Since b76d336549
pads are deactivated when going to READY but in `uridecodebin(3)`, the
sources source pads are activated while in NULL state (when PULL mode is
supported), meaning that we are ending up deactivating those pads in
NULL_TO_READY, breaking the pipeline.
The intent of the commit mentioned above is to ensure that the pads are
deactivated either in PAUSED_TO_READY or READY_TO_READY, so it should
be safe to avoid deactivating in NULL_TO_READY.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3849>
Generating the source element is done when uridecodebin is doing the
READY to PAUSED state change, so it is reasonable to set the new source
element to that state.
This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3857>
Do not store cached EGL images in GstMemory QData. Instead, use a
per-DmabufUpload GHashTable to store cache entries with a weak
reference to the GstMemory.
This allows two glupload elements on separate tee branches to have
their own EGL image cache. For this pipeline:
gst-launch-1.0 v4l2src ! tee name=t \
t. ! queue ! glupload ! fakesink
t. ! queue ! glupload ! fakesink
this gets rid of the occasional critical error message:
GStreamer-CRITICAL **: 08:26:33.194: gst_mini_object_unref: assertion 'GST_MINI_OBJECT_REFCOUNT_VALUE (mini_object) > 0' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3880>
When using qtdemux in a pipeline that should only work as a pure demuxer (not
for actual playback), qtdemux shouldn't emit new GstSegments to correct
the start time (jump to the future) to ensure that the user experiences no
playback delay. By doing so, it's generating the wrong segments when an append
of data from the past happens. When that happens, downstream elements such as
parsers (eg: aacparse) may clip those buffers laying before the GstSegment and
create problems on the GStreamer client app (eg: WebKit).
Getting buffers clipped out because of the wrong GstSegments started becoming
a problen when this commit was introduced:
ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it
This clipping makes test DASH shaka 35 from MVT tests[1] to fail in
WebKitGTK/WPE (at least) and can potentially cause a number of other problems
in the WebKit Media Source Extensions (MSE) code.
Note that this new behaviour of not emitting new GstSegments only makes sense
when qtdemux is being used as a pure demuxer and not as part of a regular
pipeline. This is why the variant field has been added. When equal to
VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode,
taking decisions that meet the expectations for an MSE-like processing mode.
This kind of tweaks have been done in the past for MSS streams, for instance.
That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of
its own dedicated boolean flag.
Co-authored by: Alicia Boya García <ntrrgc@gmail.com>
...who suggested to use "variant: mse-bytestream" in the caps to identify that
mode, as proposed in her unmerged patch:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467
[1] https://github.com/rdkcentral/mvt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3867>
The live playlists should be updated at a defined interval. The problem is that
this interval was used *after* the playlist was finally received and processed,
which resulted in a gradual shift happening in playlist updates.
Instead store and use the time at which playlists were requested to determine
when the next one should be downloaded.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
The scanning is done in a reverse order, the proper full checks to do are
therefore:
* If the position is beyond half a "segment duration", it's in the following
segment
* If the position is within the first half of a segment, it's in that one
* If the segment is the first one and the position is within half a duration
backwards, we consider the position as being within that first segment
Also handle the case where a "partial only" segment doesn't have a reliable
duration, and therefore use the playlist target duration instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
The implementation wouldn't work with regular HLS streams (i.e. the final
fallback).
Now that the implementation uses time to search for the starting
segment (instead of just the n-th from the end), we can specify the correct
hold_back fallback value from the RFC
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Avoid a deadlock if a downstream seeking query happens while the scheduler
thread is holding the manifest lock (for example during a seek back to live).
Instead, do a more elaborate fix where the external calls that need access to a
'manifest' access a copy that's updated during a manually triggered manifest
update callback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Rename track_dequeue_data_locked() to
gst_adaptive_demux_track_dequeue_data_locked(), since it's non-static.
Make find_stream_for_track_locked() static since it's only used in the main
gstadaptivedemux.c file.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
gst_adaptive_demux2_stream_finish_download() will already schedule another
fragment download if it can so don't fall through to the retry code that will
also try and schedule a download (triggering an assert).
Fix the logic in general to retry advancing into the live seek range once.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
When calculating the seek range for a live stream, use the same hold-back logic
as when choosing a starting segment, including low-latency segments if
enabled. Permits seeking closer to the live edge when re-synching or catching
up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
When playing at the live edge of a live playlist, and a download fails, we don't
expect there to be a next fragment. That case is handled lower down anyway, so
don't retry infinitely on spurious http errors at the live edge.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
gst_hls_demux_stream_has_next_fragment() can be called with a NULL
current_segment if we're past the end of the current playlist. In that case,
just return FALSE instead of hitting a critical in the playlist code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
When playing LL-HLS playlists in LL-HLS mode, update the playlist more often (on
the partial segment interval) or else we end up downloading them in bursts and
playing further from the live edge than intended.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
When playing a live stream, make the recommended buffering threshold at most the
hold-back distance from live. If we start 3 seconds from the live edge, there's
no point trying to buffer more - we'll just hit the live edge and have to wait
for more data to be available anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add a field to the DownloadRequest that reports the most recent time at which
data arrived. Update it in the DownloadHelper.
Add a method to retrieve the GST_BUFFER_OFFSET() for the DownloadRequest's data
buffer (if any).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
After cancelling a DownloadRequest, the download helper may not do so
immediately, so we can't assert on the in_use flag. Also, since there's no
refcount on the preload hint struct in the download request callback data, make
sure no callbacks will be dispatched when we're going to free the preload hint
struct.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Implement fulfilment of HTTP requests from the active preload downloads by
finding any preload request that can provide the requested data and feeding
bytes from the internal DownloadRequest to the caller provided target
DownloadRequest.
Doesn't yet calculate timestamps to make the target request have a sensible
apparent bitrate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add download_request_take_buffer_range() and
download_request_get_bytes_available() methods.
download_request_take_buffer_range() takes bytes from the front of the request
that satisfy the requested start/end byterange, and puts any remaining bytes
back into the DownloadRequest
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
