Original commit message from CVS:
2005-02-18 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_dump_byte_array):
Allocate the 1 byte more memory that was forgotten!!!!!
Flesh out the video filter base class. Make it parse the input and output caps
and turn them into GstVideoInfo. Map buffers as video frames and pass them to
the transform functions.
This allows us to also implement the propose and decide_allocation vmethods.
Implement the transform size method as well.
Update subclasses with the new improvements.
With the new video bufferpool we can now implement the propose_allocation
vmethod on some video filter elements so that we can also use video metadata and
bufferpools when not operating in passthrough mode.
GstCollectPads2 locking was changed from GstCollectPads to use
the stream lock instead of the object lock for those cases, so
change it so here as well to match.
https://bugzilla.gnome.org/show_bug.cgi?id=666379
... to also properly indicate chain's endpad if no elements are in the
chain (due to the endpad being a raw demuxer pad, or one setup without
decoders since uridecodebin or higher up decided not to need those).
Previously we always used textoverlay for rendering the output of
a parser, now the same code as for the renderers is used and the
element with the highest rank is used.
Fixes bug #663822.
We added the utf typefinder because the mp3 typefinder was a tad
overzealous when it came to typefinding things as mp3, and replaced
it with even more overzealous utf16/32 typefinders.
Fixes unit test.
This reverts commit bd539753eb.
Adding the supported metadata to the query does nothing at this stage. Proposing
allocation parameters and supported metadata for upstream should use the
propose_allocation vmethod.
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
Replace g_thread_create() with g_thread_try_new().
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.
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.
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.
After preroll the multiqueue limits are still set to the preroll
limits if use-buffering is set to TRUE. In that case we only want
time limits on the multiqueue if upstream is seekable.
Such streams were detected as seekable, as the query on the typefind
element was testing the m3u8 file listing the actual streams, and
not going through the demuxer(s).
We now check for seekability for each multiqueue following a demuxer,
so the query will flow through the elements which might prevent seeking.
https://bugzilla.gnome.org/show_bug.cgi?id=647769
API: GstVideoRate:force-fps
Changing the framerate during playback is not possible
with a capsfilter downstream if upstream is not using
gst_pad_alloc_buffer(). In that case there's no way in
0.10 to signal to videorate that the preferred framerate
has changed.
This new property will force the output framerate to
a specific value and can be changed during playback.
The ghostpad acceptcaps functions are not valid in this case because
we don't only accept the caps accepted by the target but could also
insert converters. Fixes bug #663892.
