Add AVTP Raw Video Format de-payload support. The element supports only
GRAY16_LE output format, so:
- active pixels (no vertical blanking),
- progressive mode,
- 8 and 16-bit pixel depth,
- mono pixel format,
- grayscale colorspace.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335>
Add AVTP Raw Video Format payload support. The element supports only GRAY16_LE
input format, so:
- active pixels (no vertical blanking),
- progressive mode,
- 8 and 16-bit pixel depth,
- mono pixel format,
- grayscale colorspace.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335>
Due to a bug in the VT API, attempting to encode interlaced content
with ProRes results in an error, halting the pipeline instead of
gracefully falling back to software encoding.
Should be removed in the future if Apple ever fixes this issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3222>
The VA API has not defined the scaling list entries for U/V planes
for the 4:4:4 stream. In fact, we do not meet the 4:4:4 format output
for H264 so far, and scaling list is not used frequently, so we just
print out some warning and ignore these scaling list values.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3749>
We already have functions to generate a stream-id from pads but in the
end those pads are not even used in most cases. This adds functions to
generate a stream-id even before creating the source pads for the
element that is going to use it. For example a demuxer that is properly
implements the GstStream/GstStreamCollection API will not have a Pad but
already needs to generate a stream-id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3160>
The gst-devtools project generates gstreamer-validate-1.0.pc, this
must match the dependency in gst-editing-services for detection
to work properly.
Fixes:
Run-time dependency gst-validate-1.0 found: NO (tried pkgconfig and cmake)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3859>
All the RTP src pads were sharing the same stream-id while each actually
carry a different stream.
This was causing problem for example when funneling the streams together
and then trying to split them using 'streamiddemux'.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
In theory, `dispose()` functions should be idempotent and should be
prepared not to crash or cause a double-free if an unref done from
inside caused a recursive call to `dispose()` of the same object.
https://developer.gnome.org/gobject/stable/howto-gobject-destruction.html
This patch modifies the `dispose()` method to honor these constraints.
Since the double `dispose()` call won't actually occur in qtdemux (there
is no cycle detection mechanism that could invoke it to work that way),
this is more of a code cleanup than a user-facing problem fix.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3822>
* Extend protocol so that client can notify of releasing shared memory
* Server will hold shared memory object until it's released by client
* Add allocator/buffer pool to reuse shared memory objects and buffers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3765>
With the addition of the 'keep-aspect-ratio' sizing policy, content
that doesn't fit the target size is downscaled according to its own
aspect ratio to fit that target size, and centered.
Centering might not always be the desired behaviour, however;
consumers of this API might want to align the resulting picture to
the left or to the right.
To account for any of these cases, add two new properties to the
glvideomixer pad: xalign, and yalign. They operate on normalized
coordinates (0.0 for start, 1.0 for end), and default to 0.5 which
centers content.
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3762>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3762>
If we know there's only one stream we care about and we
don't have to synchronise audio and video, or send RRs,
we might just as well not hook up all the RTCP bits and
use fewer threads and sockets and simplify the pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3531>
Spec 7.1.3:
If a memory object does not have the VK_MEMORY_PROPERTY_HOST_COHERENT_BIT
property, then vkFlushMappedMemoryRanges must be called in order to guarantee
that writes to the memory object from the host are made available to the host
domain, where they can be further made available to the device domain via a
domain operation. Similarly, vkInvalidateMappedMemoryRanges must be called to
guarantee that writes which are available to the host domain are made visible to
host operations.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3723>
There is no byte-stream/au format for AV1 but only for H264, and the
encoder actually outputs obu-stream/tu instead of the annexb
stream-format that is similar to H264 byte-stream format.
Without this the encoder can't be used with elements that require a
specific AV1 stream-format, e.g. the MP4 or Matroska/WebM muxer.
The sizing policy allows selecting between the current behavior,
which deforms the texture to fill the width and height of the
pad; and a new 'keep-aspect-ratio' sizing policy, which fits the
texture within the rectangle respecting its original aspect ratio.
The reason for this is that this allows avoiding extra elements
in the pipeline, and reduces the number of buffer passing through
the pipeline.
Most of this code is a direct port of the sizing policy handling
of the compositor element, except it is adapted to operate on GL
texture coordinates through the projection matrix.
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3760>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3760>
It's not possible to annotate a in-parameter for a return value array as
the array length. Both are assumed to have the same direction and the
current annotation causes the size parameter to be considered an out
parameter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3787>
Without this, the plugin cannot be loaded in a devenv because the
RPATH is not added to the plugin dylib. This RPATH will be stripped on
install, which is what we want.
When deploying apps, people are supposed to use `macdeployqt` to
create an AppBundle that bundles Qt for you and sets the RPATHs
correctly to point to that bundled Qt.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3708>
It is really difficult for people to figure out why nvcodec has
0 features. Even the debug log is cryptic. Also make sure the errors
go to the ERROR log level, which is more likely to be enabled by
default.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3776>
If we have caps then we can only set exactly those caps, if we have no
caps yet then negotiating anything is not very meaningful because the
caps are defined by the application and not downstream.
Avoids, among other things, an unnecessary allocation query and spurious
useless caps being set before the first buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3757>
_alloca CRT function is deprecated. Moreover, stack allocation
for string is not a good idea. We can use _malloca inline
function instead, but all use of _alloca in d3d11 library/plugin
are not performance critical path at all.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3652>
Due to the dynamic nature of multiqueue, when `use-interleave` is used we can't
report a maximum tolerated latency (when queried) since it is calculated
dynamically.
When in such live pipelines, we need to make sure multiqueue can handle the
lowest global latency (provided by this event). Failure to do that would
result in not providing enough buffering for a realtime pipeline.
Fixes#1732
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3772>
Pads are activated automatically when they are added if the element
state is >=PAUSED, so it's not necessary to activate them manually
anymore.
This patch removes manual pad activation from gstaggregator, gstconcat,
gstfunnel, and gstinputselector.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3636>
Deserialize socket control messages as GstSocketTimestampMessage only
if (level, type) is (SOL_SOCKET, SCM_TIMESTAMPNS).
Without this patch, messages with types SCM_RIGHTS or SCM_CREDENTIALS
could be deserialized as GstSocketTimestampMessage instead of
GUnixFDMessage or GUnixCredentialsMessage from gio.
Fixes#1736
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3777>
Handling mouse navigation events in glvideomixer element, if no
pixel-aspect-ratio info in the caps, an assertion error is produced
inside gst_util_fraction_multiply because default denominator is zero.
Error fixed:
```
(gst-launch-1.0:102654): GStreamer-CRITICAL **: 00:47:51.598: gst_util_fraction_multiply: assertion 'b_d != 0' failed
```
Simple pipeline to reproduce the issue:
```
gst-launch-1.0 -v glvideomixer name=mix ! glimagesinkelement gltestsrc ! mix.sink_0
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3766>
Allow requesting an offer from the peer if we're joining a call with a
peer, and allow the peer to request an offer from us if waiting for an
incoming call.
This implements all 4 variants the protocol allows for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
AVC-Intra is a range of H.264-compliant intra-only codecs from
Panasonic. The codes and descriptions have been taken from VLC.
The (encumbered) sample I have here produces byte-stream H.264,
including SPS and PPS and no `avcC` box.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3739>
We create a new context in `gst_gl_context_create_thread()` and then
activate it on the current thread. Thereafter we assume that the
current thread continues to be the active thread for that context and
call `gst_gl_context_fill_info()` which asserts that the current
thread is the active thread.
However, if at the same time a different thread calls
`send_message_async()`, it will call into
`gst_gl_window_cocoa_send_message_async()` which will schedule the
message to be invoked using GCD. That anonymous function will also
call `gst_gl_context_activate()`, which creates a race, which can lead
to:
```
gst_gl_context_fill_info: assertion 'context->priv->active_thread == g_thread_self ()' failed
```
Fix it by using `gst_gl_context_thread_add()` to invoke `fill_info()`
on the context.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3732>
This allows handling input buffers with non-default strides, which was
already handled fine by the element code.
Without this, potentially expensive conversion was needed.
The private data is not copied over for SVT AV1 encoder so this code
path would've never worked.
Instead of relying on the PTS, which is not required to be unique or
existing at all, we always take the oldest frame as AV1 has no frame
reordering / B frames.
The goal of the "global" group-id is to fix new inputs that do not come from the
same "source" as others. In order to ensure all "current" streams have the same
group-id we distribute the first valid group-id to all streams.
This commit fixes two issues with that:
* When inputs are unlinked they weren't always properly resetted (it would only
work if parsebin is used, which is no longer the default in
uridecodebin3/playbin3).
* When computing the global group-id, take into account unset
group-id (i.e. GST_GROUP_ID_INVALID).
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1698
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3712>
No matter if they're allocated via GSlice or malloc(). The allocator is
completely irrelevant, all local tags need to be in the primer so they
can be handled.
This didn't have any effect in practice because all local tags that
appear in the muxer are allocated via GSlice. Only from the demuxer they
might be allocated via malloc().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3699>
As the path to the gir file is passed to hotdoc.generate_doc() and
not the build target itself, meson doesn't know about the dependency.
In turn, as the CI doesn't build everything before building the
documentation target, some gir files might not exist, for instance
in the case of gst-rtsp-server, causing the output documentation to
be empty.
The error occurred silently because hotdoc accepts wildcards for
*-sources arguments, thus it won't warn about a missing gir file as
it is legitimate for glob matching to resolve to nothing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3686>
It might be possible to fulfill those but not with the first caps
structure. Instead of just fixating the first caps structure, check if
the preference can be fulfilled by any of the structures as the first
step.
Without this the following pipeline negotiates to mono after the
decoder because opusenc only has a single channel in its first caps
structure.
gst-launch-1.0 audiotestsrc ! audio/x-raw,channels=2 ! opusenc \
! queue ! opusdec ! queue ! opusenc ! fakesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3689>
When calculating the presentation offset for CMAF input in live
playback, subtract the stream_time of the fragment from the
calculated presentation offset, so that the first fragment
is played at running time zero.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3680>
This should fix pipelines such as this one to work as expected
... ! opusenc ! capsfilter caps='audio/x-opus,
channels=1; audio/x-opus, channels=2' ! ...
The expectation is that the encoder will propose the first structure
before the second one to the source.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3673>
The number of expected pads was:
* Defaulting to 1
* Or being overriden by GST_MESSAGE_STREAMS_SELECTED
This fails if upstream isn't a selectable source and has multiple streams, and
would therefore cause failures with multi-stream gapless playback
Fixes#1672
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3658>
It is quite possible to have the blocking probe called from different streaming
threads when all expected pads are present.
* Notify all waiters by using g_cond_broadcast instead of g_cond_signal
* Properly remove the probe after waiting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3658>
gst_element_add_pad() is supposed to activate the pad if the element
state is >= PAUSED and the pad is not already active.
Unfortunately, before this patch, the activation was performed while the
element lock was still taken, which ended causing a deadlock in
gst_pad_start_task() as it attempted to post `stream-status` message in
the element, which also requires the element lock.
Elements could work around this bug by activating the pad manually
before adding it to the element.
This patch fixes the problem by performing pad activation only after the
element lock has been released.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3635>
Commit d3a66f9851 introduced a potential deadlock with two parallel release_pad
calls, where one could release the main multiqueue lock (qlock) while still
holding the reconf_lock and then calling other routines which in some conditions
may try to acquire qlock again. The second release_pad could already acquire the
qlock and then start waiting on reconf_lock, which may never be possible because
because the first one isn't releasing it until it can acquire qlock.
Fix it by holding reconf_lock for the whole durationg of qlock, making this
particular deadlock impossible.
Fixes#1642
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3571>
This is recommended by various specifications for such framerates, while
for integer framerates we continue using centiframes to allow for some
more accuracy.
Using N means that no rounding error accumulates, eventually leading to
outputting a packet with a different duration.
Some tools such as MediaInfo determine that a stream is variable
framerate if any packet has a different duration than the others, and
there is no reason I can see for not using the full 4 bytes of
resolution that the mp4 timescale offers.
Example problematic pipeline:
```
videotestsrc num-buffers=5001 ! video/x-raw,framerate=60000/1001,width=320,height=240 ! \
videoconvert ! x264enc bitrate=80000 speed-preset=1 tune=zerolatency ! h264parse ! \
video/x-h264,profile=high-10 ! mp4mux ! filesink location="result2.mp4"
```
This results in a media file that MediaInfo detects as variable
framerate because the 5000th packet has duration 99 instead of 100.
With this patch, the timescale is 60000 and all packets have duration
1001.
Related issue for context: https://bugzilla.gnome.org/show_bug.cgi?id=769041
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3049>
The VAAPI vaQueryVideoProcPipelineCaps() requires the context as the
parameter. So far, we always pass VA_INVALID_ID and it can succeed.
But the API does not say that and in theory, a valid context is required.
Now the new platform really needs a valid context and so we have to
delay that query until the context is created.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3613>
NVDEC launches CUDA kernel function (ConvertNV12BLtoNV12 or so)
when CuvidMapVideoFrame() is called. Which seems to be
NVDEC's internal post-processing kernel function, maybe
to convert tiled YUV to linear YUV format or something similar.
A problem if we don't pass CUDA stream to the CuvidMapVideoFrame()
call is that the NVDEC's internel kernel function will use default CUDA stream.
Then lots of the other CUDA API calls will be blocked/serialized.
To avoid the unnecessary blocking, we should pass our own
CUDA stream object to the CuvidMapVideoFrame() call
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3605>
Using the "GstBin" flags to check if an adaptive demuxer is streams-aware isn't
a good idea since it prevents using elements which aren't bins.
Instead we see if a collection was posted by the demuxer by the time a pad is
added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3601>
If a discontinuity is detected in push mode, we need to clear the cached section
observations since they might have potentially changed.
This was only done properly when operating with TIME segments (dvb, udp,
adaptive demuxers, ...) but not with BYTE segments (such as with custom app/fd
sources).
We still don't want to flush out the PCR observations, since this might be
needed for seeking in push-based BYTE sources.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1650
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3584>
This reverts the decision from
https://bugzilla.gnome.org/show_bug.cgi?id=754230
where it was decided that we rather play safe and only use the `tfdt` if
it is "significantly different" to the sum of sample durations.
As the specification says
If the time expressed in the track fragment decode time (‘tfdt’) box
exceeds the sum of the durations of the samples in the preceding
movie and movie fragments, then the duration of the last sample
preceding this track fragment is extended such that the sum now
equals the time given in this box.
we have to use the `tfdt` in general to allow for it to signal gaps in
the stream.
A muxer producing fragments might not yet know the full duration of the
last sample of a previous fragment if the next fragment starts with a
gap, and knowing the actual start of the next fragment would potentially
require to violate latency requirements.
Additionally, the existence of `tfdt` allows to avoid accumulating
rounding errors from summing up the durations.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3586>
... when rendering on external HWND. ShowWindow() will cause
synchronous message passing to window thread and then can be blocked.
At the same time, window thread can wait for GStreamer thread.
Instead of the synchronous call, queue the task to window message
and performs from the window thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3583>
The rtpjitterbuffer test drop_messages_interval uses a GstClockTime for
the message drop interval. This property is defined as a guint. On
systems with 64-bit time_t but 32-bit uint, this can cause the
g_object_set function to fail to read the arguments properly.
Fixes: #1656
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3580>
Deadlock sequence:
* From a streaming thread, d3d11videosink sends synchronous message
to the parent window, so that internal (child) window can be
constructed on the parent window's thread
* App thread (parent window thread) is waiting for pipeline's
state change (to GST_STATE_NULL) but streaming thread is
blocked and waiting for app thread
To avoid the deadlock, GstD3D11WindowWin32 should send message
to the parent window asynchronously.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3570>
If we keep the old events they can be end up being passed to the app, that could
discard the protection information because it has been seen before.
Drive by improvement: use g_queue_clear_full instead of foreach+clear for
protection events.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3547>
On macOS, a Cocoa event loop is needed in the main thread to ensure
things like opening a GL window work correctly. In the past, this was
patched into glib via Cerbero, but that prevented us from updating it.
This workaround simply runs an NSApplication and then calls the
main function on a secondary thread, allowing GStreamer to correctly
display windows and/or system permission prompts, for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3532>
When a session request is coming in, ERROR occurs when the callee is busy.
But peer_status is the status of the caller, which is of course None when
calling someone, while self.peers[callee_id][2] is that of the callee.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2460>
jpegdec is capable to parse input frames, but if jpegparse is before,
there's no need to reparse frames. This patch configure jpegdec as
packetized, skipping parsing, if negotiated sink caps has the boolean
field 'parsed' as true.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2464>
Systems like musl libc don't support ISO 6937 in iconv. This ensures
that the MPEG-TS plugin can cope with that. There is existing support
in the plugin for other methods, so it seems to have been the original
intent anyway.
Fixes: #1314
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3245>
According to comment in gst_pulsering_stream_latency_cb, latency updates
happen every 100 ms. The code in gst_pulsering_stream_latency_cb does
not care about the actual state of the ringbuffer, pbuf->acquired or
not.
Thus, every 100 ms the ringbuf->segdone may be set, even though the
object itself might be in 'destroyed' state. On next
gst_pulseringbuffer_acquire the segdone is not touched, so playback may
resume with invalid/wrong segdone value. This finally leads to a period
of silence playing after resuming the pipeline.
The problem was found on 'not-yet-released'-hardware and so far was not
reproducible on desktop computer.
Removing the callback as long as the ringbuffer is not in 'acquired'
state solves the problem reliably on the hardware device that the issue
was detected on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3082>
Apparently mesa 22.3.0 has updated the egl headers, and eglplatform.h now
contains commit
3670d645f4
after which xlib headers don't get included by default anymore but are
dependent upon whether USE_X11 was defined.
This breaks headless builds of gstreamer-vaapi because we always define
an internal define USE_X11 as either 1 or 0.
Change these defines to GST_VAAPI_USE_XYZ instead to avoid this.
Fixes#1634
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3555>
This wasn't really done, and is needed in order to detect potential section
changes for sections that have got identical information (such as when switching
between streams that have the same PAT/PMT pid and subtable information).
Other checks exist in tsbase to detect if the "new" PAT/PMT really is an update or not.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3530>
gst_plugin_load_by_name() assumed a plugin has a filename,
which isn't true for static plugins, leading to criticals.
If a plugin is already loaded, just return the loaded plugin,
which makes it work for static plugins as well as saving a
moment for already-loaded dynamic plugins.
Add locking in gst_plugin_is_loaded(), as a plugin may be
still being loaded in another thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3552>
Currently the element calls abort when failed to prepare reference
picture set. This can happent when the input stream is somehow
corrupted, like a rtsp strem with lost packets. Now it will only
return with GST_FLOW_ERROR instead of terminating whole process.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3505>
An end packet is only produced once for the last subtitle, so multiple
GAP events between subtitles would result only in a single end packet
and nothing else otherwise. This would potentially starve downstream
then, so instead forward the GAP events in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3534>
Starting with glib 2.75, `NULL` is `nullptr`, which cannot be
implicitly coerced to `0`, unlike `NULL`. So explicitly pass `0`.
```
[3206/4524] Compiling C++ object subprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj
FAILED: subprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj
"cl" "-Isubprojects\gst-plugins-bad\sys\directshow\gstdirectshow.dll.p" "-Isubprojects\gst-plugins-bad\sys\directshow" "-I..\subprojects\gst-plugins-bad\sys\directshow" "-Isubprojects\gst-plugins-bad" "-I..\subprojects\gst-plugins-bad" "-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" "-Isubprojects\orc" "-I..\subprojects\orc" "-I..\subprojects\gst-plugins-bad\sys\directshow\strmbase\baseclasses" "-Isubprojects\gst-plugins-base\gst-libs\gst\video" "-Isubprojects\gstreamer\gst" "-Isubprojects\gst-plugins-base\gst-libs\gst\audio" "-Isubprojects\gst-plugins-base\gst-libs\gst\tag" "-IC:/gst-install/include/glib-2.0" "-IC:/gst-install/lib/glib-2.0/include" "-IC:/gst-install/include" "/MD" "/nologo" "/showIncludes" "/utf-8" "/W2" "/EHsc" "/O2" "/Zi" "/wd4018" "/wd4146" "/wd4244" "/wd4305" "/utf-8" "/we4002" "/we4003" "/we4013" "/we4020" "/we4027" "/we4029" "/we4033" "/we4045" "/we4047" "/we4053" "/we4062" "/we4098" "/we4101" "/we4189" "/utf-8" "-D_MBCS" "/wd4189" "/wd4456" "/wd4701" "/wd4703" "/wd4706" "/wd4996" "-DHAVE_CONFIG_H" "/Fdsubprojects\gst-plugins-bad\sys\directshow\gstdirectshow.dll.p\dshowvideosink.cpp.pdb" /Fosubprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj "/c" ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(62): warning C5051: attribute 'noinline' requires at least '/std:c++20'; ignored
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(123): error C2664: 'LRESULT SendMessageA(HWND,UINT,WPARAM,LPARAM)': cannot convert argument 3 from 'nullptr' to 'WPARAM'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(123): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type
C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(3690): note: see declaration of 'SendMessageA'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(635): error C2664: 'BOOL SystemParametersInfoA(UINT,UINT,PVOID,UINT)': cannot convert argument 2 from 'nullptr' to 'UINT'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(635): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type
C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(13153): note: see declaration of 'SystemParametersInfoA'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(1593): error C2664: 'LRESULT SendMessageA(HWND,UINT,WPARAM,LPARAM)': cannot convert argument 3 from 'nullptr' to 'WPARAM'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(1593): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type
C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(3690): note: see declaration of 'SendMessageA'
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3528>
The stream selection is done on the currently outputting tracks, but in order to
(de)activate the backing streams we can only do it if the input and output
period are identical.
Fixes crash when doing stream selection during period migration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3525>
This adds "id" variants to most debugging functions, and allows providing a
string identifier instead of a GObject.
This allows providing unified and clearer debug logs for all the
non-gobject-based items, and opens the way for more unified logging.
As an extension, copying the object name is avoided as much as possible, by
using it directly instead of going through another copy.
* API : gst_debug_message_get_object_id
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3483>
This regression was introduce by fix for making buffer pool thread safe. When
we renegotiate, the pool will be setup after we set the format. But the code
has been simplified to only get the pool once before, which caused a null
pointer deref.
Fixes 94ba019 ("v4l2: Fix SIGSEGV on 'change state' during 'format change'")
Related to !3481Fixes#1626
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3513>
There are cases where user might want to be in full control of the
timeline and not be limited by the checks that are being done by GES
to go from one timeline layout to another, this should be doable as
it is a valid use case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3501>
When coloring is in use, those escape codes are going to be created many times
for almost all debug lines.
Don't create plenty of temporary allocations, and instead build the escape code
ourselves statically
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3498>
Because of the asynchronous resolving of mDNS ICE candidates it is
possible that GstWebRTCICE outlives webrtcbin. This in turn prolongs
the lifetime of the GstWebRTCNiceStream objects via refs in
nice_stream_map. Thus the GstWebRTCICETransport objects held in
GstWebRTCNiceStream may be invalid at the time they are accessed by
the _on_candidate_gathering_done() callback since GstWebRTCNiceStream
doesn't take a reference to them. Doing so would create a circular
reference, so instead this commit introduces weak references to the
transport objects and then we can check if the objects are valid before
accessing them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3502>
And even that vaav1dec doesn't use vabasedec negotiate vmethod, it should align
with the new scheme of using base's width & height for surface size and
output_info structure for downstream display size negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>
This vmethod can be used by decoders with the same VA decoder reopen logic:
same profile, chroma, width and height.
Also a new public method called gst_va_base_dec_set_output_state() with the
common GStreamer code for setting the output state, which is always called by
the negotiate vmethod.
In order to do this refactoring, new variables in vabasedec have to be populated
by the decoders:
* width and height define the resolution set in VA decoder. In the case of H264
would be de coded_width and codec_height, or max_width and max_height in AV1.
* output_info is the downstream video info used for negotiation in
gst_va_base_dec_set_output_state().
* input_state, from codec parent class shall be also held by vabasedec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>
There could be multi-GPU setups where the non-first has more
entrypoints than the first one, and the elements names are not
homogeneous, leading to pipeline building error.
This patch add the render node in the elements names when they belong
to the non-first device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3491>
To fix the warning on Alderlake
vafilter gstvafilter.c:534:gst_va_filter_ensure_filters:<vafilter0>
vaQueryVideoProcFiltersCaps: list argument exceeds maximum number
Increase the number of caps to 16 as vadumpcaps does.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3473>
We want to make it so that we prefer a higher, not lower, number of
channels. Otherwise, this pipeline would convert from 2 to 1 channels:
gst-launch-1.0 audiotestsrc ! audio/x-raw,channels=2 ! opusenc ! queue ! opusdec ! queue ! opusenc ! fakesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3494>
In cases where an invalid input packet is submitted to the decoder we emit a
warning but reporting the flow error upstream would also be useful. This came up
with a case were the application interacts directly with the decoder, using a
mechanism similar to GstHarness.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3463>
Whenever the surface is resized before the stream is negotiated, we endup
with an assertion in libgstvideo.
gst_video_center_rect: assertion 'src->h != 0' failed
This fixes it, by following the style aready in place, which is to ensure
surfaces have a minimum size of 1x1.
Fixes#1139
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3467>
gst-launch-1.0 audiotestsrc ! udpsink host=127.0.0.1
gst-launch-1.0 udpsrc ! audioconvert ! autoaudiosink
would crash with a floating point exception when clipping the input
buffer owing to a division by zero because no caps event was received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3469>
Windows supports various IPC methods but that's completely
different form that of *nix from implementation point of view.
So, instead of adding shared memory functionality to existing
shm plugin, new WIN32 shared memory source/sink elements
are implemented in this commit.
Each videosink (server) and videosrc (client) pair will communicate
using WIN32 named pipe and thus user should configure unique/proper
pipe name to them (e.g., \\.\pipe\MyPipeName).
Once connection is established, videosink will create named shared memory
object per frame and client will be able to consume the object
(i.e., memory mapped file handle) without additional copy operation.
Note that implementations under "protocol" directory are almost
pure C/C++ with WIN32 APIs except for a few defines and debug functions.
So, applications can take only the protocol part so that the application
can send/receive shared-memory object from/to the other end
even if it's not an actual GStreamer element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3441>
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.
This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
- Based heavily on the existing Qt5 integration however:
- The sharing of OpenGL resources is slightly different
- The integration with the scengraph is a bit different
- Wayland, XCB and KMS have been smoke tested. Android, MacOS/iOS,
Windows may or may not work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3281>
Make sure that group-id of a given play item are made consistent from the
start (sources) and all the way through the output.
This ensures that we can reliably detect that we have switched to the next play
item on the output of decodebin3 (and we can therefore properly free/release it)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
When shutting down, we want to remove the urisourcebin blocking probes ... but
we also want to propagate a GST_FLOW_FLUSHING upstream (and not
GST_FLOW_NOT_LINKED) to make the upstream task gracefully stop instead of
posting an error message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
When `is_selection_done` is called, it checks that all the requested streams are
present in the active stream list ...
... except there could very well be a (about to be removed) stream from the
previous selection present.
Therefore filter the list of streams we add to the message by the streams which
are actually requested.
Fixes issues when switching between different stream types (ex: video-only to
audio-only).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>
RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.
Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>
There was a drm/drm_mode.h included added recently, drm/ is usually
referencing the linux kernel header, but we only requires the libdrm
headers to be installed. On top of this, including drm_mode.h is never
needed as its already included by drm.h.
Fixes#1596
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3452>
When getting a "404 Not Found" response from the DESCRIBE request, the
source produced a "No supported authentication protocol was found" error
instead of passing on the 404, which was confusing.
Only produce this error message when we're handling a response of "401
Unauthorized" without a compatible WWW-Authenticate header.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3414>
The legacy emulation in DRM/KMS drivers badly interact with GStreamer and
may cause the framerate to be halved. With this property, users can disable
vsync (which is handled internally by the emulation) in order to regain the
full framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3303>
The event type for instant-rate-change events was poorly chosen,
leading to them being re-sent too late and even after EOS.
Add a mechanism in GstPad for the sticky event order to be
different to the value of the event type to fix that up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3387>
The original BUNDLE support commit placed a queue after the
rtpfunnel that combines streams, but I don't see a good reason for
it. It has default settings, so if network output is slow might
accidentally store up to 1 second of pending data, increasing
latency.
Remove it in favour of doing any necessary buffering before
webrtcbin. If it turns out that there is a reason for it to
exist, the limits should probably be configurable and small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3437>
Currently, when rtspsrc property add-reference-timestamp-metadata=true,
a downstream rtph264depay element will attach multiple copies of the
same GstReferenceTimestampMeta to the depayloaded media buffers. This
can have signficant performance impacts further downstream in a pipeline
like the following:
rtspsrc add-reference-timestamp-metadata=true ! rtph264depay ! h264parse ! ... ! rtph264pay ! ...
For example, if there are 10 packet buffers for a frame of RTP H.264
video, each of those packet buffers will contain the same reference
timestamp meta. The rtph264depay element will then attach all 10
metadata to the depayloaded frame. And then later when we payload the
frame buffer again for proxying, we now have 10 more buffers each with
10 instance of the same metadata. Allocating/deallocating 100+ instances
of metadata @ 30fps for multiple streams has a pretty large performance
impact.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1578
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3431>
The tile width in pixel is not always available. Notably for
8L128 10bit format, the tile is 8x128 bytes, and the pixel
format is fully packed. That means that the tile contains at
least 6 pixels per line, but it also hold some bits of the
pixel from the same line on the previous or next tile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>
In current tile representation, only tiles with power of two
width and height in bytes are supported. This limitation
prevents adding more complex tiles formats.
In this patch, we deprecate tile_ws and tile_hs from GstVideoFormatInfo and
replace if with an array of GstVideoTileInfo. Each plane tiles are then
described with their pixels width/height, line stride and total size.
The helper gst_video_format_info_get_tile_sizes() that depends on the
deprecated API is also being removed. This can simply be removed as it wasn't
in any stable release yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>
Setting force_live lets aggregator behave as if it had at least one of
its sinks connected to a live source, which should let us get rid of the
fake live test source hack that is probably present in dozens of
applications by now.
+ Expose API for subclasses to set and get force_live
+ Expose force-live properties in GstVideoAggregator and GstAudioAggregator
+ Adds a simple test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3008>
The only case where we definitely need to write a new trun is when the
data_offset value does not match the end of the list of entries.
Needing multiple trun atoms is required when interleaving multiple
streams together.
All other cases can be covered by adding more entries to the existing
trun atom.
Fixes playback of fragemented mp4 in ffplay and chrome.
Using e.g. mp4mux fragment-duration=1000 fragment-mode=dash-or-mss
and
mp4mux fragment-duration=1000 fragment-mode=first-moov-then-finalise
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3426>
Attribute's value should use returned value from get_attribute for
VAConfigAttribRTFormat, since VAProfileHEVCMain10, in AMD Mesa Gallium,
can process either VA_RT_FORMAT_420 and VA_RT_FORMAT_420_10, which isn't
considered in gstreamer-vaapi design, where encoder's src pads will
expose only 4:2:0 color formats but no 4:2:0 10bit. So, this is a workaround
for this issue while new vah265enc is released.
Signed-off-by: Boyuan Zhang <boyuan.zhang@amd.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3397>
This was the intention from the start, just took me a few years *cough* to
actually implement it properly.
Gapless is handled by re-using as much as possible the same decoders and sinks
if present, and only pre-rolling switching at the sources level (with buffering
if/when needed).
In order to enable "gapless" playback, the "next" uri should be set at any time
between the moment the `about-to-finish` signal is emitted and the moment the
current play item is done. Previously this could only be done with the signal
emission.
This new implementation also allows "Instantaneous URI switching". This allows a
much faster way of switching playback entries while re-using as many elements as
possible. To enable this set `instant-uri` property to TRUE, the default being
FALSE.
API: instant-uri properties
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
When dealing with gapless input (i.e. streams with changing group-id in
GST_EVENT_STREAM_START), we need to take into account the elapsed
running-time (if applicable) in order to properly calculate levels and output
time. Without doing this all incoming data from future groups would be
considered as being "late" and would be consumed immediately.
This does **NOT** modify the actual segment and buffer times, and is only used
internally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
DecodebinInput (and their backing parsebin or identity) are no longer released
when the corresponding sinkpad is unlinked, but when it's released.
The parsebin element will be resetted:
* If incoming caps are incompatible (was the case before)
* Or when unlinking and it was previously pull-based
This opens the way to use decodebin3 with changing inputs (i.e. gapless)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
Introduce the option to have the streams be parsed with `parsebin` for
compatible sources (i.e. which are eligible for buffering in the same way as
before this commit).
By parsing the inputs directly, this allows more accurate buffering control:
* Instead of relying on potential bitrate information coming from somewhere
* and *without* being linked downstream
If `parse-streams` is activated and the stream is eligible for buffering, then a
`multiqueue` will be used on the output of `parsebin` in order to handle the
buffering.
API: `parse-streams`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
If the incoming streams are already parsed, there is no need to add yet-another
parsebin to process it *IF* that stream is compatible with a decoder or the
decodebin3 output caps.
This only applies if all the following conditions are met:
* The incoming stream can *NOT* do pull-based scheduling
* The incoming stream provides a `GstStream` and `GstStreamCollection`
* The caps are compatible with either the decodebin3 output caps or a decoder
input
If all those conditions are met, a identity element is used instead of a
parsebin element and the same code paths are taken.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
* Instead of creating temporary `PendingPad` structures, always create a
DecodebinInputStream for every pad of parsebin
* Remove never used `pending_stream` field from DecodebinInputStream
* When unblocking a given DecodebinInput (i.e. wrapping a parsebin), also make
sure that other parsebins from the same GstStreamCollection are unblocked
since they come from the same source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
Make an explicit topology/tree of structures:
* ChildSrcPadInfo is created for each source element source pad
* ChildSrcPadInfo contains the chain of optional elements specific to that
pad (ex: typefind)
* A ChildSrcPadInfo links to one or more OutputSlot, which contain what is
specific to the output (i.e. optional buffering and ghostpad)
* No longer use GObject {set|get}_data() functions to store those structures and
instead make them explicit
* Pass those structures around explicitely in each function/callback
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
The following problem could happen:
* Thread 1 : urisourcebin gets activated from READY->PAUSED
* Thread 2 : some element causes a pad to be added to urisourcebin , which gets
linked downstream, which decides to activate upstream to pull-based.
* That requires "activating" the pads from PUSH to NONE, and then from NONE to PULL
* Thread 1 : the base class state change handlers checks if all pads are
activated
The issue is that since going form PUSH to PULL requires going through NONE,
there is a window during which:
* Thread 1 : The pad was set to NONE (before being set to PULL)
* Thread 2 : The base class activates that pad (to PUSH)
* Thread 1 : The attempt to "activate" to PULL fails (silently or not)
This is very racy, so in order to avoid that, we make sure that we only add pads
once the transition from READY->PAUSED in the parent classes is done.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
This change allow output caps to be updated even though we stay in
streaming state. This is needed so that any upstream updated to fields
like framerate, hdr data, etc. can result in a downstream caps event
being pushed.
Previously, any of these changes was being ignored and the downstream
caps would not reflect it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3328>
In theory, input caps can be updated anytime at non-keyframe or
sequence boundary, such as HDR10 metadata, framerate, aspect-ratio
or so. Those information update might not trigger ::new_sequence()
or subclass may ignore the changes.
By this commit, input state change will be tracked by baseclass
and subclass will be able to know the non-decoding-essential
update by checking the codec specific picture struct
on ::output_picture()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3328>
This reverts commit fcad4cc646.
This was wrong is so many ways.
* The memcmp was badly used (it should use == 0 to check the data is identical,
and not != 0)
* There was no boundary checks on the present stream section_data when passing
it to memcmp.
* The return value should have been TRUE (i.e. we have done all checks, none of
them failed, therefore the section has been seen before)
* stream->section_data would *always* be NULL if the section had already been
processed
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1559
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3421>
There are cases where upstream will not provide a framerate, or it won't be
fixed. But if there is latency introduced by the decoder we do want to report
it.
Therefore use the framerate stored in the actual decoder, which will have a
default.
Fixes hangs when playing back such streams with decodebin3 (where the multiqueue
will not have been informed of that downstream latency and not grow accordingly)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3391>
Postpone the cleanup of any consecutive sequence of lost frames
which starts at frame 0, until frame 100 is dequeued from driver.
This allows fluster tests JVT/CVWP2_TOSHIBA_E, JVC/CVWP3_TOSHIBA_E
and HEVC/POC_A_Bossen_3 that sends out-of-order frames to successfully
complete (e.g., test of Amphion vpu driver).
Fixes#1569
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3398>
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, wavparse calls the typefinder helper
except that means it runs all typefinders.
Since it only cares about checking for DTS, we should only run the
audio/x-dts typefinder (if present). Commit 858e516 did not really
fix things.
Use the new type helper with the caps to fix this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3417>
Set udpsrc seqnums on all events sent to the udpsrc's, and before
forwarding events out of rtspsrc set the latest seek seqnum on them if
any.
Also produce a consistent seqnum in rtspsrc from the very beginning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
