The cancelled flag was only set in the stream finalize()
method, after all activity on the stream has stopped anyway.
Replace uses of cancelled with checks on the stream state.
Remove the replaced flag, which was checked but never set
to TRUE anywhere any more.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
With non-serialized sticky events, such as GST_EVENT_INSTANT_RATE, we both want
to store the event (for later re-linking) *AND* push the event in a non-blocking
way.
We therefore must *not* propagate pending sticky events if the event is "sticky
or serialized" but only if it's "serialized"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3254>
- Make the srt_epoll_wait loops more uniform.
- Error only via GError when possible; let the element send the error
message. Avoids a second error message.
- Return 0 when cancelled. Avoids an error message from the element.
- Don't send an error message from send_headers when we're a server
sink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
When using the child proxy notation (child::property=value) it may
happen that the target child does not exist at the time of parsing
(i.e: decodebin creates the encoder according to the contents of the
stream). On this cases, we want to delay the setting of the property
to later, when new elements are added. Previous logic performed a
delayed set even if the target child was found but the property
was not found in it. This should be treated as a failure because,
unlike missing elements, properties should not appear dynamically.
By not failing, typos in property names may go unnoticed to the end
user.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2908>
These null checkes are slightly misleading when double-checking
mutability for external language interop. None of the functions in
these files allow the variable at hand to become `NULL` under normal
operation, because they are checked at initialization and never (allowed
to be) reassigned to `NULL`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1615>
When matching segments across playlists with Program-Date-Times,
use the difference in segment PDTs to adjust the stream time
that's being transferred. This can fix cases where the
segment boundaries don't align across different streams
and the first download gets thrown away once the PTS
is seen and found not to match.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3309>
Check whether the init file / MAP data for a segment
is different to the current data and trigger an
update if so. Previously, the header would only
be checked in HLS after switching bitrate or
after a seek / first download.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3307>
Previously the minimum buffering threshold was hardcoded to a specific
value (10s). This is suboptimal this an actual value will depend on the actual
stream being played.
This commit sets the low watermark threshold in time to 0, which is an automatic
mode. Subclasses can provide a stream `recommended_buffering_threshold` when
update_stream_info() is called.
Currently implemented for HLS, where we recommended 1.5 average segment
duration. This will result in buffering being at 100% when the 2nd segment has
been downloaded (minus a bit already being consumed downstream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3240>
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>
