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
