Commit graph

7878 commits

Author SHA1 Message Date
Matthew Waters
3d9e705621 videoaggregator: allow selecting an alpha output from non-alpha inputs
e.g. if we have:

video-x/raw,format=I420 ! compositor ! video/x-raw,format=BGRA

This will currently produce a warning as the alpha-ness of the chosen
'best' format (I420) will be different from the value restricted by the
downstream caps filter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1059>
2021-03-31 01:55:17 +00:00
Matthew Waters
eb06907fb4 gl/wayland: provide a dummy global_remove function
Even if we don't care about any global objects being removed, wayland
will still error if globals are removed without a corresponding listener
set up for them.  e.g. wl_output hotplugging

Discovered by: Matthias Clasen

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1079>
2021-03-22 14:05:27 +11:00
Matthew Waters
98249a57db gst: don't use volatile to mean atomic
volatile is not sufficient to provide atomic guarantees and real atomics
should be used instead.  GCC 11 has started warning about using volatile
with atomic operations.

https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719

Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1073>
2021-03-19 04:20:19 +00:00
Jan Alexander Steffens (heftig)
a379e0e5f1 audioaggregator: Consider converting for equal audio formats
The converter might have a non-passthrough mix-matrix. The converter
can determine whether it should pass through, so let it, then remove it
if it's indeed a passthrough.

FIXME: Not converting when we need to but the config is invalid (e.g.
because the mix-matrix is not the right size) produces garbage. An
invalid config should cause a GST_FLOW_NOT_NEGOTIATED.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1070>
2021-03-16 13:46:56 +01:00
Jan Alexander Steffens (heftig)
43449d9fb2 audioaggregator: Clean up _convert_pad_update_converter
No functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1070>
2021-03-16 13:46:55 +01:00
Nirbheek Chauhan
9b01036664 rtspconnection: Consistently translate GIOError to GstRTSPResult
The users of this API need to be able to differentiate between EINTR
and ERROR. For example, in rtspsrc, gst_rtsp_conninfo_connect()
behaves differently when gst_rtsp_connection_connect_with_response_usec()
returns an ERROR or EINTR. The former is an element error while the
latter is simple a GST_ERROR since it was a user cancellation of the
connection attempt.

Due to this, rtspsrc was incorrectly emitting element errors while
going to NULL, which would or would not reach the application in
a racy manner.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1069>
2021-03-16 08:18:11 +00:00
Tim-Philipp Müller
f4a1428a69 tag: id3v2: fix frame size check and potential invalid reads
Check the right variable when checking if there's
enough data left to read the frame size.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/876

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1065>
2021-03-15 11:44:22 +00:00
Jakub Adam
1a87a6572e rtpbasedepayload: handle caps change partway through buffer list
While preparing a blist for pushing, some RTP header extension may
request caps change for a specific buffer in the list. When this
happens, depayloader should immediately push those buffers from the list
that precede the currently processed buffer (for which the caps change
was requested) and only then apply the new caps to the src pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
2021-03-12 18:45:04 +01:00
Jakub Adam
c222f322c0 rtphdrext: allow updating depayloader src caps
Add overridable method that updates depayloader's src caps based on
the data from RTP header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
2021-03-12 18:45:04 +01:00
Jakub Adam
899c69abad rtphdrext: allow the extension to inspect payloader's sink caps
Some header extensions may need to read information from the payloader's
sink caps. Introduce gst_rtp_header_extension_update_from_sinkcaps ()
that passes the caps to the extension, which can then use it to update
its internal state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
2021-03-12 18:45:04 +01:00
Guillaume Desmottes
b7c1810aa3 audioaggregator: fix input_buffer ownership
The way pad->priv->input_buffer reference was managed was pretty
spurious:
- it was overridden without unrefing it, which could potentially lead to
  leaks.
- we were unreffing it while keeping the pointer around, which could
  potentially lead to use-after-free or double-free.

As priv->input_buffer is actually no longer used outside of the
aggregate() method, remove it from pad->priv to simplify the code and
prevent the issues desribed above.

Fix a single buffer leak when shutting down the pipeline as the buffer
returned from gst_aggregator_pad_drop_buffer() was never unreffed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1061>
2021-03-10 16:38:03 +01:00
Guillaume Desmottes
44358f1eaf audioaggregator: fix input buffer when converting
This code path is meant to convert the current buffer to the new format
on update. It was using priv->input_buffer as input which is either
priv->buffer or a converted version of it.
Use priv->buffer instead as priv->input_buffer may no longer be a valid
reference.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1061>
2021-03-10 16:34:28 +01:00
Alexander Vandenbulcke
ccebcaa586 gl/dispmanx: assign render_rect to window before window_resize
If the `render_rect` for a dispmanx display is set after calling
`window_resize` the resize defaults to the dp_width and dp_height to
determine the location of the render rectangle instead of the correct
dimensions that should be set on the window_egl.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1056>
2021-03-02 09:13:25 +01:00
Kristofer Björkström
11b5ebd058 gstrtspconnection: correct data_size when tunneled mode
gst_rtsp_connection_send_messages_usec in tunneled mode does base64
encode messages. When calculating data_size 1 bytes is added, which
results in ending the base64 with a NULL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1051>
2021-02-25 12:21:53 +01:00
Robert Rosengren
e99a6f3142 audio: Use GST_BUFFER_PTS instead of deprecated GST_BUFFER_TIMESTAMP
GST_BUFFER_PTS already used in audio code base (e.g. gstaudiodecoder),
so migrate completely from deprecated GST_BUFFER_TIMESTAMP for better
readability, as gstcompat.h defines GST_BUFFER_TIMESTAMP directly to PTS
anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1048>
2021-02-25 02:04:44 +00:00
Sebastian Dröge
f5381ba9f5 audioaggregator: Log if the sample rate of one sinkpad is not accepted
Otherwise this can silently cause not-negotiated errors without any
direct hint about what went wrong.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1049>
2021-02-24 19:53:02 +02:00
Vivia Nikolaidou
1517b7043d video-converter: Don't upsample/downsample/dither invalid lines
This is a fallout from the conversion to support multiple threads.
convert->upsample_p is never NULL now, it's always an allocated array of
n_threads potentially-null pointers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1043>
2021-02-23 03:40:12 +00:00
Vivia Nikolaidou
2527c8f9f8 libs: audio: Handle meta changes in gst_audio_buffer_truncate
Set timestamp and duration to GST_CLOCK_TIME_NONE unless trim==0,
because that function doesn't know the rate and therefore can't
calculate them. Set offset and offset_end to appropriate values. Make it
clear in the documentation that the caller is responsible for setting
the timestamp and duration.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/869

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1039>
2021-02-18 11:25:32 +02:00
Alicia Boya García
29aeba639a videodecoder: Fix racy critical when pool negotiation occurs during flush
I found a rather reproducible race in a WebKit LayoutTest when a player
was intantiated and a VP8/9 video was loaded, then torn down after
getting the video dimensions from the caps.

The crash occurs during the handling of the first frame by gstvpxdec.
The following actions happen sequentially leading to a crash.

(MT=Main Thread, ST=Streaming Thread)

MT: Sets pipeline state to NULL, which deactivates vpxdec's srcpad,
    which in turn sets its FLUSHING flag.

ST: gst_vpx_dec_handle_frame() -- which is still running -- calls
    gst_video_decoder_allocate_output_frame(); this in turn calls
    gst_video_decoder_negotiate_unlocked() which fails because the
    srcpad is FLUSHING. As a direct consequence of the negotiation
    failure, a pool is NOT set.

    gst_video_decoder_negotiate_unlocked() still assumes there is a
    pool, crashing in a critical in gst_buffer_pool_acquire_buffer()
    a couple statements later.

This patch fixes the bug by returning != GST_FLOW_OK when the
negotiation fails. If the srcpad is FLUSHING, GST_FLOW_FLUSHING is
returned, otherwise GST_FLOW_ERROR is used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1031>
2021-02-16 16:57:54 +00:00
Jan Alexander Steffens (heftig)
297a5f09b1 libs: audio: Fix gst_audio_buffer_truncate meta handling
In the non-interleaved case, it made `buffer` writable but then changed
the meta of the non-writable buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1035>
2021-02-15 17:32:04 +01:00
Alejandro González
319da90d4c audioencoder: Fix gst_audio_encoder_get_audio_info return ownership GTK-Doc
GTK-Doc specifies that, by default, the caller owns returned objects, so that the caller should free them when it is done. However, in the case of this function, the returned GstAudioInfo is owned by the decoder, so this default choice is incorrect. This creates double free problems when using GStreamer Rust bindings, because they are generated using the information contained in the docs.

Fix this by correctly specifying that the caller does not own the returned object.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1032>
2021-02-13 21:25:18 +00:00
Alejandro González
2fd2540ea5 audiodecoder: Fix gst_audio_decoder_get_audio_info return ownership GTK-Doc
GTK-Doc specifies that, by default, the caller owns returned objects, so that the caller should free it when it is done. However, in the case of this function, the returned GstAudioInfo is owned by the decoder, so this default choice is incorrect. This creates double free problems when using GStreamer Rust bindings, because they are generated using the information contained in the docs.

Fix this by correctly specifying that the caller does not own the returned object.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1032>
2021-02-13 17:24:37 +00:00
Thibault Saunier
e1a8393ba7 encoding-profile: Plug a leak of factory list
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1002>
2021-02-10 15:56:26 +00:00
Thibault Saunier
a8fca8d040 encodebin: Add APIs to set element properties on encoding profiles
User often want to set encoder properties on encoding profiles,
this introduces a way to easily 'preset' properties when defining the
profile. This uses GstStructure to define those properties the same
way it is done in `splitmux` for example as it makes simple to handle.

This also defines a more complex structure type where we can map a set
of properties to set depending on the muxer/encoder factory that has
been picked by EncodeBin so it is quite flexible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1002>
2021-02-10 15:56:26 +00:00
Thibault Saunier
a8fdaba2ab encoding-profile: Cleanup profile serialization documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1002>
2021-02-10 15:56:26 +00:00
Alexander Vandenbulcke
57029ba098 gl/dispmanx: fix deadlock triggered by set_render_rectangle
When the gstglimagesink is started with the option `glimagesink
render-rectangle="<0,0,1920,1080>"`, the pipeline reaches a deadlock.
The reason the deadlock occurs is that the
`gst_gl_window_set_render_rectangle` takes locks on the window, in
addition it calls `window_class->set_render_rectangle(...)` which
executes the `_on_resize` function. Since the `_on_resize` function also
takes locks on the window the deadlock is achieved.

By scheduling the adjustment of the render rectangle through an async
message for `gst_gl_window_dispmanx_set_render_rectangle`, the actual
resize happens in another context and therefore doesn't suffers from the
lock taken in `gst_gl_window_set_render_rectangle`.

This solution follows the same approach as gl/wayland. The problem was
introduced by b887db1. For the full discussion check #849.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1030>
2021-02-10 09:30:27 +01:00
Vivia Nikolaidou
ca4240bd03 videoconvert: Support for alternate-field interlacing
Treat the data just like normal data with half the height. Also treat it
as progressive when converting from/to I420 because it requires
different handling for chroma subsampling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1027>
2021-02-04 18:22:07 +02:00
Havard Graff
0f866832b1 audio: add GstAudioLevelMeta
Will be used to implement RTP extension https://tools.ietf.org/html/rfc6464

Co-authored-by: Guillaume Desmottes <guillaume.desmottes@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/706>
2021-02-04 10:25:24 +01:00
Guillaume Desmottes
a48edc8372 rtpbasedepayload: add auto-header-extension property
Same property as the one I just added on rtpbasepayload.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1022>
2021-02-03 11:23:40 +01:00
Guillaume Desmottes
bad4b1711d rtpbasepayload: add auto-header-extension property
Using RTP header extensions is currently not convenient. Users have to
handle signals from the RTP payloader and instantiate the extension
element themselves, making it impossible to use with gst-launch.

Adding a property allowing the payloader to automatically try creating
extensions. This should help simple use cases and testing using
gst-launch.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1022>
2021-02-03 11:19:04 +01:00
Marijn Suijten
9ab400e267 gstaudiostreamalign: Pass self as const pointer in getter functions
It was noticed in [1] that `GstAudioStreamAlign` is a simple boxed type
that is passed as const in the copy function, but not as such in the
getters. These functions turn out to be the only users of `const = true`
overrides in `gstreamer-rs`. Since there is no locking or other advanced
caching/sharing going on (as happens with miniobjects) these functions
can safely take self as const pointer.

[1]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/683#note_783129

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1025>
2021-01-29 21:42:47 +01:00
Jakub Adam
11e6f8da92 video-hdr: Add API to check content light level equality
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/969>
2021-01-28 20:55:38 +01:00
Guillaume Desmottes
df9064fdc6 rtpbasedepayload: set attributes on newly requested extensions
Users were supposed to configure the extension themselves but it was
impossible to do so as they didn't have access to the caps.

Fix #864

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1021>
2021-01-27 09:48:49 +01:00
Guillaume Desmottes
912cf46b83 rtpbasepayload: set attributes on newly requested extensions
Users were supposed to configure the extension themselves but it was
impossible to do so as they didn't have access to the caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1021>
2021-01-27 09:48:49 +01:00
Guillaume Desmottes
5acde5568e rtpbasedepayload: fix clear-extensions signal definition
Typo as we were using the wrong enum.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1021>
2021-01-27 09:48:49 +01:00
Guillaume Desmottes
0896ccb436 rtp: fix clear-extensions signal definition
Typo as we were using the wrong enum.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1018>
2021-01-25 14:28:12 +01:00
Guillaume Desmottes
d396190b91 rtphdrext: fix typo in doc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1018>
2021-01-25 14:28:12 +01:00
Marijn Suijten
abb026ec6a gl,video: Make ptrs to VideoInfo and (GL)AllocationParams immutable
These parameters are incorrectly regarded as mutable in G-IR making them
"incompatible" with languages that are explicit about mutability like
Rust. In order to clean up the code and expected API there, update the
signatures here, right at the source (instead of overriding them in
Gir.toml and hoping for the best).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
2021-01-14 11:53:10 +00:00
Marijn Suijten
fa8b5b9a6d audio/audio-buffer: @buffer in audio_buffer_map is out caller-allocates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
2021-01-14 11:53:10 +00:00
Marijn Suijten
c70d263e48 video/video-frame: @frame in video_frame_map is out caller-allocates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
2021-01-14 11:53:10 +00:00
Marijn Suijten
a263919f06 audio,video: Add out caller-allocates to init and from_caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
2021-01-14 11:53:10 +00:00
Sebastian Dröge
7e16eed522 videosink: Add new GstVideoSink::set_info() virtual method
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/986>
2021-01-14 11:15:40 +00:00
Sebastian Dröge
198434e71a videosink: Implement more complete BaseSink::get_times() based on the framerate
This will only make use of the framerate if the subclass is chaining up
BaseSink::set_caps(). Otherwise it will have the same behaviour as the
basesink default.

Doing so is useful if video buffers don't contain a duration to
calculate a default duration, and various video sinks already implement
a custom version of this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/986>
2021-01-14 11:15:40 +00:00
Marijn Suijten
1f06cf60e7 video: Convert info_to_caps to take self as const ptr
This requires a slight modification to the function itself because it
was overwriting a member locally.

However, now this side-effect cannot be observed outside the function
anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1008>
2021-01-14 08:14:36 +00:00
Matthew Waters
b60951a4fa gl: add get_type() implementations for all of our memory types
Otherwise, various bindings can't really know the type of an object as
required.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/999>
2021-01-13 13:40:58 +00:00
Jakub Adam
f5d971a19e rtpbasepayload: fix header extension length calculation
Since ternary operator has the lowest precedence in the expressions at
hand, wordlen would always incorrectly yield 0 or 1.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1009>
2021-01-12 22:26:19 +01:00
Jakub Adam
2d198ff10b video-blend: fix blending 8-bit and 16-bit frames together
Replace hardcoded 255s with the correct max value for the given color
depth. Use 64-bit integer in calculations where overflow may occur.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1000>
2021-01-08 08:04:55 +00:00
Matthew Waters
f573d91237 gl: document some GL caps specifics
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/854
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/996>
2021-01-05 15:03:54 +00:00
Tim-Philipp Müller
89bd37f24e tagdemux: resize and trim buffer in place to fix interaction with oggdemux
Elements operating in pull mode may optionally pass a buffer to
pull_range that should be filled with the data. The only element
that does that at the moment is oggdemux operating in pull mode.

tagdemux currently creates a sub-buffer whenever a buffer pulled
from upstream (filesrc, usually) needs to be trimmed. This creates
a new buffer, however, so disregards any passed-in buffer from a
downstream oggdemux.

This would cause assertion failures and playback problems for
ogg files that contain ID3 tags at the end.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/848

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/994>
2021-01-04 14:21:43 +00:00
Mathieu Duponchelle
06c158957d appsrc: fix signal documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/992>
2020-12-31 17:01:40 +00:00