gst_tag_image_data_to_image_buffer() ->
gst_tag_image_data_to_image_sample() And make it return a GstSample.
Store the image-type into the extra sample info.
Remove a deprecated tag
Make appsink return a GstSample. Remove the pull_buffer_list method because it
is not very useful anymore.
Pass GstSample to the conversion function.
Update playbin2 and examples
The output size of a buffer does not depend on the input size but simply on the
caps of the output buffers. Don't let the base implementation deal with
unit_sizes, because input buffers might not be a multiple of that when they have
padding or non-default strides. instead, implement a transform size function
that simply calculate the natural size of an output buffer based on the caps.
Remove old useless caps code.
Make a negotiate function and use the configured caps as the caps on the appsrc
pad. If nothing was configured, fall back to the parent implementation.
Doing dynamic pipelines is hard in 0.10. As we don't have the sticky events in
0.10 and sending such events in special elements like adder and tee was outvoted
on last attempt, be graceful to the misbehaviour instead.
This happens when the internal elements are added before any NEWSEGMENT
event arrived and in that case we shouldn't send a NEWSEGMENT event
to the internal elements at all. They will get the NEWSEGMENT event
from upstream later.
If the sink supports raw audio/video, we first check
if the decoder could output any raw audio/video format
and assume it is compatible with the sink then. We don't
do a complete compatibility check here if converters
are plugged between the decoder and the sink because
the converters will convert between raw formats and
even if the decoder format is not supported by the decoder
a converter will convert it.
We assume here that the converters can convert between
any raw format.
Fixes bug #665120.
Make out args to gst_video_event_parse_{downstream|upstream}_force_key_unit
optional, update libgstvideo.def and fix docs a bit.
API: gst_video_event_new_upstream_force_key_unit
API: gst_video_event_new_downstream_force_key_unit
API: gst_video_event_is_force_key_unit
API: gst_video_event_parse_upstream_force_key_unit
API: gst_video_event_parse_downstream_force_key_unit
https://bugzilla.gnome.org/show_bug.cgi?id=607742
fix build errors:
gsttypefindfunctions.c:248:25: error: 'low' may be used uninitialized in this function [-Werror=uninitialized]
gsttypefindfunctions.c:239:24: error: 'high' may be used uninitialized in this function [-Werror=uninitialized]
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
audioresample is derived from GstBaseTransform, and one of
GstBaseTransform's traits is that if the derived element does not
produce an output buffer from some input buffer then the first output
buffer after that gets flaged as a discontinuity, whether or not the
buffer actually is discontinuous from the output buffer that preceded
it. When downsampling, the audioresample element requires more than
one input sample for each output sample, and if the ratio of input to
output sample rates is high enough and the input buffers short enough
it can come to pass that the resampler does not receive enough samples
on its input to produce any output. Currently the resampler returns
GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case,
causing the next buffer to be flagged as a discontinuity. If subsequent
elements in the pipeline reset themselves on disconts, this can cause
clicks and other undesireable behaviour.
Fixes bug #665004.
