Dropping a buffer because we have a seek pending is normal,
and will now happen when we trigger a seek while going through
the packets in a page. So this should not be an error.
A low bitrate stream which can pack more than 2 seconds of audio
in a page would cause the stream's position to be updated not
often enough, and would trigger a spurious "jump" via a GAP
event. Instead, we update the stream position after calculating
the new overall segment position.
https://bugzilla.gnome.org/show_bug.cgi?id=764966
... instead of relying on the segment. For the clipping at the start we assume
a proper value in the OpusHead, as generated by opusparse or opusenc.
Transmuxing in general is not guaranteed to produce the correct values, or
even have a OpusHead (e.g. when having RTP input).
https://bugzilla.gnome.org/show_bug.cgi?id=757153
The granulepos does not have the pre-skip subtracted while timestamps do,
and the last granulepos will be shorter by the number of samples that should
be dropped because of padding in the end.
As such, extrapolating the granule of the beginning of the first frame will
lead to a negative value, which is not a problem but intentional.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
This reverts commit 76647f2710.
Avoiding pull mode activation is a feature regression, and
demuxers should always use pull mode where that is possible,
e.g. if there's an upstream queue2 with a ring buffer or
a download buffer.
This patch made reverse playback no longer possible over http.
If the goal is to minimise seeks, then that can still be done
by making the demuxer behave differently in pull mode if
the SEQUENTIAL flag is set. If there are bugs, like the demuxer
needlessly scanning the entire file on start-up in pull mode,
then those should be fixed instead.
https://bugzilla.gnome.org/show_bug.cgi?id=746010
gst_event_replace() takes its own reference on the event so we should drop
ours after creating and storing an event using it.
This fix leaks which can be reproduced using the
validate.http.media_check.vorbis_theora_1_ogg scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=748247
When a stream has a skeleton index, the stream time is taken from that
index. However, when part of the stream is captured, the index is
invalid as its offsets are now wrong. To avoid this, we ignore the index
when the last offset points beyond the end of the stream (when its
byte length is known).
https://bugzilla.gnome.org/show_bug.cgi?id=744070
oggmux keeps a cached buffer per pad, and pulls buffers from
collectpads to this cached buffer for all pads before processing
the best pad. In some cases, the move from collectpads buffer
to cached buffer is delayed till next call. However, when there
is only one pad, this can't be delayed till next call as there
will be a deadlock since collectpads has no other pad to push to.
https://bugzilla.gnome.org/show_bug.cgi?id=740565
gstoggdemux.c:1233:11: error: format specifies type 'long' but the argument has type 'ogg_int64_t' (aka 'long long') [-Werror,-Wformat]
granule);
^~~~~~~
https://bugzilla.gnome.org/show_bug.cgi?id=746512
The code that was calculating the start granule from packet durations
was interpreting a negative value as an error, but this is actually a
valid case, to indicate clipping of data at start.
https://bugzilla.gnome.org/show_bug.cgi?id=743900
If we get EOS when we're trying to build a chain, we disable seeking
and continue instead of posting an error. This can happen for corner
cases such as a stream with a video that stops before the end, for
instance.
https://bugzilla.gnome.org/show_bug.cgi?id=745980
When looking for pages when seeking, we stop looking for non sparse
streams if we don't find one within a given threshold. This fixes
seeking filling up queues and blocking in corner cases such as an
audio file with a pathological 1 frame video stream (yes, I saw one).
https://bugzilla.gnome.org/show_bug.cgi?id=745980
This will usually deadlock, despite this patch being in master for
quite some time and working fine. Nevertheless, we deem it to be
not working, disregarding facts.
As such, we fix it by keeping track of seek events, and sending
them upstream from a separate thread. Buffers are then discarded
till we get a new segment with the expected seqnum.
READY->PAUSED can be too early as souphttpsrc can get the HTTP
headers after this. Try again in the chain function.
Also use seeking query to disable seeking if upstream reports
being unseekable.
Some resetting code has to be done in the NEW_SEGMENT
event handler, instead of the missing FLUSH_STOP one.
Segment base was also wrongly accounted for. This was hidden
by the fact that flushing resets the base.
A discontinuity is now also signalled on seeking. We have to
also ensure that the discontinuity "sticks" till a buffer
with a valid timestamp goes out, or the audio decoder base
class will ignore the discontinuity for purposes of keeping
track of the current time.
This allows using non flushing segment seeks for looping
HTML audio in particular, and more generally non flushing seeks.
https://bugzilla.gnome.org/show_bug.cgi?id=729198
The code was using the first nonnegative granulepos to seed the
granule tracking, which appeared to work since headers have zero
granulepos. However, this does not work for files with a hole at
start, which are common in live streaming.
The correct behavior is to look for the first granule, and subtract
the duration of all the packets finishing on this page.
The function which does this relies on the fact that the ogg_stream
structure can be duplicated by shallow copy, in order to pull the
packets from the first page(s) on the copy without affecting the
original stream state.
If the streaming task attempts to read a chain while the pipeline
is stopping (which can happen if the pipeline stops shortly after
start or a new URI being setup in gapless playback case), it will
see a flushing return from upstream, and should then also return
flushing to the caller, rather than emit a flow error.
https://bugzilla.gnome.org/show_bug.cgi?id=722442
It might happen that the timestamp is before the segment and the
check would succeed. In this case reducing the duration makes no
sense and would lead to broken results.
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
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.
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
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
