Commit graph

8796 commits

Author SHA1 Message Date
Jimmy Ohn
0ef9e6d139 qtdemux: Modify data type of duration in handle_src_query function
Data type of duration need to modify from guint64 to GstClockTime
for consistency in handle_src_query function.

https://bugzilla.gnome.org/show_bug.cgi?id=763965
2016-03-24 14:34:55 +02:00
Vivia Nikolaidou
dc2aafb3d4 deinterlace: Added "auto" fields mode
The "auto" fields mode will detect the upstream and downstream framerates and
will decide to deinterlace all or only top fields.

https://bugzilla.gnome.org/show_bug.cgi?id=763869
2016-03-24 14:34:11 +02:00
Havard Graff
bcbb8fc1da flvdemux: don't emit pad-added until caps are ready
In other words, gst_pad_get_current_caps should never return NULL
in a pad-added callback from the demuxer.

Added tests for the two special cases with AAC and H.264 where this
would happen every time.

https://bugzilla.gnome.org/show_bug.cgi?id=763780
2016-03-24 14:33:33 +02:00
Vineeth TM
1071309870 good: use new gst_element_class_add_static_pad_template()
https://bugzilla.gnome.org/show_bug.cgi?id=763076
2016-03-24 14:32:20 +02:00
Jihae Yi
da5c8a954c rtspsrc: avoid potentially overflowing expression
https://bugzilla.gnome.org/show_bug.cgi?id=757569
2016-03-24 14:28:50 +02:00
Jimmy Ohn
84f436f122 qtdemux: Add the function to get channels and sample rate for AAC
Add aac_get_channels and sample_rate function to get these value for
AAC.

https://bugzilla.gnome.org/show_bug.cgi?id=749110
2016-03-24 14:28:09 +02:00
Sebastian Dröge
605175b8c4 deinterleave: Use GstIterator for iterating all pads instead of manually iterating them while holding the object lock all the time
Doing queries while holding the object lock is a bit dangerous, and in this
case causes deadlocks.

https://bugzilla.gnome.org/show_bug.cgi?id=763326
2016-03-17 21:12:29 +02:00
Vivia Nikolaidou
5d8e7598ac deinterlace: Fix typo to not change the input caps but our filtered caps
Changing the input caps and not using them anymore afterwards is useless, and
it breaks negotiation in pipelines like:

gst-launch-1.0 videotestsrc ! "video/x-raw,framerate=25/1,interlace-mode=interleaved" !
  deinterlace fields=all ! "video/x-raw,framerate=50/1,interlace-mode=progressive" !
  fakesink
2016-03-17 21:11:36 +02:00
Nirbheek Chauhan
78847d03cf rtpmanager: Some comment and documentation clarifications/fixes 2016-03-15 09:32:47 +00:00
Sebastian Dröge
66e9e4c202 Revert "flacparse: push tags in pre_push_frame"
This reverts commit 4065fcb80a.

flacparse should not push tags by itself, the base class is going to do that
while properly merging in upstream tags. It just didn't because of a bug in
the base class, which was hidden by this commit.

https://bugzilla.gnome.org/show_bug.cgi?id=763553
2016-03-13 10:33:13 +02:00
Nirbheek Chauhan
bbde949e8e win32: Don't use __attribute__ on MSVC
Use MSVC-equivalents for alignment and packing compiler directives when building
on MSVC
2016-03-10 10:01:19 +00:00
Nirbheek Chauhan
63803bfac0 win32: Don't try to include xmath.h on newer Visual Studio 2016-03-10 10:01:19 +00:00
Nirbheek Chauhan
5d93844676 gst Factor out endian-order RGB formats
MSVC seems to ignore preprocessor conditionals inside static pad
template macros.
2016-03-10 10:00:58 +00:00
Thiago Santos
d8fb7a9c96 qtdemux: reset pending segment if we are already pushing one
When upstream is running in bytes in push-mode, qtdemux will
convert seeks from time to bytes and send it upstream. Upstream
element will perform a byte seek and send a byte segment to qtdemux
that will convert it to time and push it downstream.

There is, however, the pending_segment variable that stores a new
segment event to be pushed before the next data. When handling seeks
as mentioned above this variable was being ignored and, if it contained
some segment event, it would override the one resulting from the seek.
This would restore a previous segment and would cause the seek segment
to be discarded downstream.

This patch fixes this issue by unrefing any pending segment as the
seek from upstream should contain the latest one that should be
used, as requested by the application.

https://bugzilla.gnome.org/show_bug.cgi?id=763165
2016-03-07 15:26:13 -03:00
Thiago Santos
b46af7fda7 qtdemux: run gst-indent
Otherwise commits will fail with our indent check hook
2016-03-07 15:26:13 -03:00
Sebastian Dröge
49be64e571 udpsrc: Fix multicast group joining with provided sockets on Windows
On Windows the socket will be bound to ANY instead of the multicast group,
as binding to a multicast group does not work. Which would mean that we
override src->addr to become ANY and won't automatically join a multicast
group anymore on Windows.

On Linux we would automatically join a multicast group, keep it consistent.

https://bugzilla.gnome.org/show_bug.cgi?id=763093
2016-03-04 15:31:51 +02:00
Sebastian Dröge
b6e10be278 Revert "rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases"
This reverts commit a7fb7b5359.

The mutex is taken by the caller, we should keep it locked when returning so
the caller can unlock it again.
2016-03-02 13:13:24 +02:00
Luis de Bethencourt
4065fcb80a flacparse: push tags in pre_push_frame
Push a tag event before pre-roll if we have tags.

https://bugzilla.gnome.org/show_bug.cgi?id=762660
2016-03-01 19:23:02 +00:00
Tim-Philipp Müller
a7fb7b5359 rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases 2016-03-01 14:14:36 +00:00
Luis de Bethencourt
5dcf1a4f69 matroska-demux: remove impossible condition
It is impossible for a guint to have a negative value, no need to check for
this. Introduced in commit 6861d11c49

CID 1354509
2016-02-29 10:11:38 +00:00
Petr Viktorin
d089cd5a12 alpha: Fix sample pipeline
Use the zorder pad property to make sure the semitransparent
video is on top of the background.

https://bugzilla.gnome.org/show_bug.cgi?id=762809
2016-02-28 11:52:14 -05:00
Tim-Philipp Müller
a4d64b5caa rgvolume: make tag list writable before modifying it
Making the event itself writable is not enough, it won't make
the actual taglist in the event writable as well. Instead, just
make a copy of the taglist and then create a new tag event from
that if required, replacing the old one. Before we would
inadvertently modify taglists upstream elements might still
be holding on to. Add unit test for this as well.

https://bugzilla.gnome.org/show_bug.cgi?id=762793
2016-02-28 14:44:39 +00:00
Sebastian Dröge
bf5a72a6dd rtspsrc: Properly error out if binding the UDP sockets fails
udpsrc is not returning us a socket in that case.
2016-02-28 13:01:34 +02:00
Sebastian Dröge
03d2ae154e goom: Use goom_set_resolution() instead of recreating the goom instance when the resolution changes
https://bugzilla.gnome.org/show_bug.cgi?id=762765
2016-02-27 20:33:32 +02:00
Sebastian Dröge
bd0d2a3d7d Revert "goom: Initialize the goom struct only once we know width/height and recreate it if those change"
This reverts commit cc6e102643.
2016-02-27 20:32:45 +02:00
Sebastian Dröge
cc6e102643 goom: Initialize the goom struct only once we know width/height and recreate it if those change
Fixes crash when the width and/or height is changing.

https://bugzilla.gnome.org/show_bug.cgi?id=762765
2016-02-27 20:31:15 +02:00
Tim-Philipp Müller
fb0bc126c9 rtp: opus: move Opus RTP payloader/depayloader from -bad to -good
https://bugzilla.gnome.org/show_bug.cgi?id=756282
2016-02-25 22:45:16 +00:00
Tim-Philipp Müller
3b970e9b5e Merge branch 'plugin-move-rtp-opus'
Move Opus RTP depayloader/payloader from -bad to -good.

https://bugzilla.gnome.org/show_bug.cgi?id=756282
2016-02-25 22:45:15 +00:00
Philippe Normand
9c47c0da59 qtdemux: cenc aux info parsing from mdat support in PULL mode
This is already supported for PUSH mode but was failing in PULL mode.
The aux info is sometimes stored in the mdat before the first sample,
so the loop task needs to pull data stored at that location and
perform the aux info cenc parsing.

https://bugzilla.gnome.org/show_bug.cgi?id=761700

https://bugzilla.gnome.org/show_bug.cgi?id=762516
2016-02-25 12:46:27 +02:00
Philippe Normand
67f3fc1748 qtdemux: prevent buffer flow if any stream failed to be exposed
In some cases the stream configuration can fail, for instance if the
stream is protected and no decryptor was found. For those situations
the demuxer shouldn't emit any data on the corresponding source pad of
the stream and bail out.

https://bugzilla.gnome.org/show_bug.cgi?id=762516
2016-02-25 12:46:27 +02:00
Philippe Normand
fb5d50cd07 qtdemux: don't push encrypted buffer without cenc metadata
When the cenc metadata is stored outside of the moof box and the
stream is exposed it is possible that the cenc metadata hasn't been
processed yet while the first buffer is being pushed. When this
happens the buffer can't possibly be decrypted downstream so don't
push it.

https://bugzilla.gnome.org/show_bug.cgi?id=762516
2016-02-25 12:46:27 +02:00
Philippe Normand
459ef195bb qtdemux: read saio aux_info_type as a FOURCC
https://bugzilla.gnome.org/show_bug.cgi?id=756897
2016-02-24 10:54:23 +02:00
Sebastian Dröge
49f4631909 gst: Handle gst_pad_get_current_caps() returning NULL gracefully 2016-02-23 18:27:47 +02:00
Dave Craig
9b2e1f9f36 rtph265depay: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
Remove calls to gst_pad_has_current_caps() which then go on to call
gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
use gst_pad_get_current_caps() and check for NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=759539
2016-02-23 18:12:54 +02:00
Dave Craig
211c8492b3 gst: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
Remove calls to gst_pad_has_current_caps() which then go on to call
gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
use gst_pad_get_current_caps() and check for NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=759539
2016-02-23 18:11:42 +02:00
Dave Craig
6cdbf40622 aacparse: Handle gst_pad_get_current_caps() returning NULL gracefully
This can happen when the pipeline is currently shutting down.

https://bugzilla.gnome.org/show_bug.cgi?id=759539
2016-02-23 18:11:42 +02:00
Linus Svensson
a5691af319 matroska-demux: Don't handle seek until ready
https://bugzilla.gnome.org/show_bug.cgi?id=762542
2016-02-23 17:54:43 +02:00
Linus Svensson
1a3986d016 matroska-demux: Unref seek event
https://bugzilla.gnome.org/show_bug.cgi?id=762542
2016-02-23 17:54:43 +02:00
Aurélien Zanelli
84e441d268 multifilesink: close file on write error with next-file mode is set to buffer
If we have an error during fwrite call, file stays open and thus next
incoming buffer will trigger an assert when trying to opening a new
file.
This happens if we do not restart element, file is closed at stop, and
if application handles the returned GST_FLOW_ERROR to keep bin alive.

https://bugzilla.gnome.org/show_bug.cgi?id=762434
2016-02-23 11:34:31 +02:00
Matej Knopp
8657987f8f matroskamux: don't output empty tags/tag elements
Such files will not play on Android, because of bug in libwebm matroska parsing, which is still present in 6.0.1

https://bugzilla.gnome.org/show_bug.cgi?id=762349
2016-02-23 11:00:05 +02:00
Vincent Penquerc'h
6861d11c49 matroska-demux: make up an OpusHead block if possible when missing
https://bugzilla.gnome.org/show_bug.cgi?id=761489
2016-02-23 10:47:43 +02:00
Vincent Penquerc'h
565607107f matroska-mux: make up an OpusHead block if possible when missing
This block is needed in the Matroska file, but data coming from
RTP may not have one.

https://bugzilla.gnome.org/show_bug.cgi?id=761489
2016-02-23 10:47:43 +02:00
Mark Nauwelaerts
afad769c78 matroskademux: make stream-id more readable and order-friendly
... as streams are so ordered by id by e.g. decodebin
(and as typically already honoured by other demuxers).
2016-02-22 16:06:11 +01:00
Mark Nauwelaerts
7456ee1e1b matroska: remove confusing duplicate track uid field 2016-02-22 16:05:41 +01:00
Luis de Bethencourt
93cd4be8d5 rtpvp9pay: add missing break
VP9_PAY_PICTURE_ID_7BITS and VP9_PAY_PICTURE_ID_15BITS are mutually
exclusive options of the picture-id-mode. We can break after the
first case.

1 or 2 bytes need to be added to the header length depending on the
PictureID size.
https://tools.ietf.org/html/draft-uberti-payload-vp9-00#section-4.2

CID 1353479
2016-02-22 14:06:02 +00:00
Vineeth TM
7150b89c59 avidemux: Fix buffer memory leak
buffer being mapped is not being unmapped in some cases

https://bugzilla.gnome.org/show_bug.cgi?id=762420
2016-02-22 10:14:44 +02:00
Stian Selnes
5a2cc41398 rtpmanager: Don't warn for duplicate/reordered packets
This is a normal scenario and should not be a warning.

https://bugzilla.gnome.org/show_bug.cgi?id=762208
2016-02-21 22:37:57 +00:00
Tim-Philipp Müller
13a9a7543d win32: remove outdated build cruft
This hasn't been touched for generations, doesn't work,
and is just causing confusion. We also don't want to
maintain these files manually.
2016-02-21 09:47:43 +00:00
Matej Knopp
f96c9eb6bc qtdemux: workaround for files with wrong color_table_id value
Instead of erroring out, just use the default color table.

https://bugzilla.gnome.org/show_bug.cgi?id=761637
2016-02-19 16:00:59 +00:00
Tim-Philipp Müller
df341f41dc flvmux, rtpvp9depay: fix indentation 2016-02-19 15:04:15 +00:00
Havard Graff
7787f439fc flvmux: plug leak(s) in error-scenario
https://bugzilla.gnome.org/show_bug.cgi?id=762210
2016-02-19 14:59:09 +00:00
Havard Graff
1e09e5bfe9 flvdemux: fix eos event leak
https://bugzilla.gnome.org/show_bug.cgi?id=762209
2016-02-19 14:54:04 +00:00
Philippe Normand
52b16768a2 qtdemux: plug leaks in cenc aux info parsing 2016-02-19 10:30:46 +02:00
Sebastian Dröge
a7c3f353bd matroskademux: Unmap wavpack header buffer after creating it
Otherwise it will be mapped writable all the time and we can't read from it
anywhere.

https://bugzilla.gnome.org/show_bug.cgi?id=762239
2016-02-18 11:10:14 +02:00
Tim-Philipp Müller
d6685b247a rtp: sprinkle some G_GNUC_INTERNAL for internal utils functions 2016-02-17 15:07:37 +00:00
Alex Ashley
97f6f7c713 qtdemux: only transform protected caps once
Commit 7873bede31
(https://bugzilla.gnome.org/show_bug.cgi?id=760774) changed the
behaviour of qtdemux to call gst_qtdemux_configure_stream() for
every new moof.

When playing a protected stream, gst_qtdemux_configure_stream()
calls gst_qtdemux_configure_protected_caps(). The
gst_qtdemux_configure_protected_caps() function takes the original
media format, puts this in a field called "original-media-type"
and then changes the caps to "application/x-cenc".

The gst_qtdemux_configure_protected_caps() did not handle the case
of being called multiple times, causing it to incorrectly set the
caps. The second call was causing the caps to be set to:

    application/x-cenc, original-media-type"application/x-cenc"

This commit makes gst_qtdemux_configure_protected_caps() check that
the caps have already been transformed, so that it only gets
changed once.

    https://bugzilla.gnome.org/show_bug.cgi?id=761769
2016-02-17 17:04:25 +02:00
Sebastian Dröge
01342378b5 opus: Add proper support for multichannel audio
https://bugzilla.gnome.org/show_bug.cgi?id=757152
2016-02-17 14:58:01 +00:00
Sebastian Dröge
0472d9f8b2 opus: Copy metadata in the (de)payloader, but only the relevant ones
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without tags or
with only the audio tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
2016-02-17 14:58:01 +00:00
Sebastian Dröge
ff51629c9a opusdepay: Set multistream=FALSE on the Opus caps
The RTP Opus mapping only allows mono/stereo, and not multistream Opus
streams.
2016-02-17 14:58:01 +00:00
Olivier Crête
89b172b3ed rtpopuspay: Forward stereo preferences from caps upstream
https://bugzilla.gnome.org/show_bug.cgi?id=746617
2016-02-17 14:58:01 +00:00
Olivier Crête
4df223f325 rtpopuspay: Set the number of channels to 2 as per RFC draft
https://bugzilla.gnome.org/show_bug.cgi?id=746617
2016-02-17 14:58:01 +00:00
Sebastian Dröge
bbb1143ca3 opus: Handle sprop-stereo and sprop-maxcapturerate RTP caps fields
https://bugzilla.gnome.org/show_bug.cgi?id=746617
2016-02-17 14:58:01 +00:00
Vincent Penquerc'h
4b5ad70924 rtpopuspay: default encoding name to OPUS
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2016-02-17 14:58:01 +00:00
Vincent Penquerc'h
755289ed0c rtpopuspay: make caps writable before truncating them
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2016-02-17 14:58:01 +00:00
Vincent Penquerc'h
e427369840 rtpopuspay: negotiate the encoding name
Chrome uses a different encoding name that gstreamer.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2016-02-17 14:58:01 +00:00
Nicolas Dufresne
9e4511edf4 rtpopus: Use OPUS encoding name
Both Firefox and Chrome uses OPUS as the encoding in their SDP.
Adding this now defacto standard name remove the need for special
case in SDP parsing code.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2016-02-17 14:58:01 +00:00
Wim Taymans
b310393916 opuspay: fix timestamps
Copy timestamps to payloaded buffer.
Avoid input buffer memory leak.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692929
2016-02-17 14:58:00 +00:00
Tim-Philipp Müller
117e30c47e Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2016-02-17 14:58:00 +00:00
Wim Taymans
5d893c7ea2 opuspay: remove pointless caps serialization
Remove the caps serialization in the rtp caps. the spec nor the receiver
does anything with it.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686547
2016-02-17 14:58:00 +00:00
Tim-Philipp Müller
17742d2347 Use gst_element_class_set_static_metadata()
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
2016-02-17 14:58:00 +00:00
Olivier Crête
18638c9c4e rtpopuspay: Allocate the rtp buffer correctly
Use the right functions to allocate the rtp buffer
2016-02-17 14:58:00 +00:00
Mark Nauwelaerts
ad261f64c3 replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2016-02-17 14:58:00 +00:00
Mark Nauwelaerts
d196562755 opus: port to updated 0.11 2016-02-17 14:58:00 +00:00
Edward Hervey
77ea437507 Merge remote-tracking branch 'origin/master' into 0.11-premerge
Conflicts:
	docs/libs/Makefile.am
	ext/kate/gstkatetiger.c
	ext/opus/gstopusdec.c
	ext/xvid/gstxvidenc.c
	gst-libs/gst/basecamerabinsrc/Makefile.am
	gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
	gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
	gst-libs/gst/video/gstbasevideocodec.c
	gst-libs/gst/video/gstbasevideocodec.h
	gst-libs/gst/video/gstbasevideodecoder.c
	gst-libs/gst/video/gstbasevideoencoder.c
	gst/asfmux/gstasfmux.c
	gst/audiovisualizers/gstwavescope.c
	gst/camerabin2/gstcamerabin2.c
	gst/debugutils/gstcompare.c
	gst/frei0r/gstfrei0rmixer.c
	gst/mpegpsmux/mpegpsmux.c
	gst/mpegtsmux/mpegtsmux.c
	gst/mxf/mxfmux.c
	gst/videomeasure/gstvideomeasure_ssim.c
	gst/videoparsers/gsth264parse.c
	gst/videoparsers/gstmpeg4videoparse.c
2016-02-17 14:58:00 +00:00
Vincent Penquerc'h
8df374108a opusenc: add upstream negotiation for multistream ability
This will help elements that cannot deal with multistream,
such as the RTP payloader.

The caps now do not include a "streams" field anymore, but
a "multistream" boolean, since we have no real use for knowing
the exact amount of streams.

https://bugzilla.gnome.org/show_bug.cgi?id=665078
2016-02-17 14:58:00 +00:00
Danilo Cesar Lemes de Paula
c207bdf1e7 Adding opus RTP payloader/depayloader element
Adding OPUS RTP module based on the current draft:
http://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.txt

https://bugzilla.gnome.org/show_bug.cgi?id=664817
2016-02-17 14:58:00 +00:00
Luis de Bethencourt
f2f31ec50f rtp: h264/h265: avoid duplication of read_golomb()
There is no need to have two identical implementations of the read_golomb
function.

https://bugzilla.gnome.org/show_bug.cgi?id=761606
2016-02-17 14:18:16 +00:00
Ognyan Tonchev
750b7c72fe matroskademux: Simple implementation of TRICKMODE_KEY_UNITS
When the trickmode key-units flag is set on the segment, simply skip
any sample on a video stream that isn't a keyframe

https://bugzilla.gnome.org/show_bug.cgi?id=762185
2016-02-17 16:17:13 +02:00
Tim-Philipp Müller
77403d0afe matroska-demux: send GAP events for lagging audio and video streams too
Send GAP events for non-subtitle streams too if they lag too much
behind, but use a higher threshold than for subtitles.

This helps with fixing prerolling with a file where one of the
audio streams only has data starting from 19s onwards. It's not
a complete fix yet, it also requires changes elsewhere, such as
in baseparse, to make sure caps are propagated.

https://bugzilla.gnome.org/show_bug.cgi?id=614460
https://bugzilla.gnome.org/show_bug.cgi?id=753899
2016-02-16 17:11:39 +00:00
Stian Selnes
5faa9c11a6 rtpvp9pay: rtpvp9depay: Initial implementation of draft 01
Quick and dirty implementation of an RTP payloader and depayloader
for VP9. In particalur it assumes no spatial or temporal layering,
non-flexible mode, and some other bits and pieces.

https://bugzilla.gnome.org/show_bug.cgi?id=754773
2016-02-16 15:54:06 +02:00
Vineeth TM
dc70bfd36a avidemux: Fix string memory leak
codec_name is not being freed in all conditions leading to memory leak

https://bugzilla.gnome.org/show_bug.cgi?id=762117
2016-02-16 11:43:24 +00:00
Miguel París Díaz
92affe2dec rtpbin: add "get-session" signal
This gets the GstRTPSession element, as compared to the RTPSession object
that is returned by get-internal-session.

https://bugzilla.gnome.org/show_bug.cgi?id=759293
2016-02-16 13:39:52 +02:00
Tim-Philipp Müller
9d0f127703 rtp: h265: hook up move RTP H.265 payloader/depayloader to build
https://bugzilla.gnome.org/show_bug.cgi?id=761606
2016-02-16 00:25:50 +00:00
Tim-Philipp Müller
7f9f3d38b2 rtp: h265: use common meta utility functions
https://bugzilla.gnome.org/show_bug.cgi?id=761606
2016-02-16 00:25:46 +00:00
Tim-Philipp Müller
714d31ce30 rtp: h265: remove codecparser dependency from h265 payloader/depayloader
Looks like it just uses the NAL enums and nothing else from
the codecparsers, and that's the only reason it had to be
moved from -good to -bad when it was originally added. We
can probably keep those NAL enums up to date enough, so let's
remove the codecparser dependency so it can be moved back into
-good.

https://bugzilla.gnome.org/show_bug.cgi?id=761606
2016-02-16 00:25:41 +00:00
Tim-Philipp Müller
a70c75782b Merge branch 'plugin-move-rtp-h265'
Move RTP H.265 payloader/depayloader from -bad to -good.

https://bugzilla.gnome.org/show_bug.cgi?id=761606
2016-02-16 00:24:58 +00:00
Luis de Bethencourt
139108c83a gstrtph265depay: keep consistency with rtph264depay
Use gst_rtp_drop_meta() and the same function prototype for
gst_rtp_copy_meta() to keep consistency with the RTP elements in
gst-plugins-good
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
403ac009fa rtph265depay: fix termination of access unit
Only consider the access unit complete when the next-occurring VCL NAL unit
has the first bit after its NAL unit header equal to 1.
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
983e30f658 rtph265depay: fix unneeded sub-buffer creation
We create a sub-buffer just to copy over its metas and then throw it
away immediately, just use the original input buffer directly.
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
4ee6c17edb rtph265pay: add "send VPS/SPS/PPS with every key frame" mode
It's not enough to have timeout or event based VPS/SPS/PPS information
sent in RTP packets. There are some scenarios when key frames may appear
more frequently than once a second, in which case the minimum timeout
for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough.
It might also be desirable in general to make sure the VPS/SPS/PPS is
available with every keyframe (packet loss aside), so receivers can
actually pick up decoding immediately from the first keyframe if
VPS/SPS/PPS is not signaled out of band.

This commit adds the possibility to send VPS/SPS/PPS with every key frame.
This mode can be enabled by setting "config-interval" property to -1. In
this case the payloader will add VPS, SPS and PPS before every key (IDR)
frame.

https://bugzilla.gnome.org/show_bug.cgi?id=757892
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
64ca3b26d9 rtph265pay: change config-interval property type from uint to int
This way we can use -1 as special value, which is nicer than MAXUINT.

https://bugzilla.gnome.org/show_bug.cgi?id=757892
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
698e5bbfb5 rtph265depay: make sure we call handle_nal for each NAL
Call handle_nal for each NAL in the STAP-A RTP packet. This makes sure
we correctly extract the SPS and PPS.

https://bugzilla.gnome.org/show_bug.cgi?id=730999
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
1e55d0d725 rtph265pay: Copy metadata in the payloader, but only the relevant ones
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
8611645af6 rtph265pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()
https://bugzilla.gnome.org/show_bug.cgi?id=753228
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
df724c410b rtph265pay: fix potential crash when shutting down
A race condition in the state change function may cause buffers to be
unreffed while they are still used by the streaming thread in
gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the
parent class first in the state change function to make sure streaming
has stopped and only then free those buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=741381
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
f2bae3ab59 rtph265pay: fix buffer leak when using SPS/PPS
Fixes a buffer leak that would occur if the pipeline was shutdown while a
SPS/PPS header was being created.

https://bugzilla.gnome.org/show_bug.cgi?id=741271
2016-02-16 00:24:41 +00:00
Luis de Bethencourt
f1e2849438 rtph265depay: copy metadata in the depayloader, but only the relevant ones
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
3bede1c95b rtph265depay: checking if depay has sps/pps nals before insertion
Related to: https://bugzilla.gnome.org/show_bug.cgi?id=753430

https://bugzilla.gnome.org/show_bug.cgi?id=753228
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
18b628824b rtph265depay: only update the srcpad caps if something else than the codec_data changed
h264parse and gstrtph264depay do the same, let's keep the behaviour
consistent. As we now include the codec_data inside the stream, this causes
less caps renegotiation.

https://bugzilla.gnome.org/show_bug.cgi?id=753228
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
3979ffa6a3 rtph265depay: PPS replaces old PPS if it has the same id
https://bugzilla.gnome.org/show_bug.cgi?id=753228
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
d10b6f1e3a rtph265depay: Insert SPS/PPS NALs into the stream
rtph264depay does the same and this fixes decoding of some streams with 32
SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255),
but the field in the codec_data for the number of SPS or PPS is only 5
(or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.

This looks like a mistake in the part of the spect about the codec_data.
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
0bfa97b047 rtph265depay: implement process_rtp_packet() vfunc
For more optimised RTP packet handling: means we don't need to map the
input buffer again but can just re-use the mapping the base class has
already done.

Based on: https://bugzilla.gnome.org/show_bug.cgi?id=750235

https://bugzilla.gnome.org/show_bug.cgi?id=753228
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
a526d014db rtph265depay: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
Switching to GST_BUFFER_TIMESTAMP() to be consistent with other rtp code.
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
470c8b3720 rtph265depay: prevent trying to get 0 bytes from adapter
This causes an assertion and would lead to getting a NULL instead
of a buffer. Without proper checking this would easily lead to a
segfault.

Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
7ae49b46ff rtp: remove dead assignment
Value set to ret will be overwritten at least once at the end of the while
loop, removing assignment.
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
693a924461 remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused.
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
51791d8fe2 rtp: donl_present variable unused
donl_present is not implemented, yet the value is set and checked a few times.
Cleaning this.

CID #1249687
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
e3d8d8cedb rtp: value truncated too short creates dead code
type is truncated to 0-31 with "& 0x1f", but right after that it is checks if
the value is equivalent to GST_H265_NAL_VPS, GST_H265_NAL_SPS, and
GST_H265_NAL_PPS (which are 32, 33, and 34 respectively). Obviously, this will
never be True if the value is maximum 31 after the truncation.
The intention of the code was to truncate to 0-63.
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
59fea44503 rtp: fix nal unit type check
After further investigation the previous commit is wrong. The code intended to
check if the type is 39 or the ranges 41-44 and 48-55. Just like gsth265parse.c
does. Type 40 would not be complete.
2016-02-16 00:24:40 +00:00
Luis de Bethencourt
d215b18a20 rtp: fix dead code and check for impossible values
nal_type is the index for a GstH265NalUnitType enum. There are two types of dead
code here:
First, after checking if nal_type is >= 39 there are two OR conditionals that
check if the value is in ranges higher than that number, so if nal_type >= 39
falls in the True branch those other conditions aren't checked and if it falls
in the False branch and they are checked, they will always also be False. They
are redundant.
Second, the enum has a range of 0 to 40. So the checks for ranges higher than 41
should never be True.
Removing this redundant checks.

CID 1249684
2016-02-16 00:24:40 +00:00
Thijs Vermeir
544c0d75ce rtp: add h265 RTP payloader + depayloader 2016-02-16 00:24:40 +00:00
Stefan Sauer
af29e77858 monoscope: rework the scaling code
The running average was wrong and the resulting scaling factor was only held in
place using the CLAMP. In addtion we are now convering quickly to volume
changes.

FInally now with this change, we can change the resolution defines and
everythign adjusts.
2016-02-12 21:01:03 +01:00
Stefan Sauer
5e68873d22 monoscope: use constants in the drawing code
Make all the drawing ops be based on the constants. This way we can change
the fixed size at least at compile time.
2016-02-12 21:01:03 +01:00
Stefan Sauer
292d44316e monoscope: replace hardcoded values by constants
This at least establishes the relationship.
2016-02-12 21:01:03 +01:00
Stefan Sauer
8d17911b33 monoscpe: make the convolver use dynamic memory
Replace all #defines with members and initialize the convolver with a parameter.
2016-02-12 21:01:03 +01:00
Stefan Sauer
3d23ceebae monoscope: update README
We can already create multiple instances.
2016-02-12 21:01:03 +01:00
Stefan Sauer
daea0540fd monoscope: code cleanup
Use constants more often. Cleanup comments and add more to explain how things
work.
2016-02-12 21:01:03 +01:00
Luis de Bethencourt
3738ce8ba1 deinterlace: remove check for impossible condition
Commit bd27a1f30b added a few error handling
memory management checks. These check srccaps to see if it needs to be
unreferenced before returning, in the case of invalid_caps this goto jump
always happens before srccaps is set, so it will always be NULL in this
error label.

CID #1352035
2016-02-08 23:48:28 +00:00
Tim-Philipp Müller
f301e3f236 matroska: get rid of _stdint.h include 2016-02-08 00:11:55 +00:00
Sebastian Dröge
e244b9be87 rtpjpegpay: Skip APP and JPG markers and print warnings for unknown markers
For APP/JPG markers the size is following and we have to skip that. This is
not really a problem unless the marker contains e.g. a preview JPEG or
something else that we might interprete as another marker.
2016-01-31 11:05:05 +11:00
Seungha Yang
7873bede31 qtdemux: fix framerate calculation for fragmented format
qtdemux calculates framerate using duration and the number of sample.
In case of fragmented mp4 format, however, the number of sample can
be figure out after parsing every moof box. Because qtdemux does not
parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect
framerate calculation.

This patch will triger gst_qtdemux_configure_stream() for every new moof.
Then, framerate will be calculated by using duration and n_samples of the moof.

https://bugzilla.gnome.org/show_bug.cgi?id=760774
2016-01-29 11:01:44 +01:00
Seungha Yang
0391a93a35 qtdemux: handling zero segment-duration edit list
Based on document ISO_IEC_14496-12, edit list box can have
segment duration as zero. It does not imply that media_start equals to
media_stop. But, it just indicates a sample which should be presented
at the first. This patch derives segment duration using media_time
and duration of file. And set derived duration to segment-duration.

https://bugzilla.gnome.org/show_bug.cgi?id=760781
2016-01-29 10:57:05 +01:00
Seungha Yang
d8bb6687ea qtdemux: expose streams with first moof for fragmented format
In case of push mode, qtdemux expose streams after got moov box.
We can not guarantee that a moov box has sample data such as sample duration
and the number of sample in stbl box for fragmented format case.
So, if a moov has no sample data, streams will not be exposed until get the first moof.

https://bugzilla.gnome.org/show_bug.cgi?id=760779
2016-01-29 10:53:39 +01:00
Sebastian Dröge
3edf0737d6 deinterlace: Check for subset instead of non-empty intersection for ACCEPT_CAPS 2016-01-27 18:48:17 +01:00
Sebastian Dröge
c7d90c1112 deinterlace: Unset RECONFIGURE flag on srcpad whenever we configure new caps
Prevents double-negotiation during startup and in some other cases.
2016-01-27 18:44:23 +01:00
Vivia Nikolaidou
bd27a1f30b deinterlace: Do passthrough in auto mode if downstream only supports interlaced
If the following conditions are met:
1) upstream and downstream caps are compatible
2) upstream is interlaced
3) downstream doesn't support progressive mode
then deinterlace will just do passthrough instead of failing to link.

This is done with the following scenario in mind:

videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
queue ! deinterlace name=dein_desktop ! autovideosink
In this case, dein_src will do the deinterlacing. However,

videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
queue ! deinterlace name=dein_desktop ! autovideosink t. ! queue !
"video/x-raw,interlace-mode=interleaved" ! fakesink

In this case, caps auto-negotiation will make dein_file and dein_desktop do
the deinterlacing, while dein_src will be passthrough.

https://bugzilla.gnome.org/show_bug.cgi?id=760995
2016-01-27 16:45:29 +01:00
Sebastian Dröge
46735f8de9 deinterlace: Add mode=auto-strict
In this mode we will passthrough all progressive caps but interlaced caps must be
caps where we actually support deinterlacing.

This is the only difference between auto and auto-strict, auto would
passthrough all unsupported interlaced caps.

https://bugzilla.gnome.org/show_bug.cgi?id=720388
2016-01-27 16:45:29 +01:00
Sebastian Dröge
2e8d4e8c7a deinterlace: Implement reconfiguration a bit better
And e.g. consider reconfiguration caused by RECONFIGURE events too.

https://bugzilla.gnome.org/show_bug.cgi?id=720388
2016-01-27 16:45:29 +01:00
Sebastian Dröge
8c1c091439 deinterlace: Rewrite caps negotiation
Previously the result of the CAPS query and ACCEPT_CAPS depended on what kind
of caps were last set, and e.g. if we last had interlaced caps or not. That's
just broken.

Also previously the handling of non-sysmem caps features was rather random and
unusuable.

Now the behaviour is the following, depending on the mode property:
1) mode=disabled
  Completely do passthrough of everything
2) mode=interlaced
  Only accept formats we can actually deinterlace, and accept interlaced
  and progressive content and always run the deinterlacer and output
  progressive content
3) mode=auto (i.e. playbin)
  Accept all progressive formats as passthrough, accept all formats that we
  can deinterlace ourselves (which we do then), but also accept everything
  else for which we then just passthrough. In auto mode, deinterlacing is best
  effort: If we can, we deinterlace, if we can't we just output interlaced
  content.

https://bugzilla.gnome.org/show_bug.cgi?id=720388
https://bugzilla.gnome.org/show_bug.cgi?id=760553
2016-01-27 16:45:29 +01:00
Sebastian Dröge
1053af6d0c deinterlace: Remove unused, obsolete bufferalloc code 2016-01-27 16:45:29 +01:00
Matej Knopp
e7460d9c06 matroskamux: use A_AAC instead of A_AAC/MPEGx/y
Some GoogleCast compatible devices ignore A_AAC/MPEGx/y tracks; Also according to http://wiki.multimedia.cx/index.php?title=Matroska A_AAC/MPEGx/y is obsolete

https://bugzilla.gnome.org/show_bug.cgi?id=761144
2016-01-27 13:50:21 +01:00
Víctor Manuel Jáquez Leal
e1834d1512 gst: Fix unintialized variable warnings
While cross-compiling with Linaro GCC 5.1-2015.08, it complained
about a couple unitialized variables.

This patch initializes them to zero.

https://bugzilla.gnome.org/show_bug.cgi?id=761094
2016-01-27 13:46:07 +01:00
George Kiagiadakis
eafa9f08f7 splitmuxsrc: print potentially negative offset with a sign 2016-01-25 15:36:29 +01:00
Tim-Philipp Müller
5d14746792 taginject: fix sample pipeline in docs
https://bugzilla.gnome.org/show_bug.cgi?id=679571
2016-01-21 15:30:42 +00:00
Tim-Philipp Müller
aeed2e550c rtp: fix compiler warnings with gcc-6
In file included from gstrtpL16depay.h:27:0,
                 from gstrtp.c:73:
gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable]
 static const GstRTPChannelOrder channel_orders[] =
2016-01-19 13:04:39 +00:00
Sebastian Dröge
7927f49ca0 wavparse: Don't play anything after the end of the data chunk even when seeking
Especially in push mode we would completely ignore the size of the data chunk
when not stop position is given for the seek. Instead make sure that the end
offset is at most the end of the data chunk if known.

Without this we would output anything after the data chunk, possibly causing
loud noises if the media file is followed by an INFO chunk or an ID3 tag.
2016-01-19 14:57:03 +02:00
Sebastian Dröge
322bdf5136 wavparse: Don't do calculations with -1 offsets when handling SEGMENT events
We use that to signal "infinity", taking the difference between that and some
other value is not going to give us any useful result for the end offsets of
segments.
2016-01-19 14:55:57 +02:00
Sebastian Dröge
366bbffcd8 Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling"
This reverts commit 271501f657.

It wasn't meant to be pushed yet as the commit message indicates.
2016-01-18 11:30:45 +02:00
Aleix Conchillo Flaqué
665d14a2a0 rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
We check the stream profile and use the proper RTCP caps:
application/x-srtcp if we are using a secure profile and
application/x-rtcp otherwise.

https://bugzilla.gnome.org/show_bug.cgi?id=760556
2016-01-18 11:29:25 +02:00
Sebastian Dröge
271501f657 WIP: rtpjitterbuffer: Add RFC7273 media clock handling 2016-01-18 08:58:59 +02:00
Sebastian Dröge
53c797d604 wavparse: When flushing on EOS, don't process more data than the "data" size
Even if we have more data queued up when flushing than the size of the data
chunk, don't process and output it. If the data size is known, this likely
contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just
outputting them as if they were data is going to cause unexpected behaviour
and unpleasant audio noises.
2016-01-13 23:42:31 +01:00
Antonio Ospite
bdcc0390af interleave: Fix the example by setting channel-masks in the sink pads
The current example does not work, it fails with:

ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error.
gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0:
streaming task paused, reason not-negotiated (-4)

This is because negotiation with wavenc gets messed up by the missing
channel positions configuration.

The proper way to define the channel layout when using the interleave
element in code would be to set the channel-positions property, but
gst-launch-1.0 does not know how to deal with arrays; so the example
pipeline works around the issue by setting the channel-masks in the sink
pads.

Also fix a repetition in the deinterleave example description

https://bugzilla.gnome.org/show_bug.cgi?id=735673
2016-01-12 22:11:30 +00:00
Tim Sheridan
205565ccd9 sbcparse: Fix frame length calculation
SBC frame length calculation wasn't being rounded up to the nearest byte
(as specified in the A2DP 1.0 specification, section 12.9). This could
cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly
calculated frame lengths.

Incorrect frame length calculation causes frame coalescing to fail, as
subsequent frames in the stream aren't found in the expected locations.

https://bugzilla.gnome.org/show_bug.cgi?id=742446
2016-01-12 21:52:12 +00:00
Reynaldo H. Verdejo Pinochet
0bb8000874 flacparse: demote warning on wrong reserved value to fixme
We are likely just parsing a backward-compatible stream we
don't fully support.
2016-01-10 22:54:12 -08:00
Thiago Santos
4ac0a49308 imagefreeze: simplify caps selection
The downstream caps query with a filter alraedy gives us the possible
intersection so there is no need to check it again with downstream
if it is supported. Just try to set it directly.
2016-01-08 16:29:29 -03:00
Tim-Philipp Müller
3aa0dd8629 rtph264depay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
6171b0a675 rtpdvdepay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
c75f94c8f5 rtpamrdepay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
a8b8643977 rtpvrawdepay: fix major memory leak and performance issue
We call gst_rtp_buffer_get_payload() which creates a sub-buffer
of each input buffer, just to copy over metas, and then leak it.

https://bugzilla.gnome.org/show_bug.cgi?id=760289
2016-01-08 16:40:28 +00:00
Tim-Philipp Müller
6dab3ece07 flacparse: don't map buffer multiple times when parsing 2016-01-07 16:24:09 +00:00