Assigning TRUE (1) to a signed 1 bit integer will cause truncation
from 1 to -1 because the only non-zero value that can be stored is -1
due to how two's-complement works.
As this is a proper GObject let's not bother with all this and simply
use a normal gboolean instead.
../subprojects/gst-plugins-good/ext/pulse/pulsesink.c:1490:19: warning: implicit truncation from 'int' to a one-bit
wide bit-field changes value from 1 to -1 [-Wsingle-bit-bitfield-constant-conversion]
pbuf->in_commit = TRUE;
^ ~~~~
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4617>
The qt5 and qt6 plugins will now correctly error out if you enable the
option, and you can also now explicitly ensure that wayland, x11,
eglfs support is actually functional by enabling the options. It was
too easy to build non-functional support for these.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4537>
jackaudiosink and jackaudiosrc have a rank and might be plugged
as part of auto-plugging inside playbin and playsink or the
autoaudiosink/autoaudiosrc elements, so we don't really want to
spew ERROR log messages in that case, which is consistent with
what alsasink and pulseaudiosink do.
This is less noticable on Linux because pulseaudiosink has a
higher and alsasink which has the same rank comes before jack
in the alphabet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4545>
In order to provide build and provide the jack plugin with the prebuilt
binaries of gstreamer we distribute with releases, we can not depend
on an external dependency nor can we ship plugins linking to libraries
we don't provide.
We can also not provide jack ourselves, as it would likely cause a
mismatch with the jack daemon on the host.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4350>
Short-circuit parsing and recreating the playlist URI if
no HLS directives are going to be applied to it.
Fixes problems playing some streams (YouTube) that have
unneeded escaped characters in the URI and then complain
when GStreamer removes the escaping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4335>
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>
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>
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>
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>
Without this, the plugin cannot be loaded in a devenv because the
RPATH is not added to the plugin dylib. This RPATH will be stripped on
install, which is what we want.
When deploying apps, people are supposed to use `macdeployqt` to
create an AppBundle that bundles Qt for you and sets the RPATHs
correctly to point to that bundled Qt.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3708>
When calculating the presentation offset for CMAF input in live
playback, subtract the stream_time of the fragment from the
calculated presentation offset, so that the first fragment
is played at running time zero.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3680>
jpegdec is capable to parse input frames, but if jpegparse is before,
there's no need to reparse frames. This patch configure jpegdec as
packetized, skipping parsing, if negotiated sink caps has the boolean
field 'parsed' as true.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2464>
According to comment in gst_pulsering_stream_latency_cb, latency updates
happen every 100 ms. The code in gst_pulsering_stream_latency_cb does
not care about the actual state of the ringbuffer, pbuf->acquired or
not.
Thus, every 100 ms the ringbuf->segdone may be set, even though the
object itself might be in 'destroyed' state. On next
gst_pulseringbuffer_acquire the segdone is not touched, so playback may
resume with invalid/wrong segdone value. This finally leads to a period
of silence playing after resuming the pipeline.
The problem was found on 'not-yet-released'-hardware and so far was not
reproducible on desktop computer.
Removing the callback as long as the ringbuffer is not in 'acquired'
state solves the problem reliably on the hardware device that the issue
was detected on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3082>
The stream selection is done on the currently outputting tracks, but in order to
(de)activate the backing streams we can only do it if the input and output
period are identical.
Fixes crash when doing stream selection during period migration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3525>
- Based heavily on the existing Qt5 integration however:
- The sharing of OpenGL resources is slightly different
- The integration with the scengraph is a bit different
- Wayland, XCB and KMS have been smoke tested. Android, MacOS/iOS,
Windows may or may not work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3281>
Instead of returning a "const gchar" or a "gchar" that should not be freed, always
return a duplicated string as those functions were used together with g_strdup anyway.
This is needed to prepare support for returning modified strings in next commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>
Unlike the legacy elements, GstAdaptiveDemuxStream is a GObject now,
so a bunch of things that were actually stream methods on the
parent demux object can directly become stream methods now.
Move the stream class out to a header of its own.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
Sometimes g_input_stream_read_all_finish() can return
0 bytes, but still succeed (return TRUE) and have more
data available later. Only finish the transfer
if it returns 0 bytes *and* FALSE with no error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
The cancelled flag was only set in the stream finalize()
method, after all activity on the stream has stopped anyway.
Replace uses of cancelled with checks on the stream state.
Remove the replaced flag, which was checked but never set
to TRUE anywhere any more.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
When matching segments across playlists with Program-Date-Times,
use the difference in segment PDTs to adjust the stream time
that's being transferred. This can fix cases where the
segment boundaries don't align across different streams
and the first download gets thrown away once the PTS
is seen and found not to match.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3309>
Check whether the init file / MAP data for a segment
is different to the current data and trigger an
update if so. Previously, the header would only
be checked in HLS after switching bitrate or
after a seek / first download.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3307>
Previously the minimum buffering threshold was hardcoded to a specific
value (10s). This is suboptimal this an actual value will depend on the actual
stream being played.
This commit sets the low watermark threshold in time to 0, which is an automatic
mode. Subclasses can provide a stream `recommended_buffering_threshold` when
update_stream_info() is called.
Currently implemented for HLS, where we recommended 1.5 average segment
duration. This will result in buffering being at 100% when the 2nd segment has
been downloaded (minus a bit already being consumed downstream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3240>
These values will be referred to as timestamp relative to period start
so need to subtract period start time from the values.
Fixes a problem with determining the start position when playing Live content
with SegmentTimeline, presentationTimeOffset and a non-0 period start time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3025>
Change the way streams are woken up to download more data.
Instead of checking the level on tracks that are being
output as data is dequeued, calculate a 'wakeup time'
at which it should download more data, and wake up
the stream when the global output position crosses
that threshold.
For efficiency, compute the earliest wakeup time
for all streams and store it on the period, so the
output loop can quickly check only a single value
to decide if something needs waking up.
Does the same buffering as the previous method,
but ensures that as we approach the end of
one period, the next period continues incrementally
downloading data so that it is fully buffered when
the period starts.
Fixes issues with multi-period VOD content where
download of the second period resumes only after
the first period is completely drained.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3055>
Some servers can return playlists with "old" media playlists and different
Discont Sequence.
In those cases, the segment stream times would be negative when creating a new
time mapping. In order to properly handle such scenarios, shift the values to
stored accordingly to end up with non-negative reference stream time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3054>
When advancing fragment in live, it's normal to return
GST_FLOW_EOS when playing at the live edge of the available
fragments. In that case, we still want to adjust bitrate
dynamically.
Fixes issue with dashdemux2 where the current bitrate of
each adaptation set is changed to the lowest one when
updating the mpd for a live stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3020>
Just like for the seconds field, there are no limitations on the hours and
minutes fields. The specification for xml schema duration fields doesn't forbid
specifying durations with only (huge) minutes or hours values.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2951>
When updating a manifest during live playback, preserve the current
representation for each stream.
During update_fragment_info, if the current representation changed
because it couldn't be matched, trigger a caps change and new
header download.
This reverts commit e0e1db212f
and reapplies "dashdemux: Fix issue when manifest update sets slow start
without passing necessary header & caps changes downstream" with
changes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2920>
libsoup 3.0.x dispatches using a single source attached when the session
is created, so we need to create the session with the same context that
our download thread is later using.
2.74 or 3.1 will dispatch a response using the context which sent the
request. However, for any context other than the one that created the
session, this will also create and destroy sources, so there's still
some slight performance benefit.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1384
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2913>
Handle select-streams and seek events in an element
level send_event() vfunc, so they can be received
before any source pads are created.
This allows preferred streams to be selected before
segment downloading starts.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2912>
When playing live, it's possible that one stream reaches
the end of the available playback window and goes to sleep
waiting for a manifest update, and the manifest update
introduces a new period. In that case, the sleeping
stream needs to wake up and go 'properly' EOS before we
can advance the input to the new period.
Accordingly, make sure that a stream's last_ret value
is not marked as EOS if it's just sleeping waiting for a live
manifest update.
Also fix the output loop to go back and re-check if it's
time to switch to the next period after dequeuing and
discarding an EOS event.
https://livesim.dashif.org/livesim/periods_20/testpic_2s/Manifest.mpd
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2895>
When returning GST_ADAPTIVE_DEMUX_FLOW_RESTART_FRAGMENT
for the first segment data, we might need to requeue the
header.
This was leading to occasional prerolling stalls on
HLS live streams with renditions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2849>
Make sure gst_adaptive_demux_loop_cancel_call()
never tries to operate on an invalidated main context. Make
sure to clear the main context pointer while holding the lock,
and to check it in gst_adaptive_demux_loop_cancel_call()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2847>
