In a case of a scr different from 0, after a seek,
the src_segment.stop has been updated with the duration
not including the base_time (scr). The segment position
needs to be tested upon segment.stop + base_time (scr)
to check for an EOS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2048>
After the VA filter creation, when changing the element's state from NULL
to READY, immediatly checks for any filter operation requested by the user.
If any, the passthrough mode is disabled early, so there's no need for a
future renegotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2094>
Useful when having a service that runs a GStreamer pipeline
or application in Google Cloud to avoid storing the inputs
and outputs in the running container or service. For example
when analyzing a video from a Google Cloud Storage bucket
and extracting images or converting the video and then uploading
the results into another Google Cloud Storage bucket.
- gssrc allows to read from a file located in Google Cloud
Storage and it supports seeking.
- gssink allows to write to a file located in Google Cloud
Storage. There are 2 modes, one similar to multifilesink and
the other similar to filesink.
Example:
gst-launch-1.0 gssrc location=gs://mybucket/videos/sample.mp4 ! decodebin ! glimagesink
gst-launch-1.0 playbin uri=gs://mybucket/videos/sample.mp4
gst-launch-1.0 videotestsrc num-buffers=5 ! pngenc ! gssink object-name="img/img%05d.png" bucket-name="mybucket" next-file=buffer
gst-launch-1.0 filesrc location=sample.mp4 ! gssink object-name="videos/video.mp4" bucket-name="mybucket" next-file=none
When running locally simply set GOOGLE_APPLICATION_CREDENTIALS. But
when running in Google Cloud Run or Google Cloud Engine, just set the
"service-account-email" property on each element.
Closes#1264
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1369>
Add a new property so that clocksync can setup "ts-offset" value
based on the first buffer and pipeline's running time when the
first arrived. Newly update "ts-offset" in this case would be
a value that allows outputting the first buffer without clock waiting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/702>
In order to always insert a PCR packet right on time, we need to
check whether one is needed when outputting any packet, not only
a packet for the PCR stream. Most of the PCR packets will remain
data-carrying packets, but as a last resort we may insert stuffing
packets on the PCR stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2073>
Change PCR / SI scheduling so that instead of checking if
the current PCR is larger than the next target time, instead
check if the PCR of the next packet would be too late, so PCR
and SI are always scheduled earlier than the target, not later.
There are still cases where PCR can be written too late though,
because we don't check before each output packet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2073>
Prior to that, cccombiner's behaviour was essentially that of
a funnel: it strictly looked at input timestamps to associate
together video and caption buffers.
This patch instead exposes a "schedule" property, with a default
of TRUE, to control whether caption buffers should be smoothly
scheduled, in order to have exactly one per output video buffer.
This can involve rewriting input captions, for example when the
input is CDP sequence counters are rewritten, time codes are dropped
and potentially re-injected if the input video frame had a time code
meta.
Caption buffers may also get split up in order to assign captions to
the correct field when the input is interlaced.
This can also imply that the input will drift from synchronization,
when there isn't enough padding in the input stream to catch up. In
that case the element will start dropping old caption buffers once
the number of buffers in its internal queue reaches a certain limit
(configurable).
The property is exposed so that existing users of cccombiner can
revert back to the original behaviour, but should eventually be
removed, as that behaviour was simply inadequate.
This commit also disallows changing the input caption type, as
this would needlessly complicate implementation, and removes
the corresponding test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2076>
When we transform the caps from the sink to src, or vice versa, the
"caps" passed to us may only contain parts of the features. Which
makes our vpp lose some feature in caps and get a negotiation error.
The correct way should be:
Cleaning the format and resolution of that caps, but adding all VA,
DMA features to it, making it a full feature caps. Then, clipping it
with the pad template.
fixes: #1551
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2081>
Since glib 2.62, the accumulated return values in RUN_CLEANUP override the
accumulated return values in RUN_FIRST. Since:
1. We have a default handler that always returns TRUE, and
2. User handlers are only run in RUN_FIRST, and
3. Our accumulator just takes the latest return value
We were discarding the return value from the user handler and always
sending messages even if the user handler said not to. See
https://gitlab.gnome.org/GNOME/glib/-/issues/2352 for more details.
This signal does not need RUN_CLEANUP or RUN_FIRST, so just change it
to RUN_LAST so that it's emitted exactly once and accumulated once.
With this fix, this signal can now be used to intercept PAUSE when
going to GST_STATE_NULL so that the server does a TEARDOWN (if
necessary) and not a PAUSE, which will confuse other RTSP clients when
playing shared media.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>
Directly setting rtspsrc to the NULL state before putting the pipeline
in the NULL state usually works, but it can cause a deadlock in some
cases, so it's not a reliable mechanism to fix this.
This reverts commit f37afdafff:
"rtspsrc: Fix state changes from PAUSED to PLAYING"
and commit 76d624b2df:
"rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/908>
