Commit graph

411 commits

Author SHA1 Message Date
Hosang Lee
041e0c6cab qtdemux: Fix reverse playback for pcm audio stream
Some raw lpcm or adpcm may have larger sample sizes than the max
buffer size value set.
Trimming the buffer causes bogus size error on reverse playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5742>
2023-12-01 15:11:04 +09:00
Robin Gustavsson
38a8411bdf rtpklvdepay: Recover after invalid fragmented KLV unit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4816>
2023-11-17 09:01:10 +00:00
Sebastian Dröge
db77deef00 rtpjitterbuffer: Add new "rfc7273-reference-timestamp-meta-only" property
If this property is enabled then the jitterbuffer will do the normal PTS
calculations according to the configured mode instead of making use of
the RFC7273 media clock.

The timestamp calculated from the RFC7273 media clock will only be
stored in the reference timestamp meta, if addition of that meta is enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
eae3ef7461 rtpjitterbuffer: Add new rfc7273-use-system-clock property
When this property is used, it is assumed that the system clock is
synced close enough to the media clock used by an RFC7273 stream.

As long as both clocks are at most a few seconds from each other this
will give the correct results and avoids having to create an actual
network clock that has to sync first.

If the system clock is actually synchronized to the media clock then
everything will behave exactly the same, otherwise the reference
timestamp meta will be correct but the buffer timestamps will be off by
the difference between the two clocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
2956ba48fc rtpjitterbuffer: Improve handling of media clocks
Do more checks for clock equality than just checking pointers. The same
NTP/PTP clock might be used as pipeline clock but a new instance, so
instead also check what clock they are synced to.

Also handling setting / resetting of the media clock and pipeline clock
correctly by resetting the media clock's state accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Piotr Brzeziński
4037334143 qtdemux: Ignore raw audio streams when adjusting seek
Because we treat raw audio chunks/samples as keyframes, they were interfering
with seek time adjustment.
Became apparent when the accompanying video stream was I-frame only,
for example ProRes.
Since raw audio streams can be seeked freely, it's fine to just ignore them here,
giving priority to the real keyframes in the video stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4946>
2023-11-15 07:55:27 +00:00
Dongyun Seo
8db184085a dcaparse: keep upstream buffer meta
Some audio decoders cannot decode DTS stream if there is no
valid timestamp. So, keep upstream buffer meta.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5655>
2023-11-14 16:51:44 +09:00
Olivier Crête
c2a357c867 rtpopusdepay: set resync flag
- Set re-sync flag on output buffer when rtp had the marker flag set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5529>
2023-11-10 21:45:13 +00:00
Johan Adam Nilsson
808c27b4cc wavparse: fix buffer leak with adtl tag
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3020
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5595>
2023-11-03 19:38:38 +00:00
Tim-Philipp Müller
bce1d121ba rtpac3depay: should output audio/x-ac3 not audio/ac3
audio/x-ac3 is the canonical media format in GStreamer.
audio/ac3 is sometimes accepted as input (e.g. in rtpac3pay
or ac3parse), but shouldn't be output.

Fixes #3038.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5472>
2023-10-19 13:27:58 +00:00
Stéphane Cerveau
7c7a90b99d imagesequencesrc: fix regular image deadlock
With one regular image file path provided (without %05d),
the element was stuck in a dead loop counting the frames:

gst_image_sequence_src_count_frames

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Fixes #2843

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

The property is also exposed on rtpbin and sdpdemux

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Sponsored by Living Optics

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

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

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

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

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4792>
2023-06-07 20:30:36 +00:00