low/high-watermark are of type double, and given in range 0.0-1.0. This
makes it possible to set low/high watermarks with greater resolution,
which is useful with large multiqueue max sizes and watermarks like 0.5%.
Also adding a test to check the fill and watermark level behavior.
https://bugzilla.gnome.org/show_bug.cgi?id=770628
To make the code clearer, and to facilitate future improvements, introduce
a distinction between the buffering level and the buffering percentage.
Buffering level: the queue's current fill level. The low/high watermarks
are in this range.
Buffering percentage: percentage relative to the low/high watermarks
(0% = low watermark, 100% = high watermark).
To that end, get_percentage() is renamed to get_buffering_level(). Also,
low/high_percent are renamed to low/high_watermark to avoid confusion.
mq->buffering_percent values are now normalized in the 0..100 range for
buffering messages inside update_buffering(), and not just before sending
the buffering message. Finally the buffering level range is parameterized
by adding a new constant called MAX_BUFFERING_LEVEL.
https://bugzilla.gnome.org/show_bug.cgi?id=770628
When calculating the high_time, cache the group value in each singlequeue.
This fixes the issue by which wake_up_next_non_linked() would use the global
high-time to decide whether to wake-up a waiting thread, instead of the group
one, resulting in those threads constantly spinning.
Tidy up a bit the waiting logic while we're at it.
With this patch, we go from 212% playing a 8 audio / 8 video file down to less
than 10% (most of it being the video decoding).
https://bugzilla.gnome.org/show_bug.cgi?id=770225
This just confuses people, they look at it and try to call it
directly by name, instead of using the public GstElement API.
It stands to reason that it goes without saying that when an
element provides request pads that they can actually be
requested using the standard API, and there's no point in
printing internal implementation details of the element.
In many parts of the code we raise streaming error when the flow
goes wrong, and each time we create more or less similare error
message. Also that message does not let the application know what
has actually gone wrong. In the new API we add a "flow-return" detail
field inside the GstMessage so that the application has all the information
if it needs it.
API:
GST_ELEMENT_FLOW_ERROR
https://bugzilla.gnome.org/show_bug.cgi?id=770158
We only use GST_EXPORT consistently when building with MSVC by using the
visual studio definitions files (win32/common/*.def), so always disable
it when building with Autotools and only enable it with Meson when
building with MSVC.
This allows you to use MinGW to link to a GStreamer built with MSVC and
get the correct function prototypes to find functions and variables in
DLLs.
Fixes g-i warning "Gst: Constructor return type mismatch
symbol='gst_element_message_new_details' constructed='Gst.Element'
return='Gst.Structure'".
This is a newly-added function in git that has not been in a stable
release yet, so it's fine to rename it. It's also only used indirectly
via macros.
low/high-watermark are of type double, and given in range 0.0-1.0. This
makes it possible to set low/high watermarks with greater resolution,
which is useful with large queue2 max sizes and watermarks like 0.5%.
Also adding a test to check the fill and watermark level behavior.
https://bugzilla.gnome.org/show_bug.cgi?id=769449
To make the code clearer, and to facilitate future improvements, introduce
a distinction between the buffering level and the buffering percentage.
Buffering level: the queue's current fill level. The low/high watermarks
are in this range.
Buffering percentage: percentage relative to the low/high watermarks
(0% = low watermark, 100% = high watermark).
To that end, get_buffering_percent() is renamed to get_buffering_level(),
and the code at the end that transforms to the buffering percentage is
factored out into a new convert_to_buffering_percent() function. Also,
the buffering level range is parameterized by adding a new constant called
MAX_BUFFERING_LEVEL.
https://bugzilla.gnome.org/show_bug.cgi?id=769449
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
This makes gstconfig.h completely arch-independent. Should cover all
compilers that gstreamer is known to build on, and all architectures
that I could find information on. People are encouraged to file bugs if
their platform/arch is missing.
In ringbuffer mode we need to make sure we post buffering messages *before*
blocking to wait for data to be drained.
Without this, we would end up in situations like this:
* pipeline is pre-rolling
* Downstream demuxer/decoder has pushed data to all sinks, and demuxer thread
is blocking downstream (i.e. not pulling from upstream/queue2).
* Therefore pipeline has pre-rolled ...
* ... but queue2 hasn't filled up yet, therefore the application waits for
the buffering 100% messages before setting the pipeline to PLAYING
* But queue2 can't post that message, since the 100% message will be posted
*after* there is room available for that last buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=769802
A new event which precedes EOS in situations where we
need downstream to unblock any pads waiting on a stream
before we can send EOS. E.g, decodebin draining a chain
so it can switch pads.
https://bugzilla.gnome.org/show_bug.cgi?id=768995
Redirection messages are already used in fragmented sources and in
uridecodebin, so it makes sense to introduce these as an official message
type.
https://bugzilla.gnome.org/show_bug.cgi?id=631673
Other pads that are waiting for the stream on the selected
pad to advance before they finish waiting themselves
should be given the chance to do so when the selected pad
goes EOS. Fixes problems where input streams can end up
waiting forever if the active stream goes EOS earlier than
their own end time.
In some corner cases, the error 'code' part passed to
GST_ELEMENT_ERROR() is a valid define as well, in which
case it won't survive two levels of macro expansion, but
only one. Fixes:
oss4-sink.c: In function ‘gst_oss4_sink_open’:
error: ‘GST_RESOURCE_ERROR_0x00000002’ undeclared (first use in this function)
GST_ ## domain ## _ERROR_ ## code, __txt, __dbg, __FILE__,
which is from GST_ELEMENT_ERROR(el,RESOURCE,OPEN_WRITE,..)
and OPEN_WRITE happens to be defined to 2 here.
https://bugzilla.gnome.org/show_bug.cgi?id=756806https://bugzilla.gnome.org/show_bug.cgi?id=769117
gst_structure_id_get() returns a new reference so the returned object is
actually (transfer full).
The unit tests was already unreffing the objects.
https://bugzilla.gnome.org/show_bug.cgi?id=768776
gst_structure_id_get() returns a new reference so the returned device is
actually (transfer full).
The code using this API was already correct but the code example in
comments was not.
https://bugzilla.gnome.org/show_bug.cgi?id=768776
If segment.stop was given, and the subclass provides a size that might be
smaller than segment.stop and also smaller than the actual size, we would
already stop there.
Instead try reading up to segment.stop, the goal is to ignore the (possibly
inaccurate) size the subclass gives and finish until segment.stop or when the
subclass tells us to stop.
When dealing with small-ish input data coming into queue2, such as
adaptivedemux fragments, we would never take into account the last
<200ms of data coming in.
The problem is that usually on TCP connection the download rate
gradually increases (i.e. the rate is lower at the beginning of a
download than it is later on). Combined with small download time (less
than a second) we would end up with a computed average input rate
which was sometimes up to 30-50% off from the *actual* average input
rate for that fragment.
In order to fix this, force the average input rate calculation when
we receive an EOS so that we take into account that final window
of data.
https://bugzilla.gnome.org/show_bug.cgi?id=768649
