This avoids triggering plenty of extra code/methods/overhead downstream when
we can just quickly check whenever we want to set caps whether they are
identical or not
https://bugzilla.gnome.org/show_bug.cgi?id=706600
Either there was a flush before that resets everything anyway,
or resetting would make us lose information we might need if
it's just a segment update.
In the unlikely case that the decoder drops a frame before the first
input frame is outputted, use the input segment (since it wasn't
carried over to the output segment yet)
https://bugzilla.gnome.org/show_bug.cgi?id=702502
We have no way of tell the caller of the exact error (e.g. if we're flushing),
so will have to wait until the caller uses API that returns a GstFlowReturn,
for example when pushing this buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=700006
For this release the corresponding GstVideoCodecFrame before
pushing the buffer. The buffer will now be writable unless
the subclass still holds another reference to the buffer or
the frame.
When we get a new buffer, always call the parse function, even if it is a 0
sized buffer. For theora we need to also decode 0 sized buffers.
Ideally we would like to make theoradec be packetized but that fails currently
because of oggdemux and because of the assumptions that the base class makes.
DTS and PTS usually have a non-zero offset between them in MPEG-TS,
so assigning DTS to PTS is almost always wrong. The other, newer
timestamp recovery code does it correctly if we leave it as invalid.
And only return the proportion. The earliest time already can be
retrieved from get_max_decode_time() and by renaming we allow this
to be more extensible in the future.
Add a getter for the QoS proportion and earliest_time to help
subclasses do better estimations based on the proportion.
API: gst_video_decoder_get_qos_info()
https://bugzilla.gnome.org/show_bug.cgi?id=687991
Monitor for reordered output timestamps, and then avoid oldest DTS
as PTS approach, and try for an oldest PTS as out PTS approach,
if at least all valid PTS available.
Avoids bogus estimating upon sparse available input PTS, and tries
to handle all-keyframe input, or input PTS which are actually DTS.
Hold both the stream and the object lock to modify the output_state,
this way it can be safely modified while hold either one or the other.
Also, only hold the object lock in the query
https://bugzilla.gnome.org/show_bug.cgi?id=684832
... by having some more timestamp tracking in a private frame field.
Not doing so would lead to (a.o.) losing the needed minimum timestamp in
an earlier sent frame.
Don't try to take STREAM_LOCK on upstream events such as QOS.
Protect qos-related variables with object lock instead. Fixes
possible deadlock when shutting down in certain situations.
https://bugzilla.gnome.org/show_bug.cgi?id=684658
Drain out the decoder when encountering a gap. Needed for DVD 'still'
sequences which consist of a single video frame, and a large gap
while audio plays.
Only hold back events until the first buffer is generated, then just
send them directly. Otherwise, important events like 'still-frame' are
held forever, waiting for a frame that'll never arrive.
Sometimes the decoder would need to use the pool or the allocator for
something else than just allocating output buffers. For example, the querying
for different parameters, such as asking for a bigger number of buffers to
allocate in the pool.
This patch expose a two getters accessors: one for the buffer pool and the
other for the memory allocator.
Unifies the code and ensures that:
* subclasses needing to use the frame_number on a void* field will
always work
* wraparounds will be automatically taken care of if we have to deal
with more than 2**32 frames
Check that we have a valid output_state before attempting to use it to calculate
the duration of a buffer. It is possible that we don't have a state yet, for
example when we are dropping the first buffers.
Make sure the frame deadline was set before calculating the
max_decode_time. Fixes problems with ffmpeg skipping frames when
it doesn't need to, when the input doesn't have full timestamping
(divx in avi)
Interpolating the timestamps from the picture numbers
does more harm than good, getting it wrong in a lot of
cases (especially reverse playback). Removing it in favour
of simply incrementing the timestamps until there's
something better
Use g_list_free_full instead of walking lists twice when freeing
them.
Remove pointless clause in gst_video_decoder_chain that doesn't
actually have any effect.
Other changes to make the code slightly more like the 0.11
version.
Move processing of the gather list into the flush_parse function.
Add a last ditch attempt to apply timestamps to outgoing buffers
when walking backwards through decoded frames. Requires that each
gathered region has at least one timestamp.
Make sure to remove decoded packets from the decode list when
they are sent - otherwise the list just grows on each cycle, with
more and more frames being decoded and then clipped away.
Break out of the processing loop early on a bad flow return to make
seeking more responsive.
Use the gst_video_decoder_clip_and_push_buf function in reverse
mode, instead of pushing all buffers arbitrarily.
A couple of small efficiency gains in the list handling, by moving
list elements directly and not reallocating, and by reversing
and concatenating the gather list instead of moving it one node
at a time.
Rename the gst_video_decoder_do_finish_frame function to
gst_video_decoder_release_frame.
Rename gst_video_decoder_have_frame_2 to
gst_video_decoder_decode_frame and pass the frame to process
directly, rather than using the current_frame pointer as a holding
pen.
Move the negative rate handling out of the function to where it
is needed, and remove the process flag.
The frames are the owners of the buffers. In cases where a decoder
would keep around reference frames, we need to ensure they don't
disappear early.
To handle this, we pass downstream a complete sub-buffer of the output
buffer, ensuring that the buffer will only be released when downstream
is done with it *AND* the frame is no longer used.
Conflicts:
gst-libs/gst/video/gstvideodecoder.c
Don't replace the initial frame's timestamp with a bogus
one calculated from the (incorrect for Ogg) frame number just
because the 'sync time' hasn't changed.
Also, don't output a bogus warning about the output_frame being
NULL when it's being dropped/skipped due to QoS.
When need to push out all the previously received events, concatenate all the
events from the previous frames (instead of leaking the old ones)
Improve debugging a little
Conflicts:
gst-libs/gst/video/gstvideodecoder.c
Frames receive a refcount when added to the frames list so release that refcount
in gst_video_decoder_do_finish_frame(). Also release the ref on the frame
because gst_video_decoder_do_finish_frame() takes ownership of the passed frame.
This allows subclasses to override it, as is necessary for e.g. the
video-crop meta. It is now necessary that after decide_allocation()
there is always a allocator and a configured buffer pool inside the
query.
Some container formats (like AVI) set DTS on the buffers instead of
PTS.
We detect this by:
* detecting if input timestamps are non-increasing
* detecting if the order the frames come out is the same as the order
they were inputted (meaning the implementation is reordering frames).
If the decoder reorders frames, but input buffer timestamps were not
reordered, that means the buffers has DTS and not PTS as their timestamp.
If this is the case, we use set the PTS of the outgoing frames in the
same order as they were given to the decoder.
This fixes the issue for any decoder using this base class (yay).
