| Commit message (Collapse) | Author | Age | Files | Lines |
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Those are just invalid and just reflect what we sent. We'd need to parse the
RTCP XR packets from the other side to know more about those.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
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And expose them as standardised webrtc statistics
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
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Store them as web refs to avoid having to worry about freeing later and because
the new-jitterbuffer is on a different thread
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
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Just document all the missing fields and document which ones will never
be implemented because they depend on the codec or depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
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The receiver doesn't have the information to compute it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
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RTCP mux is now always required by the WebRTC spec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1824>
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Only want to check for json-glib when libqrencode was found,
but also it shouldn't be required but depend on the option.
Fixes #1465
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1824>
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They're just subsets of the high profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
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Don't assume the available EGL display is a wayland display -
instead, check the the GStreamer GL context is EGL, and then
use gst_gl_display_egl_from_gl_display to create a
GstGLDisplayEGL from that, which also adds refcounting
around the underlying EGLDisplay.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1385
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1752>
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The decoder can simply read this offset after decoding
to know where to blit the stripe to the full frame
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1800>
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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>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1800>
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If encoder is in stripe mode, then downstream must also support stripe
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1800>
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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>
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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>
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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>
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Since !1366 nice_agent_get_sockets() is used, which requires 0.1.17.
Update the version requirement accordingly.
Fixes #1459.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1792>
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When switching variants, try to re-use existing streams/pads instead of creating
new ones. When dealing with urisourcebin and decodebin3 this is not only the
expected way but also avoids a lot of buffering/hang issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1757>
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The rest of the code might want to use this
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1757>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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This moves `gstqroverlay.c` to `gstdebugqroverlay.c` and implements
a simple `gstqroverlay` element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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The element is specially focus on debugging purposes and not a generique QR overlay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Otherwise zbar isn't able to read the produced qrcodes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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And reindent the .h file removing tabs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Matching usual namings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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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>
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Don't hold the object lock on the vk physical device while
constructing a GstVulkanDevice around it, as
GstVulkanDevice can make calls on the physical device that
require the object lock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1754>
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In the most of case, this typo would work. But for
ARGB8888 and XRGB8888, which shm format is not based on fourcc,
which would never appear in format enumeration.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1751>
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Call the parent state_change function first when changing state
downward, to make sure that the element has stopped before cleaning
it up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
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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>
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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>
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This matches how the WebRTC javascript API works and the Chrome implementation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
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When ttmlparse is in, e.g., an MPEG-DASH pipeline, there may be
whitespace between successive TTML documents in ttmlparse's accumulated
input. As libxml2 will fail to parse documents that have whitespace
before the opening XML declaration, ensure that any preceding whitespace
is not passed to libxml2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1539>
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The parser handles only one TTML file at a time, therefore if there are
multiple TTML documets in the input, parse only the first.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1539>
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Wrapper on the iSAC reference encoder and decoder from webrtc,
see https://en.wikipedia.org/wiki/Internet_Speech_Audio_Codec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1124>
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We would use a frame callback from the surface to indicate
that last buffer is rendered, but when we destroy the surface
and that callback is not back yet, it may cause the wayland event
queue crash.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1729>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1709>
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Seems like the examples don't appear in the generated docs because the Since tag
was badly positioned in the doc blurb.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1706>
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