This is an additional quality parameter. In the default configuration this
quality switch is deactivated because it would cause a workload increase
which might be significant. If workload is not an issue in the application
it can be recommended to activate this feature.
A flush request is done when set_format is called to empty internal bit
buffer maintained by fdk-aac. When this happens, during the explicit
call to handle_buffer, decodeFrame does not return a AAC_DEC_OK. This
gets reported as a decoding error while no decoding error in fact took
place. Since this can be confusing, just return a GST_FLOW_OK and log
that an explicit flush was requested.
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, right now we call the typefinder helper
which runs all typefinders.
Speed up this type finding process by specifying the extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3294>
GST_TRACERS="leaks" GST_DEBUG="GST_TRACER:7,leaks:6" gst-play-1.0 --use-playbin3 test.mkv
When running a pipeline like above, leaks are observed.
0:00:56.882419132 240637 0x5562c528ccc0 TRACE GST_TRACER :0:: object-alive, type-name=(string)GstConcatPad, address=(gpointer)0x7efd7c0d20a0, description=(string)<'':sink_0>, ref-count=(uint)1, trace=(string);
0:00:56.882429131 240637 0x5562c528ccc0 TRACE GST_TRACER :0:: object-alive, type-name=(string)GstConcatPad, address=(gpointer)0x7efd7c0d2be0, description=(string)<'':sink_0>, ref-count=(uint)1, trace=(string);
0:00:56.882437056 240637 0x5562c528ccc0 TRACE GST_TRACER :0:: object-alive, type-name=(string)GstConcatPad, address=(gpointer)0x7efd7c0d3720, description=(string)<'':sink_0>, ref-count=(uint)1, trace=(string);
gst_element_release_request_pad does not unref the pad. It needs to
be followed by gst_object_unref. Doing that fixes the above leaks.
Use g_ptr_array_new_with_free_func with gst_object_unref as the free
function to unref the pad after release.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3177>
Posting latency messages causes a full and potentially expensive latency
recalculation of the pipeline. While subclasses should check whether the latency
really changed or not before calling this function, we ensure that we do not
post such messages if it didn't change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3282>
The picture parameter picture->top_field_first is reused in this mode
to signal the TOP fields. As a side effect, it will change every frame
and current code assumed that if this changes then a renegotiation is
needed. Fixed this by ignoring that change whenever we are decoding one field
only.
Fixes#1523
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3276>
In fact, all the h264 bit writer have byte aligned output except
the slice header. So we change the API from bit size in unit to
byte size, which is easy to use. For slice header, we add a extra
"trail_bits_num" to return the unaligned bits number.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3193>
We use va pool as msdkvpp's bufferpool, which means both va memory
and dma memory will be allocated by va pool. Considering drm modifier
stuff is not ready, we use va memory with higher priortiry than
dma memory when deciding vpp caps.
Besides, this patch removes the specified "interlace-mode" in vpp caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3253>
The gap handling was in place, but there was no event handler to trigger it.
Implement the alpha sink event handler for the gaps. This fixes handling of
valid streams which may not refresh the alpha frames for every video frames.
It will also allow a clean error if the stream was missing the initial
alpha frame, at least until we find a better way to handle these
invalid frames.
Related to #1518
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3264>
Handle when encoder doesn't support rate control, which is set as
VA_RC_NONE, and if the set rate control mode is not supported by the
GStreamer element, the element configuration fails.
Also it logs out max and target bitrate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3063>
The entrypoint is set when the encoder helper is constructed,
nonetheless it was also passed as parameter when opening. That's
buggy.
In order to simplify the code, the entrypoint at construction is
honored.
But gst_va_encoder_has_profile_and_entrypoint() now doesn't rely in
the internal list of profiles since it only contains those that
belongs to codec and entrypoint, thus it queries directly the VA
driver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3063>
Need to put the actual profile in the output caps otherwise any
capsfilter after the encoder that was used to force the output
profile will fail, such as
fdkaacenc ! audio/mpeg,stream-format=adts,profile=he-aac-v1 ! ..
because we put profile=lc in there to match the profile signaled
in the ADTS header. This is expressed through the base-profile=lc
in the GStreamer caps though, the profile needs to carry the
'real' profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
