There's no guarantee it will *actually* be the URI which refered to what we are
downloading. It could be a stream URI or anything else.
Instead of putting something wrong, put no (specific) referer as a better choice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3972>
The flowcombiner and active_streams shouldn't be cleared in the
mse-bytestream variant, only in the mss-fragmented one. Otherwise the
soft reset leaves qtdemux in a state where it still believes that it has
streams, but they've been cleared. In that case, a null pointer
dereference happens and the app crashes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4199>
The abort() method of SourceBuffer in Media Source Extensions is
expected to flush the demuxer and discard the current fragment,
if any. The configuration of tracks, if any, should be preserved.
qtdemux has different behavior for flush events depending on the
context.
This patch activates the intended behaviour only for streams of the
VARIANT_MSE_BYTESTREAM type, conformant to the ISO BMFF Bytestream
specification[1]. This flush behaviour is the same as the one
already in use for adaptivedemux sources.
[1] https://www.w3.org/TR/mse-byte-stream-format-isobmff/https://bugzilla.gnome.org/show_bug.cgi?id=795424
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4101>
This patch prevents a possible race condition from taking place between the EOS event handling and rtcp send
function/thread.
The condition starts by getting the GST_EVENT_EOS event on the send_rtp_sink pad, which causes two core things
to happen -- the event gets pushed down to the send_rtp_src pad and all sessions get marked "bye" prior to
completion of the event handler. In another thread the rtp_session_on_timeout function gets called after an
expiration of gst_clock_id_wait in the rtcp_thread function. This results in a call to the
ess->callbacks.send_rtcp(), which is configured as a function pointer to gst_rtp_session_send_rtcp via the
RTPSessionCallbacks structure passed to rtp_session_set_callbacks in the gst_rtp_session_init function.
In the race condition, the call to gst_rtp_session_send_rtcp can have the all_sources_bye boolean set to true
while GST_PAD_IS_EOS(rtpsession->send_rtp_sink) evaluates to false. This is the result of gst_rtp_session_send_rtcp
running before the send_rtp_sink's GST_EVENT_EOS handler completes. The exact point at which this condition occurs
is if there's a context switch to the rtcp_thread right after the call to rtp_session_mark_all_bye in the
GET_EVENT_EOS handler, but before the handler returns.
Normally, this would not be an issue because the rtcp_thread continues to run and indirectly call
gst_rtp_session_send_rtcp. However, the call to rtp_source_reset sets the sent_bye boolean to false, which ends up
causing rtp_session_are_all_sources_bye to return false. This gets passed to gst_rtp_session_send_rtcp and the EOS
event never gets sent.
The race condition results in the EOS event never getting passed to the rtcp_src pad, which prevents the bin and
pipeline from ever completing with EOS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3798>
Makes "start-bitrate" work without setting "connection-speed" property. Having
another property set as a requirement for this one to work is unexpected.
This commit allows to request some initial bitrate for first segment, then
go into adaptive streaming for the rest of media playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3895>
A data offset with an offset smaller than the moof length is wrong
in smooth streaming streams. The samples will not be located and
eventually playback will error out. So compensate assuming data
is in mdat following moof.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
A data offset with an offset smaller than the moof length is wrong
in smooth streaming streams.
The samples will not be located and eventually playback will
error out. So compensate assuming data is in mdat following moof.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
this is an issue seen with musl based linux distros e.g. alpine [1]
musl is not going to change this since it breaks ABI/API interfaces
Newer compilers are stringent ( e.g. clang16 ) which can now detect
signature mismatches in function pointers too, existing code warned but
did not error with older clang
Fixes
gstv4l2object.c:544:23: error: incompatible function pointer types assigning to 'gint (*)(gint, ioctl_req_t, ...)' (aka 'int (*)(int, unsigned long, ...)') from 'int (int, int, ...)' [-Wincompatible-function-pointer-types]
v4l2object->ioctl = ioctl;
^ ~~~~~
[1] https://gitlab.alpinelinux.org/alpine/aports/-/issues/7580
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3950>
The av1C box is optional so dropping parsing does not break anything
fundamentally, and there seems to be no historical record how version 0
even looks like while the comments and the parsing disagreed with each
other.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3882>
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>
