Commit graph

400 commits

Author SHA1 Message Date
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
Xabier Rodriguez Calvar
87ae60176b qtdemux: Clear protection events when we get new ones
If we keep the old events they can be end up being passed to the app, that could
discard the protection information because it has been seen before.

Drive by improvement: use g_queue_clear_full instead of foreach+clear for
protection events.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3547>
2022-12-14 11:01:23 +01:00
Mathieu Duponchelle
fa71217502 rtpvp9depay: expose keyframe-related properties
This simply brings in the wait-for-keyframe and request-keyframe
properties from rtpvp8depay.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/909>
2022-12-10 13:28:07 +00:00
Jacek Skiba
61c17c5665 qtdemux: exit when protection caps are not defined during PIFF parsing
Reproduction testcase (uses PlayReady):
https://developers.canal-plus.com/rx-player/upc/?appTileLocation=[object%20Object]

In test streams we are using PIFF box, but caps did not had
present GST_PROTECTION_SYSTEM_ID_CAPS_FIELD. In consequence, invalid
system_id was returned which caused SIGSEGV crash.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3535>
2022-12-07 18:35:37 +00:00
Philippe Normand
b9011f3541 flacparse: Fix handling of headers advertising 32bps
According to the flac bitstream format specification, the sample size in bits
corresponding to `111` is 32 bits per sample.

https://xiph.org/flac/format.html#frame_header

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3517>
2022-12-04 11:47:57 +00:00
Aleksandr Slobodeniuk
38f6a0ba2e rtspsrc: fix seek event leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3500>
2022-12-01 23:52:40 +00:00
Matt Crane
ca7f66f9b5 rtpsession: Support disabling late adjustment of ntp-64 header ext
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.

This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
2022-11-24 08:23:03 +00:00
Johan Sternerup
9794c9bfd0 Use the correct SSRC(s) when routing a RTCB FB FIR
Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>
2022-11-23 11:31:23 +00:00
Jan Schmidt
cb225b3682 rtpsource: Track the seqnum for senders
RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.

Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>
2022-11-23 10:26:29 +00:00
Jan Alexander Steffens (heftig)
1d7c936db0 rtspsrc: Don't replace 404 errors with "no auth protocol found"
When getting a "404 Not Found" response from the DESCRIBE request, the
source produced a "No supported authentication protocol was found" error
instead of passing on the 404, which was confusing.

Only produce this error message when we're handling a response of "401
Unauthorized" without a compatible WWW-Authenticate header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3414>
2022-11-22 13:07:17 +00:00
Edward Hervey
f3c2f612ce rtspsrc: Don't leak sticky events
We have incremented the reference 2 lines above, and
gst_pad_store_sticky_event() does not take a reference, therefore release it

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>
2022-11-21 19:02:44 +00:00
Vivia Nikolaidou
f29c19be58 splitmuxsink: Avoid assertion when WAITING_GOP_COLLECT on reference context
I have seen a backtrace out in the wild where this happened. Maybe after
receiving EOS and stream-start on the reference context.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3005>
2022-11-18 15:52:03 +00:00