Commit graph

20704 commits

Author SHA1 Message Date
Stéphane Cerveau
f94d9f1cc5 audiomixer: allow per feature registration
Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1029>
2021-03-16 17:58:59 +00:00
Stéphane Cerveau
520a682c7c tcp: allow per feature registration
Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1029>
2021-03-16 17:58:59 +00:00
Stéphane Cerveau
3d3831656d rawparse: allow per feature registration
Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1029>
2021-03-16 17:58:59 +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
Devarsh Thakkar
9759810d82 ext: alsa: Set buffer time after period time
This because underlying driver may have constraint on
buffer size to be dependent on period size, so period
time needs to be set first.

For e.g. Xilinx ASoC driver requires
buffer size to be multiple of period size for it's DMA
operation.

alsa-utils also set period time first as seen in below commit :
9b621eeac4

Tested it on zcu106 board with HDMI based record and playback.
Also tested on Intel PC using Logitech C920 Webcam mic and ALC887-VD
Analog for playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1040>
2021-03-11 14:15:54 +00:00
Stéphane Cerveau
01d1bbd1da playback: remove useless ret test
Use GST_ELEMENT_REGISTER_DEFINE_CUSTOM instead
of GST_ELEMENT_REGISTER_DEFINE_WITH_CODE if a specific
init needs to be tested before registering the element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1060>
2021-03-10 20:06:20 +01:00
Stéphane Cerveau
20da00f057 ogg: remove useless ret test
Use GST_ELEMENT_REGISTER_DEFINE_CUSTOM instead
of GST_ELEMENT_REGISTER_DEFINE_WITH_CODE if a specific
init needs to be tested before registering the element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1060>
2021-03-10 20:06:20 +01:00
Stéphane Cerveau
1682161355 alsa: remove useless ret test
Use GST_ELEMENT_REGISTER_DEFINE_CUSTOM instead
of GST_ELEMENT_REGISTER_DEFINE_WITH_CODE if a specific
init needs to be tested before registering the element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1060>
2021-03-10 20:06:20 +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
david
e135961e1e Set _NET_WM_NAME property for xvimagesink and ximagesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1017>
2021-03-03 12:21:22 +00:00
He Junyan
3d96786857 gl: download: Fix a caps memory leak in _try_export_dmabuf().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1058>
2021-03-03 02:43:01 +00:00
He Junyan
f506a3e0ff gl: download: Fix a caps memory leak in prepare_output_buffer().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1058>
2021-03-03 02:43:01 +00:00
He Junyan
2f3033cebe gl: download: Fix the wrong transformed result from src direction in transform_caps().
The current manner in transform_caps() for src direction is not very correct. For example,
when the src caps is:
  video/x-raw(memory:DMABuf); video/x-raw; video/x-raw(memory:GLMemory)
this function returns:
  video/x-raw(memory:DMABuf); video/x-raw; video/x-raw(memory:GLMemory)
as the sink caps. This is not correct, because DMABuf feature is not even in the sink pad's
caps template. The correct answer should be:
  video/x-raw(memory:GLMemory); video/x-raw
only.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1058>
2021-03-03 02:43:01 +00:00
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
Mathieu Duponchelle
dd71f359be compositor: fix drawing of transparent background
When drawing the background multithreaded, y_start needs to be
scaled to obtain the correct byte offset from which to start
memsetting (yoffset).

Fixes #871

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1042>
2021-03-01 23:38:35 +00:00
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
Francisco Javier Velázquez-García
740ea66e73 videotestsrc.c: Correct left shift operator
Use the left shift operator '<<' instead of the mistakenly typed less
than operator '<'.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1047>
2021-02-23 14:53:43 +01:00
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
Jeongki Kim
fd41fca7f3 audioresample: Respect buffer layout when drain
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1045>
2021-02-22 15:36:53 +09:00
Jan Schmidt
ebad39b865 videoconvert: Only prefer upstream chroma-site with same subsampling.
If converting YUV formats with different chroma-subsampling, there's
probably no good reason to prefer the upstream chroma-siting so just use
the default for the output format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1033>
2021-02-19 09:45:07 +00:00
Jan Schmidt
eabb2c1802 videoconvert: Implement more sophisticated colorimetry caps transfer
Implement a more sophisticated transfer of colorimetry and
chroma-site fields to output caps when fixating.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1033>
2021-02-19 09:45:07 +00:00
Jan Schmidt
98bdc76fa5 videoconvert: Forward colorimetry and chroma-site from upstream.
If downstream has expressed no preference for particular colorimetry
and chroma-site configuration, transfer them from the input caps.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/614

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1033>
2021-02-19 09:45:07 +00:00
Stéphane Cerveau
8bf7816790 decodebin3: change stream selection message owner
In order to select the streams on GST_MESSAGE_STREAM_COLLECTION,
the app needs to send the select-streams event
to the decodebin and not to the parsebin.

The message should be always owned by the decodebin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1014>
2021-02-19 08:01:57 +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
Tim-Philipp Müller
c7f1fd8320 uridecodebin3: make caps property work
The caps set on uridecodebin3 via the "caps" property
were never passed to the internal decodebin3, so did
absolutely nothing.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/837

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1034>
2021-02-16 22:58:22 +00:00
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
278b10dd2e videoconvert,videoscale: Add alternate-field negotiation tests
Make sure buffers with alternate-field interlacing mode can be
negotiated

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1027>
2021-02-04 21:47:27 +02:00
Vivia Nikolaidou
b7b3ec6a6e videoscale: Support for alternate-field interlacing
Accept the negotiation, video-converter.c is aware of the half-height
already

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1027>
2021-02-04 18:28:54 +02:00
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
Sebastian Dröge
23370ec429 typefindfunctions: Consider the number and types of atoms found in a row for suggesting a probability
If there are 3 or more known atoms in a row, it's likely that this is
actually MOV/MP4 even if we don't find any other known atoms. If 5 or
more are found then this is most certainly MOV/MP4 and we can return.

Also if a moov and mdat atom is found, this is definitely a MOV/MP4 file
and can be used as such, independent of anything else following the
mdat.

Fixes typefinding of various MOV files that have no `ftyp` atom but
otherwise a valid file structure followed by some garbage.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1013>
2021-01-31 11:53:43 +02:00
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