Move the following functions to matroska-read-common.[ch] from
matroska-demux.c and matroska-parse.c:
- gst_matroska_decode_content_encodings
- gst_matroska_decompress_data
https://bugzilla.gnome.org/show_bug.cgi?id=650877
Replace the following functions with their gst_matroska_read_common_*
counterparts:
- gst_matroska_{demux,parse}_parse_index
- gst_matroska_{demux,parse}_parse_skip
- gst_matroska_{demux,parse}_stream_from_num
Introduce GstMatroskaReadCommon to contain those members of
GstMatroskaDemux and GstMatroskaParse that were used by the above
functions.
https://bugzilla.gnome.org/show_bug.cgi?id=650877
Tell GstBaseParse the duration in samples instead of time, so that
a duration query in DEFAULT format will return the correct number
of samples without rounding errors. Baseparse will convert this
into time itself when needed.
https://bugzilla.gnome.org/show_bug.cgi?id=650785
When not using the fieldanalysis element immediately upstream of deinterlace,
behaviour should remain unchanged. fieldanalysis will set the caps and flags on
the buffers such that they can be interpreted and acted upon to produce
progressive output.
There are two main modes of operation:
- Passive pattern locking
Passive pattern locking is a non-blocking, low-latency mode of operation that
is suitable for close-to-live usage. Initially a telecine stream will be
output as variable framerate with naïve timestamp adjustment. With each
incoming buffer, an attempt is made to lock onto a pattern. When a lock is
obtained, the src pad and output buffer caps will reflect the pattern and
timestamps will be accurately interpolated between pattern repeats. This
means that initially and at pattern transitions there will be short periods
of inaccurate timestamping.
- Active pattern locking
Active pattern locking is a blocking, high-latency mode of operation that is
targeted at use-cases where timestamp accuracy is paramount. Buffers will be
queued until enough are present to make a lock. When locked, timestamps will
be accurately interpolated between pattern repeats. Orphan fields can be
dropped or deinterlaced. If no lock can be obtained, a single field might be
pushed through to be deinterlaced.
Locking can also be disabled or 'auto' chooses between passive and active
locking modes depending on whether upstream is live.
... since the TEARDOWN sequence might not have had a chance to even start,
but at least connections should be closed (synchronously) and state cleaned up.
See #632504.
Simplify the command handling; passing a command to thread means we really
want it to get the message, which means to always flush provided the command
can handle being interrupted. Command thread indicates whether command
allows interruption and ensure non-flushing connection as it subsequently
needs it.
In particular, this also makes the TEARDOWN sequence interruptable
and also prevents races where _loop_ could miss a command and would
continue receiving (or at least trying to).
See #632504.
With the async state changes, it is possible that we need to open the stream
before play and pause.
Also make sure we remember a previous open failure so that we don't keep trying
again.
Simplify the command handling, only continue looping when we have not received
another command or when the previous loop was successfull.
Avoid looping on a disconnected socket.
If the lock is not released before emitting a signal, it may cause a deadlock
if any other function in the element is called.
Also removed an unused timestamp parameter
https://bugzilla.gnome.org/show_bug.cgi?id=649617
Since the segment duration is given in terms of the
GST_MATROSKA_ID_TIMECODESCALE we should only convert it into
nanoseconds when we are sure that any scale specified in the file has
been read.
https://bugzilla.gnome.org/show_bug.cgi?id=650258
If the bitrates for all but one audio/video streams are known, and the
total stream size and duration can be determined, this calculates the
unkown bitrate as (stream size / duration) - (sum of known bitrates).
While this is not guaranteed to be very accurate, it should be good
enough for most purposes.
For example, this is useful for H.263 + AAC streams where no 'btrt' atom
is available for the video portion.
https://bugzilla.gnome.org/show_bug.cgi?id=619548
This parses the 'damr' atom if present, and exports the maximum bitrate
of the stream using the mode set field to determine the highest bitrate
frame type that might be present.
https://bugzilla.gnome.org/show_bug.cgi?id=620186
Since the segment duration is given in terms of the
GST_MATROSKA_ID_TIMECODESCALE we should only convert it into
nanoseconds when we are sure that any scale specified in the file has
been read.
https://bugzilla.gnome.org/show_bug.cgi?id=650258
Otherwise wavenc will fail if upstream decides to set equivalent caps or caps
with additional information later.
Thanks to Alexander Schremmer for finding this bug.
A duration tag gets inserted only for streamable=false, so only
update/write the duration later if we actually inserted that tag,
otherwise we write garbage into other tags.
https://bugzilla.gnome.org/show_bug.cgi?id=649060
Refuse h264 caps without stream-format and codec_data fields for
now, to avoid creating broken files. This might cause some pipelines
that worked previously to fail. However, the move from -bad to -good
is our only chance to fix this up, so make it strict for now. We can
always change it back to be less strict in future.
https://bugzilla.gnome.org/show_bug.cgi?id=647919
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.
g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.
Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
We use -DG_DISABLE_ASSERT for the pre-releases, which makes these
warnings pop up in cases that were previously covered by g_assert_not_reached()
and the like:
tvtime/greedyh.c:801:14: warning: 'scanline' may be used uninitialized in this function
matroska-mux.c:501:19: warning: 'context' may be used uninitialized in this function
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.
gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
This is needed for automatic transcoding using encodebin. Our typefinder
does not always add a variant to the found caps, and encodebin needs
an *exact* match to the caps on the source pad template, so we need
to add the variant-less video/quicktime caps to the template as well
for encodebin to be able to find it. Add unit test for this as well.
https://bugzilla.gnome.org/show_bug.cgi?id=642879
... and not only when sort-of feeling like it.
In any case, if it turns out all really is in order,
and presumably DTS == PTS, then no ctts will be produced anyway.
That is, all sorts of problems arise with re-ordered input timestamps that
tend to defy automagic handling for every case, so allow for a few variations
that can be tried depending on circumstances.
Also try to document accordingly.
Also fixes#638288.
In this mode, an initial empty moov (containing only stream metadata) is written,
followed by fragments containing actual data (along with required metadata).
New fragments are started either at keyframe (if such are sparse) or when
property configured duration exceeded.
Based on patch by Marc-André Lureau <mlureau@flumotion.com>
Fixes#632911.
This writes out the optional 'btrt' atom (MPEG4BitrateBox) for H.264
media if either or both of average and maximum bitrate are available for
the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=623678
This collects the 'bitrate' and 'maximum-bitrate' tags on the
corresponding pad and uses these to populate these fields in the ESDS
where applicable.
https://bugzilla.gnome.org/show_bug.cgi?id=623678
Write uint tags that have complements (e.g. track-number/
track-count) even when we only have one of them available
and set the other one to 0.
Fixes#622484
Previously we would end up with the collectpaddata structure already freed.
This would result in a bogus iteration of mux->sinkpads (all the
GstQTPad being freed) and it wouldn't be removed from that list.
Finally, due to it not being removed from that list, we would end up
calling a bogus gst_qt_mux_pad_reset on those structures => SEGFAULT
Due to GstCollectPads sink pads list being not reliably
iteratable (when not inside the collected function) this
patch adds a sink pads list to qtmux to be used when iterating
sink pads on reset function.
Fixes#609055
Adds a new property to qtmux that sets a path to a file to write
and update data about the moov atom (that is not writen till the
end of the file). If the pipeline/app crashes during execution it
might be possible to recover the movie using the qtmoovrecover element.
qtmoovrecover is an element that is also a pipeline. It is not
meant to be used with other elements (it has no pads). It is merely
a tool/utilitary to recover unfinished qtmux files.
Fixes#601576
Following the previous qtmux commit, this patch tries
to use the new info added to the caps to fill the 'trak'
atom's fields and children atoms. This way qtmux will
use the late added 'codec_data' when h264parse adds
it in the following pipeline:
videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \
h264parse output-format=0 ! qtmux ! \
filesink location=test.mov
Qtmux can accept caps renegotiation if the new caps is a
superset of the old one, meaning upstream added new info to
the caps. This patch still doesn't make qtmux update any
atoms info from the new info, but at least it doesn't
reject the new caps anymore.
A pipeline that reproduces this use case is:
videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \
h264parse output-format=0 ! qtmux ! \
filesink location=test.mov
Reads the new caps added to qtdemux by commit
c917d65e6d
and adds its corresponding atoms.
Also adds support for image/x-jpc as it is the same
as image/x-jp2, except that the buffers need to be
boxed inside a jp2c isom box before muxing. To solve
this the QTPads now have a function that (if
not NULL) is called when a buffer is collected. This
function returns a replacement to the current collected
buffer.
Fixes#598916
Adds the mapping of 'classification' tags to writing of
'clsf' atoms for gppmux.
Based on a patch by: Lasse Laukkanen <ext-lasse.2.laukkanen@nokia.com>
Use the rounding version for improved sync between streams.
Small variations in the duration when muxing might lead to
cumullative wrong timestamping when demuxing.
Fixes#602936
Try to use timestamps even when the stream has out of order
timestamps, only fall back to durations when we detect an
out of order buffer. Improves sync between streams.
Adds support for muxing SVQ3 content. Usually this format
has decoder info that must be passed in the 'seqh' field
in the caps. It is also good to add the gama atom to make
quicktime not crash.
Fixes#587922
Do not wrongly add the result of the function to the
pointer to the buffer size. Instead, check the result
to see if the serialization was ok.
Based on a patch by: "Carsten Kroll <car@ximidi.com>"
Fixes#602106
When muxing streams, some can start later than others. qtmux
now handle this by adding an empty edts entry with the
duration of the 'lateness' to the stream's trak.
It tolerates a stream to be up to 0.1s late.
Fixes#586848
Keep track of the chunk durations to be able to add 3gr6
brand if it is a faststart file and the longest chunk is
smaller than a sec. Implemented according to 3gpp
TS 26.244 v6.4.0 (2005-09)
Fixes#584361
In faststart mode, there is no need to send the ftyp
right at the beginning of the stream. Waiting and sending it
only later (when the moov atom is ready to be sent) provides
us with more information about the stream and we can better
select the compatible brands.
Moves tags data initialization to the function that actually appends
the tags to the list. Fixes#582702
Also fixes some style caught by the pre-commit hook.
State change to READY and then back to PAUSED should still provide
the proper structures as are otherwise freshly available following
a request_new_pad.
Pointed out by Thiago Santos.
Original commit message from CVS:
* gst/quicktime/gstqtmuxmap.c: Add video/x-qt-part and video/x-m4-part
to caps so schroenc/schroparse can use it. Fixes#566958
Original commit message from CVS:
* gst/quicktime/gstqtmux.c: (gst_qt_mux_change_state):
Do not tempt or suggest to violate gst_collect_pads API specification.
Original commit message from CVS:
* ext/celt/gstceltenc.c:
* ext/celt/gstceltenc.h:
* ext/metadata/gstmetadatamux.c:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
Totally remove the internal taglists and fully use tagsetter. Fixes
various tag muxing issues.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
Even if we currently do not have a duration yet, assume seekable if
it looks like we'll likely be able to determine it later on
(which coincides with needed information to perform seeking).
Fixes#641047.
Even if VBR headers are missing, we can't guarantee that a stream is in
fact a CBR stream, so it's safer to let baseparse calculate the average
bitrate rather than assume a CBR stream. However, in order to make
/some/ metadata available before the requisite number of frames have
been parsed, this posts the bitrate from the non-VBR headers as the
nominal bitrate.
https://bugzilla.gnome.org/show_bug.cgi?id=641858
Rather than a fixed default frame count, estimate frame count to aim for
an interval duration depending on fps if available, otherwise use old
fixed default.
Also add a format flag to signal baseparse that subclass/format can provide
(parsed) timestamp rather than an estimated one. In particular, such "strong"
timestamp then allows to e.g. determine duration.
Don't unref the event if it hasn't been handled, because the caller
assumes it is still valid and might reuse it.
I ran into this problem when transcoding an AVI (with mp3 inside)
to gpp.
https://bugzilla.gnome.org/show_bug.cgi?id=639555
Fix copy'n'paste bug that made us look for the raw little endian
sync word twice instead of looking for the 14-bit LE sync word
as well. Fixes parsing of such streams (see #636234 for sample file).
That is, as such formats allow subclass to extract position from frame,
it is possible to extract duration (if not otherwise provided)
from (near) last frame, and a seek can fairly accurately target the required
position.
Fixes#631389.
Arrange for upstream as well as downstream flushing when seeking.
Also determine upstream size as well as seekability. Adjust some comments
to reality and employ debug statement in proper order.
Thanks to Felipe Contreras for the suggestion. This is partially
based on his patches and makes flacparse more than 3.5 times faster.
Looking for valid frame headers is unlikely to give false positives
because every frame header is at least 9 bytes long, contains a
14 bit sync code and a 8 bit checksum over the first 8 bytes.
Fixes bug #631200.
The first newsegment event will be send by the first call to
gst_base_parse_push_buffer() if necessary, posting the tags
before that is not a good idea. Instead do it from the
GstBaseParse::pre_push_buffer vfunc.
This reverts commit b5a3d60363.
Reverting this for now, since no one really seems to remember why this
property exists or what it could possibly be good for. It seems to have
been in the original mp3parse since the beginning of time and was back-
ported from there.
Seekability, like duration, etc is unlikely to change (frequently), and
the default assumption covers most cases, so let subclass set when needed.
At the same time, allow subclass to indicate if it has seek-metadata (table)
available, and possibly have it provide an average bitrate.
This allows the child class to chain its event handler with
GstBaseParse, so that subclasses don't have to duplicate all the default
event handling logic.
https://bugzilla.gnome.org/show_bug.cgi?id=622276
We wait to parse a minimum number of frames (10, arbitrarily) before
emiting bitrate tags so that our early estimates are not wildly
inaccurate for streams that start with a silence. If the stream ends
before that, we just emit the tags anyway.
While it _would_ be nicer to be specify the threshold to start pushing
the tags in terms of duration, this would introduce more complexity than
this merits.
https://bugzilla.gnome.org/show_bug.cgi?id=614991
This is optional because it's a quite expensive operation and it's very
unlikely that a non-frame is detected as frame after the header CRC check
and checking all bits for valid values. The overall frame checksums are
mainly useful to detect inconsistencies in the encoded payload.
When called from the GST_FLAC_PARSE_STATE_HEADERS case,
gst_flac_parse_hand_headers() does a gst_buffer_set_caps() on a buffer
with refcount > 1. This change handles this case by making the buffer
metadata_Writable.
https://bugzilla.gnome.org/show_bug.cgi?id=614037
This patch adds the get_frame_overhead() vfunc so that baseparse can
accurately calculate the min/avg/max bitrates for aacparse.
Note: The bitrate was being incorrectly calculated for ADTS streams
(it's not in the header as the code suggests).
This makes baseparse keep a running average of the stream bitrate, as
well as the minimum and maximum bitrates. Subclasses can override a
vfunc to make sure that per-frame overhead from the container is not
accounted for in the bitrate calculation.
We take care not to override the bitrate, minimum-bitrate, and
maximum-bitrate tags if they have been posted upstream. We also
rate-limit the emission of bitrate so that it is only triggered by a
change of >10 kbps.
Because config.h defines __MSVCRT_VERSION__, which should be defined
before inclusion of any system header.
Also fixes mpegdemux Makefile.am LIBADD typo.
Fixes#606665
Perform sanity check on type of seek, and only perform one that is
appropriately supported. Adjust downstream newsegment event
to first buffer timestamp that is sent downstream.
In particular, consider DISCONT == !sync, and allow subclass to query
sync state, as it may want to perform additional checks depending
on whether sync was achieved earlier on.
Also arrange for subclass to query whether leftover data is being drained.
In particular, (optionally) provide baseparse with a notion of frames per second
(and therefore also frame duration) and have it track frame and byte counts.
This way, subclass can provide baseparse with fps and have it provide default
buffer time metadata and conversions, though subclass can still install
callbacks to handle such itself.
After all, stream is as-is, and there is little molding to downstream's
taste that can be done. If subclass can and wants to do so, it can
still override as such.
Also handle the case gracefully where the subclass decides to drop
the first buffers and has no caps set yet. It's still required to
have valid caps set when the first buffer should be passed downstream.
In one case we extracted the sample rate index from the codec data
and saved it as sample rate rather than getting the real sample
rate from the table. Fix that, and also make sure we don't access
non-existant table entries by adding a small helper function that
guards against out-of-bounds access in case of invalid input data.
Create output caps from input caps, so we maintain any fields we
might get on the input caps, such as codec_data or rate and channels.
Set channels and rate on the output caps if we don't have input caps
or they don't contain such fields. We do this partly because we can,
but also because some muxers need this information. Tagreadbin will
also be happy about this.
Sending the flush-start event forward before taking the stream lock actually
works, in contrast to deadlocking in downstream preroll_wait (hunk 1).
After that we get the chain function being stuck in a busy loop. This is fixed
by updating the minimum frame size inside the synchronization loop because the
subclass asks for more data in this way (hunk 2).
Finally, this leads to a very probable crash because the subclass can find a
valid frame with a size greater than the currently available data in the
adapter. This makes the subsequent gst_adapter_take_buffer call return NULL,
which is not expected (hunk 3).
The problem is that after a discont, set_min_frame_size(1024) is called when
detect_stream returns FALSE. However, detect_stream calls check_adts_frame
which sets the frame size on its own to something larger than 1024. This is the
same situation as in the beginning, so the base class ends up calling
check_valid_frame in an endless loop.
Baseparse internaly breaks the semantics of a _chain function by calling it with
buffer==NULL. The reson I belived it was okay to remove it was that there is
also an unchecked access to buffer later in _chain. Actually that code is wrong,
as it most probably wants to set discont on the outgoing buffer.
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.
This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
If we get GAP samples, there is no need to transmitt it.
In some situations, microphone is muted, we can drop net traffick
usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s
Parse session control attributes when no media control attribute is
present. Threat * control attributes as an empty string, just like the
spec says.
Fixes#646800
... by forcing a state changed to PLAYING, which should otherwise be a
no-op as elements should already be in that state.
In particular, jitterbuffer needs new base_time as soon as possible to perform
proper timing (e.g. eos timeout handling) and can't wait for the new base_time
that will be distributed when the whole pipeline returns to PLAYING.
See bug #646397.
This option allows the videomixer2 element to output a valid alpha
channel when the inputs contain a valid alpha channel. This allows
mixing to occur in multiple stages serially.
The following pipeline shows an example of such a pipeline:
gst-launch videotestsrc background-color=0x000000 pattern=ball ! video/x-raw-yuv,format=\(fourcc\)AYUV ! videomixer2 background=transparent name=mix1 ! videomixer2 name=mix2 ! ffmpegcolorspace ! autovideosink videotestsrc ! video/x-raw-yuv,format=\(fourcc\)AYUV ! mix2.
The first videotestsrc in this pipeline creates a moving ball on a
transparent background. It is then passed to the first videomixer2.
Previously, this videomixer2 would have forced the alpha channel to
1.0 and given a background of checker, black, or white to the
stream. With this patch, however, you can now specify the background
as transparent, and the alpha channel of the input will be
preserved. This allows for further mixing downstream, as is shown in
the above pipeline where the a second videomixer2 is used to mix in a
background of an smpte videotestsrc. So the result is a ball hovering
over the smpte test source. This could, of course, have been
accomplished with a single mixer element, but staged mixing is useful
when it is not convenient to mix all video at once (e.g. a pipeline
where a foreground and background bin exist and are mixed at the final
output, but the foreground bin needs an internal mixer to create
transitions between clips).
Fixes bug #639994.
Previously the chain function was working sample frame based. In each cycle it
was checking if it is time to run a fft or if it is time to send a message.
Now we changed the data transform functions to work on a block of data and
calculate the max length until either {end-of-data, do-fft, do-msg}. This allows
us also to avoid the duplicated code for the single and multi-channel case (as
the transformers have the same signature now).
Even though we wrap around the accumulated second, we still need to add the
error in the same cycle. Increase the todo in the same conditional as afterwards
the accumulated error will be below one second.
AUTHOR only existed in an old version of the spec and ARTIST is
the new replacement for this. We are still reading both to still
be compatible with old files.
Fixes bug #644875.
Before it was possible that we run an extra fft when the time for sending a new
message is due. Only do this if we have not run the fft for the interval at all.
Don't check the format for each sample frame to read. We can make that decission
in _setup already. This is still not ideal as we call the function per frame.
Ideally we determine how many samples we can copy and have a loop in the input
reader. As an alternative we might also consider to use the fft variants for the
various formats and not convert to float for all cases - we would still need to
mix or deinterleave though.
In case server-side fails to perform seek, i.e. PLAY at non-zero requested
position, recovery so far would arrange for streaming to continue, albeit
having lost position tracking in the process. So, query position prior
to seek and use upon failed seek.
Add a boolean multi-channel property with a default of FALSE. When set to TRUE
the element won't mix all input channels to mono, but instead run a FFT on each
channel. In that case the result message would contain a 2 dimensional array
of channel x data for magnitude and phase.
API: GstSpectrum:multi-channel
https://bugzilla.gnome.org/show_bug.cgi?id=593482
Use a separate function to read a sample frame into a ringbuffer slot. In the
future we can use format-specific function pointer to avoid the reoccuring
format checks.
We now keep the fft data that is related to one channel in a separate structure
to prepare for multichannel support. We also refactor the code to operate more
often on the channel context.
When using gstrtpbin with ignore-pt=true, the free_stream function tries to
call gst_element_set_locked_state and gst_element_set_state on a stream->demux
which is NULL.
fixes#642412
Fix slightly confused tag handling in some places: make it clear when
we're taking ownership of a tag list and when not. For example,
gst_icydemux_tag_found() was taking ownership when the source pad
existed, but otherwise not (leak). Also, gst_event_parse_tag() does
not return a newly-allocated taglist, but a tag list that belongs to
the tag event, so don't give ownership of it away.
While we're at it, some minor clean-ups: don't re-invent g_strndup()
and simplify gst_icydemux_parse_and_send_tags() a bit, and don't
leak the tag list in case no valid tags where found.
https://bugzilla.gnome.org/show_bug.cgi?id=641330
* gst/qtdemux/qtdemux.c (gst_qtdemux_src_convert): Unref the qtdemux; we
weren't doing so before.
(gst_qtdemux_handle_src_event, gst_qtdemux_chain): Fix some error
cases which would leak a ref to the qtdemux.
Extract MusicBrainz tags added by MusicBrainz's Picard
tagger application. These tags (esp. the album id) are
helpful for rhythmbox et.al. to automatically downloads
cover art.
https://bugzilla.gnome.org/show_bug.cgi?id=642205
Images might have framerate=0/1 in the caps, which caused an
assertion on deinterlace. I don't know of interlaced image formats
but deinterlace might be hardcoded on some generic pipelines and
it shouldn't assert.
The fix was to set field_duration to 0 if the input has a framerate
with a 0 numerator.
This patch also adds checks for this situation on the unit tests.
https://bugzilla.gnome.org/show_bug.cgi?id=641400
Theora can only use the last frame (or the keyframe) as a reference, so in
practice. If we receive a buffer that references an unknown codebook, request
new headers. It probably means that headers were lost.
Functions that process the rtcp buffer could decide to keep a ref
on the buffer for further processing. So make the metadata writable
only after they are done.
In particular, this avoids missing the intended keyframe when first converting
from the frame's mov time to global segment time, and then back from global
time to mov time when activating the segment.
Make win32 build bot happy again, and nicefy output while we're at it.
qtdemux.c: In function 'qtdemux_parse_trun':
qtdemux.c:2162:3: error: format '%lu' expects type 'long unsigned int', but argument 9 has type 'guint32'
Check that the WAVEHEADER node is present instead of blindly using it.
If not present we won't be able to provide a more refined caps, but at
least we won't crash.
https://bugzilla.gnome.org/show_bug.cgi?id=640028
Old code was difficult to understand exactly how the neighboring
scan lines are calculated, and it appeared that some were off by
+2 or -2, depending on the field flag. Fixes#639321.
Set caps from the start so discoverer doesn't blow up on
seeing no negotiated caps between elements on preroll,
which might happen if no subtitle buffers have been
pushed yet at the time. See file from bug #603308.
The previous default, greedyh, takes 4 times as long as MPEG-2
video decoding, and is unlikely fast enough on any current CPU
to play 1080i video in real-time. greedyl isn't much faster.
linear was chosen over vfir, since the quality advantage of vfir
is minimal compared to the occasional visual artifacts and slower
processing.
