Don't use private GMutex implementation details to check
whether it has been freed already or not. Just turn dispose
function into finalize function which will only be called
once, that way we can just clear the mutex unconditionally.
the calculations for drawing the videomark is being repeated
in for loop unnecessarily. Moving this outside of for loop
such that the code need not be executed evertime the loop is executed.
https://bugzilla.gnome.org/show_bug.cgi?id=744371
Always update the segment and not only for accurate seeking and always
send a new segment event after seeks.
For non-accurate force a reset of our segment info to start from
where our seek led us as we don't need to be accurate
https://bugzilla.gnome.org/show_bug.cgi?id=743363
steal_buffer() + unref seems to be a wide-spread idiom
(which perhaps indicates that something is not quite
right with the way aggregator pad works currently).
By implementing get_live_seek_range.
As shown by :
gst-validate-1.0 playbin \
uri=http://dev-iplatforms.kw.bbc.co.uk/dash/news24-avc3/news24.php
This patch handles live seeking, by setting a live seek range
comprised between now - timeShiftBufferDepth and now.
The inteersting thing with this stream is that one can actually
ask fragments up to availabilityStartTime, but it seems quite clear
in the spec that content is only guaranteed to exist up to
timeShiftBufferDepth.
One can test live seeking this way :
gst-validate-1.0 playbin \
uri=http://dev-iplatforms.kw.bbc.co.uk/dash/news24-avc3/news24.php \
--set-scenario seek_back.scenario
with scenario being:
description, seek=true
seek, playback-time=position+5.0, start="position-600.0",
flags=accurate+flush
This example will play the stream, wait for five seconds, then seek back
to a position 10 minutes earlier.
https://bugzilla.gnome.org/show_bug.cgi?id=744362
Where possible, use the _OBJECT variants in order to track better from
which object the debug statement is coming from
Define (and use) GST_CAT_DEFAULT where applicable
Use GST_PTR_FORMAT where applicable
And use the average to go up in resolution, and the last fragment
bitrate to go down.
This allows the demuxer to react rapidly to bitrate loss, and
be conservative for bitrate improvements.
+ Add a construct only property to define the number of fragments
to consider when calculating the average moving bitrate.
https://bugzilla.gnome.org/show_bug.cgi?id=742979
Add the diff between the external time when we went to playing and
the external time when the pipeline went to playing. Otherwise we
will always start outputting from 0 instead of the current running
time.
If the src framerate and videoaggreator's output framerate were
different, then we were taking every single buffer that had duration=-1
as it came in regardless of the buffer's start time. This caused the src
to possibly run at a different speed to the output frames.
https://bugzilla.gnome.org/show_bug.cgi?id=744096
gstdecklink.cpp: In member function 'virtual HRESULT GStreamerDecklinkInputCallback::VideoInputFrameArrived(IDeckLinkVideoInputFrame*, IDeckLinkAudioInputPacket*)':
gstdecklink.cpp:498:22: error: comparison between signed and unsigned integer expressions [-Werror=sign-compare]
if (capture_time > m_input->clock_start_time)
^
gstdecklink.cpp:503:22: error: comparison between signed and unsigned integer expressions [-Werror=sign-compare]
if (capture_time > m_input->clock_offset)
^
The driver has an internal buffer of unspecified and unconfigurable size, and
it will pull data from our ring buffer as fast as it can until that is full.
Unfortunately that means that we pull silence from the ringbuffer unless its
size is by conincidence larger than the driver's internal ringbuffer.
The good news is that it's not required to completely fill the buffer for
proper playback. So we now throttle reading from the ringbuffer whenever
the driver has buffered more than half of our ringbuffer size by waiting
on the clock for the amount of time until it has buffered less than that
again.
The ringbuffer's acquire() is too early, and ringbuffer's start() will only be
called after the clock has advanced a bit... which it won't unless we start
scheduled playback.
Not from the decklink clock. Both will return exactly the same time once the
decklink clock got slaved to the pipeline clock and received the first
observation, but until then it will return bogus values. But as both return
exactly the same values, we can as well use the pipeline clock directly.
