Previously, we would create a new GstMemory per write operation
and then append them to the GstBuffer. This would cause a reallocation
every 16 Memories which is an issue since the png encoder will usually
do write in a pattern of 4, 8 and 8k bytes repeating until the frame
is done.
Instead allocate a single GstMemory and keep writting it into it
with a manual index. Much like the jpegenc does.
Doing some basic testing With a testsrc snow pattern at 4k and 8k
the same pipeline would take ~3.30s to encode a 4k frame and ~23s
for an 8k. At 4k 0.70s/33% is taken by memory allocations, while at
8k its ~10.5s/45%.
With this patch, at 4k the pipeline takes ~2.40s and at 8k only 9.60s
making this 28% and 58% faster accordingly on my laptop, and
allocation runtime is dropped to subsecond times.
Here's the test pipeline used, increase num-buffers in image freeze
to gather more samples.
```
gst-launch-1.0 videotestsrc num-buffers=1 pattern=snow ! imagefreeze num-buffers=1 ! \
video/x-raw,width=7680,height=4320 ! pngenc ! fakesink
```
Close#2717
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4944>
lc3enc:
- encodes raw audio into lc3 format
- uses the default bitrate property and frame duration
from the caps to determine the byte count of
the encoded frames if it is not specified in
the downstream caps after negotiation
- uses the same byte count value for all the channels
- all the common session configuration parameters
are passed in the src caps
lc3dec:
- decodes an lc3 encoded audio
- sink caps should contain all the common session configuration
params
- uses frame_duration and frame_bytes (byte count) in the sink
caps as parameters along with sample rate and channel count
- byte count is same for all the channels
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4376>
`srt_rejectreason_str` doesn't give us a unique string for every
possible reason. Peers can define their own reasons and SRT just gives
us the string `"Application-defined rejection reason"` for all of them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4948>
When gst_element_set_state is called in _setup_locked and errors, the
callback is already processed before we reach handle_current_async, and
the timer is started even though it's finished processing, which results
in a NULL pointer crash later in async_timeout_cb.
To fix this, we check that it's still processing before calling
handle_current_async.
Fixes#1683
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4936>
The aim of this example is to show how to make use of the accept-certificate
signal from a GTK GUI, and prompt user in case of invalid certificate.
There are two subtleties to be aware of:
1. the signal is emitted from the GStreamer streaming thread, therefore the
caller can't modify the GUI straight away, instead they must do it from the
main thread (eg. by using g_idle_add())
2. in case of a redirection, then a TLS failure, the caller won't know
about the redirection. Actually, it's possible to be notified of the
redirection by watching "message:element" and inspecting http-headers,
but even in that case, the signal will be received *after* the signal
"accept-certificate" (even though the redirection happened *before*).
This second point is tricky. It's not uncommon to have servers that redirect
http requests to https. So errors of the type "HTTP -> HTTPS -> TLS error"
happen, and if the caller doesn't care about redirection, they might prompt
users with a message such as "TLS error for URL http://...", which wouldn't make
much sense.
This example shows how to handle that right, by connecting to the signal
"message:element", inspecting the http-headers, and in case of redirection,
updating the TLS error dialog to indicate that the request was redirected.
Here are a few examples of streams that exhibit TLS failure (at the time of
this commit, of course):
* https://radiolive.sanjavier.es:8443/stream: unknown-ca
* https://am981.ddns.net:9005/stream.ogg: unknown-ca
* http://stream.diazol.hu:7092/zene.mp3: redir then bad-identity
* https://streaming.fabrik.fm/izwi/echocast/audio/index.m3u8: unknown-ca
(this one is a HLS stream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4925>
With libsoup 2.x, it was possible to know when there was a TLS failure, as
libsoup provided the "special http status code" SOUP_STATUS_SSL_FAILED.
However these special codes were dropped with libsoup 3.x: now libsoup emits
the accept-certificate signal when there's a TLS failure.
This commit adds a signal "accept-certificate" to SoupHttpSrc, which is in fact
just about forwarding the signal from SoupMessage (which is, itself, forwarded
from GTlsConnection). Note that, in case of libsoup 2.x, the signal is never
emitted.
Fixes: #2379
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4925>
Adding new subtitle overlay element. It's a bin which is wrapping
two internal elements dwritesubtitlemux and dwritetextoverlay.
* dwritesubtitlemux: A new internal element to aggregate subtitle
buffers and to attach the aggregated subtitle buffers on
video buffer as meta.
* dwritetextoverlay: Extracts/renders the subtitle meta and
discard the meta after rendering.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4934>
In the current implementation, we support for most pixel format left
and top padding by changing the offset in the video meta. Though, to
align driver bytesused to the offset, we recalculate the offset, which
removed the modification we did before.
Instead, save the plane size, and truncate the driver reported bytesused
to the expected size, which ensures that the offsets still match. This
should also fix issues were the buffer size ended up bigger then the
pool size due to driver introduced padding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4920>
There's no reason to release GstMemory manually at all.
If we do release GstMemory, corresponding GstBuffer will be
discarded by GstBufferPool baseclass because the size is changed
to zero.
Actual cause of heavy CPU usage in case of fixed-size pool
(i.e., decoder output buffer pool) and if we remove GstMemory from
GstBuffer is that GstBufferPool baseclass is doing busy wait in acquire_buffer()
for some reason. That needs to be investigated though, discarding
and re-alloc every GstBuffer is not ideal already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4935>
Fixes test: validate.uridecodebin.expose_raw_pad_caps
testsrcbin (currently part of debugutilsbad) is an useful element for
validate tests.
validate.uridecodebin.expose_raw_pad_caps makes use of it.
Unfortunately, because validate tests with GStreamer only run with
whitelisted plugins and `debugutilsbad` wasn't in the whitelist, the
test was failing and being auto-skipped.
This patch adds debugutilsbad to the whitelists used by validate tests
in subprojects with a validate/meson.build.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4931>
The current way of using gst_video_info_set_format() will change all
fields of the GstVideoInfo. We only need to change its format, stride
and offset fields.
In order to keep the consistency with th common drm API, we rename the
gst_va_video_info_from_dma_info() into gst_va_dma_drm_info_to_video_info().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4883>
The current way only selects the best video format from the first
structure of the caps. The caps like:
video/x-raw(memory:VAMemory),drm-format=(string)NV12; \
video/x-raw(memory:VAMemory),format=(string){ NV12, Y210 }
Will just choose NV12 as the result, even the bitstream is 10 bits.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4928>
ges-timeline-element property getter handler was using
g_value_take_object() with internal pointers of the element as
arguments, instead of g_value_set_object().
g_value_take_object() moves the ownership of the reference; hence,
when reading "timeline" the reference ownership of timeline is moved
away from the ges-timeline-element and into the GValue.
Since GValues are temporaries that are often discarded quickly after,
this can easily lead to a double free. This was causing
gst-editing-services / pythontests to crash when running
TestTrackElements.test_ungroup_regroup() because of an innocent read of
`clip2.props.timeline` around the end of the test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4924>
Some NLE tests were calling ges_init() from the test suite
initialization function. This was causing them to deadlock when running
with glib 2.76.3.
ges_init() causes a GThreadPool to be initialized. Even if GES at this
point doesn't request any thread to be created, since glib 2.63.4+
(see 8aeca4fa64)
the first time a GThreadPool is initialized a "pool-spawner" thread is
created, which is later used by g_thread_pool_push().
The default behavior of the GStreamer check tests is to fork for every
test case. This is not safe if any thread has been created at this
point. In this particular case, GThreadPool preserves the state that
says a "pool-spawner" thread has been created, and will have access to
its mutex and condition variable, but their queues will have different
contents as the memory has been forked. In consequence, calls to
g_thread_pool_push() will deadlock.
The deadlock will not occur if running the tests with CK_FORK=no.
This patch modifies the affected tests to only call ges_init() from
inside the test cases, fixing the deadlock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4915>
A significant portion of the NLE test suite was often timing out due to
the tests taking way longer than necessary because the sinks were
synchronizing to the clock, which is the default behavior for
fakevideosink and fakeaudiosink.
Notable was the case of nleoperation.c:test_pyramid_operations, that ran
through a 10 second stream twice. As the default timeout is 20 seconds,
this made the test flaky.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4914>
Refusing an incoming segment in < GST_MATROSKA_READ_STATE_DATA should only be
done if the incoming segment is not in GST_FORMAT_TIME.
In GST_FORMAT_TIME, we are just storing the values and returning, so we can
invert the order of the checks.
Fixes proper segment propagation in matroska/webm DASH use-cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
Is a seek is done on stream-collection post, there are no selected streams
yet. Therefore none would be chosen to adjust the key-unit seek.
If no streams are selected, fallback to a default stream (i.e. one which has
track(s) with GST_STREAM_FLAG_SELECT).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
When seeking is handled by the collection posting thread, there is a possibility
that some leftover data will be pushed by the stream thread.
Properly detect and reject those early segments (and buffers) by comparing it to
the main segment seqnum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
As VK_FORMAT_FEATURE_2_xxx are defined as static const variable, the
vscoce C compiler prevents the initialization of the vk_usage_map
structure with error "C2099: initializer is not a constant".
Init the structure separately.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4904>
The current way of using parent's copy_metadata() virtual function will
selectively filter out some meta such as crop meta. That virtual function
should be used when copying input buffer's meta data into output buffer,
not suitable when importing the input buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4887>
When the input buffer has crop meta, and we need to do copy, we
should consider the uncropped video size and copy the full size
of video memory.
The video meta in this case should contain the full uncropped
resolution info. We can use it to create full size va buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4887>
Adding Direct3D11 backend Qt6 QML videosink element, qml6d3d11sink.
Implementation details are similar to the qt6 plugin in -good
but there are a few notable differences.
* qml6d3d11sink accepts all GstD3D11 supported video formats (e.g., NV12).
* Scene graph (owned by qml6d3d11sink) will hold dedicated and sharable
RGBA texture which belongs to Qt6's Direct3D11 device, instead of sharing
GStreamer's own texture with Qt6.
* All rendering operations will be done by using GStreamer's Direct3D11 device.
Specifically, upstream texture will be copied (in case of RGBA)
or converted to the above mentioned Qt6's sharable texture.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3707>
... otherwise streams with constant size samples defined with a single
`sample_size` for all samples in the `stsz` box fall in the category
`chunks_are_samples` in `qtdemux_stbl_init`, overriding the actual
sample count.
`FOURCC_soun` would set this automatically for `compression_id == 0xfffe`,
however `compression_id` is read from the Audio Sample Entry box at an offset
marked as "pre-defined" in some version of the spec and set to 0 both by
GStreamer and FFmpeg for opus streams.
Considering the stream `sampled` flag is set explicitely by other fourcc
variants, doing so for opus seems consistent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4903>
This will cause an integer overflow a little bit further down because we
allocate a bit more memory to allow for a NUL-terminator.
The caller should've avoided passing that much data in already as it's
not going to be a valid image and there's likely not even that much data
available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4894>
This is a small optimization and avoids restarting the next parsing
iteration on already accepted data.
On its own it would also fix ZDI-CAN-20968 (see previous commit) but the
previous commit independently is also a valid fix for it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4895>
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.
In `build_opus_extension`, `gst_byte_writer_put*_le ()` variants were used,
causing audio streams conversion to Opus in mp4 to offset samples due to the
PreSkip field incorrect value (29ms early in our test cases).
[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875>
Now all codec baseclasses can inform subclasses of correct max DPB size,
and exception handling (e.g., emergency bumping in h.264) has been
improved as well. Smaller number of additional DPB frame allocation
seems to be safe.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4878>
