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>
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>
It was possible to generate a SDP that had an RTX payload type
that matched one of the media payload types when providing caps via
codec_preferences without any sink pads.
Fixes
m=video 9 UDP/TLS/RTP/SAVPF 96
...
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 nack pli
a=fmtp:96 apt=96
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2046>
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored)
add 'gathering_started' flag to avoid changing ports after gathering has started
validity checks: min port <= max port enforced, error thrown otherwise
include tests to ensure port range is being utilized (by @hhardy)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
In listener mode, gst_stats() returns an independent set of
statistics for every connected caller. Having the caller's IP and port
present in each structure allows to correlate the statistics with a
particular caller that has been announced by "caller-added" signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1772>
Problem is that unreffing the EGLImage/SHM Buffer while holding the
images_mutex lock may deadlock when a new buffer is advertised and
an attempt is made to lock the images_mutex there.
The advertisement of the new image/buffer is performed in the
WPEContextThread and the blocking dispatch when unreffing wants to run
something on the WPEContextThread however images_mutex has already been
locked by the destructor.
Delay unreffing images/buffers outside of images_mutex and instead just
clear the relevant fields within the lock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1843>
As advised by !1366#note_629558 , the nice transport should be
accessed through:
> transceiver->sender/receiver->transport/rtcp_transport->icetransport
All the objects on the path can be accessed through properties
except sender/receiver->transport. This patch addresses that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1952>
Using the object lock is problematic for anything that can dispatch to
another thread which is what createWPEView() does inside
gst_wpe_src_start(). Using the object lock there can cause a deadlock.
One example of such a deadlock is when createWPEView is called, but
another (or the same) wpesrc is on the WPEContextThread and e.g. posts a
bus message. This message propagations takes and releases the object
lock of numerous elements in quick succession for determining various
information about the elements in the bin. If the object lock is
already held, then the message propagation will block and stall bin
processing (state changes, other messages) and wpe servicing any events.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1490
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1934>
On renegotiation, or when the user has specified a mid for
a transceiver, we need to avoid picking a duplicate mid for
a transceiver that doesn't yet have one.
Also assign the mid we created to the transceiver, that doesn't
fix a specific bug but seems to make sense to me.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1902>
On an error event, epoll wait puts the failed socket in both readfds and
writefds. We can take advantage of this and avoid explicitly checking
socket state before every read or write attempt.
In addition, srt_getrejectreason() will give us more detailed
description of the connection failure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1943>
This function takes the sock lock. This can result in a deadlock when
another thread holding the sock lock is trying to take the object lock.
Thread A (Holds object lock, wants sock lock):
#2 gst_srt_object_get_stats at gst-plugins-bad/ext/srt/gstsrtobject.c:1753
#3 gst_srt_object_get_property_helper at gst-plugins-bad/ext/srt/gstsrtobject.c:409
#4 gst_srt_sink_get_property at gst-plugins-bad/ext/srt/gstsrtsink.c:95
#5 g_object_get_property from libgobject-2.0.so.0
Thread B (Holds sock lock, wants object lock):
#2 gst_element_post_message_default at gstreamer/gst/gstelement.c:2069
#3 gst_element_post_message at gstreamer/gst/gstelement.c:2123
#4 gst_element_message_full_with_details at gstreamer/gst/gstelement.c:2259
#5 gst_element_message_full at gstreamer/gst/gstelement.c:2298
#6 gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1407
#7 gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
#8 gst_srt_object_write_to_callers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
#9 gst_srt_object_write at gst-plugins-bad/ext/srt/gstsrtobject.c:1598
#10 gst_srt_sink_render at gst-plugins-bad/ext/srt/gstsrtsink.c:179
Fixes d2d00e07ac.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1861>
Various software, including ffmpeg's Decklink support, fails parsing CDP
packets that contain anything but CC data in the CDP packets.
Based on this property, timecodes are not written into the CDP packets
even if they're present.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
The base class is now a bin which wraps the `overlaycomposition`
element and implements the `draw` signal.
This way we support all the video formats the GstVideoOverlayComposition
API supports and the blending code can be reused. It is also possible
to have the blending happen in the sinks now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1829>
We calculate minimum of (stripe height * sub sampling) across all components
to ensure that all component dimensions are consistent with sub-sampling.
The last stripe for each component is simply the remaining height.
limit wavelet resolutions for "thin" stripes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1800>
Storing it per-stream requires taking the manifest lock which can apparenly be
hold for aeons. And since the QoS event comes from the video rendering thread
we *really* do not want to do that.
Storing it as-is in the element is fine, the important part is knowing the
earliest time downstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1021>
If an error happened switching to a new variant, we switch back to the previous
one ... except it will be unreffed when settin git.
In order to avoid such issues, keep a reference to the old variant until we're
sure we don't need it anymore
Fixes cases of double-free on variants and its contents
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1799>
LDAC is an audio coding technology developed by Sony that enables the
transmission of High-Resolution (Hi-Res) audio contents over Bluetooth.
Currently Adaptive Bit Rate (ABR) as supported by libldac encoder is not
implemented.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
libnice doesn't touch the kernel buffer sizes. When dealing with RTP data,
it's generally advisable to increase them to avoid dropping packets locally.
This is especially important when running multiple higher bitrate streams at
the same time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1366>
On shutdown, a previous iteration of dtsl_connection_process()
might be incomplete and leave a partial bio_buffer behind.
If the DTLS connection is already marked closed, drop out
of dtls_connection_process early without asserting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
The main context can disappear in gst_webrtc_bin_enqueue_task()
between checking the is_closed flag and enqueueing a source on the
main context. Protect the main context with the object lock instead
of the PC lock, and hold a ref briefly to make sure it stays alive.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
