Commit graph

313 commits

Author SHA1 Message Date
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