Commit graph

303 commits

Author SHA1 Message Date
Jan Alexander Steffens (heftig)
5bac956b58 tsmux: Add missing include
We need `GstMpegtsPMTStream` here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5496>
2023-10-19 22:34:28 +00:00
Jan Alexander Steffens (heftig)
db88612853 tsmux: Simplify tsmux_section_write_packet
- Don't try to make the parameters match `GHFunc`. Use a dedicated
  callback for `g_hash_table_foreach`.
- Don't try to be clever with buffer memories. We're allocating a full
  packet anyway, might as well memcpy and save on a lot of complexity.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5496>
2023-10-19 22:34:28 +00:00
Jan Alexander Steffens (heftig)
950789f61b tsmux: tsmux_packet_out should always consume its buffer
Consuming the buffer only when successful is an antipattern and easily
leads to leaks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5496>
2023-10-19 22:34:28 +00:00
Jan Alexander Steffens (heftig)
55658ad166 tsmux: Don't memset in pad_stream when writing a PCR packet
tsmux_write_ts_header will write a stuffing adaptation field, so we
don't need to prefill the buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5496>
2023-10-19 22:34:28 +00:00
Jan Alexander Steffens (heftig)
2dbd89b036 tsmux: Move out parameters of tsmux_write_ts_header
Move them to the end of the parameter list and also allow passing NULLs
to ignore the payload information, but assert that the payload length is
zero in this case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5496>
2023-10-19 22:34:27 +00:00
Jan Alexander Steffens (heftig)
5128975a36 tsmux: Fix two more uses of gst_buffer_map
The buffers should be used for writing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5496>
2023-10-19 22:34:27 +00:00
Diego Nieto
e290555367 mpegtsdemux: Fix comment about the jitter description
According to the information provided below, the Jitter (J) is
defined by a network delay (D) + a noise(i)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5494>
2023-10-17 08:22:41 +00:00
Jan Alexander Steffens (heftig)
8a7d0efd96 tsmux: Fix error handling in pad_stream
Don't leak the map or the buffer if we encounter an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5490>
2023-10-16 15:46:00 +02:00
Jan Alexander Steffens (heftig)
b1810d83bc tsmux: Fill padding packets with stuffing bytes
Instead of leaving it uncleared, emitting probably old packet data but
potentially also random or sensitive application data.

Also fix the mapping mode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5490>
2023-10-16 15:46:00 +02:00
Guillaume Desmottes
dd0896f05a audiobuffersplit: disable max-silence-time if set to 0
According to the property documentation max-silence-time is supposed to be
disabled when set to 0 but it was not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5442>
2023-10-06 14:48:46 +02:00
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
Jonas K Danielsson
f4fb4a5606 ristsrc: Add support for dynamic payload
This commit ports functionality from the `rtpsrc` to make the `ristsrc`
work with dynamic payload types.

It adds two properties:
  - `caps`
  - `encoding-name`

These can be used to make the `ristsrc` receive other payload types than
the MPEG TS one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5422>
2023-10-03 20:17:07 +00:00
Philippe Normand
886bd7e4e0 interaudiosink: Ensure adapters don't store buffers with audio meta
The interaudiosrc might take buffers of different sizes from the audio adapter,
so keeping metas consistency would be an issue. So the sink now strips the audio
metas away and the src adds them back (for non-interleaved layouts only) when
taking buffers from the adapter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5324>
2023-09-28 10:26:33 +00:00
Philippe Normand
46dbe2a372 interaudiosrc: Add audio meta to buffers containing non-interleaved samples
Without this a downstream audioconverter wouldn't be able to map the
GstAudioBuffer prior to conversion.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5324>
2023-09-28 10:26:33 +00:00
Sebastian Dröge
72742dee30 mxfdemux: Check number of channels for AES3 audio
Only up to 8 channels are allowed and using a higher number would cause
integer overflows when copying the data, and lead to out of bound
writes.

Also check that each buffer is at least 4 bytes long to avoid another
overflow.

Fixes ZDI-CAN-21661, CVE-2023-40475

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2897

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5362>
2023-09-20 15:40:07 +00:00
Sebastian Dröge
ce17e968e4 mxfdemux: Fix integer overflow causing out of bounds writes when handling invalid uncompressed video
Check ahead of time when parsing the track information whether
width, height and bpp are valid and usable without overflows.

Fixes ZDI-CAN-21660, CVE-2023-40474

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2896

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5362>
2023-09-20 15:40:07 +00:00
Sebastian Dröge
889a3fe932 rtmp2: Set default flash version to NULL
This is consistent with the librtmp-based old rtmp plugin and ffmpeg.
While some servers require a valid flash-version, others are failing
with a too long or any flash-version at all.

By changing to the same default as in the old plugin and in ffmpeg,
GStreamer will at least behave the same and will work and fail with the
same servers without setting a flash-version.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5293>
2023-09-11 08:23:33 +00:00
Seungha Yang
efe35a3f6c h264parse, h265parse: Fix potential integer overflow
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2961
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5295>
2023-09-08 12:45:12 +00:00
Seungha Yang
2c4cb82afc h264parse, h265parse: Fix timecode parsing
The scaling factor for nFrame part should be "(1 + nuit_field_based_flag) / 2"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5254>
2023-09-01 06:44:52 +00:00
Jan Alexander Steffens (heftig)
14c097e87d rtmp2: Allow NULL flash version, omitting the field
rtmpsink omits it by default. Allow us to do the same.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5241>
2023-08-25 09:50:39 +02:00
Guillaume Desmottes
17ebfc7cb7 rtmp2sink: fix crash if message conversion failed
The message pointer is not set so we can't display it in logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5197>
2023-08-18 09:27:36 +02:00
Jan Schmidt
8b5833c546 audiolatency: Fix event refcounting bug handling latency events
Fix a refcounting bug introduced in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5146
If upstream returns FALSE when processing a latency event, it will
be unreffed an extra time

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5150>
2023-08-06 08:00:56 +00:00
Jan Schmidt
fd95f5682e audiolatency: Forward latency query and event upstream
Make sure the pipeline still configures the latency that it would configure
if audiolatency was not in the pipeline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5146>
2023-08-04 14:13:12 +00:00
Guillaume Desmottes
501e53b033 rtmp2src: add 'no-eof-is-error' property
There is currently no way for applications to know if the stream has
been properly terminated by the server or if the network connection
was disconnected as EOS is sent in both cases.

Adding a property so connection errors can be reported as errors
allowing applications to distinguish between both scenarios.

Fix #2828

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5115>
2023-08-04 11:07:36 +00:00
Philippe Normand
c506748c6f transcodebin: Fixes for upstream selectable support
The upstream selectable query was not performed in all situations where we
handle the stream-start event. This could potentially lead to unlinked pads
between decodebin3 and encodebin later on.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5089>
2023-08-02 17:12:48 +00:00
Seungha Yang
3679713519 rtponviftimestamp: Fix drop-out-of-segment=false mode
Fixing unexpected buffer dropping and flow error in case that:
* use-reference-timestamps=false
* drop-out-of-segment=false
* Calculated utc offset is not valid because buffer is out-of-segment

The above case should be considered as a valid data flow without returning
errors.

Fixing regression introduced by
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1683

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5116>
2023-07-28 23:36:34 +09:00
He Junyan
4e47a73ddf fakevideosink: Add DMA_DRM format into sink template caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5094>
2023-07-25 19:34:58 +00:00
Fabian Orccon
dd47fa53d8 h265parse: Parse SEI unregistered user data
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5070>
2023-07-25 18:28:26 +00:00
Jakub Adam
f7b719ae91 av1parse: calculate framerate from AV1 timing info
When framerate info isn't provided by upstream elements, try to extract
it from AV1 timing info, if present.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5041>
2023-07-19 20:45:05 +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
Stéphane Cerveau
2974c18a5c codecparsers: keep naming consistency in GST_H264_LEVEL
GST_H264_LEVEL_2 should be used instead of GST_H264_LEVEL_2_0

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4737>
2023-06-26 10:47:36 +00:00
Sebastian Dröge
0dabf0eb00 dvdspu: Avoid integer overflow when checking if enough data is available
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4896>
2023-06-20 15:36:03 +00:00
Sebastian Dröge
7ed446dca9 dvdspu: Make sure enough data is allocated for the available data
If the size read from the stream is smaller than the currently available
data then the size is bogus and the data should simply be discarded.

Fixes ZDI-CAN-20994
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2660

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4896>
2023-06-20 15:36:03 +00:00
Michael Olbrich
3f24a38c8e sdpdemux: ensure that only one srcpad is created per stream
If two senders use the same multicast IP and port then new_session_pad()
may try to add a srcpad to the same stream twice.

stream->srcpad is updated but gst_element_add_pad() fails the second
time. As a result stream->srcpad points to a deleted object and
access in gst_sdp_demux_stream_free() fails with a segfault.

Just ignore the second pad. Nothing useful can be done with it anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4603>
2023-06-15 23:35:30 +00:00
Aaron Boxer
a72ca72a27 mpegtsmux: add stream-number property on GstBaseTsMuxPad
This new property allows setting of PES stream number for AAC audio
and AVC video streams.

The stream number is subject to the following constraints:

1. it must be between 0 and 15 for video
2. it must be between 0 and 31 for audio

Currently the PES stream number is hard-coded to zero for these
stream types.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4822>
2023-06-15 10:34:25 +00:00
Marek Vasut
4c92d4096e bayer2rgb: Support video/x-bayer 10/12/14/16 bit depths
Add support for 10/12/14/16 bit depths . This consists of multiple parts.
First is the parsing of caps, which pulls out the bitness and endianness
from the video/x-bayer format.

Second, gst_bayer2rgb_split_and_upsample_horiz() is split into two similar
functions, one for 8bit bayer handling and another for 16bit bayer handling.
The content is basically identical, except one uses 8bpp and the other 16bpp
inputs and outputs, and they each use different ORC code to match. The 16bpp
variant also handles endian swapping. There is now a wrapper called
gst_bayer2rgb_split_and_upsample_horiz() which selects the correct function
based on bpp from the parser.

Third, gst_bayer2rgb_process() is extended to handle both 8bit and 16bit
bayer data. Yet again there are matching ORC functions to handle the 16bit
data. This time however the 16bit handling of data is slightly special. The
ORC is not able to emit opcodes for 'x2 mergelq', so the trick here is to
store the BG and GR longs into separate 'dtmp' temporary buffer, and then
do one more ORC post-processing step, compensate for the less-than-16bpp
bitness using left shift, and reorder them into the destination frame
using 'mergelq' .

Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
    video/x-bayer,width=512,height=512,format=bggr16le ! \
    bayer2rgb ! \
    video/x-raw,format=RGBA64_LE ! \
    videoconvert ! \
    autovideosink
```

Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
    video/x-raw,width=512,height=512,format=ARGB ! \
    rgb2bayer ! \
    video/x-bayer,format=bggr12le ! \
    bayer2rgb ! \
    video/x-raw,format=RGBA64_LE ! \
    videoconvert ! \
    autovideosink
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
9d1a750117 bayer2rgb: Add comments explaining gst_bayer2rgb_process()
Add comments regarding which LINE()s point to which data in the
temporary buffer and a large comment explaining how the buffer
is processed. This will hopefully be useful to someone, as the
code is not obvious. No functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
920851945f bayer2rgb: Add comment on bayer_orc_horiz_upsample
Explain how the bayer_orc_horiz_upsample function works and
what it does with the pixels, as this may not be obvious.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
4efe11a705 bayer2rgb: Pass filter pointer into gst_bayer2rgb_split_and_upsample_horiz()
Instead of passing a single element of GstBayer2RGB structure into the
gst_bayer2rgb_split_and_upsample_horiz(), pass the entire pointer and
let the funciton pick out whatever it needs out of the structure. This
is a preparatory patch. No functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
fbd02b3e2a bayer2rgb: Pass all parameters to LINE() macro
Pass all three parameters used by the LINE() macro to the LINE() macro
and unroll the code for readability. Add more comments regarding which
of these LINE()s point to which data in the temporary buffer to make
the code less confusing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
8bec6828f4 bayer2rgb: Fold src_stride into gst_bayer2rgb_process()
The source stride parameter can be easily obtained from GstBayer2RGB
structure, do it within gst_bayer2rgb_process() and drop the parameter.
No functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
ddcb45ffc0 bayer2rgb: Inline the j=0 value
The j variable is used as an iterator further down in this code, but
here it can be just inlined in the macro parameters to make the code
easier to read. This is done in preparation for further changes. No
functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
484e31c1d9 bayer2rgb: Disable in-place transform
The bayer2rgb process implemented doesn't support in-place tranform.
This element doesn't implement a "transform_ip" vmethod of
GstBaseTransform it will revert to using the "tranform" vmethod.
It's misleading to set it to TRUE, here. Change this to FALSE.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
0763fb107d rgb2bayer: Support video/x-bayer 10/12/14/16 bit depths
Add support for conversion to 10/12/14/16 bit bayer pattern.
The implementation is rather simplistic, just take the ARGB
input, generate 16-bit data out of it instead of 8-bit, shift
them as required by the output bitness, and apply endian swap.

Example usage:
```
$ gst-launch-1.0 videotestsrc num-buffers=1 ! \
    video/x-raw,width=512,height=512,format=ARGB ! \
    rgb2bayer ! \
    video/x-bayer,format=bggr12le ! \
    filesink location=/tmp/bayer12.raw
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Tim-Philipp Müller
a9c5e5e239 asfmux: fix potentially unaligned write on 32-bit ARM
Fixes #2665

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4842>
2023-06-14 04:59:05 +00:00
Nicolas Dufresne
4402a8044f fakesinks: Fix recursive notify loop
The proxy callback for the notify::last-message was emiting the signal
again on the child, which caused an infinit loop. We could swap the child
and the user data to signal to the bin instead, but it was found that proxying
this signal was not very useful. Typical use case it to set silent=0 and use
deep-notify feature. Proxying that signal just duplicate that output which
isn't very useful.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4766>
2023-06-13 23:33:08 +00:00
Jan Alexander Steffens (heftig)
6e9d67bbc1 mpegtsmux: Use terminological ISO 639-2 language codes
These are preferred in most circumstances.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2649
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4795>
2023-06-12 08:51:10 +00:00
Vivia Nikolaidou
0e62bb2ba6 basetsmux: Fix language crash when ts_pad->stream is NULL
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4785>
2023-06-07 16:58:38 +00:00
Vivia Nikolaidou
0a331402d6 tsdemux: Detect language from ac3 descriptor
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4709>
2023-06-07 13:04:03 +00:00
Vivia Nikolaidou
395e0c3925 tsmux: Resend PMT whenever the language changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4711>
2023-06-01 17:05:11 +00:00