Effective for the case where we have a platform that does not
require a native window. We require a mainloop to run the GL
commands which is currently operated by GstGLWindow.
Intel drivers require the display handles be the same for context
sharing to occur. Also solves some cases of use after free of the
display when integrating with gstreamer-vaapi.
See https://bugs.freedesktop.org/show_bug.cgi?id=41736 for the intel bug.
GstGLDisplayX11 holds the display connection and name. Each thread requires
it's own X11 Display connection (initialised from name) due to the fact that
we do not want to call XInitThreads(). Doing so would result in segfaults
when integrating with GUI toolkits Gtk, Qt, etc.
The Display connection is for OpenGL platforms where a constant display is
required in order to share contexts (egl). In the case of a wrapped context
(added later), we do not have GstGLWindow to retreive the display from so a
'master' connection is used instead.
We can transform from any input in our caps to any output.
With the following pipeline snippet:
... ! vaapidecode ! glcolorscale ! xvimagesink
GstVideoGLTextureUploadMeta was being used on both src and sink
pads causing linking to fail. This allows the usage of the meta
on either pad without affecting whether the meta is chosen on the
other pad.
the 16bit data is uploaded as LUMINANCE_ALPHA, then expanded, composed
in shader. value weight is a little complicate, high byte weight is
255*256/65535 (denormalize to [0~255] ,shift to high byte,then normalize
to [0~1]), low byte weight is 255/65535(similar)
https://bugzilla.gnome.org/show_bug.cgi?id=722670
We create our textures (in Desktop GL) with GL_TEXTURE_RECTANGLE,
vaapi attempts to bind our texture to GL_TEXTURE_2D which throws a
GL_INVALID_OPERATION error and as thus, no video.
Also, by moving exclusively to GL_TEXTURE_2D and the npot extension
we also remove a difference between the Desktop GL and GLES2 code.
https://bugzilla.gnome.org/show_bug.cgi?id=712287
gstgldownload.c:846:22: error: argument to 'sizeof' in 'memcpy' call is the same expression as the source; did you mean to provide an explicit length? [-Werror=sizeof-pointer-memaccess]
Write forward declarations in another way to avoid
repeated typedefs "error: redefinition of typedef".
Raised when using i686-apple-darwin11-llvm-gcc-4.2
It seems that C apple compiler does not support
C11 feature.
https://bugzilla.gnome.org/show_bug.cgi?id=703885
Indeed 'glXMakeCurrent (window_x11->device, None, NULL);'
has to be called in the thread where the glcontext is actually
current.
Otherwise glXDestroyContext may crash.
1) Need to include GL/glext.h on W32 to provide necessary macros
2) W32 macros that pre-processor has by default are different for different
toolchains. Borland uses __WIN32__, everyone else has _WIN32, so check both.
Side-note: glext.h is not supplied by mingw-w64 at the moment, but can be
downloaded from http://www.opengl.org/registry/https://bugzilla.gnome.org/show_bug.cgi?id=703364
We cannot use _finalize as this will be called before GstGLWindow's
_finalize. This causes us to try and perform operations with the
wayland server when the connection has been severed.
The window's loop could be still running when _finalize is called
and if we destroy the display connection it will never be closed.
Add _open and _close vfuncs to GstGLWindow so that subclasses can
start up and shutdown at the right time.
previously, on context creation, when we failed to get a valid context
we would still atempt to run the window mainloop as well as setting an error
on the display. This would cause the calling element to finalize the display
and therefore attempt to quit the window mainloop. However the mainloop
may not have started running. Thus when the window mainloop ran it would
never a get a quit message and never end.
GNUstep has a signal handler for SIGINT which runs in the main thread.
However that thread has never been 'registered' with GNUstep so
the autorelease magic of objective-c causes a segfault from accessing
a null NSThread object somewhere deep in GNUstep.
- Make GstGLWindow subclassablerather than specified at compile time.
- Add GstGLWindowX11 for x11 windows and two subclasses, GstGLWindowX11GLX
and GstGLWindwX11EGL for GLX and EGL repectively. (win32 and cocoa
ports to come)
- Also cleanup GL library detection in configure.ac
rename functions so that other code can create textures in the GL thread
change upload functions to take a GstVideoFrame
default to GLSL for upload conversion
Fix bug #310775
gst-launch audiotestsrc ! libvisual_gl_projectM ! glimagesink is working
but for now you cannot append any other opengl filters between
libvisual_gl_projectM and glimagesink because our FBO is turned OFF.
It would require that libvisual allows to split rendering between
pass1,2,3... and final rendering. In order to unbind our FBO before
the passN, and then rebind it just before the final libvisual rendering.
- cmake could not find glib
- put gtk variables at the beginning to avoid GL conflicts
- update examples to clutter-1.8
- use const instead of deprecated G_CONST_RETURN
- set max pending events to 0 to make cube example works again
Add a new convenience function in GstGLFilter that just draws an input
texture to a target texture using a simple shader with just a "tex"
uniform sampler.
Move draw_texture from glfiltersobel to glfilter. Still need to update
other plugins to this.
Allow to use the opengl context sharing feature if the original
context has been used already.
This is only possible on win32 with wglShareLists because on other
backends, the sharing is made when a context is created.
This new feature is only usefull when using external context
because gst-gl internally makes sure that the orginal context is not
already used when sharing it.
Fix bug #611067
First, inactivate shared gl contexts known by each sink pad.
Then, destroy the gl context known by the glmixer.
Finally, re-activate shared gl contexts.
This is to satisfy the fact that no shared gl context must be current
when an opengl context is destroyed.
Moreover the application may hang or crash without those steps.
Before, the window size was given at its creation. Now, it's done at
the drawing step because it's only relevant when there is a glimagesink
element in the pipeline.
glmixer can be seen as a glfilter except it handles N requested
sink pads.
Each sink pad and the src pad are video/x-raw-gl.
glmixer is responsible for managing different framerates from inputs.
It uses OpenGL context sharing. It means that each input is in its
own OpenGL context shared together and shared with the OpenGL context
of the ouput gl chain.
Also add a glmosaic which is an example of implementation of glmixer.
For now glmosaic is a cube but it will be fixed in the next commits.
For now the glmixer has some weird behaviours in some configurations
but it will be improved in the next commits.
The autotools builds is temporarly broken since those changes
have been made on win32.
Before, a gstgldisplay was instancied by the gl src in terms of gl chain.
And then the next element got it through the first gstglbuffer.
Now, this is done though queries.
All glelements get their ref on a gstgldisplay in READY state.
This rewrite is mainly a first step to be able to share OpenGL context hold
by the gstgldisplay using more complex glelements.
For example, with a glvideomixer. The associated gstgldisplay of each gl chain
of the sink pads will share their OpenGL context.
A texture is not destroyed when when we are done with it.
This texture is just added to the texture pool in order to be
re-used. In this case no OpenGL code is executed so we do not need to
request gl thread.
Add a pkg-config check for opengl and if not found assume opengl-es. If user has
none of both one still get build error later on (there is no pkg-config for
opengl-es).
Add more files to EXTRA dist and build the opengles variant if selected.
Simmilar changes could be done for the winCE backend.
Default implementation of NSOpenglView::update is not safe because it
just calls update on the opengl context whereas we are not in the gl thread.
Also fix the white flickering when resizing, because now we need to call
the draw callback manually when resizing.
- All gstglwindow members are now modified only in the gl thread
to avoid thread concurrency
- OpenGL context is now properly clean
- fix a couple of things in implementation of xoverlay interface
It works with both gst-launch and a cocoa app (non-embedded and embedded)
But there is still some problems:
- sometimes crash when closing
- flickering when resizing
- embedded mode not perfect
I will first make the CMake build work with cocoa backend
in order to generate a XCode project.
Then it should be easier to fix those issues.
Fixes bug #584877
Before this commit calling "gst_x_overlay_set_xwindow_id" more
than one time, had no effect.
It mainly affects the glimagesink implementation.
But on win32 (and CE), some stuff has to be done to
release the old parent.
And add a switchxoverlay example where the user
can click on left/right part of the main window to
switch the xoverlay.
The external opengl context must be specify when creating
our OpenGL context (glx) or just after (wgl).
When calling glXCreateContext or wglShareLists, the
external opengl context must not be current.
Then our gl context can be current in the gl thread while
the external gl context is current in an other thread.
See tests/examples/clutter/cluttershare.c
In OpenGL 2.x for Embedded System, a lot of basic scene/draw functions
have been removed. It means that everything is made using vertex and
fragment shaders.
I have also added a gstglwindow backend for winCE that uses EGL
(Native Platform Graphics Intercace) (which is a full part of
OpenGL ES specification). It remove the use of wgl/glx functions.
Althought the XEvent's xclient.data.l array is an array of
longs they will be constrained to 32 bit by the X11 protocol.
On 64 bit architectures use two elements of the array to store
one pointer.
This fixes segfaults that happen at least for every example
on startup.
This reverts commit 96e4ab18c2cf9876f6c031b9aba6282d0bd45a93.
You should have asked first. And you would have been told "no",
because it causes people on development branches to do a huge
amount of extra work.
Althought the XEvent's xclient.data.l array is an array of
longs they will be constrained to 32 bit by the X11 protocol.
On 64 bit architectures use two elements of the array to store
one pointer.
This fixes segfaults that happen at least for every example
on startup.
Registering GstRTSPMessage as a boxed type allows it to be conveniently
used as an argument to signals, a-la GstSDPMessage, and general usage
from bindings.
https://bugzilla.gnome.org/show_bug.cgi?id=762884
The while loop should end when all buffers "and" the preroll
buffer are consumed but this means to continue waiting if there
are still some pending buffers "or" preroll buffer.
The unit test was correct but racy because of this mistake.
I.e. because of the wrong "and" the while could finish too early.
cd tests/check && GST_CHECKS=test_query_drain make elements/appsink.forever
https://bugzilla.gnome.org/show_bug.cgi?id=789763
There is a race going on somewhere when we attempt to remove elements
*while* the parent container is switching to PLAYING.
In order to avoid this issue with discoverer, make sure we never
remove elements while switching to PLAYING.
This would result in a lot of warnings regarding elements not being
in NULL state when removed, or even leaked elements.
Instead make sure we take the lock and check whether we are processing
or not before allocating or adding anything to the pipeline
The attribute can be defined without value regardless session-level
or media-level.
Although `gst_sdp_message_insert_attribute` can be used to set NULL,
it would be easier if `gst_sdp_message_add_attribute` accepts NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=789841
RTCP XR provides supplements information of the report blocks
from SR and RR. This patch is for downgrading warnings when
XR is detected before implementing entire block types of RFC3611.
https://bugzilla.gnome.org/show_bug.cgi?id=789743
The lock is already taken before calling the flush method and can lead to
deadlock for some encoders that need to take the same lock from another
thread while flushing
https://bugzilla.gnome.org/show_bug.cgi?id=787311
It allows encoders to detect and drop input frames which are already
late to increase the chance of the pipeline to catch up.
The QoS logic and code is directly copied from gstvideodecoder.c.
https://bugzilla.gnome.org/show_bug.cgi?id=582166
When an empty mix matrix is passed, audio-channel-mixer
will now generate a (potentially truncated) identity matrix,
this replicates the behaviour of audiomixmatrix in first-channels
mode.
https://bugzilla.gnome.org/show_bug.cgi?id=788833
Properly handle protocol version in the connection
Add the following headers types:
* Pipelined-Request
* Media-Properties
* Seek-Style
* Accept-Ranges
https://bugzilla.gnome.org/show_bug.cgi?id=781446
This is the same code that is in decklinkaudiosrc, audioringbuffer,
audiomixer and various other places. Have it once instead of copying it
everywhere.
https://bugzilla.gnome.org/show_bug.cgi?id=787560
+ Refactor previous constructor to call on that new constructor
+ Reimplement is_passthrough to strictly check whether the matrix
is an identity matrix, comparing channel-masks was incorrect:
the mixer may be remixing from a list of positions to the same
list of positions, but ordered differently, and reciprocally,
the mixer may be remixing from a list of positions to another
list of positions identically ordered
+ Remove unused tmp field, must have been a refactoring leftover
https://bugzilla.gnome.org/show_bug.cgi?id=785471
If someone calls gst_app_sink_try_pull_sample they are
probably no longer interested in any preroll samples.
Useful if the user has not registered a preroll appsink callback.
Also added unit test 'test_do_not_care_preroll'
make elements/appsink.check
that fails without this patch.
https://bugzilla.gnome.org/show_bug.cgi?id=786740
There is no reason for appsink to hang onto the preroll buffer.
If needed, the application can just keep a ref on this buffer
after calling gst_app_sink_try_pull_preroll.
Also added unit test 'test_pull_preroll'
make elements/appsink.check
https://bugzilla.gnome.org/show_bug.cgi?id=786740
This is used to indicate upstream the requirement in buffers
while no buffer pool can be provided. In this case, only
configure the pool with caps/size/min/max if we have caps,
which we only parsed when there was no allocation pool.
https://bugzilla.gnome.org/show_bug.cgi?id=730758
__gst_video_element_proxy_caps is called by
__gst_video_element_proxy_getcaps with caps set to the caps
allowed downstream. As we didn't set colorimetry or
chroma-site on the resulting caps, upstream considered it
possible to use whatever values it wanted, leading to
not negotiated errors later on.
As the description for that function is:
"Takes caps and copies its video fields to tmpl_caps",
it seems legitimate to set these fields there.
https://bugzilla.gnome.org/show_bug.cgi?id=786172
Only adjusting the base_ts might lead to a negative ts and as such integer
overflow into a huge timestamp which then propagates into the granulepos
and so on. Instead, resync to incoming buffer timestamp using both base_ts
and sample count rather than only base_ts.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=785948
The g-i stuff for this helper lib was never usable from bindings
anyway and there are problems with the latest gobject-introspection,
so we might just as well remove the g-i integration entirely for
this lib.
And monitor no_more_pads.
With live sources such as rtsp, uridecodebin only creates its
child decodebins between PAUSED and PLAYING.
This means that the ASYNC_DONE it posts when getting NO_PREROLL
in its change_state method gets immediately propagated by the
GstBin parent class, as opposed to a situation where a
decodebin has been added to it already, and has posted ASYNC_START.
The proposed solution, instead of simply waiting for ASYNC_DONE,
and finishing prematurely in that case, waits for three conditions
to be true:
* the uridecodebin needs to have emitted no_more_pads
* its current state must be PAUSED if not live, PLAYING otherwise
* There must be no "pending subtitle pads", ie pads where we haven't
received tags yet.
All these conditions are checked in the message handler, as we
post custom messages on it when we get subtitle tags or no_more_pads.
https://bugzilla.gnome.org/show_bug.cgi?id=783257
When the input is TRICKMODE_KEY_UNITS, we expect to only receive keyframes
which we want to decode/push immediately. Therefore don't queue them.
If upstream didn't send just keyframes (which is the ideal situation), two
different things can happen:
1) Either the subclass checks the segment flags and properly configures
the decoder implementation to only decode/output keyframes,
2) Or the subclass really decodes and outputs everything, in which case
the reverse frames will end up arriving "late" downstream (and will
be dropped). If upstream did properly send GOP in reverse order, we
still end up just showing keyframes (but at the overhead of decoding
everything).
https://bugzilla.gnome.org/show_bug.cgi?id=777094
Always put multiview-caps onto the output caps, assuming
mono if we've got no other information. It's still easy for
downstream elements to override using a capssetter or event
probe if desired.
https://bugzilla.gnome.org/show_bug.cgi?id=776172
Child streams could have more accurate width/height or various other
information added. If they have the same name, they are likely to be the
same streams.
https://bugzilla.gnome.org/show_bug.cgi?id=782697
This is now needed as GstClock does not do that internally anymore,
because that broke bindings.
And mark the function correctly as (transfer full), which it already was
before.
https://bugzilla.gnome.org/show_bug.cgi?id=743062
Performing a gst_sdp_media_get_caps_from_media() would result in
changing fields in the GstSDPMedia violating the const tag in the
function declaration.
Before there would be a line with a=rtpmap:96 VP8/90000
after, that attribute would only contain a=rtpmap:96
Fix by performing modifications on duplicated strings instead of on
the internal values.
Also add a simple test for checking that the representation doesn't
change by a gst_sdp_media_get_caps_from_media()
The GSource for dealing with timeouts in
gst_video_convert_sample_async() might be attached to a non-default
context, so we should not be using g_source_remove() on the returned ID.
The correct thing to do is to keep a reference to the actual GSource and
then call g_source_destroy() on it.
https://bugzilla.gnome.org/show_bug.cgi?id=780297
Track how long it takes to generate the first buffer after a flush
as a simple measure of how efficient the decoder is at skipping /
rushing to get to the first decode.
When initializing a timecode from a GDateTime, and the remaining time
until the new second is less than half a frame (according to the given
frame rate), it would lead to the creation of an invalid timecode, e.g.
00:00:00:25 (at 25 fps) instead of 00:00:01:00. Fixed.
https://bugzilla.gnome.org/show_bug.cgi?id=779866
Use G_GUINT64_FORMAT for guint64 values.
Introduced by fcb63e77a9
Found by Alexander Larsson
gstvideodecoder.c: In function 'gst_video_decoder_have_frame':
gstvideodecoder.c:3312:51: error: format '%u' expects argument of type 'unsigned int', but argument 8 has type 'guint64 {aka long long unsigned int}' [-Werror=format=]
Don't guess a timestamp of the start of the segment when running
in reverse mode, as more likely it means we're discontinuous somewhere
in the middle of the segment, and we'll fix up timestamps once
the frames are decoded and reversed.
When a PTS is not set, we still want to store the rest of the
buffer information, or else we lose important things like the
duration or buffer flags when parsing.
This adds a property to select the maximum number of threads to use for
conversion and scaling. During processing, each plane is split into
an equal number of consecutive lines that are then processed by each
thread.
During tests, this gave up to 1.8x speedup with 2 threads and up to 3.2x
speedup with 4 threads when converting e.g. 1080p to 4k in v210.
https://bugzilla.gnome.org/show_bug.cgi?id=778974
In gst_video_time_code_is_valid, also check for invalid
ranges when using drop-frame TC. Refactor some code which
broke after the check was added.
https://bugzilla.gnome.org/show_bug.cgi?id=779010
It was taking the initial input y-offset from the output value, which
only works for y=0 (in which case both are the same). If y > 0, we would
always stay behind the requested input offset and never ever read
anything from the input.
The parser might do some conversion on a stream but the stream keeps
being the same, and we need to make sure GstDiscoverer detects it is the
case.
https://bugzilla.gnome.org/show_bug.cgi?id=778298
There was already a check for that, but it failed because
subformat_guid[0] is a guint32 and that is then casted implicitely to a
guint16 when recursing... just that we checked the uncasted value.
This caused an infinite recursion and thus stack overflow.
https://bugzilla.gnome.org/show_bug.cgi?id=777265
Sometimes there is a human-oriented timecode that represents an
interval between two other timecodes. It corresponds to the human
perception of "add X hours" or "add X seconds" to a specific timecode,
taking drop-frame oddities into account. This interval-representing
timecode is now a GstVideoTimeCodeInterval. Also added function to add it to
a GstVideoTimeCode.
https://bugzilla.gnome.org/show_bug.cgi?id=776447
It is often usefull to make sure that you get a full copy of a profile.
For example you want to let the user modify it in the user interface
but still keep an unchanged version for later use.
API:
gst_encoding_profile_copy
Initialize min and max _get_property() to gets rid of these
compiler warnings:
gstappsrc.c:741:7: error: 'max' may be used uninitialized in this function
g_value_set_int64 (value, max);
^
gstappsrc.c:733:7: error: 'min' may be used uninitialized in this function
g_value_set_int64 (value, min);
^
Which happens because gcc doesn't know that GST_IS_APP_SRC will never
fail here.
https://bugzilla.gnome.org/show_bug.cgi?id=752052
This way special characters such as '@' can be used in
usernames or passwords, e.g.
rtsp://view:%40dm%4An@<IP-ADDR>/media/camera1
will now parse username and password into:
User: view
Pass: @dm:n
https://bugzilla.gnome.org/show_bug.cgi?id=758389
When parsing NUL-terminated strings, do not include the terminating
NUL byte(s). Depending on the encoding used, either g_utf8_validate()
failed due to this, or worse the call to g_utf16_to_utf8() would
return 0 items read on an empty string, causing it to fail parsing
certain frames.
https://bugzilla.gnome.org/show_bug.cgi?id=770355
encoding-profile.c: In function ‘get_profile_format_from_possible_factory_name’:
encoding-profile.c:1532:6: error: ‘fact’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
if (fact)
^
encoding-profile.c: In function ‘profile_from_string’:
encoding-profile.c:1720:6: error: ‘res’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
if (profile)
^
cc1: all warnings being treated as errors
Instead of enforcing the user to know and understand caps to describe
the encoding format, let him use element factory names directly.
This also makes it possible to ensure that a specific encodore/muxer
is used instead of letting the ranking system do it.
It is now possible to describe an encoding format simply specifying:
matroskamux:x264enc:vobisenc
Factor out functions in the parsing, cleaning up the whole thing.
Update documentation.
We used to only care about the name of the files even if the name
is defined in the encoding target serialized file.
That commit also allows user to define several names for a single
target file (using a ';' between the names) which allows us to have
a target for youtube that is called 'youtube;yt' or a target for
'ogg;ogv;oga' file extension.
We checked this already earlier, so this is dead code.
Leave an assert in place for consistency with the other
branch and in case the rest of the code changes.
CID 1397350.
The caps put into the stream topology by decodebin are the caps at the
moment the pads are exposed on it. This is usually before decoders
received any buffers.
In discoverer we however wait for pre-roll, which ensures that each
decoder handled buffers already. At this point, there might be more
information known about the caps already that we could make use of.
One example here is extra information stored in the SEI of H264, like
the multiview-mode. This will be known if there is a SEI before the
first keyframe, but decodebin won't put this into the topology as it
only waits for the initial caps of h264parse (which come directly after
SPS/PPS).
With this change, the multiview-mode is in the caps reported by
discoverer in many cases.
To make the structs usable in bindings, and fix
gstrtspmessage.c:1188: Warning: GstRtsp:
gst_rtsp_message_parse_auth_credentials: return value: Invalid
non-constant return of bare structure or union; register as
boxed type or (skip)
https://bugzilla.gnome.org/show_bug.cgi?id=774416
Pass the frame data and size explicitly to
id3v2_add_id3v2_frame_blob_to_taglist() and add a
comment that it's being deliberately / manually
passed the full ID3v2 frame including header.
Ensure that nothing is in any of the streaming thread functions
anymore when going from PAUSED to READY. While the parent's state change
function has deactivated all pads, there is nothing preventing
downstream from activating our srcpad again and calling the getrange()
function. Although we're in READY!
https://bugzilla.gnome.org/show_bug.cgi?id=775687
The flags and field order weren't properly initialized in the
gst_video_info_init().
Furthermore in gst_video_info_from_caps() we might set unitiliazed
values previously, this only sets them if valid.
Optimize LE<->BE conversion by adding a dedicated fast path instead of
using the generic converter. Implement transform_ip function in order to do the
endian swap in place.
This saves buffer allocation for the intermediate format, can be done in place
and also performs the conversion in one step instead of unpack-convert-pack.
For all bit widths the naive algorithm is implemented, which provides the best
performance when compiled with -O3. ORC was considered but eventually removed
as it requires a dedicated function for in-place conversion (due to the
"restrict" parameters).
A more complex algorithm for the 24-bit conversion with unrolled loop and
32-bit processing is implemented in the #if 0 section. It performs better if
compiled with -O2. With -O3 however the naive algorithm performs better.
https://bugzilla.gnome.org/show_bug.cgi?id=773073
It is not needed to store a pointer to every single chain element to free it.
Instead walk the channel list backwards and free the chain elements one by one.
Rename GstAudioConverter->chain_pack to chain_end.
https://bugzilla.gnome.org/show_bug.cgi?id=773073
The caps might not be fixated (which is required by GstVideoInfo) and we
would assert otherwise. However the caps often contain useful
information in the already-fixed parts that we can use here.
When gst_rtp_buffer_add_extension_onebyte_header() is used over a
GstRtpBuffer that only contains a memory for the whole packet,
ensure_buffers function crashes at the next point:
mem = gst_memory_copy (rtp->map[i].memory, offset, rtp->size[i]);
when i==2 because the payload is not mapped.
In addition the offset is calculated subtracting in the wrong direction.
https://bugzilla.gnome.org/show_bug.cgi?id=774959
For example mmap can fail with EACCES if the the fd has been open
with read only mode. And mapping the memory might be the only way
to check that. So no need to print out an error.
Ex: ioctl(dev, DRM_IOCTL_PRIME_HANDLE_TO_FD, flags & ~DRM_RDWR)
https://bugzilla.gnome.org/show_bug.cgi?id=765600
This class was made subclassable, though for future growth of the code,
it's better if we have some room for add class members. Using the small
padding since this is unlikely.
For drop-frame timecodes, the nsec_since_daily_jam doesn't necessarily
directly correspond to this many hours/minutes/seconds/frames. We have
to get the frame count as per frames_since_daily_jam and then convert.
https://bugzilla.gnome.org/show_bug.cgi?id=774585
Rename function parameter and make sure the name in the
declaration matches the name in the implementation, to
avoid g-i warnings. Also add Since markers for gtk-doc.
gstappsink.c:1248: Warning: GstApp: gst_app_sink_set_buffer_list_support:
unknown parameter 'buffer_list' in documentation comment, should be 'drop'
Refuse to answer BYTES queries ourselves. The only
time they make sense is on raw elementary streams,
in which case upstream would already have answered.
https://bugzilla.gnome.org/show_bug.cgi?id=757631
Refuse to answer BYTES queries ourselves. The only
time they make sense is on raw elementary streams,
in which case upstream would already have answered.
They especially don't make sense for encoders to answer
based on upstream values - although perhaps later
we could make it do TIME->BYTES conversion on the source
pad based on bitrate.
https://bugzilla.gnome.org/show_bug.cgi?id=757631
It is actually needed as we need some symbols. We do not link
to libgstsdp as the user of the lib should do it (same with
autotools build).
This reverts previous commit
gst_audio_buffer_reorder_channels() was always mapping the buffer read-write
regardless whether any reordering was needed. If the from and to channel order
is identical return immediately without remapping the buffer.
Add a small helper function gst_audio_channel_positions_equal() which is used
in both gst_audio_reorder_channels() and gst_audio_buffer_reorder_channels().
https://bugzilla.gnome.org/show_bug.cgi?id=773833
It adds a third argument to pass GstBufferPoolAcquireParams
to gst_buffer_pool_acquire_buffer.
If a user subclasses GstBufferPoolAcquireParams, this allows to
pass an updated param to the underlying buffer pool at each
gst_video_decoder_allocate_output_frame_with_params call.
https://bugzilla.gnome.org/show_bug.cgi?id=773165
Adds "memory:DMABuf" caps feature. Since 1.11 tag.
Useful when the the dma-buf buffer cannot be mapped to CPU for r/w requests.
Example: protected content or platform constraints.
https://bugzilla.gnome.org/show_bug.cgi?id=759358
Usually this information is static for the whole stream, and various
container formats store this information inside the headers for the
whole stream.
Having it inside the caps for these cases simplifies code and makes it
possible to express these requirements more explicitly with the caps.
https://bugzilla.gnome.org/show_bug.cgi?id=771376
All the GstAudioClock method declarations required object of GstClock type
as a first argument, but in fact, required GstAudioClock object (runtime
check in function body). Instead of checking type in run-time, we can
change functions declaration, to accept only GstAudioClock methods. Then,
runtime check is not necessary anymore, since always GstAudioClock object
is passed to a function.
https://bugzilla.gnome.org/show_bug.cgi?id=756628
Also the format must be fixed on the default raw caps. If not
gst_video_info_from_caps() will fail and
gst_video_decoder_negotiate_default_caps() return FALSE.
The test simulates the use case where a gap event is received before
the first buffer causing the decoder to fall back to the default caps.
https://bugzilla.gnome.org/show_bug.cgi?id=773103
Seen on the Jenkins CI:
FAILED: subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o
ccache cc '-Isubprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta' '-fdiagnostics-color=always' '-I../subprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/.' '-I../subprojects/gst-plugins-base/.' '-Isubprojects/gst-plugins-base/gst-libs' '-I../subprojects/gst-plugins-base/gst-libs' '-Isubprojects/gstreamer/libs' '-I../subprojects/gstreamer/libs' '-Isubprojects/gstreamer/.' '-I../subprojects/gstreamer/.' '-pipe' '-Wall' '-Winvalid-pch' '-DHAVE_CONFIG_H' '-msse4.1' '-fPIC' '-O0' '-g' '-fPIC' '-I/usr/include/glib-2.0' '-I/usr/lib/glib-2.0/include' '-pthread' '-Isubprojects/gstreamer/gst' '-MMD' '-MQ' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' '-MF' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o.d' -o 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' -c ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c
In file included from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h:24:0,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-private.h:23,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-macros.h:25,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.h:23,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c:24:
../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory
#include <gst/audio/audio-enumtypes.h>
^
compilation terminated.
Workaround source_root being the root directory of all projects
in the subproject case.
Remove now unneeded getpluginsdir and define c++ tests in the same loop.
Bump meson requirement to 0.35
This makes sure that we only build files that need explicit SIMD support
with the relevant CFLAGS. This allows the rest of the code to be built
without, and specific SSE* code is only called after runtime checks for
CPU features.
https://bugzilla.gnome.org/show_bug.cgi?id=729276
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.
They are false positive overflows, because coverity doesn't realize that
hours <= 24, minutes < 60 and seconds < 60 in all functions. Also casting the
number 60 (seconds in minute, minutes in hour) to guint64 for the
calculations, in order to avoid overflowing once we allow more than 24-hour
timecodes.
CIDs #1371459, #1371458
_stdint.h is generated by Autotools and we don't really need it.
stdint.h is now available on all supported platforms.
This really only makes a difference for MSVC, which has it starting from
Visual Studio 2015.
Add GST_FD_MEMORY_FLAG_DONT_CLOSE to avoid closing the fd when the
memory is freed. When you can guarantee the lifetime of the fd is
longer than the memory, this can save a dup() call.
Most of them are overflow related and false positives, but coverity can't know
that these can't overflow without us giving it more information. Add some
assertions for this.
One was an actual issue with flags comparison.
CIDs #1369051, #1369050, #1369049, #1369048, #1369045
WAV is too generic to impose more-or-less arbitrary boundaries on the
sample rate and channel count caps. For example, there are 384 kHz WAV
files. Another example: it is in theory possible that somebody puts DSD
data into a WAV file, which will then have a sample rate of ~2.8 MHz.
For this reason, get rid of the rate and channel caps unless they are
fixed values. Downstream anyway usually knows the limitations better.
https://bugzilla.gnome.org/show_bug.cgi?id=761514
gst_rtp_buffer_add_extension_onebyte_header() and
gst_rtp_buffer_add_extension_twobytes_header() can have a const argument for
the actual extension data.
The _pull_sample() and _pull_preroll() functions block
until a sample is available, EOS happens or the pipeline
is shut down (returning NULL in the last two cases).
This adds _try_pull_sample() and _try_pull_preroll()
functions with a timeout argument to specify the maximum
amount of time to wait for a new sample.
To avoid code duplication, wait forever if the timeout is
GST_CLOCK_TIME_NONE and use that to implement
_pull_sample/_pull_preroll with the original behavior.
Add also corresponding action signals "try-pull-sample"
and "try-pull-preroll".
https://bugzilla.gnome.org/show_bug.cgi?id=768852
Remove unnecessary helper struct for callbacks. The bclass
member of the helper struct was not used, so we can just
remove it and the GET_CLASS() call and simplify the whole
affair by passing the depayloader directly to the callback.
parsed, framed, stream-format and alignment are only relevant for parsers and
should not matter here. We still want to be able to use an encoder that can
only output byte-stream if the input was avc.
https://bugzilla.gnome.org/show_bug.cgi?id=768566
Caps are cleaned up for missing plugins, and for creating encoding profiles
and caps descriptions.
Fields like streamheader, parsed, framed, stream-format and alignment are not
relevant here. The last ones all because a parser will take care of them.
https://bugzilla.gnome.org/show_bug.cgi?id=768566
We need to take into account the input segment flags to know whether
we should drain the decoder after a new keyframe in trick mode.
Otherwise we would have to wait for the next frame to be outputted (and
the segment to be activated) which ... well ... kind of beats the whole
point of this draining :)
And especially don't use the stream lock for that, as otherwise non-serialized
queries (CONVERT) will cause the stream lock to be taken and easily causes the
application to deadlock.
https://bugzilla.gnome.org/show_bug.cgi?id=768361
By default we'll wait for a certain amount of data before
attempting typefinding. However, if the stream is fairly
short, we might get EOS before we ever attempted any
typefinding, so at this point we should force typefinding
and output any pending data if we manage to detect the
type.
https://bugzilla.gnome.org//show_bug.cgi?id=768178
In 0.10 the source pad was a dynamic pad that was only added once
the type had been detected, but in 1.x it's an always source pad,
so checking whether it's still NULL won't work to detect if the
type has been detected.
Makes tagdemux error out when we get EOS but haven't managed to
identify the format of the data after the tag.
https://bugzilla.gnome.org//show_bug.cgi?id=768178
Fix problem with the line cache where it would forget the first line in
the cache in some cases.
Keep as much backlog as we have taps. This generally works better and we
could do even better by calculating the overlap in all taps.
Allocated enough lines for the line cache.
Use only half the number of taps for the interlaced lines because we
only have half the number of lines.
The pixel shift should be relative to the new output pixel size so scale
it.
Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=767921
gst_buffer_copy_region() does not copy the duration if it doesn't start
with the first byte. We just skip the tag here, so the duration is still
valid.
https://bugzilla.gnome.org/show_bug.cgi?id=767791
Elements inherited from GstAudioDecoder, supporting PLC and introducing
delay produce invalid timestamps. Good example is opusdec with in-band FEC
enabled. After receiving GAP event it delays the audio concealment until
the next buffer arrives. The next buffer will have DISCONT flag set which
will make GstAudioDecoder to reset it's internal state, thus forgetting
the timestamp of GAP event. As a result the concealed audio will have the
timestamp of the next buffer (with DISCONT flag) but not the timestamp
from the event.
The serialization of double typed geographical
coordinates to DMS system supported by the exif
standards was previously truncated without need.
The previous code truncated the seconds part of
the coordinate to a fraction with denominator
equal to 1 causing a bug on the deserialization
when the test for the coordinate to be serialized
was more precise.
This patch applies a 10E6 multiplier to the numerator
equal to the denominator of the rational number.
Eg. Latitude = 89.5688643 Serialization
DMS Old code = 89/1 deg, 34/1 min, 7/1 sec
DMS New code = 89/1 deg, 34/1 min, 79114800UL/10000000UL
Deserialization
DMS Old code = 89.5686111111
DMS New code = 89.5688643
The new test tries to serialize a higher precision
coordinate.
The types of the coordinates are also guint32 instead
of gint like previously. guint32 is the type of the
fraction components in the exif.
https://bugzilla.gnome.org/show_bug.cgi?id=767537
For reverse playback it is important to handle correctly the frame sync
points, which is set when the input buffer doesn't have the DELTA_UNIT flag.
This is handled correctly when decoder is packetized, but when it is not the
frame's sync point is not copied, and the reverse playback never decodes frame
batches.
The current patch adds the buffer's flags to the Timestamp list, where the
timestamp and duration of the input buffers are hold.
There were two consecutive log messages in gst_video_decoder_decode_frame().
Given the information they provide, it is more efficient to squash them into a
single one.
The playback rate is hold in the input_segment member variable, not in the
output_segment, and the parse_gather list was never filled because of that.
This patch changes the comparison with input_segment.
The output segment is only set up after data is output, which might be far in
the future for reverse playback. Also we are here interested in the state at
the current *input* frame (which is the keyframe), not any possible output.
For reverse playback the same behaviour was already implemented in
flush_parse().
For reverse playback, chain_forward() is only used to gather frames and not
for decoding, and it is actually called by the draining logic, causing an
infinite recursion.
While it's a bit tricky to discard frames *before* decoding (because
we might not be sure which data is needed or not by the decoder), we
can discard them after decoding if they are too late anyway.
Any following basetransform based element or similar would drop the frame too.
When asked to just decode keyframe, if we got a keyframe drain out
the decoder straight away.
This avoids having to wait for the next frame and reduces delay even
more.
https://bugzilla.gnome.org/show_bug.cgi?id=767232
This ensures the decoder is properly drained out when receiving a
DISCONT buffer. The optimal way of doing this would have been to
receive a GAP event before hand but it is not always possible.
Fixes big delays with some decoders (ex gst-libav) that will not
drain out data when only decoding keyframes.
https://bugzilla.gnome.org/show_bug.cgi?id=767232
gst_buffer_copy_region() does not copy the timestamp if it doesn't start
with the first byte. We just skip the tag here, so the timestamp is still
valid.
https://bugzilla.gnome.org/show_bug.cgi?id=767173
The base class was setting the DISCONT flag before checking whether the buffer
would be in segment or not.
Fix issues with DISCONT flags not being properly propagated downstream when
decoders buffers were out of segment.
https://bugzilla.gnome.org/show_bug.cgi?id=766800
gst_video_sink_center_rect() can be called without a GstVideoSink
having been instantiated so we can't relly on the video sink
class_init function to init the category.
Fix a warning when running:
GST_CHECKS=test_video_center_rect GST_DEBUG=6 G_DEBUG=fatal_warnings make libs/video.check-norepeat
https://bugzilla.gnome.org/show_bug.cgi?id=766510
If we e.g. have AVI with DV container with video/audio inside the DV
container, we can't handle this at this point with an encoding profile.
Instead of erroring out, flatten the container hierarchy.
https://bugzilla.gnome.org/show_bug.cgi?id=765708
This tag match the EXIF_TAG_FOCAL_LENGTH_IN_35_MM_FILM exif tag and is
stored on a short. Hence there is a precision loss compared to the
GstTag which is a double value.
https://bugzilla.gnome.org/show_bug.cgi?id=753930
It is the 35 mm equivalent focal length of the lens, mainly used in
photography. Tag value is stored in a double value to be consistent with
GST_TAG_CAPTURING_FOCAL_LENGTH.
https://bugzilla.gnome.org/show_bug.cgi?id=753930
When converting discoverer output to an encoding profile, it makes sense to
omit these. It's very very unlikely that our encoder is going to produce bit
by bit the same codec_data or streamheader.
https://bugzilla.gnome.org/show_bug.cgi?id=765534
We add a couple of new functions gst_sdp_media_parse_keymgmt and
gst_sdp_media_parse_keymgmt. We also implement
gst_sdp_message_attributes_to_caps and gst_sdp_media_attributes_to_caps
in terms of these new functions and also gst_mikey_message_to_caps.
Current implementation requires all srtp and srtcp parameters to be
given in the caps. MIKEY uses only one algorithm for encryption and one
for authentication so we now allow passing srtp or srtcp parameters. If
both are given srtp parametres will be preferred.
https://bugzilla.gnome.org/show_bug.cgi?id=765027
As said in its doc GST_AUDIO_CHANNEL_POSITION_NONE is meant to be used
for "position-less channels, e.g. from a sound card that records 1024
channels; mutually exclusive with any other channel position".
But at the moment using such positions would raise a
'g_return_if_reached' warning as gst_audio_get_channel_reorder_map()
would reject it.
Fix this by preventing any attempt to reorder in such case as that's not
what we want anyway.
https://bugzilla.gnome.org/show_bug.cgi?id=763799
We currently don't log much about channel positions making debugging
harder as it should be. This is the first step in my attempt to improve
this.
https://bugzilla.gnome.org/show_bug.cgi?id=763985
There is a small window of time where the audio ringbuffer thread
can access the parent thread variable, before it's initialized
by the parent thread. The patch replaces this variable use by
g_thread_self().
https://bugzilla.gnome.org/show_bug.cgi?id=764865
Since the allocation query caps contains memory size and the pad's caps
contains the display size, an audio encoder or decoder might need to allocate
a different buffer size than the size negotiated in the caps.
This patch splits this logic distinction for audiodecoder and audioencoder.
Thus the user, if needs a different allocation caps, should set it through
gst_audio_{encoder,decoder}_set_allocation_cap() before calling the negotiate()
vmethod. Otherwise the allocation_caps will be the same as the caps in the
src pad.
https://bugzilla.gnome.org/show_bug.cgi?id=764421
Since the allocation query caps contains memory size and the pad's caps
contains the display size, a video encoder or decoder might need to allocate
a different frame size than the size negotiated in the caps.
This patch splits this logic distinction for videodecoder and videoencoder.
The user if needs a different allocation caps, should set the allocation_caps
in the GstVideoCodecState before calling negotiate() vmethod. Otherwise the
allocation_caps will be the same as the caps set in the src pad.
https://bugzilla.gnome.org/show_bug.cgi?id=764421
Doing so prevents us dropping buffers in the rare, but possible, situations,
when the stream changes SSRC and new sequence numbers does not differ
much from the last sequence number from previous SSRC. For example:
ssrc - 0xaaaa 101,102,103,104 ssrc - 0xbbbb 102, 103, 104, 105...
In the scenario above we don't want to drop the first 3 packets of
0xbbbb stream.
https://bugzilla.gnome.org/show_bug.cgi?id=764459
The function gst_discoverer_info_copy doesn't copy the data members seekable
and result of the source GstDiscovererInfo.
In the case of copying a GstDiscovererInfo for later use, the seekbale will be
undefined, which in practice usually will be false, even though the seekable of
the original GstDiscovererInfo is true.
https://bugzilla.gnome.org/show_bug.cgi?id=762710
The parameter type was wrongly documenting that a GstVideoInfo structure
pointer was needed, while it needs a GstVideoFormatInfo structure
pointer.
https://bugzilla.gnome.org/show_bug.cgi?id=764414
P010 is a YUV420 format with an interleaved U-V plane and 2-bytes per
component with the the color value stored in the 10 most significant
bits.
https://bugzilla.gnome.org/show_bug.cgi?id=761607
---
Changes since v2:
- Set bits=16 in DPTH10_10_10_HI
Changes since v1:
- Fixed x-offset calculation in uv.
- Added 6-bit shifts to FormatInfo.
Store the filter in the desired sample format so that we can simply do a
linear or cubic interpolation to get the new filter instead of having to
go through gdouble and then convert.
Remove some unused variables from the inner product functions.
Make filter coefficients by interpolating if required.
Rename some fields.
Try hard to not recalculate filters when just chaging the rate.
Add more proprties to audioresample.
Rearrange the oversampled taps in memory to make it easier to use
SIMD instructions on them. this simplifies some sse code.
Add some more optimizations
Improve int16 resampling by using pmaddwd
Use intrinsics to scale and pack int16 samples
Align the coefficients so that we can use aligned loads
Add padding to taps and samples so that we don't have to use partial
loads for the remainder of the loops.
Remove copy_n, we can reuse the plain copy function with some new
parameters.
Align and pad the sample array.
Remove the consumed/produced output fields from the resampler and
converter. Let the caler specify the right number of input/output
samples so we can be more optimal.
Use just one function to update the converter configuration.
Simplify some things internally.
Make it possible to use writable input as temp space in audioconvert.
Base class type name should not reference libvisual since not all child
elements use this. This was an oversight when merging audiovisualizers into
a common base class.
libgstreamer currently exports some debug category
symbols GST_CAT_*, but those are not declared in any
public headers.
Some plugins and libgstvideo just use GST_DEBUG_CATEGORY_EXTERN()
to declare and use those, but that's just not right at
all, and it won't work on Windows with MSVC. Instead look
up the categories via the API.
If we don't have writable memory, make sure to make a copy of the input
samples into a temporary (writable) buffer, even if we are dealing with
a native intermediate format that we don't need to call the unpack
function for.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=761655
gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
In that case, use gst_pad_peer_query_caps() with template caps as filter
to have negotiated output caps properly before forwarding GAP event.
https://bugzilla.gnome.org/show_bug.cgi?id=761218
Add missing ':' to tile_ws and tile_hs fields documentation to avoid
bad render of these two fields, mark reserved bytes as private to hide
field and avoid gtkdoc warning and add parameters description to
documented macro to avoid gtkdoc warnings.
https://bugzilla.gnome.org/show_bug.cgi?id=761132
In gst_video_info_to_caps(), make sure we end up with an RGB matrix for
RGB formats and warn when the GstVideoInfo colorimetry is wrong.
In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
for RGB formats and warn about inconsistent caps.
See https://bugzilla.gnome.org/show_bug.cgi?id=759624
For RGB formats, the matrix in the colorimetry (conversion from YUV to
RGB) is irrelevant and we should ignore it and assume the identity
transform for everything we do.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624
It's quite unexpected behaviour that various subclass settings are just
reset before set_format(). Unfortunately changing this now has the risk
of breaking existing code but we should reconsider this for 2.0.
When the input and output formats are the same and in a possible
intermediate format, avoid unpack and pack.
Never do passthrough channel mixing.
Only do dithering and noise shaping in S32 format
Add support for float and int16 mixing
Remove in-place processing, this simplifies things as we won't be using it.
Don't do clipping for float audio formats
Process as many samples as we can from the input and return the number
of processed samples from the chain. This simplifies some code.
Fix the IN_WRITABLE handling, don't overwrite the flags.
Pass flags in _converter_new() so that we can configure ourselves
differently depending on some options.
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
Simplify the API, we don't need the consumed and produced output
arguments. The caller needs to use the _get_in_frames/get_out_frames API
to check how much input is needed and how much output will be produced.
When we run the loop for another time and do not have a preset name, we would
try to print the preset name of a previous iteration that is already freed.
Also move some other variables into the block where they are actually used
to prevent similar mistakes in the future.
CID 1346536
We did not take the sample size into account. Rearrange the tests to have more
conversion test and an extra test case for passthrough operations.
Fixes#759890
Rename samples to num_samples, since we also have samples in chain, but that is
the data pointer. Always use gzize for num_samples. Make the log output a bit
more homogenous.
The EOS event can be propagated to the downstream elements when
is_eos flag remains set even after leaving the flushing state.
This fix allows this element to normally restart the streaming
after receiving the flush event by clearing the is_eos flag.
https://bugzilla.gnome.org/show_bug.cgi?id=759110
Rework the main processing loop. We now create an audio processing
chain from small core functions. This is very similar to how the
video-converter core works and allows us to statically calculate an
optimal allocation strategy for all possible combinations of operations.
Make sure we support non-interleaved data everywhere.
Add functions to calculate in and out frames and latency.
Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we
are using a proxy server. Also, send Host header for compatability with
HTTP/1.1 and some HTTP/1.0 servers.
https://bugzilla.gnome.org/show_bug.cgi?id=758922
gst_rtsp_connection_connect_with_response accepts a response pointer
which it fills with the response from setup_tunneling if the
connection is configured to be tunneled. The motivation for this is to
allow the caller to inspect the response header to determine if
additional authentication is required so that the connection can be
retried with the appropriate authentication headers.
The function prototype of gst_rtsp_connection_connect has been
preserved for compatability with existing code and wraps
gst_rtsp_connection_connect_with_response.
https://bugzilla.gnome.org/show_bug.cgi?id=749596
Depayloaders will look at rtpbuffer->buffer for the discont flag.
When we set the discont flag on a buffer in the rtp base depayloader
and we have to make the buffer writable, make sure the rtpbuffer
actually contains the newly-flagged buffer, not the original input
buffer. This was introduced with the addition of the process_rtp_packet
vfunc, but would only trigger if the input buffer wasn't flagged
already and was not writable already.
When we detect a discont and the input buffer isn't already flagged
as discont, handle_buffer() does a gst_buffer_make_writable() on the
input buffer in order to set the flag. This assumed it had ownership
of the input buffer though, which it didn't. This would still work
fine in most scenarios, but could lead to crashes or mini object
unref criticals in some cases when a discont is detected, e.g. when
using pcapparse in front of a depayloader. This problem was
introduced in bc14cdf529.
In the case where the stream doesn't have a framerate set and the frames
don't have a duration set, we still want to use the clipping path to
make sure we don't push buffers outside of the segment.
The problem was the previous iteration was setting a duration of 2s, which
meant that any buffer which was less than 2s before the segment start would
end up getting pushed.
Instead, use a saner 40ms (25fps single frame duration) to figure out whether
the frame could be within the segment or not
Some servers incorrectly parse header names with strict case-sensitivity. For
compatibility with these systems change X-Sessioncookie to x-sessioncookie.
https://bugzilla.gnome.org/show_bug.cgi?id=758921
Any latency query before this will not get the correct latency so a new
latency query should be triggered once the audio sink know its own latency.
Without this the initial latency query from the pipeline arrives too early
sometimes and the resulting latency is too short.
https://bugzilla.gnome.org/show_bug.cgi?id=758911
Some Opus files found on the wild have 0 in the version field of the
OpusHead header, instead of the correct value of 1. The files still
play, don't make this error fatal.
https://bugzilla.gnome.org/show_bug.cgi?id=758754
Commit ff6d1a2a25 changed sample's type from
gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
which means it can never be a negative value and the check making sure that
in_samples is >= 0 is never going to be false. Removing it.
CID 1338689
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.
API: gst_audio_channel_get_fallback_mask()
Add a TRUNCATE_RANGE flag for unpack functions to fill the least
significate bits with 0 (as did the old code). Also add functions
that don't truncate. Use the TRUNC flag in audioconvert for
backwards compatibility for now.
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
