.weba (audio) and .webv (video) were speculation on my part before
the public launch. As of yet no decision has been made on the
file extension for audio-only WebM, and I'm pretty sure there will
never be one for video-only.
Fixes bug #623837.
Fixes spurious errors that happen after an error and playing a working
stream afterwards or signals that are emitted for non-active groups.
Fixes bug #624266.
This reverts commit 9d7538247f.
If the DVD subpicture caps are not part of the raw caps, uridecodebin
doesn't qualify resindvdbin as raw source and plugs decodebins, which
causes broken DVD playback because of bugs elsewhere.
This change was originally added to only expose supported, raw subtitles,
e.g. if the subtitle sink did not support DVD subpictures but a converter
to some supported format exists. It's not very important right now because
we have nothing (that is autoplugged) to convert from plaintext/pango-markup
or DVD subpictures to something else.
Fixes bug #623583.
Otherwise the uridecodebin will be still a child of playbin2 and
its signals will still be connected. In future state changes this
will then emit unrelated signals that will confuse playbin2 or,
even worse, cause crashes and assertions.
Fixes bug #623318.
If an error happens, the PAUSED state will never be reached. If an
application re-uses decodebin2 (like totem) where one would normally
set to READY between each file, the cleanup that normally happens in
the PAUSED=>READY codepath will never be called, resulting in the
following file to re-use the previous demuxer/decoder/...
https://bugzilla.gnome.org/show_bug.cgi?id=622807
We need to clear the pointer to our ts-offset element when we destroy the video
chain elements to make sure nobody derefs it to invalid memory afterwards.
Otherwise we would end up with a bogus ->audiochain->ts_offset field
which would cause segfaults/assertions when trying to modify the
'ts-offset' property in update_av_offset().
Was easy to trigger when using a list of audio+video files mixed with
video-only files in totem.
Use the pad caps when they are available to continue the autoplugging. If the
pad caps are set, they are fixed and then we can directly continue autoplugging.
Use an accumulator for the autoplug-sort signal so that we can stop the emission
when a signal handler produced a valid result. This avoids the object handler
to overwrite the results from user signals.
Fixes#621161
Scan a bit into the data when checking for dts frames instead
of expecting the frame sync to be right at the start of the
data. This is needed for some dts-disguised-as-pcm-in-wav files.
See #413942.
Orc is not a hard requirement. Things should still compile and
work without orc, but slow fallback code may be used in this
case. Fix up configure to not error out if orc is not installed
and wrap use of orc profiling in audioresample in #ifdefs.
Fixes#620136 some more.
Make jpeg typefinder check more than just the first two bytes
plus Exif or JFIF marker. This allows us to report MAXIMUM
probability in cases where there's no Exif or JFIF marker,
making typefinding stop early. Also extract width and height,
because we can.
Fix typo that made the AC-3 typefinder not actually check for a
second frame, but rather compare the sync point found to itself,
which resulted in the AC-3 typefinder reporting an overly optimistic
MAXIMUM or VERY_LIKELY probability when it found a possible frame
sync.
Move the convert_frame function to playsink and make it part of the API. This is
in preparation to add the convert_frame signal to playsink.
See #620279
If a file contains raw streams (not requiring a decoder) that we do
not want (expose-all-streams == FALSE), we would previously consider
those of unknown-type (missing a decoder) ... whereas in fact it was just
because they don't need decoders.
This only applies if expose-all-streams is FALSE.
* don't re-create our possible caps every single time, just use the
template caps.
* don't intersect the caps against the template, basetransform has already
done that for us.
62% speedup of _transform_caps() (instruction calls, measured with callgrind)
API : expose-all-streams
If disabled:
* only the streams that CAN be decoded and match the final caps will have a
decoder plugged in and be exposed.
* the streams that COULD HAVE BEEN decoded but do not match the finals caps
will not have a decoder plugged in and will not be exposed.
If no decoder is available to decode a certain stream, then the missing element
message will still be emitted regardless of the value of the property.
https://bugzilla.gnome.org/show_bug.cgi?id=617868
Adder was using always incrementing timestamps. Seeking was done by setting the
position in the newsegment event. This was failing when doing segmented seeks
with rate<0.0, as offset (and thus timestamp) would go below 0.
Now we take both cur and end from the seek event. We construct newsegment events
depending including cur and end from the seek event. We set position to the
start of the segment. Timestamp is set to start or end of segment depending on
rate. Offset is recalculated.
Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to.
This should make sure arguments are passed to the linker in the right
order, and makes LDFLAGS usable again.
Based on initial patch by Brian Cameron <brian.cameron@oracle.com>
Fixes#615697.
This adds code to calculate the level for a given AAC stream and export
it in the stream caps. For AAC LC streams, the level is calculated
according to the definition under the AAC Profile. For other streams,
the definition under the Main Profile is used.
HE-AAC support is still to be done, and is dependent on detecting the
presence of SBR and PS in the stream.
Level is added as a field of type string because that's the way it's
done in H.264 caps as well. There are only a few possible levels, so
not using a numerical type is not too painful in this case, and
consistency is nice.
Fixes#613589.
This looks at the AAC profile for ADTS streams and adds the profile as a
string in the corresponding caps.
Profile is the actual profile, base-profile denotes the minimum codec
requirements to decode this stream. In this case they're always the
same, but they may differ e.g. in case of certain HE-AAC streams that
can be partially decoded by LC decoders (with loss of quality of course)
if no suitable HE-AAC decoder is available.
Fixes#612312.
Decrement sample counter when playing backwards. Set proper segment when playing
backwards (0..cur instead or cur..-1). Add more logging and fix a format string.
Unreffing it whenever the sinks are removed will make the volume
element unavailable after a playbin reuse because it is only
recreated if the audio sink has changed.
Fixes bug #614288.
In reverse mode we want use the next next timestamp (and not the other way
around). Fixes the tests again. Also readd a log line that was dropped with
previous commit.
We know our plugins and examples are independent of each other, so may
just as well build them in parallel. Makes the output a bit messy, but
that shouldn't be a problem and can easily be avoided with make -j1.
And fix the resulting compile failures.
I'm sorry about the patch necessary to gstclockoverlay.h but after
talking to Tim we decided we can live with it.
Change playbin2 to not error out if there are subtitles and audio
but no video. If visualizations are enabled the subtitles are rendered on top
of the visualization stream, otherwise the subtitles are not linked at all and
only the audio is played (and a warning message is posted).
If there are only subtitles but neither audio nor video an error message is
still posted.
Fixes bug #610866.
