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Author SHA1 Message Date
Nicolas Dufresne
afb412b75d v4l2src: Refactor to use PreferredCapsInfo structure 
Avoid passing around a bare structure for the preference, this removes
the need to copy and free that structure and simplify the code. Also
fix a type in the structure name, Prefered -> Preferred.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
c0fdaffc55 v4l2src: Stub preferred resolution support 
This stubs the ability to use preferred resolution from digital
video timings, analog TV standards or driver reported native
resolution.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
1b21c1b15b v4l2: Subscribe source_change for the current input 
When we subscribe for source-change event, we need to specify for which
input. Make sure we subscribe for the current input.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
c386aa43cc v4l2src: Add input signal status detection 
As part of the support to select a preferred size, we can also
detect the signal status. This is a split patch so that feature
is separated to ease review.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
293abe0675 v4l2: Add helper to query input status 
This is a wrapper around ENUM_INPUT renamed for readability.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
ba3eddebc8 v4l2: Fix input/output index sign 
This is an unsigned integer in the kernel API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
1094e2548e v4l2src: Add source resolution change support 
This patch adds support for source resolution change detection.
Resolution change is signaled by drivers when a change in the detected
signal have been detected. This is notably seen on HDMI receivers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
bb1d9b398e v4l2bufferpool: Handle resolution change event 
This patch adds the detection, dequeuing and reporting of the SOURCE_CHANGE
event when the CH_RESOLUTION flag is set. The acquire function will now return
a new custom success called GST_V4L2_FLOW_RESOLUTION_CHANGE. In order to use
this new feature, elements must enable it by calling:

  gst_v4l2_buffer_pool_enable_resolution_change (pool);

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
4be9bf4085 v4l2object: Add event helpers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
954e38a6b9 v4l2bufferpool: use FLOW_LAST_BUFFER 
This uses the GST_V4L2_FLOW_LAST_BUFFER alias instead of
GST_FLOW_CUSTOM_SUCCESS to make the code more readable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 15:22:47 -05:00
Lucas Stach
1b1ad32b90 v4l2object: prefer NV12 over I420 
Considering NV12 an 'odd' format is a historical artifact. This format
is now quite common, and usually preferable to I420 due to more memory
friendly access patterns.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/857>
 2021-02-19 13:12:54 -05:00
Guillaume Desmottes
1796f3f5e4 wavparse: fix seeking in READY state 
wavparse claims to be able to support seeking in the READY state by
saving the pending seek event and actually seeking later after having parsed the
header.
Problem was that this seek event was reset on the READY to PAUSED
transition, making all this code useless. Fixing it by stop resetting
on READY to PAUSED transition as we already reset on PAUSED to READY
and when initiating the element.

Note that DTS marker detection isn't support in such scenario as
gst_type_find_helper_for_buffer() needs a buffer containing the
beginning of the stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>
 2021-02-18 16:32:24 +01:00
Guillaume Desmottes
4aa39da2d3 tests: wavparse: factor out create_pipeline() 
No semantic change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>
 2021-02-18 10:38:18 +01:00
Mathieu Duponchelle
f07fe93202 docs: update plugins cache with new h264 / vp8 depay properties 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
 2021-02-18 01:54:03 +00:00
Mathieu Duponchelle
49de1e6679 rtph264depay: expose request-keyframe property 
When set, the depayloader will request new keyframes on packet
loss

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
 2021-02-18 01:54:03 +00:00
Mathieu Duponchelle
20fc6da913 rtpvp8depay: expose request-keyframe property 
When set, the depayloader will request new keyframes on packet
loss

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
 2021-02-18 01:54:03 +00:00
Mathieu Duponchelle
69a43dd00b rtph264depay: expose wait-for-keyframe property 
Similar to rtpvp8depay, when packet loss occurs, the depayloader
starts waiting for a keyframe.

We try to only stop waiting when all the packets for the new keyframe
have been received, by only resetting waiting_for_keyframe when
encountering the first packet of a keyframe, this is slightly
fragile because there is no bit that explicitly marks the start
of an access unit, so we rely on the existing picture_start
detection code.

As a consequence, the property is only meaningful when outputting
access units, and is ignored when outputting NALs directly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
 2021-02-18 01:54:03 +00:00
Mathieu Duponchelle
e71648e214 videomixer: document as deprecated 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/878>
 2021-02-18 01:48:24 +01:00
Ashley Brighthope
2a4c63036b wavenc: Fixed INFO chunk corruption, caused by odd sized data not being padded. Code style was updated. 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/873>
 2021-02-17 09:54:40 +02:00
Jakub Adam
5fe0aa03eb rtpopuspay: add info regarding (non-standard) multichannel support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Jakub Adam
748a1866af docs: update plugins cache for rtpopus 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Jakub Adam
b105797163 tests: add rtpopus multichannel test cases 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Jakub Adam
8f6969429d rtpopusdepay: support libwebrtc-compatible multichannel payload 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Jakub Adam
b9ed5c7fa0 rtpopuspay: support libwebrtc-compatible multichannel payload 
When the audio has more than 2 channels, add optional fields to output
caps from which webrtcbin can generate SDP in the syntax recognized by
"multiopus" codec present in libwebrtc [1].

e.g. for 5.1 audio:

a=rtpmap:96 multiopus/48000/6
a=fmtp:96 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5

[1] https://webrtc-review.googlesource.com/c/src/+/129768

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Jakub Adam
8b4147c757 rtpopuspay: make use of gst_rtp_base_payload_set_outcaps_structure() 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Olivier Crête
45d105bea2 effectv: Remove redundant license file 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/869>
 2021-02-09 19:31:28 -05:00
Kevin Song
c63ff9c06c Apply 1 suggestion(s) to 1 file(s) 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
 2021-02-05 00:55:49 +00:00
Kevin Song
fd6c296021 Apply 1 suggestion(s) to 1 file(s) 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
 2021-02-05 00:55:49 +00:00
Bing Song
025b43e512 v4l2videoenc: support resolution change stream encode. 
Resolution change stream transcoding will drain before send new video
frame buffer. Need encode video frame after process EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
 2021-02-05 00:55:49 +00:00
Xabier Rodriguez Calvar
61d204ab22 qtdemux: added support for cbcs encryption scheme 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/865>
 2021-02-04 12:30:22 +01:00
Guillaume Desmottes
7b7e49de31 rtp: add rtphdrextrfc6464 
Header Extension for Client-to-Mixer Audio Level Indication as
defined in RFC 6464.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
 2021-02-04 11:12:51 +01:00
Guillaume Desmottes
4b6c3c9a1b level: add GstRTPAudioLevelMeta on buffers 
This meta can be used by a RTP payloader to send the level information
to the peer.

Part of https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/446

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
 2021-02-04 11:12:47 +01:00
Robert Swain
25f98ab134 deinterlace: Provide documentation for GST_DEINTERLACE_BUFFER_STATE 
More information available in
https://gstconf.ubicast.tv/videos/interlacing-and-telecine-in-gstreamer/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
 2021-02-03 17:10:20 +02:00
Vivia Nikolaidou
c7b11482d0 deinterlace: Fix telecine/onefield mixup 
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/838

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
 2021-02-03 16:30:15 +02:00
Vivia Nikolaidou
4c4e1b580e deinterlace: Better alternate support 
Improve line offset halving based on whether this field is top or
bottom.

Also handle the buffer state the same as mixed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
 2021-02-03 16:30:15 +02:00
Bing Song
5f00810ee0 v4l2h265codec: fix HEVC profile string issue. 
Keep HEVC profile compatible with other module.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/850>
 2021-01-27 01:22:41 +00:00
Bing Song
b87db31fbe v4l2object: Need keep same transfer as input caps. 
GST_VIDEO_TRANSFER_BT2020_12 and GST_VIDEO_TRANSFER_BT2020_10 will
be mapped to V4L2_XFER_FUNC_709. Need check input caps when map
V4L2_XFER_FUNC_709 back to GST_VIDEO_TRANSFER_BT2020_12 and
GST_VIDEO_TRANSFER_BT2020_10

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/816
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/841>
 2021-01-18 04:26:04 +00:00
Tobias Ronge
706d91371c rtspsrc: Do not wait for response while flushing 
Due to the may_cancel flag in GstRTSPConnection, receiving might not get
cancelled when supposed to. In this case, gst_rtsp_src_receive_response
will have to wait until timeout instead but if busy receiving RTP
data, this timeout will never occur.

With this patch, gst_rtsp_src_receive_response returns GST_RTSP_EINTR
if flushing is set to TRUE instead of continuing to receive.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/831>
 2021-01-15 09:24:51 +00:00
Tim-Philipp Müller
c854e6bd39 meson: allow libdv subproject fallback 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/854>
 2021-01-14 19:13:03 +00:00
Xabier Rodriguez Calvar
c5ebaadf9d qtdemux: Allow streams with no specified protection system ID 
This is necessary in cases like CMAF where there won't be any events
passing thru.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/852>
 2021-01-14 10:11:10 +01:00
Hou Qi
386b785e48 v4l2object: Map correct video format for RGBA 
Map V4L2_PIX_FMT_RGBA32 pixel format to GST_VIDEO_FORMAT_RGBA instead of
GST_VIDEO_FORMAT_RGB video format to support RGBA.

Fixes #823

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/848>
 2021-01-11 09:05:05 +08:00
Sanchayan Maity
79efd372c1 udpsrc: Fix marker links 
These should be with a single ':'. The double '::' results in a CI with
build failure message like below.

ERROR: [links]: (mandatory-link-not-found): Mandatory link Link GstSocketTimestamp -> None (GstSocketTimestamp) could not be resolved
ERROR: [check-missing-since-markers]: (missing-since-marker): Missing since marker for udpsrc:socket-timestamp
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/828>
 2021-01-04 15:23:22 -05:00
Sanchayan Maity
e0b09a1612 udpsrc: Allow use of socket control message timestamps for DTS 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/828>
 2021-01-04 15:23:22 -05:00
Matthew Waters
db15ec9286 videoflip: fix possible crash when setting the video-direction while running 
A classic case of not enough locking.

One interesting thing with this is the interaction between the
rotation value and caps negotiation.  i.e. the width/height of the caps
can be swapped depending on the video-direction property.  We can't lock
the entirety of the caps negotiation for obvious reasons so we need to
do something else.  This takes the approach of trying to use a single
rotation value throughout the entirety of the negotiation and then
subsequent output frame in a kind of latching sequence.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/792
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
 2021-01-04 12:10:12 +00:00
Matthew Waters
35018d67ef tests: add tests for videoflip 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
 2021-01-04 12:10:12 +00:00
Ignacio Casal Quinteiro
219b659320 deinterlace: force -DPREFIX on macos 
This is due to a bug in meson where it will not detect properly
the compiler if the symbols need an undercore.
https://github.com/mesonbuild/meson/issues/5482

Fixes #821

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/845>
 2020-12-30 13:40:35 +01:00
Sebastian Dröge
39c6bc0507 rtspsrc: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/842>
 2020-12-21 09:59:43 +00:00
Vivia Nikolaidou
81d2f67ba5 splitmuxsink: Avoid deadlock when releasing a pad from a running muxer 
Might not drain correctly

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/838>
 2020-12-16 06:17:08 +00:00
Hou Qi
ff1dd307f2 v4l2object: Use active resolution during fallback colorspace probe 
For legacy drivers that don't implement ENUM_FRAMESIZE, use active
resolution to probe colorspace. This can improve the accuracy of the
result when the colorspace depends on the resolution. This fixes a
wrong colorspace issue on board with vendor bsp at resolution 2560x1440.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/830>
 2020-12-14 09:15:18 +08:00
Mathieu Duponchelle
6d4dcb430d rtpst2022-1-fecdec: don't xor out of bounds 
When reconstituting packets from a stream with variable packet
sizes, don't xor larger packets past the length of the protected
packet

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
 2020-12-12 09:29:15 +00:00