This reverts commit 8e923a8e2d.
This caused regressions, see #3303.
Without this commit, osxaudiosrc ! osxaudiosink won't work
right, but since that hasn't really been a huge problem
for years it's probably best to revert this until a proper
solution can be figured out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6356>
Memory from gst_adapter_map() could live shorter than the GstMemory that the GstBuffer wraps around it, which in lucky
cases 'just' caused a re-use of the same memory for multiple (potentially still in use!) input buffers, but could easily
end up pointing to an already-freed memory.
Manifested when an AudioToolbox encoder kept getting silence inserted in seemingly random circumstances, turned out
to be the memory being re-used by GStreamer at the same time that the AT API was processing it...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6349>
Cocoa version of glwindow only checks the preferred size upon window creation. glimagesink sets the size right before
calling gst_gl_window_show(), which might be way after the window is created in some cases. If the size was set too
late, glimagesink on macOS would remain 320x240 unless manually resized.
This change makes sure to resize the existing window when _show() is called.
Curiously, this has always been an issue, but went from manifesting every once in a while to being almost completely
broken once old event loop workarounds were removed and gst_macos_main() was introduced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6280>
Upon creating a window, glimagesink and osxvideosink now set the policy to
NSApplicationActivationPolicyRegular, which lets us show an icon in the Dock
for convenience and appear in the top menu bar like other apps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6103>
* Bump the rank of the musepack v7/v8 FFmpeg demuxers to SECONDARY
* Bump the rank of the musepack v7/v8 FFmpeg audio decoders to SECONDARY
* Demote the rank of the musepackdec element to MARGINAL
This is for two reasons:
* The musepack library is no longer maintained, whereas the FFmpeg
implementation can/will receive fixes
* The `musepackdec` implementation was a all-in-one "parsing and decoding" blob
which doesn't play nicely with decodebin3 and others
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3033
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6085>
Converting from RGB to YUV: When comparing the info.colorimetry to
GST_VIDEO_COLORIMETRY_BT709 it does not make sense to look at the input
signal because that is of type of RGB. The plugin needs to look at the
output YUV-type and compare GST_VIDEO_COLORIMETRY_BT709 to that, because
that is the YUV-type the plugin needs to convert input-RGB into.
Converting from YUV to RGB: Comparing to the input is correct, but because
here the color encoding info BT601/BT709 is on input side of the plugin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6046>
Don't call wait_event() at all for gap events, as basesink will
end up waiting for the time that the gap event would be rendered
out at the audio device. There's no need to render it at all,
just treat it as a handy point to resync the audio if needed,
let the ringbuffer render silence, and place the next buffer
into the ringbuffer where it belongs.
The only thing we really need to do is make sure the ringbuffer
and clock are running, and wait for preroll.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2749
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5953>
If GST_PAD_SINK is passed in this means that we're supposed to convert
from sink caps to src caps, not the other way around. In other words, if
GST_PAD_SINK is passed we're supposed to produce the possible output
caps.
Previously this was inverted. This had the effect that glcolorconvert
pretended to be able to convert *to* I420 without glDrawBuffers, which is
not possible, and pretended not to be able to convert *from* I420
without glDrawBuffers, which it always supports.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5947>
The propose and decide allocation vfuncs are called directly from
basetransform and need to use the locked accessor function for
retrieving a reliable reference to the GstGLContext (if available)
Fixes spurious crashes on shutdown during pad reconfiguration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5518>
This was causing a memory leak in cases like `gltestsrc ! gltransformation scale-x=0.5 ! glimagesink`.
Parent meta was being added in assumption that those buffers are different, which was not the case here,
creating a reference loop and never freeing the buffer.
Co-authored-by: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5453>
When using `gst_sdp_media_set_media_from_caps` on `application/x-rtp` caps
without `clock-rate` it wrongly reports missing payload type even if `payload`
is present in the caps.
This seems to be a copy&paste error from the error message for missing payload
type.
When using payload=10, both `clock-rate` and some other media properties are
defined by the RTP standard so I was wondering whether I could omit `clock-rate`
and was confused about the error message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5252>
Add gst_audio_ring_buffer_set_errored() that will mark the
ringbuffer as errored only if it is currently started or paused,
so gst_audio_ringbuffer_stop() can be sure that the error
state means that the ringbuffer was started and needs stop called.
Fixes a crash with osxaudiosrc if the source element posts
an error, because the ringbuffer would not get stopped and CoreAudio
would continue trying to do callbacks.
Also, anywhere that modifies the ringbuffer state, make sure to
use atomic operations, to guarantee their visibility
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5216>
The drop-frame rules are specified in “SMPTE ST 12-3:2016” and are
consistent with the traditional ones:
“
To minimize fractional time deviation from real time, the first two
super-frame numbers (00 and 01) shall be omitted from the count at the
start of each minute except minutes 00, 10, 20, 30, 40, and 50. Thus the
first eight frame numbers (0 through 7) are omitted from the count at
the start of each minute except minutes 00, 10, 20, 30, 40, and 50.
”
Where “super-frame” is a group of 4 frames for 120 FPS.
Fixes#2797
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5061>
The current implementation copies metas without checking if the buffer
is writable.
The operation that needs to be done, replacing the input buffer and
copying the metas, is only part of that process. We create a new function
that does both.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5054>
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/4897>
glfilter will unref input buffer after _transform() call immidiately,
but gpu may still reading input buffer for rendering because gl
api is executed async. Need hold reference for input buffer by
adding parent meta to output buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4846>
Appsink will unref prev sample in dispose function. Which is later
when V4L2 video decoder link with appsink as V4L2 video decoder
will close V4L2 device fd during GST_STATE_CHANGE_READY_TO_NULL.
If the video buffer return to V4L2 video decoder after the decoder
closed V4L2 device fd, V4L2 can't release the video frame buffer
which allocated with MMAP mode as application can't call
VIDIOC_REQBUFS 0 to release the video frame buffer by V4L2 driver.
The memory of the video frame will leak.
Unref the gstbuffer in stop() function, so V4L2 video decoder
can received all video frame buffers and release it before close
V4L2 device fd.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4820>
When the alignment contains nothing, all its fields are 0 and always
can be satisfied. So there is no need to validate it in this case.
And there are a lot of places just setting this alignment to default
all zero value, this validation generates lots of warnings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4704>
Add d3d11 conversion path to make gst_video_convert_sample() work
for GstD3D11Memory.
Note that just adding "d3d11download" to the exisitng code is
suboptimal from GstD3D11 point of view because:
* d3d11convert element can support crop/colorspace-conversion/scale
all at once while existing software pipeline needs intermediate steps
for the conversion
* "Process everything on GPU then download it to CPU memory" would be likely
faster than "download GPU memory to CPU then processing it on CPU"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4689>
The proxy and queue are created in the gst_gl_window_wayland_egl_open()
function and will be recreated on open. This leaks both objects, the
wayland client documentation mentions that they should be destroyed
using the appropriate destroy functions.
Found during valgrind memory leak testing, these blocks were marked as
definitely lost.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4355>
These parameters are not actually `out` parameters but must
be allocated and zero-initialized by the calling function.
Marking them as `out caller-allocates` will cause memory
corruptions when calling these APIs from e.g., Python code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4061>
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/4020>
