Commit graph

611 commits

Author SHA1 Message Date
Jan Schmidt
57013e1a7c rtspsrc: Handle queries and events with no manager
When doing direct output with no session manager, we still
want to respond to queries and events from downstream, so
install the handlers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
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
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
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
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
Stéphane Cerveau
7c7a90b99d imagesequencesrc: fix regular image deadlock
With one regular image file path provided (without %05d),
the element was stuck in a dead loop counting the frames:

gst_image_sequence_src_count_frames

This allows to display any image file out of the element
for a given number of buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5471>
2023-10-12 22:06:02 +00:00
Guillaume Desmottes
a56aabc773 flvmux: set the src segment position as running time
We were already converting the pad last timestamp to running time but
not the segment position.
This segment position is used by gst_aggregator_simple_get_next_time()
to compute the waiting time when aggregating.

Those waiting times were wrong in my live pipeline using the system
clock, resulting in the aggregator to never wait at all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5460>
2023-10-11 15:20:18 +00:00
Nicolas Dufresne
aaed9272c1 video-filters: Fix passthrough with ANY caps feature
With the support for DRM modifiers, passthrough caps must now include DMA_DRM
format, otherwise pipeline using thhese filters unconditionally may fail
to negotiate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5386>
2023-10-03 21:13:00 +00:00
Xavier Claessens
0ab48250a9 GstCustomMeta: Use simplified API where possible
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5385>
2023-09-27 18:46:34 +00:00
Daniel Moberg
0e6cd64232 rtspsrc: Property for adding custom http request headers
This commit adds a property which enables adding custom http request headers to
the rtspsrc element. Added headers will be appended to http requests
made during http tunneling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5268>
2023-09-26 06:35:43 +00:00
Stijn Last
4bda59f88d deinterlace: greedy, improve quality
scanlines->m1 = same line of the previous field
scanlines->t0 = line above of the current field
scanlines->b0 = line below of the current field
scanlines->mp = same line of the next field

Deinterlacing a field weaved frame:
When deinterlacing the top field, the next bottom field is available
(part of the same frame). but when deinterlacing the bottom field,
the next top field (part of the next frame) is not available and
scanlines->mp equals NULL.

In this case it's better to use greedy algorithm using the prevous field
(twice) rather then linear interpolation of the current field.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5331>
2023-09-25 06:40:47 +00:00
Sebastian Dröge
2a2ef23829 rtpsource: Don't store invalid running times and calculate with it
If we end up with GST_CLOCK_TIME_NONE as running time for an RTP packet
then this can't be used for bitrate estimation, and also not for
constructing the next RTCP SR. Both would end up with completely wrong
values, and an RTCP SR with wrong values can easily break
synchronization in receivers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5329>
2023-09-23 07:39:00 +00:00
Sebastian Dröge
fcd591c1af rtpjitterbuffer: Avoid integer overflow in max saveable packets calculation with negative offset
The timestamp offset can be negative, and it can be a bigger negative
number than the latency introduced by the rtpjitterbuffer so the overall
timeout offset can be negative.

Using the negative offset for calculating how many packets can still
arrive in time when encountering a lost packet in an equidistant stream
would then overflow and instead of considering fewer packets lost a lot
more packets are considered lost.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5296>
2023-09-12 08:38:53 +00:00
Jonas K Danielsson
749652e60c rtp: Add rtppassthroughpay element
This elements pass RTP packets along unchanged and appear as a RTP
payloader element.

This is useful, for example when using the gstreamer-rtsp-server
library, in the case where you are receiving RTP packets from a
different source and want to serve them over RTSP. Since the
gst-rtsp-server library expect the element marked as payX to be a RTP
payloader element and assumes certain properties are available.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5204>
2023-08-22 14:01:09 +00:00
Guillaume Desmottes
bc06c2109c flvmux: add 'enforce-increasing-timestamps' property
The hack enforcing strictly increasing timestamps was, according to the
code comments, because librtmp was confused with backwards timestamps.

rtmp2sink is not using librtmp as rtmpsink did, so this is no longer
required.
Also changing the timestamps is causing audio glitches when streaming to
Youtube.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5212>
2023-08-21 14:26:06 +02:00
Sebastian Dröge
09045da073 rtpgstpay: Enable hdrext aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
2023-08-15 07:12:03 +02:00
Jochen Henneberg
a97d3acb90 rtp/vp8depay+vp9depay: Enable hdrext aggregation for VP8 and VP9
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
2023-08-15 07:12:03 +02:00
Jochen Henneberg
2673a66e60 rtp/h264depay+h265depay: Enable hdrext aggregation for H264 and H265
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
2023-08-15 07:12:03 +02:00
Vivia Nikolaidou
3257ee4374 deinterlace: Fix vfir 16-bit orc calculations
memcpy works in bytes, but orc works in items, so given that the size
arguments is in bytes, we need to divide by the pixel stride.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5172>
2023-08-11 17:47:27 +00:00
Vivia Nikolaidou
6145a5c7cb deinterlace: Fix greedyh crash for alternate-field interlacing
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2645

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5172>
2023-08-11 17:47:27 +00:00
Stéphane Cerveau
1e4cc59a3f isomp4: update isml documentation
Closing #2893

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5165>
2023-08-09 09:15:30 +00:00
Tim-Philipp Müller
be2a3780c1 flvmux: use version template in metadata creator properties
Avoids documentation churn when the version changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5090>
2023-08-07 10:41:16 +00:00
Tim-Philipp Müller
5bbd8c2d71 rtspsrc: use version template in user-agent property
Avoids documentation churn when the version changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5090>
2023-08-07 10:41:16 +00:00
Charlie Blevins
05cffc19dd rtpjitterbuffer: Allow earlier reference-timestamp-meta
Allow reference-timestamp-meta to be added earlier if an RTCP sender
report is sent before the first RTP packet.

Fixes #2843

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5084>
2023-08-03 17:26:42 +00:00
Alicia Boya García
5fd3c8a16c qtdemux: Fix premature EOS when some files are played in push mode
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2771

This EOS branch exists so that if a seek with a stop is made, qtdemux
stops accepting bytes from the sink after the entire requested playback
range is demuxed, as otherwise we could keep download content that is
not being used.

This patch fixes two flaws that were present in that EOS check:

1) A comparison was made between track time and movie time without conversion.
This made the check trigger early in files with edit lists. This patch fixes
this by converting the track PTS to movie PTS (stream time) for the check.

2) To avoid sending a EOS prematurely when the segment stop is within a GOP and
B-frames are present, the check for EOS should only be done for keyframes. I
gather this was already the intention with the existing code, but because it
used `stream->on_keyframe` instead of the local variable `keyframe` the old
code was checking if the *previous* frame was a keyframe.

It's interesting to note that these two flaws in the old code mask each other
in most cases: the track PTS will have reached the movie end PTS, but EOS would
only be sent if the previous frame was a keyframe. A simple case where they
wouldn't mask each other, reproducing the bug, is a sequence of 3 frame GOPs
with structure I-B-P.

The following validateflow tests have been added to future-proof the
fix:

 * validate.test.mp4.qtdemux_ibpibp_non_frag_pull.default
 * validate.test.mp4.qtdemux_ibpibp_non_frag_push.default

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5021>
2023-07-26 19:14:43 +00:00
Guillaume Desmottes
6b339b5d39 videoflip: fix concurrent access when modifying the tag list
We were checking if the tag list is writable, but it may actually be
shared through the same event (tee upstream or multiple consumers).

Fix a bug where multiple branches have a videoflip element checking the
taglist. The first one was changing the orientation back to rotate-0
which was resetting the other instances.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5097>
2023-07-25 15:18:05 +02:00
Xabier Rodriguez Calvar
5114fb4170 qtdemux: attach cbcs crypt info at the right moment
Before it was always added but that can cause issues when the stream begins
unencrypted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5085>
2023-07-25 10:06:48 +00:00
David Craven
c79d16ae80 matroska: demux: Strip signal byte from encrypted blocks
Removes the signal byte when the frame is unencrypted to
be consistent with when the frame is encrypted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4997>
2023-07-11 10:26:36 +00:00
Guillaume Desmottes
7b31c89f25 videoflip: fix critical when tag list is not writable
Fix this pipeline where the tag list is not writable:

gst-launch-1.0 videotestsrc ! taginject tags="image-orientation=rotate-90" ! videoflip video-direction=auto \
  ! autovideosink

GStreamer-CRITICAL **: 12:34:36.310: gst_tag_list_add: assertion 'gst_tag_list_is_writable (list)' failed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4987>
2023-07-07 11:17:43 +00:00
Seungha Yang
794cde703c rtspsrc: Fix crash when is-live=false
The pad's parent (i.e., rtspsrc) can be nullptr since we add pads
later.

Co-authored-by: Jan Schmidt <jan@centricular.com>

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2751
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4965>
2023-07-05 06:48:37 +00:00
Peter Stensson
33fb3bfd60 rtpvp9pay: Only mark first outgoing packet as non delta-unit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4937>
2023-06-29 09:48:41 +00:00
Peter Stensson
af43648bdf rtpvp8pay: Only mark first outgoing packet as non delta-unit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4937>
2023-06-29 09:48:41 +00:00
Peter Stensson
b40b4ffb81 rtph265pay: Only mark first NAL as non delta-unit
When the input buffer contained multiple NAL's the second one would keep
the non delta-unit flag for a key frame.

The delta-unit flag will now be set per NAL when preparing the buffer
list to payload.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4937>
2023-06-29 09:48:41 +00:00
Mathieu Duponchelle
7445b73e76 rtpsession: expose timeout-inactive-sources property
In some situations it is not desirable to time out when no data is
received any longer, users can opt in to this behavior via a new
property.

The property is also exposed on rtpbin and sdpdemux

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4880>
2023-06-28 18:45:25 +00:00
Edward Hervey
2f95cbd551 matroska-demux: Properly handle early time-based segments
Refusing an incoming segment in < GST_MATROSKA_READ_STATE_DATA should only be
done if the incoming segment is not in GST_FORMAT_TIME.

In GST_FORMAT_TIME, we are just storing the values and returning, so we can
invert the order of the checks.

Fixes proper segment propagation in matroska/webm DASH use-cases

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
2023-06-22 06:56:33 +00:00
François Laignel
1d00f726a0 qtdemux: opus: set entry as sampled
... otherwise streams with constant size samples defined with a single
`sample_size` for all samples in the `stsz` box fall in the category
`chunks_are_samples` in `qtdemux_stbl_init`, overriding the actual
sample count.

`FOURCC_soun` would set this automatically for `compression_id == 0xfffe`,
however `compression_id` is read from the Audio Sample Entry box at an offset
marked as "pre-defined" in some version of the spec and set to 0 both by
GStreamer and FFmpeg for opus streams.

Considering the stream `sampled` flag is set explicitely by other fourcc
variants, doing so for opus seems consistent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4903>
2023-06-20 17:15:22 +00:00
Sebastian Dröge
dbbfc917fe flacparse: Avoid integer overflow in available data check for image tags
If the image length as stored in the file is some bogus integer then
adding it to the current byte readers position can overflow and wrongly
have the check for enough available data succeed.

This then later can cause NULL pointer dereferences or out of bounds
reads/writes when actually reading the image data.

Fixes ZDI-CAN-20775
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4894>
2023-06-20 10:02:19 +00:00
François Laignel
fa30504ec2 qtdemux: parse Opus and dOps as qtdemux nodes and add size checks
This allows checking the nodes conformity and dumping parsed values.

Note: Audio Sample Entry version parsing and offset handling is handled as part
of `FOURCC_soun` common processing and in `qtdemux_parse_node`.

Also, only read `stream_count` and `coupled_count` when
`channel_mapping_family` != 0. See:

https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875>
2023-06-19 14:31:55 +00:00
François Laignel
439717ab65 qtdemux: fix byte order for opus extension and version field type
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.

[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875>
2023-06-19 14:31:55 +00:00
François Laignel
f3496ea3bf qtmux: fix byte order for opus extension
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.

In `build_opus_extension`, `gst_byte_writer_put*_le ()` variants were used,
causing audio streams conversion to Opus in mp4 to offset samples due to the
PreSkip field incorrect value (29ms early in our test cases).

[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875>
2023-06-19 14:31:55 +00:00
Mark Hymers
1ae8af4909 matroska: Add support for more pixel formats
- Add support for GRAY16_LE (using ffmpeg fourcc mapping)
- Update documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4824>
2023-06-14 13:40:58 +00:00
Daniel Morin
00178cbd89 matroska: Add new pixels format support
- Add support for GRAY10_BE32
- Add support for RGBA64_LE and BGRA64_LE

Sponsored by Living Optics

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4824>
2023-06-14 13:40:57 +00:00
Tim-Philipp Müller
f3c126d07c matroska-demux: fix accumulated base offset in segment seeks
When doing a segment seek, the base offset in the new segment
would be increased by segment.position which is basically the
timestamp of the last packet. This does not include the duration
of the last packet though, so might be slightly shorter than the
actual duration of the clip or the requested segment.

Increase the base offset by the segment duration instead when
accumulating segments, which is more correct as it doesn't cut
off the last frame and makes the effective loop segment duration
consistent with the actual duration returned from a duration
query.

In case a segment stop was specified it's also possible that
some data was sent beyond the stop that's necessary for decoding
so the base offset increment should be based on that then and
not on the timestamp of the last buffer pushed out.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4604>
2023-06-13 18:19:48 +00:00
Jochen Henneberg
fd1d208446 rtspsrc: Cleanup code for next pending command
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4792>
2023-06-07 20:30:36 +00:00
Jochen Henneberg
4790a8d2be rtspsrc: Do not try send dropped get/set parameter
If the set_get_param_q has been emptied we have to reset the cached
pending command to CMD_LOOP as we will not have the request parameters
anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4792>
2023-06-07 20:30:36 +00:00
Xabier Rodriguez Calvar
bdff780fe9 qtdemux: Fix critical message on cenc sample grouping parsing
Inside qtdemux_parse_sbgp there is already a check to ensure the fragment group
properties are not null but it is being hit in some examples and it is better to
directly avoid the critical.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4576>
2023-06-07 11:01:20 +00:00
Guillaume Desmottes
9b0736c85d videoflip: update orientation tag in auto mode
The frames are flipped according to the tag orientation so it's no longer accurate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4778>
2023-06-06 19:28:09 +00:00
Jan Alexander Steffens (heftig)
93699123b4 isomp4: Fix (E)AC-3 channel count handling
The muxer used a fixed value of 2 channels because the TR 102 366 spec
says they're to be ignored. However, the demuxer still trusted them,
resulting in bad caps.

Make the muxer fill in the correct channel count anyway (FFmpeg already
does) and make the demuxer ignore the value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4739>
2023-06-02 19:07:58 +00:00
Stéphane Cerveau
dd17beb681 gstreamer-full: add full static support
Allow a project to use gstreamer-full as a static library
and link to create a binary without dependencies.

Introduce the option 'gst-full-target-type' to
select the build type, dynamic(default) or static.

In gstreamer-full/static build configuration gstreamer (gst.c)
needs the symbol gst_init_static_plugins which is defined
in gstreamer-full.
All the tests and examples are linking with gstreamer but the
symbol gst_init_static_plugins is only defined in the gstreamer-full
library. gstreamer-full can not be built first as it needs to know what plugins
will be built.

One option would be to build all the examples and tests after
gstreamer-full as the tools.

Disable tools build in subprojects too as it will be built at the end of
build process.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
2023-05-31 15:17:11 +00:00
Hyung Song
d68a7fbd94 aacparse: parse GASpecificConfig for channels
Some media have valid channel information in GASpecificConfig which is
not yet implemented in gstaacparse. Parse data according to ISO/IEC
14496-3 just enough to get channels.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4720>
2023-05-30 09:09:16 +00:00
Guillaume Desmottes
0fd3c28620 flvmux: push metadata on caps change
The metdata contains tags but also caps dependent info such as the
resolution and the framerate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4730>
2023-05-30 09:35:43 +02:00
Guillaume Desmottes
3ae2904f3d flvmux: rename 'new_tags' to 'new_metadata'
The metadata contains more than just tags: resolution, framerate, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4730>
2023-05-30 08:27:18 +02:00
Guillaume Desmottes
853fad001e flvmux: add some logs when input is changing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4730>
2023-05-30 08:27:18 +02:00
Michael Olbrich
2197cdc289 flvmux: use the correct timestamp to calculate wait times
Since c0bf793c05 ("flvmux: Set PTS based on
running time") the timestamp of the output buffer is already in running
time. So using that for 'srcpad->segment.position' does not work correctly
because gst_aggregator_simple_get_next_time() will convert it again with
gst_segment_to_running_time().
This means that the timestamp returned by
gst_aggregator_simple_get_next_time() may be incorrect. For example, if
flvmux is added to a already runinng pipeline then the timestamp is too
small and gst_aggregator_wait_and_check() returns immediately. As a result,
buffers may be muxed in the wrong order.

To fix this, use the PTS of the incoming buffer instead of the outgoing
buffer. Also add the duration as get_next_time() is supposed to return the
timestamp of the next buffer, not the current one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4701>
2023-05-29 14:56:13 +00:00
Ruben Gonzalez
059965fe53 doc: Fix newline char between authors
Found running `gst-inspect-1.0 -a |& grep -v ":" | grep @`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4682>
2023-05-20 05:48:23 +00:00
Sebastian Dröge
d5a0cfc563 qtdemux: Add support for SpeedHQ video codec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3982>
2023-05-19 07:16:03 +00:00
Sebastian Dröge
99285bb566 qtmux: Fix extraction of CEA608 data from S334-1A packets
The index is already incremented by 3 every iteration so multiplying it
by 3 additionally on each array access is doing it twice and does not
work.

This caused invalid files to be created if there's more than one CEA608
triplet in a buffer, and out of bounds memory reads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4634>
2023-05-16 11:29:45 +00:00
Jan Schmidt
131d59518e splitmuxsrc: Make PTS contiguous by preference
Make splitmuxsrc deal better with stream reordering by
making the largest observed PTS contiguous in the
next fragment. Previously, it selected DTS, but then
aligned that with the segment start of the next fragment,
which holds PTS values - leading to glitches in
streams that don't have PTS = DTS at the start.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4637>
2023-05-16 04:34:16 +00:00
Sebastian Dröge
f9a3b3eacf rtpjitterbuffer: Fix uninitialized variable compiler warning
It could indeed be used uninitialized, but only if one of the
g_return_val_if_fail() caused an early return.

../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c: In function ‘rtp_jitter_buffer_append_query’:
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c🔢10: warning: ‘head’ may be used uninitialized
      [-Wmaybe-uninitialized]
 1234 |   return head;
      |          ^~~~
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c:1232:12: note: ‘head’ was declared here
 1232 |   gboolean head;
      |            ^~~~

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4616>
2023-05-14 14:26:05 +00:00
Mathieu Duponchelle
3d3d2ed447 Revert "qtdemux: fix conditions for end of segment in reverse playback"
This reverts commit 9deb3c27ac.

The test case that was described in the associated MR
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/262)
remains adequately fixed by a related MR that was merged later
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/275).

It introduced incorrect logic that broke edit lists as described in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2549

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2549
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4560>
2023-05-11 16:45:37 +00:00
François Laignel
6675ed9aae rtpmanager/rtsession: data race leading to critical warnings
This is a fix for a data race leading to:

> GLib-CRITICAL: g_hash_table_foreach:
>   assertion 'version == hash_table->version' failed

Identified sequence:

* `rtp_session_on_timeout` acquires the lock on `session` and proceeds with its
  processing.
* `rtp_session_process_rtcp` is called (debug log : received RTCP packet) and
  attempts to acquire the lock on `session`, which is still held by
  `rtp_session_on_timeout`.
* as part of an hash table iterator, `rtp_session_on_timeout` transitively
  invokes `source_caps` which releases the lock on `session` so as to call
  `session->callbacks.caps`.
* Since `rtp_session_process_rtcp` was waiting for the lock to be released, it
  succeeds in acquiring it and proceeds with `rtp_session_process_rr` which
  transitively calls `g_hash_table_insert` via `add_source`.
* After `source_caps` re-acquires the lock and gives the control flow back to
  `rtp_session_on_timeout`, the hash table iterator is changed, resulting in the
  assertion failure.

This commits copies `sess->ssrcs[sess->mask_idx]` and iterates on the copy so
the iterator is not affected by a concurrent change due to the lock being
released in the `source_caps` callback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4555>
2023-05-09 16:05:29 +00:00
Philippe Normand
fd194a0a2b rtpdtmfdepay: Classify as RTP element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4582>
2023-05-09 15:18:47 +00:00
Philippe Normand
a51fd006e6 rtpdtmfsrc: Classify as RTP source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4582>
2023-05-09 15:18:47 +00:00
Mathieu Duponchelle
020fd3d14d videoflip: fix setting of method property at construction time
Since c2f890ab, element properties are gathered from the parse-launch
line and passed at object construction.

This caused the following issue to happen in videoflip:

* videoflip installed a CONSTRUCT property named method, now deprecated
* videoflip now also overrides that property with a video-direction
  property

GObject construction causes method to be set first at construct time,
with the user-provided value, then video-direction with the default
value.

The user-provided value was thus overridden, causing a regression.

Fix by not installing the properties as CONSTRUCT, and explicitly
implementing constructed() instead in order to ensure that we do still
call gst_video_flip_set_method() at least once during construction.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2529

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4536>
2023-05-05 08:57:04 +00:00
Camilo Celis Guzman
0cee3cd833 rtpvp8pay: rtpvp9pay: access picture_id property atomically
Atomically set and get the picture_id. This changeset only atomically gets
the picture-id when such property is queried on the element, on every other
place where it is accessed internally it is accessed directly.

This is because there is no MT scenario where we would be modifying this value
and reading it internally in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
e4d8cda9a1 rtpvp8pay, rtpvp9pay: increment PictureID on FLUSH_START
In recent versions of Chrome (M106) a change on their jitter buffer means that
they are very susceptible to PictureID discontinuities.

Then avoid at all cost resetting the PictureID. Moreover, according to
the RFCs for VP8 and VP9 payloads; the PictureID can start off at any
random value. So there is no logical problem of incrementing it here
rather than resetting it, as long as it is a different PictureID.

WebRTC's recent corruption issue:
https://bugs.chromium.org/p/webrtc/issues/detail?id=15101

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
f159fd8568 rtpvp8pay, rtpvp9pay: expose picture-id as a property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
11187a81c3 rtpvp9pay: add picture-id-offset property
Bring the VP9 payloader in sync in this regard to the VP8 payloader

Allowing setting the picture id to a known value is useful when testing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
7cffb40c2e rtpvp9pay: minor refactor of PictureID logic
This brings the logic inline with the vp8pay

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
a79616ea7a rtpvp8pay: avoid reseting PictureID if NO_PICTURE_ID mode is set
There is no logical change here, as `& (1 << nbits) - 1` would produce also 0
when NO_PICTURE_ID mode is choosen. However, this avoid computing a random
integer that is actually unused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
7dd6375c5e rtpvp8pay, rtpvp9pay: use GType like name for PictureIDMode
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Xabier Rodriguez Calvar
021572de93 qtdemux: emit no-more-pads after pruning old pads
If we don't do that, clients can rely on this signal to see the final pad
topology but it won't be the real one as some of them will disappear after
emitting that signal. This can happen after injecting a different init segment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4535>
2023-05-03 12:06:00 +00:00
Carlos Rafael Giani
0071c97128 qtdemux: Add audio clipping meta when playing gapless m4a content
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4200>
2023-05-03 08:47:55 +00:00
Carlos Rafael Giani
51ebda4df5 qtdemux: use qtdemux debug category instead of default in qtdemux_tags.c
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4200>
2023-05-03 08:47:55 +00:00
François Laignel
5ef2ce69ff rtpmanager/rtsession: race conditions leading to critical warnings
While testing the [implementation for insertable streams] in `webrtcsink` &
`webrtcsrc`, I encountered critical warnings, which turned out to result from
two race conditions in `rtpsession`. Both race conditions produce:

> GLib-CRITICAL: g_hash_table_foreach:
>   assertion 'version == hash_table->version' failed

This commit fixes one of the race conditions observed.

In its simplest form, the test consists in 2 pipelines and a Signalling server:

* pipelines_sink: audiotestsrc ! webrtcsink
* pipelines_src: webrtcsrc ! appsrc

1. Set `pipelines_sink` to `Playing`.
2. The Signalling server delivers the `producer_id`.
3. Initialize `pipelines_src` to establish a session with `producer_id`.
4. Set `pipelines_src` to `Playing`.
5. Wait for a buffer to be received by the `appsrc`.
6. Set `pipelines_src` to `Null`.
7. Set `pipelines_sink` to `Null`.

The race condition happens in the following sequence:

* `webrtcsink` runs a task to periodically retrieve statistics from `webrtcbin`.
  This transitively ends up executing `rtp_session_create_stats`.
* `pipelines_sink` is set to `Null`.
* In `Paused` to `Ready`, `gst_rtp_session_change_state()` calls
  `rtp_session_reset()`.
* The assertion failure occurs when `rtp_session_reset` is called while
  `rtp_session_create_stats` is executing.

This is because `rtp_session_create_stats` acquires the lock on `session` prior
to calling `g_hash_table_foreach`, but `rtp_session_reset` doesn't acquire the
lock before calling `g_hash_table_remove_all`.

Acquiring the lock in `rtp_session_reset` fixes the issue.

[implementing insertable streams support]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4528>
2023-05-02 21:56:39 +00:00
Xabier Rodriguez Calvar
66c15bc753 qtdemux: Fix segfault in cenc sample grouping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4523>
2023-05-02 11:32:01 +02:00
Sebastian Dröge
3044b0992f Revert "splitmuxsink: Avoid assertion when WAITING_GOP_COLLECT on reference context"
This reverts commit f29c19be58. If this is
called for the reference context then we would run into an infinite
loop, which is not really better than an assertion.

By fixing up DTS to never be ahead of the PTS in the previous commit
this situation should be impossible to hit now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4498>
2023-04-28 11:00:19 +00:00
Sebastian Dröge
de907c225b splitmuxsink: Catch invalid DTS to avoid running into problems later
DTS > PTS makes no sense, so we clamp DTS to the PTS. Also if there's a
PTS but no DTS, then assume that PTS=DTS to make sure we're not working
with a much older DTS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4498>
2023-04-28 11:00:19 +00:00
Sebastian Dröge
ef89bac181 rtspsrc: Fix handling of * control path
Regression introduced by 7f9d689572.
Thanks to Tristan Matthews for reporting this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4497>
2023-04-27 13:47:56 +00:00
Sebastian Szczepaniak
277a9f0cef qtdemux: Add support for cenc sample grouping
Co-authored-by: Xabier Rodriguez Calvar <calvaris@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3551>
2023-04-26 18:51:56 +00:00
Guillaume Desmottes
d4a9106499 videoflip: check that stream actually changed when resetting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4377>
2023-04-22 14:03:16 +02:00
Guillaume Desmottes
7c4e36acfd videoflip: reset orientation if not present in a tag update
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4377>
2023-04-22 14:02:13 +02:00
Guillaume Desmottes
c0fa04fcaf videoflip: handle tag list scopes
STREAM taglist can now overrides the orientation from the GLOBAL
taglist, but not the other way around.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4377>
2023-04-22 14:02:13 +02:00
Guillaume Desmottes
96afec6253 videoflip: reset orientation on new stream
Fix the following use:
- upstream sends a video with a rotation tag, say 90°
- upstream switches to another video without rotation
- the second video was still rotated by videoflip

Fix this by resetting the orientation when receiving STREAM_START.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4377>
2023-04-22 14:02:13 +02:00
Seungha Yang
52cb42f4bb deinterlace: Add support for high bitdepth planar YUV formats
Add C implementation for high bitdepth planar YUV formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1476>
2023-04-18 01:32:25 +09:00
Seungha Yang
aabe9136f6 deinterlace: yadif: Prettify indentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1476>
2023-04-18 01:25:45 +09:00
Edward Hervey
4c6f41a00a qtdemux: Fix av1C parsing
This is a regression introduced by
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3882

The av1c codec configuration parsing would always fail due to an off-by-one
error, the content of an atom starting at offset 8 (i.e. the 9th byte) and not
9 (the 10th byte).

Also introduce a break in order to not get stray warnings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4433>
2023-04-17 09:28:43 +02:00
Mathieu Duponchelle
6a27fe8955 docs: mark GstRTPMux as plugin API
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4408>
2023-04-13 21:46:59 +00:00
Jan Alexander Steffens (heftig)
ac83e121a7 imagesequencesrc: Properly set default location
Noticed this because the generic_states test kept segfaulting at random.
GLibC 2.37 can crash when NULL is supplied as a format string.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4109>
2023-04-13 01:55:23 +00:00
Tim-Philipp Müller
b020d399cb multifile: error out if no filename was set
Fixes #2483

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4404>
2023-04-12 18:55:26 +00:00
Seungha Yang
3374f2f44d udpsrc: Add support for IGMPv3 SSM
Adding "multicast-source" property to support Source Specific Muliticast
RFC 4604. The source can be multiple address with '+' (for positive
filter) or '-' (negative filter) prefix, or URI query can be used.
Note that negative filter is not implemented yet and it will be
ignored

Example:
gst-launch-1.0 uridecodebin \
  uri=udp://{ADDRESS}:PORT?multicast-source=+SOURCE0+SOURCE1

Inspired by:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2620

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3485>
2023-04-12 16:32:07 +00:00
Edward Hervey
7e619f7e83 twcc: Better handle duplicate packets
The previous code would only check if two packets in a row were duplicates. If
not (i.e. a packet is a duplicate of a packet received slightly before) the code
would generate completely bogus FCI because it assumes there were no duplicates
present in the array.

In order to be efficient, just store all received packets and remove the
duplicates just before the FCI is generated once the array of observations have
been sorted by seqnum.

Fixes TWCC usage with moderate to high packet duplication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4328>
2023-04-10 09:37:51 +00:00
Sebastian Dröge
43e4db9fc9 rtspsrc: Skip PTs with caps incompatible to the global caps
Otherwise empty caps are created while all following code assumes that
the caps will have exactly one structure, and then run into assertions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4339>
2023-04-04 22:13:59 +00:00
Tim-Philipp Müller
ba417b0e07 rtpjpegdepay: fix logic error when checking if an EOI is present
We wouldn't add the missing EOI marker if the frame ended with
either 0xFF NN or 0xNN D9.

Fixes #2407

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4256>
2023-03-24 19:39:33 +00:00
Piotr Brzeziński
5beef42922 qtdemux: Fix seek adjustment with SNAP_AFTER flag
With GST_SEEK_FLAG_SNAP_AFTER present, the previous version would
adjust seek time based on the keyframe farthest away from desired_time.
This was incorrect, because we always want the *earliest* suitable keyframe
to seek to, not the last one.
With this fix, in case of the SNAP_AFTER, we now look for the closest keyframe
that can be found after desired_time. Behaviour for SNAP_BEFORE should remain
unchanged.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4183>
2023-03-22 13:05:53 +00:00
Edward Hervey
ee759fb4bf plugins: Fix wrong enum usage
gcc 13 now detects conflicting enum usages. Fix the various cases where it was wrong

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4225>
2023-03-20 11:40:30 +00:00
Sebastian Dröge
621ec7b6e8 matroskademux: Make gst_byte_reader_get_data() usage less confusing
This is effectively the same behaviour but retrieving 0 bytes of data is
confusing to read.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4210>
2023-03-18 16:34:19 +02:00
Enrique Ocaña González
735dac9d2f qtdemux: Fix crash on MSE-style flush
The flowcombiner and active_streams shouldn't be cleared in the
mse-bytestream variant, only in the mss-fragmented one. Otherwise the
soft reset leaves qtdemux in a state where it still believes that it has
streams, but they've been cleared. In that case, a null pointer
dereference happens and the app crashes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4199>
2023-03-17 15:33:49 +00:00
Tim-Philipp Müller
0fc568c6b1 gst-plugins-good: re-indent with GNU indent 2.2.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4182>
2023-03-17 03:18:54 +00:00
Arun Raghavan
82b892ba3e matroskamux: Set rate/channels in Opus template caps
For some reason these were missed, and if caps didn't have them, we would emit
an invalid Matroska file with a 0 value for Sampling Frequency or channels.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4151>
2023-03-14 11:09:08 -04:00
Arun Raghavan
0ed51294e0 rtpopusdepay: Assume 48 kHz if sprop-maxcapturerate is missing
This matches 7587, section 6.1:

>   sprop-maxcapturerate:  a hint about the maximum input sampling rate
>      [...]
>      bandwidths (Table 1).  By default, the sender is assumed to have
>      no limitations, i.e., 48000.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4151>
2023-03-14 11:09:08 -04:00
Itamar Marom
b8730bc98e splitmuxsink: Fix docs support version
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4138>
2023-03-09 15:08:19 +02:00
Matt Feury
224030ff0c rtspsrc: Consider "451: Parameter Not Understood" when handling broken control urls
similar to https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3854

it seems that some implementations return this when
the server does not implement URL handling correctly

this fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2334

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4123>
2023-03-07 10:32:32 -05:00
Alicia Boya García
c1f4bd5a3f qtdemux: Add MSE-style flush
The abort() method of SourceBuffer in Media Source Extensions is
expected to flush the demuxer and discard the current fragment,
if any. The configuration of tracks, if any, should be preserved.

qtdemux has different behavior for flush events depending on the
context.

This patch activates the intended behaviour only for streams of the
VARIANT_MSE_BYTESTREAM type, conformant to the ISO BMFF Bytestream
specification[1]. This flush behaviour is the same as the one
already in use for adaptivedemux sources.

[1] https://www.w3.org/TR/mse-byte-stream-format-isobmff/

https://bugzilla.gnome.org/show_bug.cgi?id=795424

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4101>
2023-03-02 17:54:41 +00:00
Shengqi Yu
83576690b6 matroskademux: Consider TrackUID==0 a warning and not handle it as error
some special files whose trackUID is 0 can be played on the other
player. But it cannot be played in GStreamer, because trackUID 0 will be
treated as an error in matroskademux.

So, it makes sense to only consider trackUID==0 a warning and not handle
it as error

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1821

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4036>
2023-03-01 07:38:24 +00:00
Scott Kanowitz
2e4fd325e7 rtpsession: fix a race condition during the EOS event in gstrtpsession.c
This patch prevents a possible race condition from taking place between the EOS event handling and rtcp send
function/thread.

The condition starts by getting the GST_EVENT_EOS event on the send_rtp_sink pad, which causes two core things
to happen -- the event gets pushed down to the send_rtp_src pad and all sessions get marked "bye" prior to
completion of the event handler. In another thread the rtp_session_on_timeout function gets called after an
expiration of gst_clock_id_wait in the rtcp_thread function. This results in a call to the
ess->callbacks.send_rtcp(), which is configured as a function pointer to gst_rtp_session_send_rtcp via the
RTPSessionCallbacks structure passed to rtp_session_set_callbacks in the gst_rtp_session_init function.

In the race condition, the call to gst_rtp_session_send_rtcp can have the all_sources_bye boolean set to true
while GST_PAD_IS_EOS(rtpsession->send_rtp_sink) evaluates to false. This is the result of gst_rtp_session_send_rtcp
running before the send_rtp_sink's GST_EVENT_EOS handler completes. The exact point at which this condition occurs
is if there's a context switch to the rtcp_thread right after the call to rtp_session_mark_all_bye in the
GET_EVENT_EOS handler, but before the handler returns.

Normally, this would not be an issue because the rtcp_thread continues to run and indirectly call
gst_rtp_session_send_rtcp. However, the call to rtp_source_reset sets the sent_bye boolean to false, which ends up
causing rtp_session_are_all_sources_bye to return false. This gets passed to gst_rtp_session_send_rtcp and the EOS
event never gets sent.

The race condition results in the EOS event never getting passed to the rtcp_src pad, which prevents the bin and
pipeline from ever completing with EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3798>
2023-02-28 17:01:08 +00:00
Sebastian Dröge
269915a51e rtspsrc: Use the correct vfunc for the push-backchannel-sample action signal
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/446

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4050>
2023-02-23 09:22:23 +00:00
Seungha Yang
1f0528b428 qtmux: Fix assertion on caps update
GstQTMuxPad.configured_caps should be protected since it's
updated from streaming thread and accessed in aggregate thread

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4042>
2023-02-22 19:16:52 +00:00
Hosang Lee
88f16ebd2a qtdemux: compensate wrong data offset for MSS fragments
A data offset with an offset smaller than the moof length is wrong
in smooth streaming streams.

The samples will not be located and eventually playback will
error out. So compensate assuming data is in mdat following moof.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Seungha Yang
f7c2602d41 splitmuxsrc: Proxy latency query to part reader
splitmuxsrc can respond to the latency query

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3566>
2023-02-15 23:47:50 +00:00
Vivia Nikolaidou
4e7a5ebb11 qtdemux: Handle moov atom length=0 case by reading until the end
Previously it would fail to demux the file by trying to read G_MAXUINT64
bytes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3934>
2023-02-11 02:20:39 +00:00
Vivia Nikolaidou
3a9acff978 qtdemux: Fix guint vs gsize type confusion
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3934>
2023-02-11 02:20:39 +00:00
Sebastian Dröge
5486ed24a5 qtmux: Implement writing of av1C version 1 box
Version 0 is ancient and not specified in any documents. Take it
directly from the `codec_data` if presents or otherwise try to construct
a reasonably looking `av1C` box.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3882>
2023-02-09 14:04:06 +00:00
Sebastian Dröge
8593a58916 qtdemux: Drop av1C version 0 parsing and implement version 1 parsing
The av1C box is optional so dropping parsing does not break anything
fundamentally, and there seems to be no historical record how version 0
even looks like while the comments and the parsing disagreed with each
other.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3882>
2023-02-09 14:04:06 +00:00
Patricia Muscalu
c3e52d5c4f rtph264pay: Don't insert SPS/PPS before the second image slice
Only the first slice, for which fist_mb_in_slice is set to 0,
should trigger insertion of SPS and PPS buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3402>
2023-02-08 12:10:11 +00:00
Enrique Ocaña González
92a4cfe20f qtdemux: Don't emit GstSegment correcting start time when in MSE mode
When using qtdemux in a pipeline that should only work as a pure demuxer (not
for actual playback), qtdemux shouldn't emit new GstSegments to correct
the start time (jump to the future) to ensure that the user experiences no
playback delay. By doing so, it's generating the wrong segments when an append
of data from the past happens. When that happens, downstream elements such as
parsers (eg: aacparse) may clip those buffers laying before the GstSegment and
create problems on the GStreamer client app (eg: WebKit).

Getting buffers clipped out because of the wrong GstSegments started becoming
a problen when this commit was introduced:

ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it

This clipping makes test DASH shaka 35 from MVT tests[1] to fail in
WebKitGTK/WPE (at least) and can potentially cause a number of other problems
in the WebKit Media Source Extensions (MSE) code.

Note that this new behaviour of not emitting new GstSegments only makes sense
when qtdemux is being used as a pure demuxer and not as part of a regular
pipeline. This is why the variant field has been added. When equal to
VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode,
taking decisions that meet the expectations for an MSE-like processing mode.
This kind of tweaks have been done in the past for MSS streams, for instance.
That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of
its own dedicated boolean flag.

Co-authored by: Alicia Boya García <ntrrgc@gmail.com>

...who suggested to use "variant: mse-bytestream" in the caps to identify that
mode, as proposed in her unmerged patch:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467

[1] https://github.com/rdkcentral/mvt

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3867>
2023-02-06 12:42:49 +00:00
Guillaume Desmottes
3d1390d31a rtpptdemux: set different stream-id on each src pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
2023-02-01 09:17:33 +00:00
Guillaume Desmottes
cc2b8f6ae8 rtpssrcdemux: set different stream-id on each src pad
All the RTP src pads were sharing the same stream-id while each actually
carry a different stream.

This was causing problem for example when funneling the streams together
and then trying to split them using 'streamiddemux'.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
2023-02-01 09:17:33 +00:00
Sebastian Dröge
3ca85189fd rtspsrc: Also consider "Method Not Valid In This State" error in broken control URL handling workaround
Some servers send a 455 error instead of any reasonable error when using
a correctly constructed control URL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3854>
2023-02-01 07:55:24 +00:00
Alicia Boya García
8a6023a38a qtdemux: Use safer clearing functions in dispose()
In theory, `dispose()` functions should be idempotent and should be
prepared not to crash or cause a double-free if an unref done from
inside caused a recursive call to `dispose()` of the same object.

https://developer.gnome.org/gobject/stable/howto-gobject-destruction.html

This patch modifies the `dispose()` method to honor these constraints.

Since the double `dispose()` call won't actually occur in qtdemux (there
is no cycle detection mechanism that could invoke it to work that way),
this is more of a code cleanup than a user-facing problem fix.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3822>
2023-01-28 00:32:57 +00:00
Daniel Knobe
5e9a32ed8c imagefreeze: add bayer support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3807>
2023-01-26 21:30:51 +00:00
Mathieu Duponchelle
2048a0a4d9 redenc: fix setting of extension ID for twcc
1 was previously hardcoded in, and the bug went under the radar because
webrtcsink hardcodes the number too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3785>
2023-01-24 22:52:48 +00:00
Tim-Philipp Müller
74e103e53f xingmux: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
fc82621e09 multiudpsink: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
8222b97331 rtpmanager: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
e66f8cff26 rtp: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
56d3beed0b multifile: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
e256472ca6 matroska: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
172c6ca1dc flv: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
9f4c514c52 dtmf: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
David Svensson Fors
d0edc1ad6a udpsrc: GstSocketTimestampMessage only for SCM_TIMESTAMPNS
Deserialize socket control messages as GstSocketTimestampMessage only
if (level, type) is (SOL_SOCKET, SCM_TIMESTAMPNS).

Without this patch, messages with types SCM_RIGHTS or SCM_CREDENTIALS
could be deserialized as GstSocketTimestampMessage instead of
GUnixFDMessage or GUnixCredentialsMessage from gio.

Fixes #1736

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3777>
2023-01-24 10:49:01 +01:00
Hiero32
145d362129 taginject: Add scope property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3697>
2023-01-24 00:20:53 +00:00
Sebastian Dröge
067b5d92b4 matroska: Add stream-format = (string) obu-stream to AV1 caps
Anything else is not allowed in Matroska/WebM.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3740>
2023-01-19 12:10:40 +02:00
Sebastian Dröge
4c8141a0c3 isomp4: Add stream-format = (string) obu-stream to AV1 caps
Anything else is not allowed in MP4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3740>
2023-01-19 12:10:40 +02:00
Jan Alexander Steffens (heftig)
211191564e qtdemux: Add basic support for AVC-Intra video
AVC-Intra is a range of H.264-compliant intra-only codecs from
Panasonic. The codes and descriptions have been taken from VLC.

The (encumbered) sample I have here produces byte-stream H.264,
including SPS and PPS and no `avcC` box.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3739>
2023-01-18 10:01:30 +00:00
Olivier Crête
c593930055 rtopuspay: Use GstStaticCaps to cache parsed caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Olivier Crête
46a6f72f03 rtopuspay: Ignore the stereo parameter in multiopus caps
Also add unit tests for the various variants

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Olivier Crête
f1cf457811 rtpopuspay: Leave original caps as-is
This should make it work if someone specifies stereo with MULTIOPUS
somehow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Olivier Crête
c52c66b575 rtpopuspay: Return upstream channel filter based on OPUS vs MULTICAPS
Only allow 1 or 2 channels if the caps are OPUS, or 3+ if they are
MULTIOPUS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Olivier Crête
c51ae6112d rtpopus: Put MULTIOPUS in all caps
The RTP payload encoding-name are always in caps in GStreamer.
In SDP, they are not case-sensitive, but since caps are, we need to pick
a caps, and we picked upper-case along time ago.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Seungha Yang
6540c4e89c rtspsrc: Fix string leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>
2022-12-28 04:39:18 +09:00
Seungha Yang
9b305df1cc rtptimerqueue: Fix memory leak
Should chain up to parent's finalize

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>
2022-12-27 19:31:16 +00:00
Patricia Muscalu
d752bf1b46 qtmux: Fix buffer leak in fragment_buffers
When pushing buffers from one of the sink pads fail,
make sure that all buffers added to fragment_buffers on other pads
are freed as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3624>
2022-12-22 14:11:10 +00:00
Mathieu Duponchelle
194dcd91e0 qtmux: For video with N/1001 framerates use N as timescale instead of centiframes
This is recommended by various specifications for such framerates, while
for integer framerates we continue using centiframes to allow for some
more accuracy.

Using N means that no rounding error accumulates, eventually leading to
outputting a packet with a different duration.

Some tools such as MediaInfo determine that a stream is variable
framerate if any packet has a different duration than the others, and
there is no reason I can see for not using the full 4 bytes of
resolution that the mp4 timescale offers.

Example problematic pipeline:

```
videotestsrc num-buffers=5001 ! video/x-raw,framerate=60000/1001,width=320,height=240 ! \
videoconvert ! x264enc bitrate=80000 speed-preset=1 tune=zerolatency ! h264parse ! \
video/x-h264,profile=high-10 ! mp4mux ! filesink location="result2.mp4"
```

This results in a media file that MediaInfo detects as variable
framerate because the 5000th packet has duration 99 instead of 100.

With this patch, the timescale is 60000 and all packets have duration
1001.

Related issue for context: https://bugzilla.gnome.org/show_bug.cgi?id=769041

Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3049>
2022-12-22 12:31:06 +02:00
Sebastian Dröge
066558cba1 qtdemux: Always use tfdt if available in BYTE segments
This reverts the decision from
  https://bugzilla.gnome.org/show_bug.cgi?id=754230
where it was decided that we rather play safe and only use the `tfdt` if
it is "significantly different" to the sum of sample durations.

As the specification says

    If the time expressed in the track fragment decode time (‘tfdt’) box
    exceeds the sum of the durations of the samples in the preceding
    movie and movie fragments, then the duration of the last sample
    preceding this track fragment is extended such that the sum now
    equals the time given in this box.

we have to use the `tfdt` in general to allow for it to signal gaps in
the stream.

A muxer producing fragments might not yet know the full duration of the
last sample of a previous fragment if the next fragment starts with a
gap, and knowing the actual start of the next fragment would potentially
require to violate latency requirements.

Additionally, the existence of `tfdt` allows to avoid accumulating
rounding errors from summing up the durations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3586>
2022-12-17 19:26:19 +02:00