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Author SHA1 Message Date
Nirbheek Chauhan
76d624b2df rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL 
This usually doesn't matter, but it is disruptive when streaming from
a shared media since it will pause all other clients when any client
exits.

This new behaviour is opt-in and should be safe because you need to
set the NULL state on rtspsrc directly, instead of just on the
pipeline. See the updated documentation for an explanation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/901>
 2021-03-15 12:59:09 +05:30
Philipp Zabel
78dec1e403 v4l2object: handle GST_VIDEO_TRANSFER_BT601 
V4L2 makes no difference between the BT.601 and BT.709 transfer
functions [1], but GStreamer does since 1.18 [2].

Adapt gst_v4l2_object_get_colorspace() and
gst_v4l2_object_set_format_full().

[1] https://linuxtv.org/downloads/v4l-dvb-apis-new/userspace-api/v4l/colorspaces-details.html#colorspace-smpte-170m-v4l2-colorspace-smpte170m
[2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/724

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/856>
 2021-03-12 16:57:50 +00:00
Mathieu Duponchelle
f2d909eb4e rtspsrc: fix title of a few properties docstrings 
GstRtspSrc -> GstRTSPSrc

This would have been noticed by the since checker, but those
properties were introduced prior to that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/899>
 2021-03-11 22:22:15 +01:00
Vladimir Menshakov
4de3ddad15 wavpackdec: Add floating point format support 
This commit negotiate F32 audio format if MODE_FLOAT used in wavpack file.
Wavpack float mode is always in 32-bit IEEE format.

The following pipeline plays distorted audio if source file is encoded in float mode:
gst-launch-1.0 filesrc ... ! wavpackparse ! wavpackdec ! pulsesink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/894>
 2021-03-08 15:19:57 +02:00
Matthew Waters
e73559f815 matroska: also support push-mode from seek events sent to the element 
Otherwise sending seek events would fail to actually seek.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/891>
 2021-03-04 13:28:55 +00:00
Marc Leeman
c7356c70ee gstrtspsrc: 551 should not result in an unhandled error 
Some cameras (e.g. HikVision DS-2CD2732F-IS) return "551 Option
not supported" when a command is sent that is not implemented
(e.g. PAUSE). Instead; it should return "501 Not Implemented".

This is wrong, as previously, the camera did announce support for PAUSE
in the OPTIONS.

In this case, handle the 551 as if it was 501 to avoid throwing errors
to application level. */

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/885>
 2021-03-04 13:07:49 +01:00
Hou Qi
fc5a1e468b v4l2videodec: Do not expose profiles/levels in vp8/vp9 template caps 
Vp8/vp9 supported profiles/levels are listed in decoder sink caps, but
there is no parser for these two formats and the demuxers also don't have
these information. It causes negotiation fail between demuxers and decoder
when check caps "accept = gst_caps_is_subset (caps, template_caps);".
To fix this, need to remove profiles/levels for vp8/vp9 formats in decoder
sink caps.

Fix #854

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/887>
 2021-03-03 15:17:58 +00:00
Seungha Yang
614f4ec5b5 rtpmanager: Fix an MSVC compile warning 
We don't expect this object is a part of public library.

gstrtphdrext-twcc.c(45): warning C4273: 'gst_rtp_header_extension_twcc_get_type': inconsistent dll linkage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/889>
 2021-03-03 18:30:39 +09:00
Philipp Zabel
e885e0e1d7 v4l2videodec: fix src side frame rate negotiation 
Negotiating v4l2h264dec ! v4l2h264enc transcoding pipelines fails in
case the encoder does not accept framerate=(fraction)0/1.
The acquired caps used for downstream negotiation are determined from
gst_v4l2_object_acquire_format(), which sets the GstVideoInfo::fps_n
and ::fps_d fields to 0.
To fix this, copy the frame rate from the sink side.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/882>
 2021-02-24 17:15:49 +01:00
Jordan Petridis
1257e50d49 rpicamsrc: depend on posix threads and vchiq_arm 
Could only test on rpi 3b+

Close #839

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/875>
 2021-02-21 00:18:34 +00:00
Nicolas Dufresne
bb29b8f8d2 v4l2bufferpool: Silence traces around unsupported source change 
Don't be too spamy about unsupported source change flags as these will be
commonly extended in the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
409ea3ac84 v4l2src: Move preferred resolution query before the probe 
As we lock the DV_TIMINGS (and standards in the future), we need to probe the
caps after, otherwise, we may endup fixating to an unsupported resolution,
which would lead to a not-negotiated error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
b530c0ef4e v4l2src: Calculate framerate from DV timings 
And use this framerate in our preference. Note that we also flush
the probed caps as it seems that the format enumeration may change
when a new source change event get triggered.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
25696203c3 v4l2rc: Add DV_TIMINGS query and locking 
This adds support to DV_TIMINGS query and locking. The timing width and
height is then used as a preference.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
b750fb2d5b v4l2src: Force renegotiation on resolution change 
As mandated by the specification, make sure to cycle through streamoff
/ streamon regardless if the caps have changed or not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
Nicolas Dufresne
7b8fe18d06 v4l2object: Remove unused streaming member 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
 2021-02-19 16:02:02 -05:00
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