The previous code was setting keytarget to target
to make sure the keyframe found for each pad was
indeed before the target.
Then if target == keytarget, it assumed a keyframe had been
found, which was not the case if target was before the first frame
in the file.
This patch checks that a keyframe was indeed found, and if not
seeks to 0, without bisecting again.
Assuming default gst qa assets in $HOME/gst-validate
seek_before_first_frame.scenario:
description, seek=true, handles-states=true
pause, playback-time=0.0
seek, playback-time=0.0, start=0.0, flags=accurate+flush
seek, playback-time=0.0, start=0.01, flags=accurate+flush
seek, playback-time=0.0, start=0.1, flags=accurate+flush
GST_DEBUG=*theoradec*:2 gst-validate-1.0 playbin \
uri=file://$HOME/gst-validate/gst-qa-assets/medias/ogg/vorbis_theora.0.ogg \
--set-scenario seek_before_first_frame.scenario
https://bugzilla.gnome.org/show_bug.cgi?id=741097
When encoding, libvorbis will tell us how many samples are encoded
in the buffer it returns. This number may be less than the maximum
of samples in the block, if this is the last packet. In we have no
segment end time, we set it to the end time of that last sample to
tell downstream that the buffer contains less samples.
Samples may be clipped at the end, and this is conveyed by a
granulepos that's smaller than it would otherwise be. Use the
segment stop time to detect this, and calculate the right
granulepos.
When the textoverlay is set outside the video frame by deltax or deltay the
calculation segfaults, but it is also unnecessary since it doesn't need to be
displayed. So we should clip the text.
https://bugzilla.gnome.org/show_bug.cgi?id=738242
vorbis_reorder_map is defined for eight channels max. If we have more
than eight channels, it's the application which shall define the order.
Since we set audio position to none, we just interleave all the channels
without any particular reordering.
https://bugzilla.gnome.org/show_bug.cgi?id=737742
The allocation query failure doesn't mean that the negotiation
has failed as the element can allocate buffers itself.
Instead, only fail if the pads are flushing and the allocation
query failed.
https://bugzilla.gnome.org/show_bug.cgi?id=735844
The source pad might be flushing while negotiating, resulting in
set_caps or the ALLOCATION query failing. In this case set the
reconfigure flag on the source pad so that negotiation is retried on the
next buffer.
When downstream claims to accept the overlay meta but fails to
provide it in the allocation query, properly fallback to setting
a new caps without the overlay meta as that is not going to be used.
Only do this if the original caps doesn't have the overlay already,
otherwise there isn't much that can be done.
https://bugzilla.gnome.org/show_bug.cgi?id=735800
Setting segment.base in the segment sent from gst_ogg_demux_handle_page() is
enough to ensure that chained oggs are played corretly (see bgo#706569).
Tweaking the base in gst_ogg_pad_submit_packet() as well result in delays when
playing a file with start != -1.
https://bugzilla.gnome.org/show_bug.cgi?id=735808
It's not done in any other code calling negotiate and will cause deadlocks
as it is sending events and queries in the pipeline.
Specifically this pipeline was deadlocking:
gst-launch-1.0 videotestsrc ! textoverlay ! textoverlay ! fakesink
Base time should be accumulated so non flushing seeks have the expected base.
Not accumulating result in segments appearing as "too late" and so are not
played by the sink.
https://bugzilla.gnome.org/show_bug.cgi?id=735509
Make textoverlay negotiate caps more correctly.
1) Check what caps we received in the video-sink
2) If it already has the overlay meta -> use it directly
3) If it doesn't, textoverlay try adding the overlay meta and using it,
if downstream doesn't support it, just use what is received in the
video-sink
4) Check if the allocation query also supports the meta to enable
really using it
Before it wasn't really doing renegotiation of any kind, just
re-checking if it should use the overlay meta or not
Also had to update the caps in the test as memory:SystemMemory seems
to be required when you use a caps feature otherwise intersection/subset
checks will fail.
https://bugzilla.gnome.org/show_bug.cgi?id=733916
When a pipeline using alsasink and push mode upstream fails
to preroll, the following state will be the case:
- A loop upstream will be PAUSED, pushing a first buffer
- alsasink will be READY, pending PAUSED, because async
On error, the pipeline will switch to NULL. alsasink is in
READY, so goes to NULL immediately. It zeroes its cached
caps. Meanwhile, the upstream loop can cause a caps query,
conccurent with the state change. This will use those cached
caps. If the zeroing happens between the NULL test and the
dereferencing, GStreamer will critical down in the GstValue
code.
Since it appears that such a gap between states (PAUSED
and pushing upstream, and NULL downstream) is expected, we
need to protect the read/write access to the cached caps.
This fixes the critical.
See https://bugzilla.gnome.org/show_bug.cgi?id=731121
This lets oggdemux determine they are not delta units, and removes
spurious per packet warnings about being unable to determine the
packet's keyframeness.
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
This should not cause any actual bug since Theora and Daala have
a maximum shift of 31, and a packet duration of 2^31 seems very
implausible. But it fixes:
Coverity 1139804, 1139803, 1139802
Add an extra function to the oggstream map to inform it about
the incoming buffers. This way oggmux can keep a count on the
vp8 invisible frames and calculate the granulepos correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=722682
vp8 stream header shouldn't be assumed to be provided in caps always
as this would repeat the same code in all demuxers/encoders. Instead,
make oggmux generate them if they are not supplied.
https://bugzilla.gnome.org/show_bug.cgi?id=722682
When seeking back to original state after duration seeks, let
upstream know that we want the whole file, including the last
byte that wasn't requested on the duration seeks.
https://bugzilla.gnome.org/show_bug.cgi?id=724633
video time uses the 'segment' and the text time should use
the 'text_segment'.
If different segments are used for video and text it would
lead to out of sync video/subtitles.
A change in gst_ogg_demux_do_seek caused oggdemux to wait for
a page for each of the streams, including a skeleton stream if
one was present. Since Skeleton only has header pages, that
was never going to end well.
Also, the code was skipping CMML streams when looking for pages,
so would also have broken on CMML streams.
Thus, we change the code to disregard Skeleton streams, as well
as discontinuous streams (such as CMML and Kate). While it may
be desirable to consider Kate streams too (in order to avoid
losing a subtitle starting near the seek point), this may be
a performance drag when seeking where no subtitles are. Maybe
one could add a "give up" threshold for such discontinuous
streams, so we'd get any page if there is one, but do not end
up reading preposterous amounts of data otherwise.
In any case, it is important that the code that determines
the amount of streams to look pages for remains consistent with
the "early out" conditions of the code that actually parses
the incoming pages, lest we never decrease the pending counter
to zero.
This fixes seeking on a file with a skeleton track reading all
the file on each seek.
https://bugzilla.gnome.org/show_bug.cgi?id=719615
Ogg data is read chunk by chunk, and the chunk size used was
originally taken from libvorbisfile. However, this value leads
to poor performance when used on an Ogg file with large pages
(Ogg pages can be close to 64 KB).
We can't just use a larger chunk size, since this will decrease
performance on small page streams, so we use an adaptive scheme
where the chunk size is twice the largest page size we've seen
so far in the stream. For "typical" Ogg/Vorbis, this gives us
almost the same chunk size (a bit lower), and this lets us get
better performance on streams with large pages.
In case we receive a flush event before having our caps set, we will
end up trying to create a theora encoder even though we are not ready.
Avoid that situation making sure we are initialized before accepting to
be flushed.
https://bugzilla.gnome.org/show_bug.cgi?id=709858
The initial support for the new ALSA chmap API.
Just translate the current chmap to GstAudioChannelPosition during the
setup. No function to specify the channel map manually yet, so still
impossible to assign any non-standard positions or to configure in a
different order even if the hardware allows.
https://bugzilla.gnome.org/show_bug.cgi?id=709755
Store the seek stop and seqnum and properly restore them when
receiving the corresponding Segment from upstream. Also fixes
seqnum for converted seek events.
When bisecting after an earliest time has been found, we need
to only consider the stream for which the earliest time was found.
Before, the following scenario could be and was encountered:
a) Find the earliest time for stream X
b) bisect and find a page which granuletime is indeed < target, but
contains another stream.
c) decide to seek at the wrong offset, sometimes inferior to
the real one, in which case the error was undected or
d) the offset was superior, and thus the actual target keyframe was
not processed, and packets were skipped waiting
for a granulepos.
https://bugzilla.gnome.org/show_bug.cgi?id=700537
