gcc 4.5 warns when comparing some integer with an enum value, in
the case of GstFlowReturn this is valid though. We should later
add GST_FLOW_CUSTOM_OK1, GST_FLOW_CUSTOM_OK2, etc. after new core
is released.
Adds video-capture-width, video-capture-height and
video-capture-framerate properties to allow applications to
get/set those values. Getting was not possible before this patch,
and setting was done through the set-video-resolution-fps
action, which sets the properties and promptly resets the
video source to use them.
Fixes#614958
Adds image-capture-width and image-capture-height properties
to camerabin, allowing the user to get/set them. Getting was
not possible before and setting was done through the
set-image-resolution action, which shouldn't now just set
the properties.
Fixes#614958
Adds block-after-capture property to block running viewfinder after capturing.
This property is useful if application wants to display capture preview and avoid
running viewfinder on background.
Based on a patch by Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
Adds a new property called viewfinder-filter to camerabin.
This property is used to add a filter to process the video
flow right before the viewfinder sink.
Also updates test to check property exists.
Add video-source-filter property that can be used to inject application
specific gstreamer element to camerabin pipeline. The video-source-filter
element will process all frames coming from video source.
One could add image analyzers to collect information about the stream,
or add image enhancers to improve capture quality, for example.
If source-resize flag is disabled then set resolution to image capture caps
according to capture resolution video source element produces. Otherwise we
write wrong resolution to image metadata.
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
The current code just uses table id, subtable extension and version number to
check if the section has been seen before. However, this comparison is not
sufficient, causing actually new tables being dismissed.
Fixes bug #614479.
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.
Note that this one isn't a problem with normal trace macros, but causes problems with
some replacement trace macros that I use, which expect the format string to be
appendable (ie "foo "fmt in the macro)
https://bugzilla.gnome.org/show_bug.cgi?id=612454
The frei0r documentation says that these functions must not be called
on the same instance from different threads at the same time. All
other functions are guaranteed to be threadsafe.
Automatic inverse telecine element. Right now, it clumsily attempts
to rearrange video fields into frames that don't have combing effects,
and only works with 60i/24p content at 720x480. Later, it will handle
other pulldown variations, change caps and smooth timestamps
appropriately.
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
If no video sink is set and autovideosink is not available for some
reason, post a proper error message on the bus when failing to
change state, and don't try to gst_object_ref() NULL pointers. Fixes
generic/states unit test when distchecking.
The current code is comparing timestamps with different clock.
Let's use only the clock for PTS values.
Also rename frequency to interval, to avoid confusion. And remove
documentation about value 0, which won't work like documented.
https://bugzilla.gnome.org/show_bug.cgi?id=608896
This patch address the issue observed with KF timestamps
and delta flag. When a section is appended before the keyframe,
it is not marked as non-delta. It's preferable to mark the
first buffer non-delta.
This patch also simplify the initial patch written by thomas,
since it does not clutter tsmux/ with a delta flag passed
around only for GStreamer convenience.
https://bugzilla.gnome.org/show_bug.cgi?id=604908
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
Some H264 packets can be as small as 5 bytes for repeated frames.
In such a situation the output buffer size was not big enough (5*2) to fit the
SPS/PPS header and the start codes. This corrupts the ES stream.
We now generate the SPS/PPS only once which is much more optimal and we now
know the size of the header to calculate the output buffer size more safely.
Previously jpegparse was failing in decodebin as the caps we were setting where not
setting all caps fields. We need the own getcaps function to report what we actualy
accept.
Following the ed4d08189ea6e19a50e029e60da52d3583c39fbb
commit, this one fixes rtpasfpay to use packet length
as the payloaded data length, but also accepting it
as the full packet size for compatibility with
other implementations due to the lack of clarity of the
spec in this part.
Makes the asfmux content compatible with WMSP and does
some hacks to make it playable in WMP, it doesn't accept
data objects with 0 size indicating that we don't know
its size, though the spec says it should be possible.
Fixes#607555
This function is not supposed to dispose the element in the case of failure
as the caller is using the elements name in the error message. Also add
some more input parameter checks in the form of g_return_val_if_fail
When fixed_vop_rate is not set we can not set a framerate based on
vop_time_increment_resolution as it would most likely be wrong.
Don't set any framerate on the caps in that case.
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
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.
When the stream type is set to private data, gst-mpegtsdemux is trying to find
audio descriptors in PMT and look for AC3 (tag 0x6a) but doesn't look for EAC3
(tag 0x7a). Handle this case too.
Fixes bug #605904.
Does some general improvements with the internal sink handling.
1) Do not remove and re-add the ghostpad when changing
internal sink
2) Only instantiate the default sink when changing from NULL
to READY if there is no other available
3) Avoid changing the internal sink if not on NULL state
Fixes#598682
Downgrade a warning message to debug. Remove an
already fixed FIXME and add a note about (not-)using
fpsdisplaysink in autovideosink. Change the created
ghostpad to use the name "sink" as it is advertised in
the pad template.
Use GST_IS_BIN instead of G_OBJECT_TYPE to check if the
internal sink is a bin. Using the later won't work when
the sink is not a bin directly (but inherits from one, like
autovideosink).
Fixes#604280
Follow-up on 4111d6321f, the video
sink(s) used by fpsdisplaysink might not have the sync property. So we
check its existence to avoid warning from g_object_set() at runtime.
Fixes#604280
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>
Exposes the internally used sink as video-sink property and
makes the default one to be autovideosink instead of
the hardcoded xvimagesink
Fixes#604280
Because of an allocated priv (GstRTPMuxPadPrivate), the element will
leak memory if not gst_rtp_mux_release_pad() is called. This would
previously only happen if release_request_pad() was called explicitly,
somthing that should not be neccesary.
Fixes#604099
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.
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.
Fix order, fix variables that don't exist, like GST_LIBS_LIBS,
use $(LIBM) instead of -lm, and move _LIBS from LDFLAGS to LIBADD.
Spotted by Havard Graff.
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
Prevents losing sync when remuxing streams with different
start times. The smallest start time is selected as
the base time and all timestamps are subtracted
from it to get the actual time to be used when
muxing and building indexes
Fixes#586848
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
Using the end time makes it impossible to replace buffers, which is
a big problem for subtitles that could have very long durations.
Merged from gst-plugins-base, 27034be461.
Scaletempo was missing an update of 'stop' in
new segment parameters when pushing it downstream,
which caused files to end earlier when rate < 1.
Fixes#599903
Based on patch by: Bastian Hecht <hechtb@gmail.com>
It looks at raw audio data and emits messages when DTMF is detected.
The dtmf detector is the same Goertzel implementation used in FreeSwitch
and Asterisk. It is in the public domain.
There is unfortunately no G_*_FORMAT conversion specifier for differences of
pointers in glib, and we can't rely either on all platforms being 64bit.
So let's just cast the difference to a gint and be done with it.
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
Merged from gst-plugins-base, 6f4c1ac583.
Replaced with "GStreamer maintainers
<gstreamer-devel@lists.sourceforge.net>" or just removed,
depending on the number of other authors.
Merged from gst-plugins-base, 0e9bc5125a.
Set the output caps on the srcpad before pushing the buffer because else core
will do a rather expensive check to see if we can actually accept those caps on
the srcpad.
Merged from gst-plugins-base, bdfb4b46d7.
Install a custom acceptcaps function instead of using the default expensive
check. We accept whatever downstream accepts so we pass along the acceptcaps
call to the downstream peer.
Merged from gst-plugins-base, 5b72f2adf9.
Clarify the ownership of the internal plugin feature list by making
a copy of any passed list. Avoids crashes when freeing a passed list,
or leaks caused by not freeing any internally built list.
Also remove GST_PLUGINS_BASE_LIBS from LIBADD since we don't
need to link against any of the -base libs (we just use a define
from the gstaudio headers).
When sending new-segment to a stream, ensure that there is either a valid
PCR, or else wait until there's a PTS on the stream (dropping packets if
needed) in order to avoid generating an invlaid new-segments event.
https://bugzilla.gnome.org/show_bug.cgi?id=595161
g_convert seems to add a single null terminating byte to
the end of the string, even when the output is UTF16, we
force the second 0 byte when copying to the output buffer.
This issue was causing random crashes because it was
assumed that the string resulting from g_convert had
2 extra bytes, but it has only one.
Add the 'initial-identity' property, which inserts identity for
at startup for event passing, and replaces it with a new child
when the first buffer (and caps) actually arrives.
https://bugzilla.gnome.org/show_bug.cgi?id=599469
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.
Align element initialisation. This should be re-thought, g_object_new zeros things already.
Harmonize the element getters for the src/sinks to return what we actualy use.
This uses same approach like in playbin, namely checking for user defined
element, auto{audio,video}{sink,src} and finally DEFAULT_{AUDIO,VIDEO}{SRC,SINK}
defines from config.h.
