The audio packet times can be completely unrelated to the video stream
time, depending on the card. While this looks like a bug in the driver,
just always using the video stream time (which is correct) works as a
workaround for now.
This patch bumps the required meson to 0.40.1 as gstreamer core just
did, and cleanup some code to use a feature from 0.37 that allow
specifying version range when checking dependency.
https://bugzilla.gnome.org/show_bug.cgi?id=780654
Otherwise fall back to glDrawBuffers. Also check if glReadBuffer exists
before using it.
glDrawBuffer does not exist for GLES, only glDrawBuffers does.
https://bugzilla.gnome.org/show_bug.cgi?id=782376
This commit fixes the following assumptions with live seeking:
1) start was always valid and of type GST_SEEK_TYPE_SET
2) direction was always forward
3) stop should be offsetted when handling non-accurate seeks before
the range start position.
In order to handle more live seeking use-cases (including reverse playback),
only do non-accurate start/stop value clamping for GST_SEEK_TYPE_SET values.
Also add a bit more debugging lines for issues
https://bugzilla.gnome.org/show_bug.cgi?id=782330
When dealing with live streams, we can't rely on GstSegment calculation
since it uses the segment duration to calculate the absolute values.
But since we are dealing with live *and* we know the ranges, we can
compute the absolute seeking values using the range stop (i.e. "now")
as the END position.
Allows seeking back to "live" by using start_type:GST_SEEK_TYPE_END
and start:0
https://bugzilla.gnome.org/show_bug.cgi?id=782228
We were only ignoring the listed msvc warnings for C language
files and not C++. This was working by the coincidence that we did
not have any instances of these warnings in C++ files. Lately the
build of decklink has been fixed on windows, and it has an
instance of one of these warnings in a C++ file.
https://bugzilla.gnome.org/show_bug.cgi?id=782345
Earlier, the plugin was ignoring those settings and blindly setting
buffer-time to 2 seconds and latency-time to 200ms, which forced all
pipelines to have a minimum latency of 200ms + sink latency.
The values of segsize and segtotal were also not derived correctly.
Now we obey these values, and you can get close to the previous
behaviour by setting buffer-time and latency-time manually. Note that
they are set in microseconds.
As a consequence, when we haven't received enough data from the
device, we now sleep for a time proportional to the data remaining.
However, Directsound is a deprecated API so it maintains its own
software ringbuffer which updates at arbitrary intervals. Hence we
might have to wait a full segsize to get the last 10% of data. To
avoid tight loops, we clamp our sleep floor at 10ms.
In my testing, this keeps the wakeups not-too-high (proportional to
the latency-time set on the source). Further improvements should be
made by fixing the WASAPI audio source plugin instead of this.
Directsound is deprecated and as the comments explain, it is
impossible to get low latency, decent quality, or good performance
from it.
Based on a patch by Sebastian Dröge <sebastian@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=781249
The allowed live seek ranges returned by subclasses are "inclusive", that is
to say that the "range_stop" value they return is the highest acceptable position
one can seek to (i.e. "now").
Allow seeking to exactly that value
This reverts commit 845832263b.
The commit broke cross-mingw CI:
https://ci.gstreamer.net/job/GStreamer-master/8659/console
It seems that cross-mingw on Autotools and native-mingw on Meson
disagree about the size of HRESULT. Revert for now till I can
investigate the Meson side of things some more.
MinGW does not provide comsupp.lib, so there's no implementation of
_com_util::ConvertBSTRToString. Use a fallback implementation that
uses wcstombs() instead.
On MinGW we also truncate the name to 100 chars which should be fine.
The QTKit framework had been deprecated for long in favour of AVFundation
framework and we already have avfvideosrc that provides the same
functionality.
https://bugzilla.gnome.org/show_bug.cgi?id=782078
MediaCodec gives us a presentation timestamp of 0 if it does not know
anything, but GStreamer gives us GST_CLOCK_TIME_NONE. Don't mix up these
two.
https://bugzilla.gnome.org/show_bug.cgi?id=780190
A common subtitling use case is live-generated subtitles, in which each
new word is contained in its own span, and the spans are displayed
sequentially, with the effect that lines of displayed subtitles are
built up word-by-word.
This can, however, cause problems when the number of words in a block is
greater than the number of allowed GstMemorys in a GstBuffer.
Since in this use case each span will have the same styling as adjacent
spans, we can join adjacent spans (and other inline elements, such as
breaks) into a single element containing the concatenated text of each,
thus avoiding the limit of GstMemorys in a GstBuffer and also reducing
the amount of styling/layout metadata that is attached to each buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=781725
The parser stores the text from each inline element of a scene in its
own GstMemory, which is inserted in the GstBuffer containing the scene
data. However, GstBuffers can contain only a limited number of
GstMemorys. Therefore, don't add more than the maximum number of
GstMemorys to each buffer, and warn if this is attempted.
https://bugzilla.gnome.org/show_bug.cgi?id=781725
When parsing <br> elements, store an actual newline in the text field of
the created TtmlElement. They then don't need to be treated as a
separate case from anon-span elements when being processed.
https://bugzilla.gnome.org/show_bug.cgi?id=781725
Encapsulates in a function the code that warns of an illegally
positioned element, rather than repeating the same code multiple times.
Also frees a string allocated by ttml_get_element_type_string, which was
previously being leaked.
https://bugzilla.gnome.org/show_bug.cgi?id=781725
../subprojects/gst-plugins-bad/ext/smoothstreaming/gstmssdemux.c: In function ‘gst_mss_demux_requires_periodical_playlist_update’:
../subprojects/gst-plugins-bad/ext/smoothstreaming/gstmssdemux.c:729:16: error: unused variable ‘mssdemux’ [-Werror=unused-variable]
GstMssDemux *mssdemux = GST_MSS_DEMUX_CAST (demux);
^~~~~~~~
cc1: all warnings being treated as errors
Rationale is to allow the manifest update task to continue running while
seeks are occurring. Otherwise, if the user reliably performs a seek
before the manifest is updated, then as the manifest task is reset on
seeks (and thus the time to wait between manifest updates), the manifest
would never be updated.
This fix makes the manifest update task free-running and continously
update even during seeks.
Without this, for streams where the content is stored indefinitely and
can be seeked on, the duration would never increase when in paused or,
until we reached near the end of the currently advertised stream (where
the internal fragment parser would see descriptions of new fragments).
The TTML spec has an issue in which tab (U+0009) characters that are
first in a sequence of whitespace characters are not suppressed at the
start and end of line areas. This issue was reported in [1] and the
editor of the TTML specs confirmed that this was not the intention
behind the spec.
The editor has created an issue to fix this in both the TTML1 and TTML2
specs [2], giving a proposal of what the spec should say. This patch
updates ttmlparse to implement the intended behaviour as proposed, in
which tabs in the input are converted to spaces before processing.
[1] https://github.com/w3c/imsc/issues/224
[2] https://github.com/w3c/ttml1/issues/235https://bugzilla.gnome.org/show_bug.cgi?id=781539
