Add a helper that submits and handles blocking preload requests for future
PART/MAP data from live playlists. Add handling in the hlsdemux stream to submit
preload requests when hitting the end of the available segments in a live
playlist.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add a flag to hlsdemux to enable or disable LL-HLS handling.
When LL-HLS is enabled and an LL-HLS playlist is loaded, use the part-hold-back
threshold to choose a starting segment.
For live streams that aren't LL-HLS, use the provided hold-back attribute, or
fall back to landing 3 segments from the end.
Make the gst_hls_media_playlist_seek() method able to choose a partial segment
within 2 target durations of the end of the playlist when requested.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Fix an off-by-one in gst_hls_media_playlist_sync_to_playlist() that would ignore
the first fragment in the reference playlist. The error was harmless, since we
expect the reference playlist to be older than the playlist we're
synchronising (so the first/oldest segment in the reference playlist will likely
not exist in the new playlist), so this is just for correctness.
Also fix a segment leak in gst_hls_media_playlist_advance_fragment() when
ignoring the partial_only segment.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add a function for synchronising current position with the contents of a
playlist that is specifically for that and can handle synchronising to a partial
segment.
gst_hls_media_playlist_seek() will be used only when performing external seek
requests, to find the best segment or partial segment at which to resume
playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Fixes for stream_time recalculation and handling in partial segments.
Disallow bitrate switching when in the middle of partial segments - only at a
full segment (or right before the first partial segment of a segment).
It's possible but more difficult to switch bitrates in the middle of a partial
segment group, since they are less likely to have aligned keyframes. In any
case, the seek code can't do that right now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
This will be used for CUDA stream sharing.
* Adding GstCudaPoolAllocator object. The pool allocator will
control synchronization of allocated memory objects.
* Modify gst_cuda_allocator_alloc() API so that caller can specify/set
GstCudaStream object for the newly allocated memory.
* GST_CUDA_MEMORY_TRANSFER_NEED_SYNC flag is added in addition to
existing GST_CUDA_MEMORY_TRANSFER_NEED_{UPLOAD,DOWNLOAD}.
The flag indicates that any GPU command queued in the CUDA stream
may not be finished yet, and caller should take care of the
synchronization.
The flag is controlled by GstCudaMemory object if the memory holds
GstCudaStream. (Otherwise, GstCudaMemory will do synchronization
as before this commit). Specifically, GstCudaMemory object will set
the new flag automatically when memory is mapped with
(GST_MAP_CUDA | GST_MAP_WRITE) flags. Caller will need to unset
the flag via GST_MEMORY_FLAG_UNSET() if it's already synchronized
by client code.
* gst_cuda_memory_sync() helper function is added to perform synchronization
* Why not use CUevent object to keep track of synchronization status?
CUDA provides fence-like interface already via CUevent object,
but cuEventRecord/cuEventQuery APIs are not zero-cost operations.
Instead, in this version, the status is tracked by using map and
object flags.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3629>
Usually gst-plugin-scanner.exe will be located under libexec/gstreamer-1.0
or even somewhere user specified location via GST_PLUGIN_SCANNER
environment. So, in order for child process to be able to load
GStreamer DLLs, parent process will need to update PATH env
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3886>
Add AVTP Raw Video Format de-payload support. The element supports only
GRAY16_LE output format, so:
- active pixels (no vertical blanking),
- progressive mode,
- 8 and 16-bit pixel depth,
- mono pixel format,
- grayscale colorspace.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335>
Add AVTP Raw Video Format payload support. The element supports only GRAY16_LE
input format, so:
- active pixels (no vertical blanking),
- progressive mode,
- 8 and 16-bit pixel depth,
- mono pixel format,
- grayscale colorspace.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335>
Due to a bug in the VT API, attempting to encode interlaced content
with ProRes results in an error, halting the pipeline instead of
gracefully falling back to software encoding.
Should be removed in the future if Apple ever fixes this issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3222>
The VA API has not defined the scaling list entries for U/V planes
for the 4:4:4 stream. In fact, we do not meet the 4:4:4 format output
for H264 so far, and scaling list is not used frequently, so we just
print out some warning and ignore these scaling list values.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3749>
We already have functions to generate a stream-id from pads but in the
end those pads are not even used in most cases. This adds functions to
generate a stream-id even before creating the source pads for the
element that is going to use it. For example a demuxer that is properly
implements the GstStream/GstStreamCollection API will not have a Pad but
already needs to generate a stream-id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3160>
The gst-devtools project generates gstreamer-validate-1.0.pc, this
must match the dependency in gst-editing-services for detection
to work properly.
Fixes:
Run-time dependency gst-validate-1.0 found: NO (tried pkgconfig and cmake)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3859>
All the RTP src pads were sharing the same stream-id while each actually
carry a different stream.
This was causing problem for example when funneling the streams together
and then trying to split them using 'streamiddemux'.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
In theory, `dispose()` functions should be idempotent and should be
prepared not to crash or cause a double-free if an unref done from
inside caused a recursive call to `dispose()` of the same object.
https://developer.gnome.org/gobject/stable/howto-gobject-destruction.html
This patch modifies the `dispose()` method to honor these constraints.
Since the double `dispose()` call won't actually occur in qtdemux (there
is no cycle detection mechanism that could invoke it to work that way),
this is more of a code cleanup than a user-facing problem fix.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3822>
* Extend protocol so that client can notify of releasing shared memory
* Server will hold shared memory object until it's released by client
* Add allocator/buffer pool to reuse shared memory objects and buffers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3765>
With the addition of the 'keep-aspect-ratio' sizing policy, content
that doesn't fit the target size is downscaled according to its own
aspect ratio to fit that target size, and centered.
Centering might not always be the desired behaviour, however;
consumers of this API might want to align the resulting picture to
the left or to the right.
To account for any of these cases, add two new properties to the
glvideomixer pad: xalign, and yalign. They operate on normalized
coordinates (0.0 for start, 1.0 for end), and default to 0.5 which
centers content.
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3762>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3762>
If we know there's only one stream we care about and we
don't have to synchronise audio and video, or send RRs,
we might just as well not hook up all the RTCP bits and
use fewer threads and sockets and simplify the pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3531>
