Commit graph

10406 commits

Author SHA1 Message Date
Mathieu Duponchelle
cb75eda13b isomp4/qtmux: allow renegotiating when tier / level / profile change
Those are carried either in codec_data or in-band SPS (for avc3),
and it is OK for those to change, though decoders obviously need
to support it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
2021-08-20 00:16:43 +00:00
Mathieu Duponchelle
896c49cf49 isomp4/qtmux: accept video/x-h264, stream-format=avc3
The main difference between avc1 and avc3 is that avc3 is allowed
to contain in-band SPS / PPS. In practice decoders will always use
in-band parameter sets anyway, but it is cleaner to explicitly
advertise it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
2021-08-20 00:16:43 +00:00
Mathieu Duponchelle
fa835d686f isomp4/qtmux: make sure to switch to next chunk on new caps
For example, with single video sink pad, and new codec_data is
received, current_chunk_offset must be reset to -1 for the
aggregate loop to open a new chunk.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
2021-08-20 00:16:43 +00:00
Mathieu Duponchelle
e069824c7d isomp4/atoms: fix multiple stsd entries
stsd entries are serialized in reverse order (starting from
g_list_last()), and must be prepended to the entry list for their
index to be correct when referenced from stsc entries.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
2021-08-20 00:16:43 +00:00
Arun Raghavan
2c6be7373f matroska-mux: Add a timestamp-offset property
Adds a user-controllable timestamp offset to clusters and blocks. This
should be useful if we want to have timestamps that have significance
outside of the current file (for example, we might set the offset to the
wallclock when the file is being created, or some other common base, if
we want to correlate streams across multiple files).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1051>
2021-08-18 10:51:15 -04:00
Stéphane Cerveau
508a565163 matroska: demux: update stream_start_time
The stream_start_time can be less than the first detected.
In case of B-Frame based media, the first frame PTS might be
greater than the next one.

Need to keep the segment.start if a seek has been performed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1030>
2021-08-17 16:09:14 -04:00
Nicolas Dufresne
65deef0b0c mastrokademux: Remove redundant assignment
The segment.position is unconditionnaly set few lines below.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1030>
2021-08-17 16:08:33 -04:00
Víctor Manuel Jáquez Leal
d1cd310e42 videocrop: Fix icles tests.
Internally videcrop can call gst_video_crop_set_info() with NULL as in
caps. Then critical messages are raised when the in caps are
processed.

To fix this the in caps are checked, and if they are present, its
capsfeature is extracted, otherwise, the previous raw caps detection
remains as before.

Also the videocrop-test removes the format field in the structure
because now its always passed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1056>
2021-08-17 17:19:16 +00:00
Jakub Adam
286208576f rtp: Color Space header extension
Implements WebRTC header extension defined in
http://www.webrtc.org/experiments/rtp-hdrext/color-space.

It uses RTP header to communicate color space information and optionally
also metadata that is needed in order to properly render a high dynamic
range (HDR) video stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/853>
2021-08-17 15:28:19 +00:00
Per Förlin
9a216d0ffa rtspsrc: Add support to ignore x-server HEADER reply
When connecting to an RTSP server in tunnled mode (HTTP) the server
usually replies with a x-server header. This contains the address
of the intended streaming server. However some servers return an
"invalid" address. Here follows two examples when it might happen.

1. A server use Apache combined with a separate RTSP process to handle
   Https request on port 443. In this case Apache handle TLS and
   connects to the local RTSP server, which results in a local
   address 127.0.0.1 or ::1 in the x-server reply. This address is
   returned to the actual RTSP client in the x-server header.
   The client will receive this address and try to  connect to it
   and fail.

2. The client use a ipv6 link local address with a specified scope id
   fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via Http on port 80.
   The RTSP server receives the connection and returns the address
   in the x-server header. The client will receive this address and
   try to connect to it "as is" without the scope id and fail.

In the case of streaming data from RTSP servers like 1. and 2. it's
useful to have the option to simply ignore the x-server header reply
and continue using the original address.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1007>
2021-08-17 10:15:27 +00:00
Tulio Beloqui
9af6ce974a rtpjitterbuffer: fixed stall on gap when using rtx
Co-authored-by: Håvard Graff <havard@pexip.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1055>
2021-08-16 09:51:05 +00:00
Nirbheek Chauhan
620e9323c5 flv: use g_memdup2() as g_memdup() is deprecated
g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1052>
2021-08-16 08:00:53 +00:00
Víctor Manuel Jáquez Leal
862aa25e53 videocrop: Resurrect logging category.
Fix for a regression from commit 8f1384c9. That commit moved the debug
category definition, as static, into a gstvideocropelement.c, but that
category was used as default, in gstvideocrop.c, so it was never used
at logging, so the debug selector never showed the logs for
videocrop.

This patch move back the category definition into gstvideocrop.c and
leaving the function videocrop_element_init() as a noop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1049>
2021-08-11 16:09:06 +02:00
Víctor Manuel Jáquez Leal
9e1919c040 videocrop: Resurrect any caps feature negotiation.
Commit e31cbce4 brought a regression to negotiate featured caps. But
only by removing the entry in the caps template. This commit brings it
back.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1039>
2021-07-28 08:47:21 +00:00
Jan Schmidt
9499976fbb splitmuxsink: Fix some reference leaks in error cases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1023>
2021-07-26 17:56:42 +10:00
Jan Schmidt
b50d3b9c9f splitmuxsink: Prevent hang going back to NULL after failures
Prevent a condition where splitmuxsink won't go back to NULL state
after a child element fails to change state by making sure that
a READY->READY state change doesn't fail, and by returning
GST_FLOW_ERROR or GST_FLOW_FLUSHING upstream to shut down streaming
as quickly as possible.

This can happen after (for example) setting an invalid filename
on the sink element. In that case, the READY->PAUSED transition
fails, but with internal elements still in the NULL state. Trying
to set splitmuxsink back to NULL then ends up trying to bring
those NULL elements up to READY with a READY->READY transition,
(which fails, prevent splitmuxsink from getting to NULL)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1023>
2021-07-26 16:22:23 +10:00
Mathieu Duponchelle
2c85fd1be9 deinterlace: reduce noise when gst_pad_set_caps fails
It may be that downstream is simply flushing, in which case logging
an error is misleading.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1029>
2021-07-13 06:52:26 +00:00
Mathieu Duponchelle
a6d6e99f59 splitmuxsink: always use factory property when set
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1024>
2021-07-08 02:23:02 +02:00
Yacine Bandou
ce0be27caf qtdemux: No need for new "application/x-cbcs" caps
Instead of using the new "application/x-cbcs" caps, we are just adding
a new structure field "ciphe-mode", to indicate which encryption scheme
is used: "cenc", "cbcs", "cbc1" or "cens".

Similarly for the protection metadata, we add the "cipher-mode" field
to specify the encryption mode with which the buffers are encrypted.

"cenc": AES-CTR (no pattern)
"cbc1": AES-CBC (no pattern)
"cens": AES-CTR (pattern specified)
"cbcs": AES-CBC (pattern specified, using a constant IV)

Currently only "cenc" and "cbcs" are supported.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1013>
2021-07-06 12:12:24 +00:00
Jan Schmidt
d270fa498c matroskamux: Always write a tags element into seekhead
If there are only stream tags, we still want to write the
tags entry into the seekhead, so that tags can be found
quickly in the player.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/905

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1020>
2021-07-02 07:45:07 +00:00
Seungha Yang
adae01e4c3 qtmux: Don't need to update track per GstCaps if it's not changed
Skip GstQTMuxPad::set_caps() call for duplicated caps.
All the processing done in set_caps() method for duplicated caps
are redundant.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1019>
2021-07-02 06:22:41 +00:00
Sebastian Dröge
6e2924ff9c rtpssrcdemux: Remove pads and reset the element also in READY->NULL
Mostly for completeness.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1018>
2021-07-01 13:19:53 +03:00
Sebastian Dröge
c94469339a rtpptdemux: Remove pads also in PAUSED->READY
They're based on per-stream information and that should be reset
whenever going to READY state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1018>
2021-07-01 13:19:53 +03:00
Seungha Yang
e76218c1cb multiudpsink: Fix broken SO_SNDBUF get/set on Windows
SO_SNDBUF has been undefined on Windows because of missing WinSock2.h
include. And don't use native socket functions (e.g., setsockopt())
if code is expected to be built on Windows. We don't link ws2_32.lib
for this plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1016>
2021-06-28 15:32:51 +00:00
Olivier Crête
38e906de5d rtpmanager: Access GstRTPHdrExt fields through accessor
This way, the implementation can be private.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1017>
2021-06-24 14:57:14 -04:00
Jan Schmidt
34a6acede1 qtdemux: Refuse seeks in BYTES format
If downstream tries to seek in BYTES format, don't pass that through
to upstream. The byte positions downstream requests won't make any
sense in the muxed stream. There might be other formats we want to
pass through to upstream, but BYTES is not one of them. If we get a
seeking query about BYTES format, refuse that too.

This fixes a situation where we're playing a fragmented mp4 over http
and qtdemux refuses the initial seek (in TIME format), but then
h264parse/baseparse send a seek in BYTES format and everything falls
apart.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1014>
2021-06-22 18:05:56 +10:00
Nirbheek Chauhan
95f6c31c21 rtph265depay: update codec_data in caps regardless of format
Updating of codec_data in the caps is important to propagate changes
in sps/pps/vps via NALs. Without this, downstream does not renegotiate
when upstream changes resolution.

The comment referring to rtph264pay is from 2015 and is out of date.
rtph264pay stopped doing that in 2017 with commit
dabeed52a9

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1011>
2021-06-16 16:35:07 +05:30
Tim-Philipp Müller
21c90afd92 qtdemux: use g_memdup2() as g_memdup() is deprecated
- atom nodes/bytereader sizes are already checked
- palettes: are fixed/known size

g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.

Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>
2021-06-02 17:34:38 +01:00
Tim-Philipp Müller
05854f74c5 matroskademux: use g_memdup2() as g_memdup() is deprecated
- ebml-read: add some sanity checks when going from 64-bit
  to 32-bit length
- matroska-ids: codec_data_size has been checked via
  gst_ebml_read_binary(), is existing allocation.
- matroska-demux: alloc size is from existing allocations

g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.

Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>
2021-06-02 17:34:38 +01:00
Tim-Philipp Müller
aa4448cdd6 rtpjpegpay: fix image corruption when compiled with MSVC on Windows
On Windows with MSVC, jpeg_header_size would end up 2 bytes larger
than it should be. This then leads to the first 2 bytes of the
actual jpeg image data to be dropped, because we think those
belong to the header, which results in an undecodable image when
reconstructed in the depayloader.

What happens is that when the compiler evaluates

  jpeg_header_size = mem.offset + read_u16_and_inc_offset_by_2(&mem);

it actually uses the mem.offset value after it has been increased
by the function call on the right hand size of the equation.

From section 6.5 of the C99 spec:

  3. The grouping of operators and operands is indicated by the syntax [74].
     Except as specified later (for the function-call (), &&, ||, ?:, and
     comma operators), the order of evaluation of subexpressions and the
     order in which side effects take place are both unspecified.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/889

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/999>
2021-05-29 14:31:34 +01:00
Seungha Yang
80567ca939 deinterlace: Drop "field-order" field while transforming caps
Like other basetransform subclasses are doing, drop field
which can be converted by deinterlace.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>
2021-05-27 12:58:30 +00:00
Seungha Yang
9a8aea4a6a deinterlace: Drop field-order field if outputting progressive
Progressive with field-order doesn't make sense

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>
2021-05-27 12:58:30 +00:00
Havard Graff
26c94af2ea rtpssrcdemux: fix "data flow before segment event" crash
This crash could happen at any time a RTP and RTCP buffer arrived
simultaneously in ssrcdemux.

The problem was that sticky-event arriving while the rtp and rtcp pads
were being set up could arrive just too late to be included in the initial
forwarding.

The fix checks if the stickies have been sent on the srcpad about to be
pushed on, and if not sends them. It also blocks any stickes from
being forwarded *prior* to this happening, to avoid them arriving on
the srcpad multiple times.

Since the test loops 1000 times, this will make running under valgrind
take forever, so use the RUNNING_ON_VALGRIND variable to detect we
are running under valgrind, and reduce the loop-count to 2 in that case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
2021-05-25 22:04:41 +00:00
Havard Graff
de3a3882e9 rtpssrcdemux: refactor destruction of GstRtpSsrcDemuxPads
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
2021-05-25 22:04:41 +00:00
Havard Graff
c721c6fe72 rtpssrcdemux: make naming consistent
Use plural for GstRtpSsrcDemuxPads, since it contains two pads, and
use the variable-name 'dpads' everywhere.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
2021-05-25 22:04:41 +00:00
Tim-Philipp Müller
80966ed0a3 wavparse: use g_strndup() for copying text data
So we don't rely on NUL terminators inside the data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
2021-05-23 15:20:16 +01:00
Tim-Philipp Müller
5353ff355f wavparse: clean up adtl/note/labl chunk parsing
We were passing the size of the adtl chunk to the note/labl
sub-chunk parsing function, which means we may memdup lots of
data after the chunk string's NUL terminator that doesn't
really belong to it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
2021-05-23 15:19:41 +01:00
Tim-Philipp Müller
3dd8de1d7c wavparse: guard against overflow when comparing chunk sizes
Could be rewritten as lsize > (size - 8) a well, but the
extra check seems clearer. Doesn't look like it was problematic,
lsize wasn't actually used when parsing the sub-chunks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
2021-05-23 15:17:27 +01:00
Stéphane Cerveau
918d882021 matroskademux: fix decoder glitches with H264 content
To avoid decoder starvation causing glitches on screen,
the demuxer shall clip only when the buffer is a key frame
and the lace time is greater than the stop time.

Fixes gst-editing-services#128

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/973>
2021-05-20 15:07:07 +02:00
Nicolas Dufresne
d877f7f816 matroskademux: Advertise codec-alpha in caps
This will be used to select the appropriate decoders. We also only attach the
GstVideoCodecAlphaMeta if the AlphaMode element is set, this is to stay on the
safe side and mimic what browsers (verified in Firefox and Chromium code) do.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
2021-05-11 16:52:22 -04:00
Nicolas Dufresne
b2e857efc6 matroskademux: Store alpha stream in VideoCodecAlphaMeta
This generalize the feature over using mini object quark data. If
that feature was Matroska specifc, using the new CustomMeta would have
been enough and arguably cleaner then QData, though it seems that
similar technique is use with AV1 Image Format (AVIF).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
2021-05-11 16:06:44 -04:00
Tim-Philipp Müller
b84bad6ac3 matroska-demux: extract VP8 alpha from BlockAdditionals
And put it on buffers as qdata (which is easier in this
case than a private custom meta because it can be picked
up easily in other modules).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
2021-05-11 16:06:44 -04:00
Jan Alexander Steffens (heftig)
0ff50d6723 udpsrc: Plug leaks of saddr in error cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>
2021-05-07 10:09:38 +00:00
Jan Alexander Steffens (heftig)
e425bcada5 udpsrc: Whitespace
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>
2021-05-07 10:09:38 +00:00
Jan Alexander Steffens (heftig)
fa1cc0a81f deinterlace: Plug a method subobject leak
Changing the method would leak the previous method.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/976>
2021-05-07 09:31:48 +00:00
David Fernandez
056f8ce6ca matroska-mux: Change accepted caps width and height from [16, MAX] to [1, MAX]
There are cases where the video size might be less than 16x16.
This change allows the Matroska muxer to accept this cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/539>
2021-05-05 16:31:33 -04:00
François Laignel
39f0905a7e Use gst_element_request_pad_simple
Instead of the deprecated gst_element_get_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/958>
2021-05-05 06:17:20 +00:00
Jan Schmidt
7c5f2185a9 qtmux: Make sure to write 64-bit STCO table when needed.
qtmux attempts to choose between writing a 32-bit stco chunk offset table
when it can, but switch to a 64-bit co64 table when file offsets go over
4GB.

This patch fixes a problem where the atom handling code was checking
mdat-relative offsets instead of the final file offset (computed by
adding the mdat position plus the mdat-relative offset) - leading to
problems where files with a size between 4GB and 4GB+offset-of-the-mdat
would write incorrect STCO tables with some samples having truncated
32-bit offsets.

Smaller files write STCO correctly, larger files would switch to
co64 and also output correctly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/970>
2021-04-30 08:12:47 +10:00
Guillaume Desmottes
5fa3325335 rtpopuspay: set MARKER flag
Set MARKER flag on first buffer after DTX.

According to RFC 3551 section 4.1 the marker bit needs to be set on
"the first packet after a silence period during which packets have
not been transmitted contiguously".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>
2021-04-26 15:25:56 +02:00
Guillaume Desmottes
41ba8c1b00 rtpopuspay: add DTX support
If enabled, the payloader won't transmit empty frames.

Can be tested using:
  opusenc dtx=true bitrate-type=vbr ! rtpopuspay dtx=true

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>
2021-04-26 15:25:56 +02:00