Apparently I forgot how gobject works, there is no need to expose
it directly as one can call it from the parent_class pointer
This reverts commit 8a64592481.
Add gst_audio_decoder_set_use_default_pad_acceptcaps() to allow
subclasses to make videodecoder use the default pad acceptcaps
handling instead of resorting to the caps query that is, usually,
less efficient and unecessary
API: gst_audio_decoder_set_use_default_pad_acceptcaps
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.
API: gst_audio_decoder_sink_query_default
https://bugzilla.gnome.org/show_bug.cgi?id=753623
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
For alaw/mulaw we should also try to initialize the channel positions in the
ringbuffer's audio info. This allow pulsesink to directly use the channel
positions instead of using the default zero-initialized ones, which doesn't
work well.
https://bugzilla.gnome.org/show_bug.cgi?id=751144
This new clock slaving method allows for installing a callback that is
invoked during playback. Inside this callback, a custom slaving
mechanism can be used (for example, a control loop adjusting a PLL or an
asynchronous resampler). Upon request, it can skew the playout pointer
just like the "skew" method. This is useful if the clocks drifted apart
too much, and a quick reset is necessary.
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
https://bugzilla.gnome.org/show_bug.cgi?id=708362
We only get here if we don't have any srcpad caps, and we're going
to override the GstAudioInfo a few lines below anyway without ever
using it if for whatever reason we get caps here.
memcmp will blindly compare the reserved fields, as well as any
padding the compiler may choose to sprinkle in GstSegment.
Fixes valgrind complaints in unit tests, as well as some found via
https://bugzilla.gnome.org/show_bug.cgi?id=738216
When the ringbuffer is deactivated and then acquired, if the audio clock
provided by the sink gets reset to zero, we need to add an offset to the
clock to make sure that subsequent samples are written out at the right
times. While we need to leave this to derived classes to take care of
when they provide their own clock (since that clock may or may not be
reset to zero), we can do this ourselves if we know the provided clock
is our own (which does reset to zero on a re-acquire).
Make sure to update the output segment to track the segment
we're decoding in, but don't actually push it downstream until
after buffers are decoded.
https://bugzilla.gnome.org/show_bug.cgi?id=744806
If we have timestamps on input buffers and are in trickmode no-audio
mode, then don't pass anything to the subclass for decode and simply
send gap events downstream
Only for forward playback for now - reverse requires accumulating
GAP events and pushing out in reverse order.
https://bugzilla.gnome.org/show_bug.cgi?id=735666
In trickmode no-audio mode, or when receiving a GAP buffer,
discard the contents and render as a GAP event instead.
Make sure when rendering a gap event that the ring buffer will
restart on PAUSED->PLAYING by setting the eos_rendering flag.
This mostly reverts commit 8557ee and replaces it. The problem
with the previous approach is that it hangs in wait_preroll()
on a PLAYING-PAUSED transition because it doesn't commit state
properly.
https://bugzilla.gnome.org/show_bug.cgi?id=735666
The decoder can fail to drain on EOS if there was only one gather
set, because it will never have sent the segment event downstream
and set the output segment, and fail to detect that the rate < 0.0
Make sure to send pending events before sending all the gather data
for decode.
Don't render out silence samples to a buffer, just
start the clock running, since any buffer with the
GAP flag will be discarded in render() now anyway.
Make the base audio sink throw away buffers marked GAP, or all
incoming buffers when performing a trick play with
GST_SEGMENT_TRICKMODE_NO_AUDIO flag set, and make sure to start
the ringbuffer when that happens so the clock starts running.
Preserve the timing calculations when rendering, so state is all
updated the same, but just don't render samples.
https://bugzilla.gnome.org/show_bug.cgi?id=735666
Some audio sink sub-classes (pulsesink) don't start their clock
when the ringbuffer starts, but always have to on EOS. When we
explicitly need to start the ringbuffer, make sure sub-classes will
do it by (ab)using the existing eos_rendering flag.
