The problem experienced is that the EOS was never emitted by oggmux during a
rendering with GES. The proposed patch checks if the pad is EOS before deciding
it's the "best pad".
https://bugzilla.gnome.org/show_bug.cgi?id=699792
If our previous flow return was NOT_LINKED, don't try to push on the pads some
more. If we get a RECONFIGURE event on the pad, try to push on it again.
Changed the check to a current_time equal to the stop will produce
EOS instead of the next one. Also, segment.start can't be NONE, so removing
this check.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=696899
No need to copy buffers we put into the streamheader any more
now that we don't put caps on buffers any more, so there's no
danger of a refcount cycle.
Pick delta pad earlier during header parsing, and pick it based
on whether it's a video stream or not rather than some rather
byzantine signalling from theoraenc etc. which would set the delta
flag on header packets which oggmux would then pick up and determine
that this is a "delta-able" stream.
Since the new videodecoder-based theoraenc didn't do that any more,
we would only see the first delta flag on the second video packet,
which is after we've already muxed a few audio packets flagged as
key units, which trips up the unit test.
Fixes pipelines/oggmux unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=679958
A crafted file with invalid pages will cause repeated searches from
earlier offsets in steps of 8500 bytes, but reading till the end of
the stream. Since we know the maximum size of an Ogg page, we can
bound the search for next page, to get a linear behavior (though
still not good enough as it will read the entire file backwards if
there's no valid page till then).
Take chain lock when accessing chains. Fall back gracefully
when there's no current chain Hopefully fixes crash when
seeking in Jamendo or Magnatune streams in Amarok.
https://bugzilla.gnome.org/show_bug.cgi?id=675609
They're hardly used, and probably more confusing than anything
else, and it's not clear that anyone would really need to be
able to tell them apart at the media type level.
This never really took off - it's hardly used anywhere
and deprecated in favour of Kate. Exposing pads just
leads to confusing 'you are missing a plug-in' messages
when people come across such streams. We could still post
the data on the bus for applications to parse.
In case many packets fit on a page, we may not see a granpos for
a while, and granpos interpolation can wrap the 'frames since last
keyframe' part of the granpos, generating a granpos which is smaller
than what it should be.
This is fixed by detecting keyframe packets (at least for Theora),
and updating the last keyframe granpos from this.
This may still be generating potentially wrong granpos for streams
which have a Theora like granpos (keyframes, a max keyframe distance
and a count of frames since last keyframe), and which allow implicit
granules on packets. For these streams, a custom keyframe detection
routine should be plugged into their GstOggStream mapper.
https://bugzilla.gnome.org/show_bug.cgi?id=669164
When I first implemented push mode seeking, I removed the chain
freeing there as it could be used later. The current code does not
seem to do that though, so I'm restoring the previous freeing,
which plugs the leak while apparently not reintroducing use of
freed data with chained and normal files, both with gst-launch
playbin2 and Totem.
This reverts commit 5df30c1b90.
I must have dreamt the Valgrind logs, reverting this reintroduces
no leak, and gets rid of the test failures it introduced :S
A first hang was happening when trying to locate a page backwards,
where we'd sync forever on the same page.
With that fixed, a second hang would happen after preparing an EOS
event, but with no chain created yet to send it to, the pipeline
would stay idle forever.
An element error is now emitted for this case.
Pads are initialized twice: when requesting pads and when
initializing collectpads. Avoid double initialization by
checking if collectpads are still going to be initialized when
creating request pads.
This prevents trying to seek and failing, then ending up unable
to stream because we can't get back at the headers.
A more robust way would be to find a good place to reinject the
headers when a seek fails, but I can't seem to get this to work.
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
Replace g_thread_create() with g_thread_try_new().
If we already saw the keyframes that we need to find,
we do not need to bisect to find them.
This will always be the case for streams with audio only,
where each frame acts as a keyframe, but will occasionally
also happen for streams with video.
https://bugzilla.gnome.org/show_bug.cgi?id=662475
Opus streams outside of Ogg may not have headers, and oggstream
may be used by oggmux to mux an Opus stream which does not come
from Ogg - thus without headers.
Determining headerness by packet count would strip the first two
packets from such an Opus stream, leading to a very small amount
of audio being clipped at the beginning of the stream.
In push mode, we determine duration by doing a seek to the end of the
stream. However, a skeleton stream with an index will cause the duration
to be known already, and we end up never setting the push_time_duration
variable which we use to know duration has been determined.
https://bugzilla.gnome.org/show_bug.cgi?id=662049
The codec setup headers are a lot more likely to have correct information,
especially as it's easy to remux a skeleton in a file where streams don't
have the same parameters (I've even seen a file with two skeletons).
Still, this is useful in the case we have a codec we can't decode, so we
can at least (theoretically) convert granpos to time, so we discard this
information if the codec setup has already provided it.
This fixes playback on (at lesat) the original archive.org encoding of
"The Night of the Living Dead" (now replaced by another encoding).
https://bugzilla.gnome.org/show_bug.cgi?id=612443
This could happen when testing with navseek, and pressing
right and left at roughly the same time. The current chain
is temporarily moved away, and this caused the flush events
not to be sent to the source pads, which would cause the
data queues downstream to reject incoming data after the
seek, and shut down, wedging the pipeline.
Now, I can't really decide whether this is a nasty steaming
hack or a good fix, but it certainly does fix the issue, and
does not seem to break anything else so far.
https://bugzilla.gnome.org/show_bug.cgi?id=621897
This patch implements seeking in push mode (eg, over the net)
in Ogg, using the double bisection method.
As a side effect, it also fixes duration determination of network
streams, by seeking to the end to check the actual duration.
Known issues:
- Getting an EOS while seeking stops the streaming task, I can't
find a way to prevent this (eg, by issuing a seek in the event
handler).
- Seeking twice in a VERY short succession with playbin2 fails
for streams with subtitles, we end up pushing in a dataqueue
which is flushing. Rare in normal use AFAICT.
- Seeking is slow on slow links - byte ranges guesses could be
made better, decreasing the number of required requests
- If no granule position is found in the last 64 KB of a stream,
duration will be left unknown (should be pretty rare)
https://bugzilla.gnome.org/show_bug.cgi?id=621897
The first packet of a sparse stream may arrive after an initial
delay in the stream. If ogg_stream_packetout reports a discontinuity
in a sparse stream, do not propagate it to other streams in the
chain unnecessarily.
https://bugzilla.gnome.org/show_bug.cgi?id=621897
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
