Only update the last_stop value when we had a valid stop position for the
clipping or else the clipping code assumes the stop position extends to the end
of the segment, which makes the position reporting return weird values.
Unify the different position reporting code paths to make it more
understandable.
Use start_time to get more accurate position reporting in paused.
Fix unit tests for more accurate reporting.
Use atomic ops to read and write more properties. Taking the preroll lock in get_property
can lock up applications reading the property during preroll.
Add a new enable-last-buffer property. When false, it disables storing the last
received buffer in basesink::last-buffer. This can be useful in cases where
buffers need to be released asap.
API: GstBaseSink::enable-last-buffer
For the reason outlined at the beginning of gst_private.h (inline
functions in glib may need the g_log_domain variable). Also include
gst_private.h before using any G_OS_* defines, esp. in plugin loader.
This allows demuxers to update the segment stop of an already
finished stream. This might be needed if some stream goes to
EOS before the duration of the longest stream is known to properly
set the segment stop of all streams to the same value in the end.
Rounding errors with the floating point rate could make it so that we
don't end up exactly at the required stepping duration.
Use the segment clipping boundaries, which are not subject to rate
adjustements, instead to detect when we reached the stepping duration.
Add some debug info related to going to the PAUSED state.
When clamping the base time, correctly use 'now', instead of
'-now' - the intent is to prevent 'now-base' ever being
negative, which would cause a position report outside the segment.
Fixes: #602419
Element base_time is a signed quantity, which leads to basesink returning
a position of 0 when dealing with a negative base time - which are quite
legal when clocks (such as the audio clock) are close to 0.
This doesn't manifest in normal pipelines, of course - but can happen
(at least) when manually setting the base time on a pipeline.
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a tenth of a polar bear.
The code was previously:
* checking if ret was != OK
* .. but if it was FLOW_STEP, swith it to OK
* .. and then not using ret
Instead we just make it more compact by checking if it's OK or STEP.
Update design doc with step-start docs.
Add eos field to step done message
when stepping in reverse, update the segment time field.
Flush out the current step when we are flushing.
When we start stepping, store the start/stop values of the segment before we
install new start/stop values for clipping in non-flushing steps.
for non-flushing steps, update the element start time. For flushing steps, it
does not change because running_time does not advance
Make sure we always perform the stop_stepping operations even when we drop
frames.
Note in the docs that a flushing step in PLAYING brings the pipeline to the lost
state and skips the data before prerolling again.
Implement the flushing step correctly by invalidating the current step
operation, which would activate the new step operation.
When a subclass is blocking in _wait_preroll() in the _render method, make sure
we can unlock the subclass and detect this return value from the render method.
Update framestep document, we want to pass the flush flag in the step-done
message.
Add flush flag to the gstmessage.
Update examples to use the new step-done message api.
Implement framestep with playback rates < 0.0 too.
Make start and stop_stepping methods and move their invocation in the right
places.
Perform the atual stepping operation where we have full context about the
timestamps.
Unlock the prerolled frame and recheck if we need to step.
Keep a simple counter for the frames we're about to skip while stepping and
preroll/post step_done when stepping finished.
Due to a typo basesink didn't do any emergency rendering of late buffers
if the only buffer ever rendered was the first one with timestamp 0. This
means that in cases where the decoder is very very slow, we'd never see
any buffers but the very first one rendered. Fixes#576381.
When we are not ready to handle a latency query (we are not yet prerolled) we
also don't try to forward the latency event because that might cause unexpected
errors when upstream is not yet linked.
Fix a regression introduced by fix for #567725 in commit
1c7ab4ed4f. We should only call the preroll
function once namely when we did not yet commit the state change.
Add a unit test to check that we call the preroll function when interrupting the
clock_wait (see #567725).
Add a unit test to check that we only call the preroll function once.
Original commit message from CVS:
* libs/gst/base/gstbasesink.c: (gst_base_sink_commit_state),
(gst_base_sink_wait_clock):
* libs/gst/base/gstbasesink.h:
Fix documentation for the wait_clock method, rename basesink -> sink
for consistency.
Original commit message from CVS:
* libs/gst/base/gstbasesink.c: (gst_base_sink_get_position_last),
(gst_base_sink_get_position_paused), (gst_base_sink_get_position):
Release the object lock before calling the query convert pad functions
to avoid deadlocks.
Original commit message from CVS:
* libs/gst/base/gstbasesink.c: (gst_base_sink_class_init),
(gst_base_sink_init), (gst_base_sink_set_property),
(gst_base_sink_get_property):
Expose the render-delay as a property so things like appsink can use it
to tweak the synchronisation.
Original commit message from CVS:
* libs/gst/base/gstbasesink.c: (gst_base_sink_set_render_delay):
Changing the render delay changes the latency and so we must post a
latency message.
Original commit message from CVS:
* gst/gstbin.c: (bin_handle_async_start),
(gst_bin_handle_message_func), (gst_bin_query):
* libs/gst/base/gstbasesink.c: (gst_base_sink_render_object),
(gst_base_sink_event), (gst_base_sink_change_state):
* libs/gst/base/gstbasesrc.c: (gst_base_src_perform_seek),
(gst_base_src_loop), (gst_base_src_change_state):
Copy seqnums from events to messages so that they can all be related
back to eachother.
Original commit message from CVS:
* libs/gst/base/gstbasesink.c: (gst_base_sink_default_do_seek),
(gst_base_sink_default_prepare_seek_segment),
(gst_base_sink_perform_seek), (gst_base_sink_get_position_last),
(gst_base_sink_get_position_paused), (gst_base_sink_get_position),
(gst_base_sink_query):
Implement more seeking in pull mode.
Use pad convert functions to convert position to the requested format.
Fix position/duration reporting in pull mode.
Implement position and duration reporting in other formats than time.
* libs/gst/base/gstbasesink.h:
Add member to keep track of when the segment is playing.
Original commit message from CVS:
* libs/gst/base/gstbasesink.c: (gst_base_sink_pad_activate_pull),
(gst_base_sink_query):
Query the total number of bytes when activating the pad in pull mode.
Implement duration query in pull mode by using the installed pad convert
function to convert from bytes to the requested format.
Original commit message from CVS:
* docs/libs/gstreamer-libs-sections.txt:
* libs/gst/base/gstbasesink.c: (gst_base_sink_do_preroll),
(gst_base_sink_flush_start), (gst_base_sink_flush_stop),
(gst_base_sink_event), (gst_base_sink_perform_seek),
(gst_base_sink_loop), (gst_base_sink_pad_activate_pull),
(gst_base_sink_send_event), (gst_base_sink_change_state):
* libs/gst/base/gstbasesink.h:
Add method to commit the state in subclasses.
Refactor the flush_start and flush_stop code because we need it for
flushing while seeking too.
Implement the beginnings of seeking in pull mode.
Use the segment last_stop field for the pulling offset.
Fix the pause method in pull mode.
Configure the segment to BYTES for pull mode.
API: GstBaseSink::gst_base_sink_do_preroll()
