This simplifies the code a lot without any functional changes apart from
not closing the display connection. Closing the display connection is
not safe to do as it is shared between all other code in the same
process and no reference counting or anything happens at the platform
layer.
Move the package defines for GST_PLUGIN_DEFINE from the
command line into the source file to avoid quoting issues
(-DPACKAGE_NAME="foo" means the quotes won't actually make
it to the compiler and then it no longer gets a string constant).
1. Propagate the GstGLDisplay we create
2. Add the created GstGLContext to the propagated GstGLDisplay
Otherwise with multi-branch GL pipelines involving gtkglsink, things
will fall apart and errors will be genarated somewhere.
Except for gst/gl/gstglfuncs.h
It is up to the client app to include these headers.
It is coherent with the fact that gstreamer-gl.pc does not
require any egl.pc/gles.pc. I.e. it is the responsability
of the app to search these headers within its build setup.
For example gstreamer-vaapi includes explicitly EGL/egl.h
and search for it in its configure.ac.
For example with this patch, if an app includes the headers
gst/gl/egl/gstglcontext_egl.h
gst/gl/egl/gstgldisplay_egl.h
gst/gl/egl/gstglmemoryegl.h
it will *no longer* automatically include EGL/egl.h and GLES2/gl2.h.
Which is good because the app might want to use the gstgl api only
without the need to bother about gl headers.
Also added a test: cd tests/check && make libs/gstglheaders.check
https://bugzilla.gnome.org/show_bug.cgi?id=784779
This is useful for autoplay for example. With autoplay, it is necessary to
wait until the scene graph is fully set up. This signal is emitted once the
QML item node is ready. So, inside a connected slot, the pipeline's state
can be set to PLAYING to automatically start playback as soon as the QML
script is loaded.
https://bugzilla.gnome.org/show_bug.cgi?id=786246
This fixes a memory leak. When dropframe-threshold has been set,
libvpx may output less frames than the input ones, which causes
some GstVideoCodecFrames to queue up in GstVideoEncoder's internal
frame queue with no chance of ever being all released. And because
the frames keep references to the input buffers, the input buffer
pool keeps allocating new buffers and memory usage grows very fast.
For example the following pipeline's memory usage grows at a rate
of about 1GB per minute!
videotestsrc ! capsfilter caps=video/x-raw,width=1920,height=1080,framerate=30/1,format=I420 ! \
vp8enc target-bitrate=1000000 end-usage=cbr dropframe-threshold=95 ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=783086
Instead of just sending a sticky event with them downstream. This allows
getting the HTTP headers easily in the application, and especially also
on errors.
QML can destroy the video widget at any time, leaving
us with a dangling pointer. Use a lock and a proxy
object to cope with that, and block in the widget
destructor if there are ongoing calls into the widget.
Soup allows only up to two connections per host in a session,
if we use default value. When session sharing is used, however,
more connections might be required in a session.
(e.g., multi-audio adaptive streaming case)
https://bugzilla.gnome.org/show_bug.cgi?id=784495
Currently we only allowed HTTP proxy. Don't filter for the scheme, just check
if it looks like an URI. Soup will warn if the URI is invalid or if
proxy protocol is not supported. This enables using SOCKS 4/5 which is
directly implemented into GIO.
https://bugzilla.gnome.org/show_bug.cgi?id=783012
This allows timing out on network errors much earlier
(currently it takes ~15min to timeout) and we can still
unlock and change state in the meantime.
https://bugzilla.gnome.org/show_bug.cgi?id=571722
Tags are pushed to "videosrcpad"/"audiosrcpad" in
gst_dvdemux_add_pad() method, however they will be NULL
in this method, hence tags are not pushed.
Instead, send tag event to "pad" created gst_dvdemux_add_pad().
Signal no-more-pads when both pads are created
https://bugzilla.gnome.org/show_bug.cgi?id=743657
souphttpsrc now shares its SoupSession with other elements in the
pipeline via GstContext if possible (session-wide settings are all the
defaults), or if the context was forced by the application.
This allows multiple souphttpsrcs to reuse connections, cookies, etc.
https://bugzilla.gnome.org/show_bug.cgi?id=780140
Let libsoup handle redirection automatically.
And then, to figure out redirection uri, extract it on "restarted"
callback which will be fired before soup_session_send() is returned.
https://bugzilla.gnome.org/show_bug.cgi?id=778428
Fix a regression introduced by commit 183695c61a (refactor to use
Soup's sync API). The code previously attempted to reconnect when the
server closed the connection early, for example when the stream was put
in pause for some time.
Reintroduce this feature by checking if EOS is received before the
expected content size is downloaded. In this case, do the request
starting at the previous read position.
https://bugzilla.gnome.org/show_bug.cgi?id=776720
This check must be done only when we are sure the request was
successfully sent. soup_session_send() might fail without setting the
status code. In this case status code is 0 so we would only catch the
error after the seek range check. In this case we would report an error
saying that the seek range was not respected, instead of reporting the
underlying error that triggered the soup_session_send() failure.
https://bugzilla.gnome.org/attachment.cgi?bugid=777222
... so as to allow one clearly defined (absolute) positioning mode
that can cater for a variety of absolute but also relative positioning
with respect to edge or center.
The flow return values was stored in the element before because the
result had to be set from callbacks. This is not the case anymore, we
can return the flow result directly from functions, making the code
easier to understand.
https://bugzilla.gnome.org/show_bug.cgi?id=777222
The current code configures libsoup to handle redirections
transparently, without informing the caller, thus preventing the element
to record the redirect code and location uri.
Fix this by always setting the SOUP_MESSAGE_NO_REDIRECT, preventing
libsoup from handling the redirection. When we receive a redirection
request and libsoup can safely handle it, return a custom error which
triggers a retry with the new URI.
https://bugzilla.gnome.org/show_bug.cgi?id=777222
The encoder fails to initialise when we try to set GST_TAG_PREVIEW_IMAGEs
sent to use by qtdemux from iTunes-generated m4a files. We should
not just blindly translate the PREVIEW tag to file icon image types,
but check if the specific conditions required are met (i.e. image
type 1 must be a 32x32 PNG icon, and what we're getting is 500x500).
https://bugzilla.gnome.org/show_bug.cgi?id=776962
Especially don't put them into GstStructures in one way or another, just
ignore them or error out cleanly depending on the importance of their
content.
1. Need set use-default-fbo to qquickwindow during set property
to support change render target on the fly.
2. Calculate qmlglsrc refresh frame rate in qtglwindow
https://bugzilla.gnome.org/show_bug.cgi?id=774035
While computing the x and y offsets, it's the video resolution and
resized overlay resolution to be used instead of actual overlay image
resoltuion. Due to this, the overlay image used to get wrongly overlayed
in undesired location
https://bugzilla.gnome.org/show_bug.cgi?id=757292
souphttpsrc maintains two variables for the position:
* 'request_position' is where we want to be
* 'read_position' is where we are
During Normal operations both are updated in sync when data arrives. A seek
changes 'request_position' but not 'read_position'.
When the two positions get out of sync, then a new request is send and the
'Range' header is adjusted to the current 'request_position'.
Without this patch, if reading fails, then the source is destroyed. This
triggers a new request, but the range remains unchanged. As a result, the
old range is used and old data will be read.
Changing the 'read_position' to -1 makes it explicitly different from
'request_position' and as a result the 'Range' header is updated correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=773509
GstAudioRingBuffer doesn't needs us to have at least 2 segments. We make
sure that if our buffer parameters are such that the maxlength is not at
least 2x fragsize, we still request the ringbuffer to keep that much
space so it continues to work.
https://bugzilla.gnome.org/show_bug.cgi?id=770446
If jackd changes the buffer size or sample rate, jackaudiosink hangs
and can't be stopped. This also happens if jack is configured as slave
and a gstreamer pipeline is started on the slave machine while the jack
master isn't running yet. If the the jack master is started it changes
the buffer size / sample rate and jackaudiosink can't be stopped.
This fix calls jack_shutdown_cb when jack_sample_rate_cb or
jack_buffer_size_cb is called.
https://bugzilla.gnome.org/show_bug.cgi?id=771272
When start qmlglsink app, it will set NULL buffer to GstQSGTexture
in which case that qt_context_ will be a random value and cause
gst_gl_context_activate() fail.
https://bugzilla.gnome.org/show_bug.cgi?id=770925
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.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.
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.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.
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.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.
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.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.
At the end of a range request, we don't want to return GST_FLOW_EOS otherwise
the last bytes we just read will be dropped by basesrc.
Instead just return GST_FLOW_OK (which was set just before) and let basesrc
handle the fact we are at the end of the segment.
If we're at the end of a range request, read again to let libsoup
finalize the request. This allows to reuse the connection again later,
otherwise we would have to cancel the message and close the connection.
We have to get rid of the message on EOS when the complete stream is read to
remember that we successfully finished handling this specific message.
Otherwise we will cancel it later and close the connection instead of reusing
it at a later time.
It might also make sense to reuse connections if a non-200 response is
received. As long as there was no connection error, the HTTP connection should
be re-usable.
There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
Some elements might want to have that information.
Also remove downstream-only CAPS event handling and minimize code
Compiler would complain about include directory that didn't
exist because QPA_INCLUDE_PATH gets subst-ed regardless
(and if it didn't we'd have just an empty -I argument).
https://bugzilla.gnome.org/show_bug.cgi?id=767553
Update the blocksize depending on how much is obtained from a read
of the input stream. This avoids doing too many reads in small chunks
when larger amounts of data are available and also prevents using
a very large memory area to read a small chunk of data.
https://bugzilla.gnome.org/show_bug.cgi?id=767833
The current state of c++ ABI's on Window's and Gst's/Qt's conflicting
mingw builds means that we cannot use mingw for building the qt plugin.
Instead, a qmake .pro file is provided that is expected to be used with the
msvc binaries provided by Qt like so:
(with the PATH environment variable containing the path to the qt biniaries
and PKG_CONFIG_PATH containing the path to GStreamer modules)
cd /path/to/sources/gst-plugins-bad/ext/qt
qmake -tp vc
Then open the resulting VS project and build the library. Then
cp debug/libgstqtsink.dll /path/to/prefix/lib/gstreamer-1.0/libgstqtsink.cll
https://bugzilla.gnome.org/show_bug.cgi?id=761260
The heuristic to choose between packetise or not was changed to use the
segment format. The problem is that this change is reading the segment
during the caps event handling. The segment event will only be sent
after. That prevented the decoder to go in packetize mode, and avoid
useless parsing.
https://bugzilla.gnome.org/show_bug.cgi?id=736252
The heuristic to choose between packetise or not was change to use the
segment format. The problem is that this change is reading the segment
during the caps event handling. The segment event will only be sent
after. That prevented the decoder to go in packetize mode, and avoid
useless parsing.
https://bugzilla.gnome.org/show_bug.cgi?id=736252
Previously the segment.time was wrong, and the position was not updated
correctly, resulting in seeks in PUSH mode with upstream providing a BYTES
segment to not work at all.
https://bugzilla.gnome.org/show_bug.cgi?id=767157
This fixes seeking in DV streams where upstream operates in PUSH mode with a
TIME segment (e.g. avidemux). Without this, we would generate wrong durations
and timestamps after a seek.
https://bugzilla.gnome.org/show_bug.cgi?id=767157
When early returning in gst_soup_http_src_read_buffer() because the
element is FLUSHING, we need to unmap and unref the buffer which was just created.
https://bugzilla.gnome.org/show_bug.cgi?id=766718
Directly setting audio/x-raw caps leads to problems when the delivered
data blocks do not align properly at sample boundaries (for example, a
data block with 391 bytes). So, instead, set audio/x-unaligned-raw to
let a parser be autoplugged.
https://bugzilla.gnome.org/show_bug.cgi?id=689460
Non-blocking read will return the amount of data available without
blocking to wait for the full requested size.
The downside is that now it souphttpsrc needs to have a waiting
mechanism in case there is no data available yet to avoid busy
looping arond the inputstream.
The previous ones resulted in odd display aspect ratios and were different
from the ones used by e.g. ffmpeg. The new ones now result in display aspect
ratios of 4:3 and 16:9.
https://bugzilla.gnome.org/show_bug.cgi?id=765946
When a frame's duration is too low, calling gst_util_uint64_scale()
to scale its value can result into it being truncated to zero, which
will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error
when trying to encode.
To prevent this from happening, we simply ignore the duration when
encoding if it becomes zero after scaling, logging a warning message.
https://bugzilla.gnome.org/show_bug.cgi?id=765391
Fix syntax errors when compiling against cerbero-provided libmpg123
headers. We do the same as the libmpg123 internal visual studio
build here.
mpg123.h(1378): error C2143: syntax error: missing ')' before '('
mpg123.h(1378): error C2081: 'ssize_t': name in formal parameter list illegal
mpg123.h(1378): error C2143: syntax error: missing ')' before '*'
mpg123.h(1378): error C2091: function returns function
mpg123.h(1378): error C2143: syntax error: missing '{' before '*'
mpg123.h(1378): error C2059: syntax error: ')'
mpg123.h(1379): error C2143: syntax error: missing ')' before '*'
mpg123.h(1379): error C2365: 'off_t': redefinition; previous definition was 'typedef'
...
This reverts commit 0dd46accf6.
With some audiosinks, starting the ringbuffer on the first commit
causes audio glitches at startup by starting to output segments
from the ringbuffer before it has been filled / fully prerolled. This
doesn't usually happen with pulsesink because we map the pulseaudio
ringbuffer directly, but we should keep things consistent with
other sinks with regards to startup latency, plus it gives more
headway to avoid glitching, should the initial 2nd segment take
more than 10ms to generate.
https://bugzilla.gnome.org/show_bug.cgi?id=657076
Remove calls to gst_pad_has_current_caps() which then go on to call
gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
use gst_pad_get_current_caps() and check for NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=759539
rename gst-launch --> gst-launch-1.0
replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
fix caps in examples
https://bugzilla.gnome.org/show_bug.cgi?id=759432
The old code was using gst_caps_normalize() and was generally overly
complex. Simplify by picking sample rate and number of channels from
upstream and the sample format from the allowed caps. If the format caps
is a list of strins, just pick the first one. And if the srcpad isn't
linked yet, use the default format (S16).
https://bugzilla.gnome.org/show_bug.cgi?id=740195
It's possible that the sink element will be freed before the widget is
destroyed. When the widget was eventually destroyed, it was attempting to
access member variables of the freed sink struct which resulted in undefined
behaviour.
Fix by disconnecting our signal on finalize.
https://bugzilla.gnome.org/show_bug.cgi?id=762098
CPU waits are more expensive and are only required if the CPU is ever going to
access the data. GPU waits perform inter-context synchronisation and are cheaper
as they don't require CPU intervention.
We already pass the entire frame to the decoder. If the decoder ask for
more data, don't pass the same data again as this leads to infinit loop.
Instead, simply fail the fill function to signal the problem with that
frame. It will then be skipped properly.
https://bugzilla.gnome.org/show_bug.cgi?id=761670
All code paths for handle_frame() must somehow take ownership of the frame, be
it by actually unreffing, forwarding the frame elsewhere or storing it for
later.
http://bugzilla.gnome.org/show_bug.cgi?id=760666
