This elements pass RTP packets along unchanged and appear as a RTP
payloader element.
This is useful, for example when using the gstreamer-rtsp-server
library, in the case where you are receiving RTP packets from a
different source and want to serve them over RTSP. Since the
gst-rtsp-server library expect the element marked as payX to be a RTP
payloader element and assumes certain properties are available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5204>
The hack enforcing strictly increasing timestamps was, according to the
code comments, because librtmp was confused with backwards timestamps.
rtmp2sink is not using librtmp as rtmpsink did, so this is no longer
required.
Also changing the timestamps is causing audio glitches when streaming to
Youtube.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5212>
These 10bit formats are identical to NV12_16L32S, but 64bytes of data is being
prefixed with 16bytes data with four pixels of lower 2bits per byte. For
MT2110T, the lower two bits set so each bytes contains a column of 4 pixels,
also describe as tiled lower 2 bits. MT2110T has been chosen as a name to match
the vendor chosen name. This format is unlikely to exist for other vendors.
For MT2110R, the 2 low bits are in raster order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3444>
With libsoup 2.x, it was possible to know when there was a TLS failure, as
libsoup provided the "special http status code" SOUP_STATUS_SSL_FAILED.
However these special codes were dropped with libsoup 3.x: now libsoup emits
the accept-certificate signal when there's a TLS failure.
This commit adds a signal "accept-certificate" to SoupHttpSrc, which is in fact
just about forwarding the signal from SoupMessage (which is, itself, forwarded
from GTlsConnection). Note that, in case of libsoup 2.x, the signal is never
emitted.
Fixes: #2379
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4925>
- Adding bayer 10,12,14,16 bits components with 16 bits storage. These
changes only adds capabilities. Capability format string is a complete
description of the frame and pixels layout. Only mapping LE bayer
formats as v4l2 only define LE bayer formats.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4852>
Since c2f890ab, element properties are gathered from the parse-launch
line and passed at object construction.
This caused the following issue to happen in videoflip:
* videoflip installed a CONSTRUCT property named method, now deprecated
* videoflip now also overrides that property with a video-direction
property
GObject construction causes method to be set first at construct time,
with the user-provided value, then video-direction with the default
value.
The user-provided value was thus overridden, causing a regression.
Fix by not installing the properties as CONSTRUCT, and explicitly
implementing constructed() instead in order to ensure that we do still
call gst_video_flip_set_method() at least once during construction.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2529
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4536>
Assuming that V4L2 CAPTURE devices always use one buffer per JPEG image, we can
always mark JPEGs provided by a V4L2 element as parsed.
The V4L2 elements require that JPEG images sent to V4L2 OUTPUT devices must
always be parsed.
This is necessary to link a V4L2 CAPTURE device with a V4L2 OUTPUT device
without explicitly marking the stream as parsed or adding a jpegparse into the
pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4229>
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.
This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
Previously the minimum buffering threshold was hardcoded to a specific
value (10s). This is suboptimal this an actual value will depend on the actual
stream being played.
This commit sets the low watermark threshold in time to 0, which is an automatic
mode. Subclasses can provide a stream `recommended_buffering_threshold` when
update_stream_info() is called.
Currently implemented for HLS, where we recommended 1.5 average segment
duration. This will result in buffering being at 100% when the 2nd segment has
been downloaded (minus a bit already being consumed downstream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3240>
