Commit graph

258 commits

Author SHA1 Message Date
Haihua Hu
4e381b8901 playback: Support runtime change connection-speed of adaptivedemux(2)
Update connection-speed at runtime in playbin, uridecodebin and decodebin
also do the same thing in urisourcebin.

With contributions from Philippe Normand <philn@igalia.com> (build fixes and
rebase on mono-repo).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4713>
2023-10-12 16:06:42 +00:00
Guillaume Desmottes
8004b1650a videorate: log when rolling back previous caps
We were logging when restoring the current caps but not when it was
changed, making logs quite confusing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5433>
2023-10-04 14:19:37 +00:00
Arun Raghavan
9e137ea6a4 gio: Drop some trailing whitespace in giobasesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5372>
2023-10-04 12:56:03 +00:00
Arun Raghavan
ca337002f1 giostreamsink: Add a property to close stream on stop()
Back in the mists of time[1], we switched `giostream*` elements to not close the
stream on stop() so that applications that needed a handle to the stream after
the element stopped had it.

Unfortunately, we also have cases[2] where waiting for the element to be
finalized is too late for the stream to be closed.

In order to not change the behaviour of the element, we add a property to allow
users to select the desired behaviour.

[1]: https://bugzilla.gnome.org/show_bug.cgi?id=587896
[2]: gst-plugins-rs#423

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5372>
2023-10-04 12:56:03 +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
Aleksandr Slobodeniuk
3901984621 videotestsrc: fix max value for timestamp-offset
Compiled for x64 with msvc the timestamp-offset property
max limit is 2147483646999999999 that is smaller then
the timestamps provided by the rtspsrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3771>
2023-09-28 17:32:36 +00:00
Philippe Normand
9dbe8a1e36 videoconvertscale: Expose converter config as new property
This allows the user to have full control on the conversion parameters. If set,
the property takes precedence over the other similar conversion tweaking properties.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2263>
2023-09-28 15:02:22 +02:00
Guillaume Desmottes
0ae230c8be uridecodebin3: proxy urisourcebin::download-dir property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5227>
2023-08-23 13:52:19 +00:00
Guillaume Desmottes
8263ce2a31 urisourcebin: add 'download-dir' property
The directory were buffers are downloaded was not documented and not
configurable. Users may want to ensure buffers are saved to a specific
partition for example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5227>
2023-08-23 13:52:19 +00:00
Jan Schmidt
ccfbdcad90 videoconvertscale: Don't passthrough with dither or alpha settings
If the configured properties request dithering/quantization be applied
or alpha be set/multiplied then don't do passthrough, even if the
caps are the same.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5183>
2023-08-16 07:38:21 +00:00
Roman Lebedev
8b1500d7ff volume: support arbitrarily-large positive gains
The current limit is `x10`, which allows just `+20 dB` of gain.

While it may seem sufficient, this came up as a problem
in a real-world, non-specially-engineered situation,
in strawberry's EBU R 128 loudness normalization.
(https://github.com/strawberrymusicplayer/strawberry/pull/1216)

There is an audio track (that was not intentionally engineered that way),
that has integrated loudness of `-38 LUFS`,
and if we want to normalize it's loudness to e.g. `-16 LUFS`,
which is a very reasonable thing to do,
we need to apply gain of `+22 dB`,
which is larger than `+20 dB`, and we fail...

I think it should allow at least `+96 dB` of gain,
and therefore should be at `10^(96/20) ~= 63096`.

But, i don't see why we need to put any specific restriction
on that parameter in the first place, other than the fact
that the fixed-point multiplication scheme does not support volume
larger than 15x-ish.

So let's just implement a floating-point fall-back path
that does not involve fixed-point multiplication
and lift the restriction altogether?

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5063>
2023-08-07 13:17:45 +00:00
Philippe Normand
1afeef0e8b decodebin3: Ensure the slot is unlinked before linking to decoder
When switching from a raw stream to an encoded stream we need to make sure the
slot is unlinked, there is code in place for this but it wasn't triggered
because the slot being reconfigured wasn't advertised as linked beforehand.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5126>
2023-08-01 20:16:59 +00:00
Philippe Normand
0d5f6f3d47 decodebin3: Prevent a critical warning when reassigning output slots
Do not attempt to send a streams-selected message when reassigning
an output slot in case upstream signalled that it is handling stream selection.
In this case decodebin3 doesn't keep track of stream
collections (`dbin->collection` is NULL).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5059>
2023-07-19 15:17:46 +00:00
Philippe Normand
4404e7203c decodebin3: Fix slot input linking when the associated stream has changed
Setting the input field on the empty slot prevents future linking of it and will
result in flow errors later on.

This was observed in WebKit's MediaStream source element, when it changes the
caps on one of its associated streams, from an encoded format to a raw video
format. The associated stream-id on the sticky stream-start event doesn´t
change, but the element creates a new GstStream with a different ID and sets it
on the stream-start event. Stream parsing is disabled in urisourcebin, so
decodebin3 handles the parsing. Without this patch we would end-up with unlinked
pads in decodebin3 after switching to the raw video format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5048>
2023-07-19 01:24:30 +00:00
Edward Hervey
7e7f02f4f4 decodebin3: Rename and refactor function
It was doing a bit more than it did initially, update the name accordingly.

Refactor slightly for visibility

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5052>
2023-07-18 11:42:13 +02:00
Edward Hervey
1fd7c2c17a decodebin3: Remove dead code
Was never used since initial commit

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5052>
2023-07-18 11:42:13 +02:00
Philippe Normand
4dc503e1e4 decodebin3: Remove spurious input locking during parsebin reconfiguration
Commit 22917b140f added extra locks in
`reset_input_parsebin()` but all call sites of that function already take the
input lock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5022>
2023-07-12 18:33:00 +00:00
Matthew Waters
cae434c6ff videorate: properly handle variable framerate input and drop-only=true
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4969>
2023-07-05 19:33:59 +00:00
Guillaume Desmottes
1027180960 subtitleoverlay: fix mutex error if sink caps is not video
We were trying to unlock a mutex that was not locked.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4964>
2023-07-05 10:34:21 +00:00
Edward Hervey
f825b7aba3 uridecodebin3: Refuse sub uri in gapless mode
This is too problematic to handle properly right now.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2550 and
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2605

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4958>
2023-07-03 16:02:40 +02:00
Hou Qi
dbdbf2d256 decodebin3: fix memory leak when remove candidate decoder
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4955>
2023-07-03 07:13:13 +00:00
Thibault Saunier
c5304751ab uridecodebin: Handle non dynamic sources with several source pads
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4881>
2023-06-30 01:00:34 +00:00
Thibault Saunier
e7f13ede0f videoconvertscale: Remove the restriction on ANY memory
Our pad templates already expose ANY feature and the code supports that
case even if only for the passthrough, we should not disable that feature.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4733>
2023-06-27 08:17:33 +00:00
Carlos Rafael Giani
8c5a8f4466 dsd: Add code for DSD audio support
Related to:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/972

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3901>
2023-06-23 01:27:03 +00:00
Sebastian Dröge
069065adc4 subparse: Skip after the end of a valid closing tag instead of only skipping <
This is a small optimization and avoids restarting the next parsing
iteration on already accepted data.

On its own it would also fix ZDI-CAN-20968 (see previous commit) but the
previous commit independently is also a valid fix for it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4895>
2023-06-20 09:06:44 +01:00
Sebastian Dröge
97c6d7495e subparse: Look for the closing > of a tag after the opening <
Previously when fixing up subrip markip, we were looking from the start
of the remaining buffer instead. Due to how skipping over closing tags
works, the remaining buffer will still contain the closing `>` of the
previous tag so if a unexpected closing tag is found after another
closing tag, we would potentially do an out of bounds memmove().

Fixes ZDI-CAN-20968
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2662

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4895>
2023-06-20 09:06:38 +01:00
Marek Vasut
5ad834ce28 videotestsrc: Support video/x-bayer 10/12/14/16 bit depths
Add support for generation of 10/12/14/16 bit bayer test 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 ! \
    video/x-bayer,width=512,height=512,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
e569b8ba1e videotestsrc: Simplify ARGB to Bayer conversion
Simplify the conversion to bayer pattern as suggested by Nicolas Dufresne.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Marek Vasut
d0efb05f52 videotestsrc: Move video/x-bayer caps parsing in one place
Move all the video/x-bayer caps parsing into one place,
gst_video_test_src_parse_caps(), no functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
2023-06-15 08:26:12 +00:00
Xabier Rodriguez Calvar
4769a2ab97 playbin2: improve transference doc of get-*-pad actions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2081>
2023-06-12 10:26:26 +00:00
Hou Qi
95ac8b0cea decodebin3: filter error message and store latency message for candidate decoder
If the GST_MESSAGE_SRC of error message belongs to candidate decoders,
filter the error message and don't forward it as there might be a
following candidate decoder that can be used.

If the GST_MESSAGE_SRC of error message belongs to candidate decoders,
store the latency message and handle it after decoder is accepted.
This is to avoid the selection lock failure if decodebin3 needs to
handle latency message for candidate decoders when sending sticky event.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
2023-06-02 14:51:38 +00:00
Hou Qi
887ae4d9e0 decodebin3: try candidate decoders to select the first one that works
Send sticky events to the new created decoder after it switches
to PAUSED state. It it fails, just skip this decoder and try the
next one until finding one that works. Otherwise remove this
failing stream after trying all decoders and no one can work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
2023-06-02 14:51:38 +00:00
Hou Qi
6fc6e934aa decodebin3: send sticky event to decoder after setting it to PAUSED
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
2023-06-02 14:51:38 +00:00
Hou Qi
837169a221 decodebin3: add function remove_decoder_link()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
2023-06-02 14:51:38 +00:00
Hou Qi
536c344111 decodebin3: copy sticky event
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
2023-06-02 14:51:38 +00:00
Edward Hervey
9befb81036 urisourcebin: Set source element to READY before querying it
Generating the source element is done when urisourcebin is doing the READY to
PAUSED state change, so it is reasonable to set the new source element to that
state.

This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).

Finally it makes more sense to have an element in READY when attempting to query
information from it (such as SCHEDULING queries or probing live-ness).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3856>
2023-06-02 07:00:43 +00:00
Guillaume Desmottes
b3d27da397 streamsynchronizer: check reset-time when handling FLUSH_STOP
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4749>
2023-06-01 09:45:50 +02:00
Guillaume Desmottes
c2d8f4f729 streamsynchronizer: reset eos on STREAM_START
self->eos was never reset after streamsynchronizer has sent EOS
(except on explicit flush or switching back to PAUSED).
As a result, synchronization was broken if new streams were pushed later
as gst_stream_synchronizer_wait() does not wait if self->eos is set.

Fix this by reseting self->eos on STREAM_START as that means a new
stream is being sent upstream and so a new EOS will follow later on.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4749>
2023-06-01 09:45:15 +02:00
Edward Hervey
d6f1c517f3 decodebin3: Handle changes in stream type
While decodebin3 could handle changes in inputs (ex: changing codecs), there was
still one limitation which was when changing between sources which had
non-intersecting stream types (ex: switching from a video-only source to a
audio-only source). While the decoder *could* change to the proper codec ... it
would carry on using a `DecodebinOutputStream` associated to that stream
type (and therefore with pads with the wrong name).

In order to handle this:

* We notify the `MultiQueueSlot` of the change in `GstStreamType` if it already
  had an associated inputstream (ex: the one associated with the static sink
  pad)

* We detect such changes on the output of multiqueue as soon as
  possible (i.e. when we get the GST_EVENT_STREAM_START for the new stream type)
  by discarding the associated output.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1669

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4703>
2023-05-25 21:23:21 +00:00
Edward Hervey
f51283b57b uridecodebin3: Also re-use decodebin3 static sink pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4703>
2023-05-25 21:23:21 +00:00
Guillaume Desmottes
58f80c180c uridecodebin3: notify when 'current-(sub)uri' properties are updated
Assume both uri and suburi are changed when the main item changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4543>
2023-05-24 09:28:58 +00:00
Thibault Saunier
7eca75130d encodebin: Plug a parser before timestamper
Timestamper might not support all stream formats so we need to make sure
some element is able to convert between those formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4122>
2023-05-22 15:10:52 +00:00
Haihua Hu
9f67b866b9 decodebin3: avoid identity, sinkpad, parsebin leakage when reset input
when reset_input, need remove identity/parsebin from decodebin3
when release_pad, need call free or reset input if collection
didn't change

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4664>
2023-05-19 16:24:08 +00:00
Haihua Hu
36333a5152 decodebin3: fix random hang when remove failing stream
When reconfigure_output_stream entry missing decoder path,
requested_selection should been update with what is really
active/selected immdiately with SELECTION_LOCK hold. So
use an optional message return from reconfigure_output_stream
and post it after release SELECTION_LOCK. This can make sure
other thread call to check_slot_reconfiguration will got
a correct requested_selection.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4599>
2023-05-16 12:43:12 +00:00
Edward Hervey
0db6149880 decodebin3: Handle streams without CAPS or TIME segment
decodebin3 will do its best to figure out whether a parsebin is required to
process the incoming stream.

The problem is that for push-based stream it could happen that the stream would
not provide any caps, resulting in nothing being linked internally.

Furthermore, there is the possibility that a stream *with* caps would not be
using a TIME segment, which is required for multiqueue to properly work.

In order to fix those two issues, we force the usage of parsebin on push-based
streams:
* When the pad is linked, if upstream can't provide any caps
* When we get a non-TIME segment

Fixes #2521

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4492>
2023-04-27 12:50:21 +00:00
Hou Qi
aed4d31e67 decodebin3: fix segment fault when print decoder log
Segment fault happens when cannot find decoder but try to print
decoder name. Need to check the decoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4461>
2023-04-25 21:42:15 +08:00
Sebastian Dröge
6c429a5891 audiotestsrc: Initialize all samples in wave=ticks mode
Previously samples were only initialized in 2 out of 3 cases.

Probably fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/337

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4475>
2023-04-22 08:44:02 +00:00
Patricia Muscalu
c6bb20bba8 playsink: Fix volume leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4459>
2023-04-20 14:21:15 +00:00
Mathieu Duponchelle
be208b9f50 docs: mark GstVideoConvertScale as plugin API
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4408>
2023-04-13 21:46:59 +00:00
Ilie Halip
6846810f50 typefindfunctions: Increase xml typefinder closing brace limit
If the first XML element in a DASH manifest has its closing brance
beyond the first 512 bytes (because of, e.g. lots of attributes),
the MPD typefinder fails. Try to read a larger block, and then
smaller blocks until 512 bytes.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2385

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4241>
2023-03-29 17:50:20 +00:00