6011>
2024-01-30 09:09:22 +00:00
Thibault Saunier 77e7efe407 qtdemux: Lower verbosity of some often happenning warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6011>
2024-01-30 09:09:22 +00:00
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
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
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
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
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
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