This is done by adding a capsfilter after every parser/converter that contains
all possible caps supported by downstream elements. A capsfilter is necessary
here because the decoder is only selected after the parser selected a format
and the parser can't know what downstream would support otherwise.
This reverts commit 105814e2c7.
The general consensus seems to be that we should revert this for
now. If such behaviour is desired, we should probably enable it
via a flag. And maybe use the scaletempo plugin instead.
Make enums for the chroma siting for easier use in the videoinfo.
Make enums for the color range, color matrix, transfer function and the
color primaries. Add these values to the video info structure in a Colorimetry
structure. These values define the exact colors and are needed to perform
correct colorspace conversion. Use a couple of predefined colorimetry specs
because in practice only a few combinations are in use.
Add view_id to the video frames to identify the view this frame represents in
multiview video.
Remove old gst_video_parse_caps_framerate, use the videoinfo for this.
Port elements to new colorimetry info.
Remove deprecated colorspace property from videotestsrc.
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
Instead of just assuming all pads are created at the same time,
remember which ones are actually new (via ->pending_blocked_pads).
This allows the following use-case to properly work:
* Upstream starts with audio-only
* Only that pad gets data, blocks and a real audio sink is created
* Upstream laters adds a video stream
* A new pad is requested, blocks and reconfiguration kicks in in
order to add a new real video sink
When we don't have specific {audio|video|text}-sink properties, don't
set them on playsink when reconfiguring.
If we do that, we end up setting the previous configured sink to
GST_STATE_NULL resulting in any potentially pending push being returned
with GST_FLOW_WRONG_STATE which will cause the upstream elements to
silently stop.
https://bugzilla.gnome.org/show_bug.cgi?id=655279
When we have a multi-stream (i.e. audio and video) input and the demuxer
adds/removes pads for a new stream (common in a mpeg-ts stream when the
program stream mapping is updated), the algorithm for EOS handling was
previously wrong (it would only drop the EOS of the *last* pad but would
let the EOS on the other pads go through).
The logic has only been changed a tiny bit for EOS handling resulting in:
* If there is no next group, let the EOS go through
* If there is a next group, but not all pads are drained in the active
group, drop the EOS event
* If there is a next group and all pads are drained, then the ghostpads
will be removed and the EOS event will be dropped automatically.
This allows us to make parsers accept both parsed and unparsed input
without decodebin plugging them in a loop until things blow up, ie.
without affecting applications that still use the old playbin or the
old decodebin.
(Making parsers accept parsed input is useful for later when we want
to use parsers to convert the stream-format into something the decoder
can handle. It's also much more convenient for application authors
who can plug parsers unconditionally in transcoding pipelines, for
example).
This is especially needed when switching between a non-sparse and sparse
video stream, see bug #537382. It also lowers the time needed for switching
between streams a bit.
In particular, in audio only cases whose (estimated) metadata provides bitrate
information, the buffer-size based on such bitrate (and buffer-duration)
will be much more reasonable than queue2 default buffer-size.
For streams at low bitrates we need to set a limit in time because the limit
in bytes might not reached too late, sometimes more than 30 seconds.
This limit can only be set if upstream is seekable (see #584104)
Closes#647769
These reconfigure based on the caps and plugin in converters if
necessary. This also makes switching between compressed and raw
streams work flawlessly without loosing the states of any element
somewhere or having running time problems.
Before playbin2 would use different selectors for raw audio and
compressed audio (and the same for video) and used different
pads from playsink. This made the involved logic much more
complex and was not implemented completely in playsink, which
made it impossible to support files with a compressed and
uncompressed stream that is support by the sink.
playbin2 handles raw/non-raw streams the same now and the
decision is left to playsink, which now can also handle
caps changes from raw to non-raw and the other way around.
Fixes bug #632788.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
In addition to ensuring that an element we want to select in
autoplug-select can enter the READY state, we also now check if it can
accept the caps we wish to plug it for. This is handy for sinks that
need to perform a probe to figure out whether they can actually handle a
given format.
