Commit graph

8346 commits

Author SHA1 Message Date
Sebastian Dröge
d1c7f2e4d1 rtspsrc: Fix compiler warnings
gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
      'GstRTSPResult' [-Werror,-Wenum-conversion]
    res = gst_sdp_message_new (&sdp);
        ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
      'GstRTSPResult' [-Werror,-Wenum-conversion]
    res = gst_sdp_message_parse_uri (uri, sdp);
        ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
2014-09-26 13:46:16 +03:00
Jonas Holmberg
1371fa0c61 matroskademux: make demuxer reusable
Remove pads from flow combiner and reset last
flow return to FLOW_OK by resetting the flow combiner.
This prevents FLOW_FLUSHING when trying to re-use the
demuxer after setting it back to NULL/READY state.

https://bugzilla.gnome.org/show_bug.cgi?id=737359
2014-09-25 16:14:18 +01:00
Wim Taymans
84ec78bd86 videomixer: use video library code instead of copy 2014-09-24 16:46:36 +02:00
Sanjay NM
323683db96 audioparsers: Added index check before using the index
https://bugzilla.gnome.org/show_bug.cgi?id=736878
2014-09-24 10:21:35 +03:00
Matej Knopp
9f85dfd733 qtmux: Do not infer DTS on buffers from sparse streams.
DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later.
This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap)

https://bugzilla.gnome.org/show_bug.cgi?id=737095
2014-09-23 22:25:47 -03:00
Sanjay NM
36140ccf69 goom: Clarified precedence between % and ?
https://bugzilla.gnome.org/show_bug.cgi?id=736887
2014-09-24 00:48:09 +01:00
Sanjay NM
f62076e49c rtsp: clarify expression so operator precedence is clear
https://bugzilla.gnome.org/show_bug.cgi?id=736903
2014-09-24 00:48:09 +01:00
Sanjay NM
26a1344f37 Miscellaneous minor cleanups
Fix redundant variables and assignments,
and unreachable breaks.

https://bugzilla.gnome.org/show_bug.cgi?id=736875
https://bugzilla.gnome.org/show_bug.cgi?id=736876
https://bugzilla.gnome.org/show_bug.cgi?id=736879
https://bugzilla.gnome.org/show_bug.cgi?id=736880
https://bugzilla.gnome.org/show_bug.cgi?id=736881
https://bugzilla.gnome.org/show_bug.cgi?id=736888
https://bugzilla.gnome.org/show_bug.cgi?id=736890
https://bugzilla.gnome.org/show_bug.cgi?id=736892
https://bugzilla.gnome.org/show_bug.cgi?id=736893
https://bugzilla.gnome.org/show_bug.cgi?id=736894
2014-09-24 00:45:31 +01:00
Tim-Philipp Müller
208e12dca2 videobox: remove duplicate assignments
https://bugzilla.gnome.org/show_bug.cgi?id=736897
2014-09-24 00:12:14 +01:00
Sebastian Dröge
91a3d044f0 flacparse: Only calculate with durations != -1 2014-09-23 22:56:21 +03:00
Matej Knopp
fd3e8c5672 qtmux: collect pad for sparse stream should be created with lock set to false
Avoids waiting for buffers from sparse streams

https://bugzilla.gnome.org/show_bug.cgi?id=737095
2014-09-23 15:25:45 -03:00
Matej Knopp
6695341583 qtmux: fix subtitle buffer duration and strip null termination
Strip the \0 off the subtitle as we already know the size and also remember
to set the duration as buffer copying doesn't do it.

https://bugzilla.gnome.org/show_bug.cgi?id=737095
2014-09-23 15:25:28 -03:00
Matej Knopp
f57e9c4516 qtmux: move subtitle layer above video and set alternate group
layer -1 is above video, that is 0
And having all subtitles in alternate group 2 means that only one
should be selected at a time.

https://bugzilla.gnome.org/show_bug.cgi?id=737095
2014-09-23 15:20:37 -03:00
Matej Knopp
8a4931726d qtdemux: Handle mp4a without ESDS atom
https://bugzilla.gnome.org/show_bug.cgi?id=736986
2014-09-22 13:04:52 -03:00
Sanjay NM
89eb378598 dtmf: Removed unused structure members
https://bugzilla.gnome.org/show_bug.cgi?id=736883
2014-09-19 15:42:04 -04:00
Reynaldo H. Verdejo Pinochet
e655d47dfc isomp4: fix wrong DAR calculation for PAR <= 1
CID #1226452

https://bugzilla.gnome.org/show_bug.cgi?id=736396
2014-09-18 18:53:38 -03:00
Sanjay NM
ba4b9b22d0 flv: Removed unreachable break statements
https://bugzilla.gnome.org/show_bug.cgi?id=736884
2014-09-18 09:42:43 -04:00
Ognyan Tonchev
f7ae4288a2 rtpbin: do not leak encsink pad in error case
https://bugzilla.gnome.org/show_bug.cgi?id=736807
2014-09-18 12:49:53 +03:00
Ognyan Tonchev
3bf81ad12c multipartdemux: do not leak new stream event
https://bugzilla.gnome.org/show_bug.cgi?id=736805
2014-09-18 12:49:53 +03:00
Ravi Kiran K N
5480f6d2dd y4menc: port y4menc to use GstVideoEncoder base class
https://bugzilla.gnome.org/show_bug.cgi?id=735085
2014-09-17 18:28:00 -03:00
Ognyan Tonchev
7cd335e9b9 flacparse: do not leak uid after parsing TOC event
https://bugzilla.gnome.org/show_bug.cgi?id=736739
2014-09-17 09:51:15 +03:00
Sebastian Dröge
4bc10e755a rtpvrawdepay: Declare some more required caps fields in the sink template caps
Now only missing are width and height, which are expressed as strings
for RTP... so we can't put them into the template caps.
2014-09-16 22:47:13 +03:00
Wim Taymans
711e1407a1 capssetter: update to 1.0 transform_caps sematics
In 1.0, we pass the complete caps to transform_caps to allow for better
optimizations. Make this function actually work on non-simple caps
instead of just ignoring the configured filter caps.
2014-09-15 18:14:06 +02:00
Peter G. Baum
f8f61237f8 wavenc: use WAVE_FORMAT_EXTENSIBLE for more than 2 channels
https://bugzilla.gnome.org/show_bug.cgi?id=733444
2014-09-15 11:19:23 +03:00
Sebastian Dröge
a9d7c1d95e wavparse: Fix parsing of adtl chunks
We have to skip 12 bytes of data for the chunk, and the data size
passed to the sub-chunk parsing functions should have 4 bytes less
than the data size.

Also when parsing the sub-chunks, check if we actually have enough
data to read instead of just crashing.

https://bugzilla.gnome.org/show_bug.cgi?id=736266
2014-09-12 15:08:23 +03:00
Sanjay NM
66810a32f6 udp: include string.h for memcmp and memset
https://bugzilla.gnome.org//show_bug.cgi?id=736528
2014-09-12 10:45:39 +01:00
Anuj Jaiswal
4242495ea7 matroskamux: don't bitwise OR the same flag twice
https://bugzilla.gnome.org//show_bug.cgi?id=736543
2014-09-12 10:37:31 +01:00
Tim-Philipp Müller
4c08f2694d matroskademux: handle real audio 28_8
Fixes duplicate check for 14_4.

https://bugzilla.gnome.org//show_bug.cgi?id=736543
2014-09-12 10:35:36 +01:00
Anuj Jaiswal
86579c59bf multifilesink: don't OR the same flag twice
https://bugzilla.gnome.org/show_bug.cgi?id=736462
2014-09-11 11:05:35 +01:00
Tim-Philipp Müller
e6f77948ac udpsrc: more efficient memory handling
Drop use of g_socket_get_available_bytes() which is
not useful on all systems (where it returns the size
of the entire buffer not that of the next pending
packet), and is yet another syscall and apparently
very inefficient on Windows in the UDP case.

Instead, when reading UDP packets, use the more featureful
g_socket_receive_message() call that allows to read into
scattered memory, and allocate one memory chunk which is
likely to be large enough for a packet, while also providing
a larger allocated memory chunk just in case the packet
is larger than expected. If the received data fits into the
first chunk, we'll just add that to the buffer we return
and re-use the fallback buffer for next time, otherwise we
add both chunks to the buffer.

This reduces memory waste more reliably on systems where
get_available_bytes() doesn't work properly.

In a multimedia streaming scenario, incoming UDP packets
are almost never fragmented and thus almost always smaller
than the MTU size, which is also why we don't try to do
something smarter with more fallback memory chunks of
different sizes. The fallback scenario is just for when
someone built a broken sender pipeline (not using a
payloader or somesuch)

https://bugzilla.gnome.org/show_bug.cgi?id=610364
2014-09-09 17:38:52 +01:00
Tim-Philipp Müller
39505584e1 udpsrc: rework memory allocation bits and ensure we always have two chunks of memories to read into
First chunk is the likely/expected buffer size, second is as
fallback in case the packet is larger in the end.

Next step: actually use these.
2014-09-09 17:35:38 +01:00
Tim-Philipp Müller
305e4c2f46 udpsrc: track max packet size and save allocator negotiated by GstBaseSrc 2014-09-09 17:35:14 +01:00
Tim-Philipp Müller
8e28994207 audioecho: fix example command line 2014-09-08 16:15:32 +01:00
Tim-Philipp Müller
7271ff253b avidemux: fix crash with certain videos
This is a regression from 1.2 caused by the port
to the pad flow combiner.

https://bugzilla.gnome.org/show_bug.cgi?id=736192
2014-09-07 12:48:16 +01:00
Sebastian Dröge
a3a5530518 matroska-demux: Don't handle parse errors at the end of file as an error
But only if they happen after the Matroska segment.

https://bugzilla.gnome.org/show_bug.cgi?id=735833
2014-09-05 11:36:30 +03:00
Andrei Sarakeev
558f9a2a6f videomixer: Fix synchronization if dynamically changing the FPS
https://bugzilla.gnome.org/show_bug.cgi?id=735859
2014-09-04 11:34:26 +03:00
Ravi Kiran K N
ea43ef214a smpte: Check if input caps are the same and create output caps from video info
This makes sure that also properties like the pixel-aspect-ratio are the same
between both streams and that the output caps contain all fields necessary for
complete video caps.

https://bugzilla.gnome.org/show_bug.cgi?id=735804
2014-09-04 10:47:34 +03:00
Vineeth T M
6ff397eccc imagefreeze: replace with gst_buffer_copy
gst_buffer_ref and gst_buffer_writable is being used to create a writable copy of source buffer.

replacing the same with gst_buffer_copy as the functionality is same.

https://bugzilla.gnome.org/show_bug.cgi?id=735880
2014-09-03 21:33:09 -03:00
Tim-Philipp Müller
884f81ba28 qtdemux: mark jpeg and png as parsed so avdec_mjpeg can be used too
https://bugzilla.gnome.org/show_bug.cgi?id=735971
2014-09-03 23:08:16 +01:00
Jan Schmidt
9375e90203 qtdemux: Silence some warnings for normal file contents 2014-09-03 23:47:49 +10:00
Nicolas Huet
15894c1853 aacparse: Fix parsing issue when the buffer does not have a complete ADTS/LOAS frame
https://bugzilla.gnome.org/show_bug.cgi?id=735520
2014-09-02 09:43:14 +03:00
Vineeth T M
3a1e010221 imagefreeze: Don't call gst_caps_unref() on template caps when already unreferenced
Adding an extra condition while calling gst_caps_unref (templ)
and replacing gst_caps_make_writable (gst_caps_ref (caps)) with
gst_caps_copy (caps) in line 177, since the functionality is same.

https://bugzilla.gnome.org/show_bug.cgi?id=735795
2014-09-01 14:34:43 +03:00
Sebastian Dröge
f5df8af59e wavparse: Store size of data tag in a 64 bit integer locally too
Otherwise we will clip the DS64 value of RF64 files to 32 bits again.
2014-08-29 11:55:26 +03:00
Sebastian Dröge
d924f8a955 wavparse: Use 64 bit scaling functions now that fact is a 64 bit integer 2014-08-29 11:53:23 +03:00
Peter G. Baum
5c838af300 wavparse: support rf64 format
https://bugzilla.gnome.org/show_bug.cgi?id=735627
2014-08-29 11:49:42 +03:00
Jason Litzinger
bcbdcbf638 multipartdemux: Ensure caps before pad added.
This stores the stream-start, sets caps, and then adds the pad,
which ensures that the caps are set for the "pad-added" callback.

https://bugzilla.gnome.org/show_bug.cgi?id=735626
2014-08-29 11:38:19 +03:00
Nicolas Dufresne
356defdfea flvmux: Fallback to PTS if DTS is missing
Fixing a regression introduce when fixing:
https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-28 15:05:56 -04:00
Vineeth T M
d46631c5c7 imagefreeze: Remove impossible error condition
We return EOS after the first buffer, and GstPad will make sure now that we
won't get any other buffer afterwards until a flush happens. No need to check
for it ourselves.

https://bugzilla.gnome.org/show_bug.cgi?id=735581
2014-08-28 14:55:00 +03:00
Nicolas Dufresne
a7a3cb343a flvmux: Correctly offset timestamp
The previous method would break AV sync in the case audio or video
didn't start at the same point in running time.

https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-27 21:09:57 -04:00
Nicolas Dufresne
aa5bd99127 flvmux: Save dts from buffer
We no longer set dts in muxed buffer. This would lead to encoding tags
with timestamp 0 instead of the timestamp of previous buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-27 21:08:21 -04:00
Nicolas Dufresne
c1e7bec616 flvmux: Ensure Timestamp starts at 0
FLV documentation stipulates that timestamp must start at zero.
In order to respect this rule, keep the first timestamp around
and offset the timestamp from this value. This allow for longer
recording time in presence of timestamp that does not start
at 0 already.

https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-26 16:46:03 -04:00
Nicolas Dufresne
ff2bce7b26 flv: Tag timestamp are DTS not PTS
The tags in FLV are DTS. In audio cases, and for many video format this makes
no difference, but for AVC with B-Frames, PTS need to be computed from
composition timestamp CTS, with PTS = DTS + CTS.

https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-26 16:45:59 -04:00
Youness Alaoui
a98341397d jitterbuffer: Allow rtp caps without clock-rate
The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.

https://bugzilla.gnome.org/show_bug.cgi?id=734322
2014-08-21 18:32:58 -04:00
Thiago Santos
fa103ca5ad qtdemux: avoid crashing on dash streams
DASH/fragmented moov might have no samples as those are carried
in moof fragments. Avoid crashing or failing the stream because
of that.
2014-08-18 14:05:52 -03:00
Víctor Manuel Jáquez Leal
419332e287 udp: fix udpsrc documentation
udpsrc gtk-doc documentation refers to sockfd and closefd properties which has
been removed. This patch replaces those references to socket and close-socket
respectively.

https://bugzilla.gnome.org/show_bug.cgi?id=734987
2014-08-18 11:01:31 +01:00
Jan Schmidt
6e7930a10c qtmux: Make the default timescale 1/1800 second
The old default timescale of 1 millisecond produces irrational
numbers for a lot of framerate/audio-packet-duration multiples.
1/1800 is a nicer number, as it tends to produce better fractions
and therefore slightly higher accuracy overall
2014-08-15 13:03:52 +10:00
Jan Schmidt
f1c3a40547 matroska: Use gst_video_guess_framerate() function
Remove local framerate guessing function in favour of
the new gst_video_guess_framerate() function.
2014-08-15 01:17:27 +10:00
Jan Schmidt
ca068865c3 qtdemux: Improve framerate calculation/guessing
Change the way the output framerate is calculated
to ignore the first sample (which is sometimes truncated
in my testing) and use the new gst_video_guess_framerate()
function to recognise common standard framerates better.

Remove the code that was sorting the first 20 sample
durations and then ignoring the result.
2014-08-15 01:12:20 +10:00
Sebastian Dröge
ce1d4d9f21 videomixer: Use the best width/height/etc if downstream can handle that
Before it was always using whatever downstream preferred, while
the code and documentation claimed something different.

https://bugzilla.gnome.org/show_bug.cgi?id=727180
2014-08-14 16:36:44 +03:00
Ravi Kiran K N
61fe02a018 videomixer: Avoid double free of VideoConvert
https://bugzilla.gnome.org/show_bug.cgi?id=734764
2014-08-14 15:31:48 +03:00
Tim-Philipp Müller
6ee2665b7c flvdemux: fix indentation 2014-08-13 11:59:39 +01:00
Tim-Philipp Müller
9afeb9652b flvdemux: un-break duration querying
Commit 2b9493b5 broke this in two ways: a) we should only
pass duration queries in TIME format upstream (or at least
not those in DEFAULT or BYTE format), and b) we mustn't
overwrite the default value of 'res' from TRUE to FALSE
and not set it again later. This led to bogus durations
being reported for FLV playback from file, because TIME
queries would fail (as 'res' had been set to FALSE) and
parsers then do a BYTE query as fallback and try to
guesstimate something in return, which of course goes
horribly wrong since the BYTE size returned is for the
muxed file.
2014-08-13 11:59:39 +01:00
Sebastian Dröge
0911307d7d videobalance: Allow any raw caps in passthrough mode, not just the ones we handle 2014-08-13 13:25:36 +03:00
Sebastian Dröge
a9eda81978 videobalance: Allow ANY capsfeatures, but only in passthrough mode
When changing the properties to not be in passthrough mode anymore,
we will only accept caps we can process ourselves, potentially causing
a not-negotiated error.

https://bugzilla.gnome.org/show_bug.cgi?id=720345
2014-08-13 13:24:38 +03:00
George Kiagiadakis
9dd48c503c qtdemux: forward DISCONT from upstream to the output streams
This makes sense in DASH reverse playback, where the upstream dashdemux
will download DASH segments in reverse order, but push their buffers
forward to qtdemux and mark each segment start as DISCONT. This needs
to be forwarded downstream to the parser/decoder, otherwise it won't work.

https://bugzilla.gnome.org/show_bug.cgi?id=734443
2014-08-11 10:28:14 +02:00
Sebastian Rasmussen
70a43758bb shapewipe: Unref caps and element after usage
https://bugzilla.gnome.org/show_bug.cgi?id=734478
2014-08-10 11:09:09 +01:00
Tim-Philipp Müller
e8321af983 qtdemux: improve debug logging of fourccs
If we can't show ASCII, at least show them
in big endian order.
2014-08-09 20:50:01 +01:00
Tim-Philipp Müller
f41d03cd4d qtdemux: add support for 'wma ' mapping as found in some ismv files
e.g. To_The_Limit_720_2962.ismv
2014-08-09 20:49:53 +01:00
Tim-Philipp Müller
6183f83190 qtdemux: add support for 'vc-1' mapping as found in some ismv files
e.g. To_The_Limit_720_2962.ismv
2014-08-09 20:49:49 +01:00
Sebastian Rasmussen
276269d956 rtph263ppay: Unref pad template caps after use
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734435
2014-08-08 16:02:24 -03:00
Sebastian Rasmussen
1fa61632fe videomixer: Unref allowed caps after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734474
2014-08-08 15:59:36 -03:00
Sebastian Rasmussen
c85ae43a6e imagefreeze: Unref pad template caps after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734475
2014-08-08 15:54:39 -03:00
Sebastian Rasmussen
edf8728016 navseek: Unref peer pad after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734476
2014-08-08 15:50:55 -03:00
Sebastian Rasmussen
1a35bf9647 rtpmux: Unref pad template caps after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734473
2014-08-08 15:38:32 -03:00
Srimanta Panda
421b00cd17 rtph264pay: append packetization mode parameter to SDP
Append packetization-mode parameter to SDP description.
Packetization mode signals the properties of an RTP payload type.

https://bugzilla.gnome.org/show_bug.cgi?id=733556
2014-08-08 13:41:36 +01:00
Jan Schmidt
d9e1aa4959 isomp4/qtmux: Write correct file duration when gaps exist.
When writing out a trak with an edit list, make sure the
overall file duration is also updated to reflect the
lengthening of the stream.

Add some more debug to qtdemux to warn about streams that
are longer than the file and get truncated.
2014-08-08 04:01:19 +10:00
Sebastian Dröge
add40de469 rtspsrc: Push the correct segment in TCP mode when seeking 2014-08-05 16:28:04 +02:00
Mark Nauwelaerts
d5d28055c1 rtph264pay: unbreak au aligned byte-stream payloading 2014-08-03 14:42:45 +02:00
Srimanta Panda
dd9f716892 rtph264pay: append profile-level-id to SDP
Append profile-level-id to SDP if available.

https://bugzilla.gnome.org/show_bug.cgi?id=733539
2014-08-01 16:01:07 +01:00
Philippe Normand
b8b5704445 interleave: set output caps layout to interleaved
Set output caps layout independently from input caps layout which can
be either non-interleaved or interleaved.

https://bugzilla.gnome.org/show_bug.cgi?id=733866
2014-07-29 11:49:32 +02:00
Tim-Philipp Müller
5122410f11 qtdemux: fix language code parsing for 3-letter codes starting with 'a'
And handle special value for 'unspecified' explicitly.

https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFChap4/qtff4.html
2014-07-21 18:21:50 +01:00
Sebastian Dröge
b1f7681555 videobox: Don't overwrite the first component with the alpha value for BGRx
Instead leave the x component unset when filling the borders.

https://bugzilla.gnome.org/show_bug.cgi?id=733380
2014-07-19 11:31:45 +02:00
Sebastian Dröge
638a700463 aacparse: Properly report in the CAPS query that we can convert ADTS<->RAW
https://bugzilla.gnome.org/show_bug.cgi?id=733190
2014-07-16 17:27:57 +02:00
Sebastian Rasmussen
f45657f604 rgvolume: Avoid taking unnecessary refs
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122
2014-07-16 16:45:43 +02:00
Sebastian Rasmussen
ca22ad8da9 rtpdtmfmux: Avoid taking an unnecessary ref
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122
2014-07-16 16:45:31 +02:00
Tim-Philipp Müller
c2614e5253 rtspsrc: fix query leak
https://bugzilla.gnome.org/show_bug.cgi?id=733003
2014-07-10 17:19:42 +01:00
Sebastian Dröge
dd5144fd4e wavenc: Return not-negotiated if we got no caps or caps negotiation failed
And do it always, not inside a g_return_val_if_fail().

See https://bugzilla.gnome.org/show_bug.cgi?id=732939
2014-07-10 14:37:31 +02:00
Tim-Philipp Müller
deeef84d2c videomixer: fix double unlock in segment seek segment code path
We only want to unlock if we push an event downstream and
jump to done_unlock label afterwards. We would also unlock
in case of a segment seek and then unlock again later, and
nothing good can come of that.

(This code looks a bit dodgy anyway though, shouldn't it
also bail out with FLOW_EOS here in case of a segment seek
scenario, just without the event?)
2014-07-04 20:26:46 +01:00
Sebastian Rasmussen
d33d8cf026 avidemux, wavparse: Print invalid fourcc in hex
Previously this was printed as characters which caused later processing
of the error message to sometimes warn about non-UTF-8 characters.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732714
2014-07-04 09:21:07 +01:00
Wim Taymans
db1d9444d6 rtspsrc: fix for mikey api change 2014-07-02 16:01:47 +02:00
Vincent Penquerc'h
bbb1a8de1f videomixer: reset QoS on segment event
https://bugzilla.gnome.org/show_bug.cgi?id=732540
2014-07-01 16:35:05 +01:00
Vincent Penquerc'h
5653b1a25a matroskademux: send gap events instead of segment tricks
This fixes missing frames from being time skipped.

https://bugzilla.gnome.org/show_bug.cgi?id=732372
2014-07-01 15:14:34 +01:00
Sebastian Dröge
2f47105129 rtpbin: Don't leak caps 2014-06-29 23:55:19 +02:00
Sebastian Dröge
bbca040336 rtpssrcdemux: Fix compiler warning when compiling with G_DISABLE_ASSERT 2014-06-29 19:59:53 +02:00
Sebastian Dröge
5500dd4a20 matroskamux: Fix compiler warnings when compiling with G_DISABLE_ASSERT 2014-06-29 19:57:57 +02:00
Sebastian Dröge
b03a4d9155 deinterlace: Fix compiler warnings when compiling with G_DISABLE_ASSERT 2014-06-29 19:54:44 +02:00
Tim-Philipp Müller
155a3fec93 matroskaparse: don't error out if there's not enough data in the adapter
gst_matroska_parse_take() would return FLOW_ERROR instead of
FLOW_EOS in case there's less data in the adapter than requested,
because buffer is NULL in that case which triggers the error
code path. This made the unit test fail (occasionally at least,
because of a bug in the unit test there's a race and it would
happen only sporadically).
2014-06-28 17:39:36 +01:00
Sebastian Dröge
c0f5644b80 videomixer: Update dist generated ORC files 2014-06-28 16:56:18 +02:00
Sebastian Dröge
db43a39bbf videomixer: Update videoconvert code from -base
And also rename the remaining symbols to prevent conflicts
during static linking.

https://bugzilla.gnome.org/show_bug.cgi?id=728443
2014-06-28 16:56:18 +02:00
Tim-Philipp Müller
8b7f0ae3fe autovideosrc: use videotestsrc as fallback element instead of fakesrc
fakesrc doesn't announce video caps, so most video pipelines will
just error out with not-negotiated if a fallback element is created.
2014-06-28 14:25:25 +01:00
Tim-Philipp Müller
7dcc3ffe5a autoaudiosrc: use audiotestsrc as fallback element instead of fakesrc
fakesrc doesn't announce audio caps, so most audio pipelines will
just error out with not-negotiated if a fallback element is created.
2014-06-28 14:25:25 +01:00
Thibault Saunier
45b9ef1825 videomixer: Declare as Compositor in 'klass' 2014-06-26 17:49:23 +02:00
Tim-Philipp Müller
e9f2d63011 flvdemux: fix speex caps
Decoder complains about "notification: Invalid mode encountered.
The stream is corrupted" though, even if it works, so there's
probably something wrong with the generated codec headers.
2014-06-26 13:50:19 +01:00
Tim-Philipp Müller
d98b996523 flvmux: fix speex in FLV
Speex in FLV is always mono @ 16kHz, see
http://download.macromedia.com/f4v/video_file_format_spec_v10_1.pdf
section E.4.2.1: "If the SoundFormat indicates Speex, the audio is
compressed mono sampled at 16 kHz, the SoundRate shall be 0, the
SoundSize shall be 1, and the SoundType shall be 0"

Also see https://bugzilla.gnome.org/show_bug.cgi?id=683622
2014-06-26 13:43:33 +01:00
Jan Schmidt
8da6ee0312 isomp4: Add object type id and fourcc for DTS/DTS-HD
Enables playback for files with DTS audio tracks.
Also add an extra AC-3 variant fourcc from Nero
2014-06-26 19:57:41 +10:00
David Fernandez
4ed74d3ab0 videomixer2: Solve segmentation fault when src caps are configured
Change function pointers to NULL while holding the lock to avoid
race conditions

https://bugzilla.gnome.org/show_bug.cgi?id=701110
2014-06-25 16:44:38 +02:00
Wim Taymans
ca9cfd40dd jitterbuffer: improve SR packet handling
Implement 3 different cases for handling the SR:

 1) we don't have enough timing information to handle the SR packet and
    we need to wait a little for more RTP packets. In that case we keep
    the SR packet around and retry when we get an RTP packet in the
    chain function.

 2) the SR packet has a too old timestamp and should be discarded. It is
    labeled invalid and the last_sr is cleared.

 3) the SR packet is ok and there is enough timing information, proceed
    with processing the SR packet.

Before this patch, case 2) and 1) were handled in the same way,
resulting that SR packets with too old timestamps were checked over and
over again for each RTP packet.
2014-06-25 16:14:46 +02:00
Olivier Crête
64f28e2552 avimux: Add UYVY format 2014-06-23 19:55:29 -04:00
Miguel París Díaz
b22aed9bbc gstrtpssrcdemux: manage ssrc of RTCP RR packets
https://bugzilla.gnome.org/show_bug.cgi?id=731324
2014-06-23 16:23:00 -04:00
Sebastian Dröge
efaf996b1a wavparse: Update offset after parsing adtl chunk
Otherwise we will parse it over and over again without ever
getting past it.

https://bugzilla.gnome.org/show_bug.cgi?id=731533
2014-06-23 20:53:50 +02:00
Sebastian Dröge
daf25482ed matroskademux: Don't call GST_DEBUG_OBJECT() and other macros with non-GObject objects
It will crash with latest GLib GIT and was never supposed to work before
either.
2014-06-22 19:26:03 +02:00
Tim-Philipp Müller
41c895de4d multiudpsink: optimisation: avoid unnecessary memory ref/unrefs
We know the buffer will stay valid and we will also not
modify the buffer, we just want to send out the data.
2014-06-20 12:21:05 +01:00
Tim-Philipp Müller
3512ad3be0 multiudpsink: avoid some unnecessary run-time type checks 2014-06-20 12:06:57 +01:00
Wim Taymans
98a4ee0f92 rtspsrc: pass the stream id when asking for crypto params
This way the app can choose different parameters for each stream.
2014-06-19 16:17:23 +02:00
Aleix Conchillo Flaqué
7ce0ea3946 rtspsrc: add support for key length parameters
This patch adds supports for the incoming key management parameters for
encryption and authentication key lengths.

It also adds a new signal request-rtcp-key that allows the user to
provide the crypto parameters and key for the RTCP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=730473
2014-06-19 16:11:19 +02:00
Wim Taymans
8a78fa1ff5 vp8depay: fix header size checking
Use a different variable name to make it clear that we are calculating
the header size.
Correctly check that we have enough bytes to read the header bits. We
were checking if there were 5 bytes available in the header while we
only needed 3, causing the packet to be discarded as too small.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723595
2014-06-19 15:29:46 +02:00
Guillaume Desmottes
f00c2b7155 rtph264pay: propagate the GST_BUFFER_FLAG_DISCONT flag
Similarly to what we did with the DELTA_UNIT flag, this patch
propagates the DISCONT flag to the first RTP packet being used to transfer a
DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-19 12:22:49 +02:00
Guillaume Desmottes
4be99ec7d5 rtph264pay: propagate the GST_BUFFER_FLAG_DELTA_UNIT flag
Downstream elements may be interested knowing if a RTP packet is the start
of a key frame (to implement a RTP extension as defined in the
ONVIF Streaming Spec for example).

We do this by checking the GST_BUFFER_FLAG_DELTA_UNIT flag we receive from
upstream and propagate it to the *first* RTP packet outputted to transfer this
buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-19 12:22:38 +02:00
Guillaume Desmottes
42ff642372 gstrtpmp4gpay: propagate the GST_BUFFER_FLAG_DISCONT flag
Propagate the DISCONT flag to the first RTP packet being used to transfer
a DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-18 16:25:07 +02:00
Guillaume Desmottes
9a7479fb0d rtpjpegpay: propagate the GST_BUFFER_FLAG_DISCONT flag
Propagate the DISCONT flag to the first RTP packet being used to transfer
a DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-18 16:25:07 +02:00
Tim-Philipp Müller
460ab3dd76 avidemux: don't leak flow combiner 2014-06-18 15:03:25 +01:00
Tim-Philipp Müller
6347ec522d rtpjp2kpay: pre-allocate buffer-list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
ccb7380689 rtpjpegpay: pre-allocate buffer list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
70bfc35756 rtpmp4vpay: pre-allocate buffer list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
4b1f771e4d rtpvp8pay: allocate bitreader on the stack 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
725b8f272b rtpvp8pay: post error message on bus on error and don't use g_message() 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
f4db7443ae rtpvp8pay: couple of minor optimisations
Pre-allocate buffer list of the right size to avoid re-allocs.
Avoid plenty of double runtime cast checks and re-doing the
same calculation over and over again in rtp_vp8_calc_payload_len().
Only call gst_buffer_get_size() once.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
6c9e2194d2 rtpgstpay: pre-allocate buffer list of the right size
To avoid re-allocs.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
01ee993d8d rtph264pay: pre-allocate bufferlist of the right size
To avoid unnecessary re-allocs.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
c7c72c00b1 rtph264pay: push single buffer directly, no need to wrap it in a bufferlist
No point in a buffer list if we just have one single
buffer to push. Fix up unit test to handle that case
as well.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
0f5da64de3 rtpvrawpay: make chunks per frame configurable
Bit of a misnomer because it's really chunks per field
and not per frame, but we're going to ignore that for
the time being.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
2cf13b603f rtpvrawpay: remove unused variables 2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
a09e237b85 rtpvrawpay: pre-allocate buffer lists of sufficient size
Avoids unnecessary reallocs when appending buffers
to the bufferlist.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
15a33ccc65 rtpvrawpay: micro-optimise variable access in inner loop
Store some values that don't change during the execution
of the inner loops locally, so the compiler knows that too.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
fdf95fecbd rtpvrawpay: use buffer lists
Collect buffers to send out in buffer lists instead of
pushing out single buffers one at a time. For HD video
each frame might easily add up to a couple of thousand
packets, multiply that by the frame rate and that's a
lot of push() and sendmsg() calls per second.

A good reason to push out buffers as early as possible is
latency, so we don't accumulate the whole frame in a single
buffer list, but instead push it out in a few chunks, which
is hopefully a reasonable compromise.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
884d1af074 udp: improve element descriptions for dynudpsink and multiudpsink 2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
6c1231eed3 udp: remove suppression of compiler warnings for deprecated GLib API
Not needed any more.
2014-06-18 14:54:58 +01:00
Ravi Kiran K N
3c4c130c5e videobox: Fix caps negotiation issue
Make sure that if AYUV is received it will detect that it can produce
both RGB and YUV formats

Signed-off-by: Ravi Kiran K N <ravi.kiran@samsung.com>

https://bugzilla.gnome.org/show_bug.cgi?id=725248
2014-06-17 09:27:45 -03:00
Tim-Philipp Müller
054f774455 rtptheoradepay: fix double frees
Fix double-frees introduced to fix another coverity report.

CID 1223053
2014-06-16 12:03:38 +01:00
Tim-Philipp Müller
bb51ec5842 dynudpsink: return FLUSHING when sendto got canceled, not an error 2014-06-13 10:12:07 +01:00
Vincent Penquerc'h
25c26a4c4c rtptheordepay: fix leaks
Coverity 1212163
2014-06-12 11:24:15 +01:00
Vincent Penquerc'h
8e80478cf7 rtpg729pay: leak fixes
Coverity 1212159
2014-06-12 11:16:08 +01:00
Vincent Penquerc'h
fe4c5b92b1 rtph263pay: fix leak
Coverity 1212157
2014-06-12 11:11:38 +01:00
Vincent Penquerc'h
6ef26e4a8a rtph263pay: fix leaks
Coverity 1212149
2014-06-12 10:43:53 +01:00
Vincent Penquerc'h
c58a2d9bbb rtpdvpay: catch failures to map buffer
Coverity 1139741
2014-06-12 10:31:47 +01:00
Vincent Penquerc'h
7e278e6b22 multipartdemux: guard against having no MIME type
The code would previously crash trying to insert a NULL string
into a hash table.
It does seem a little broken that indexing is done by MIME type
and not by index though, unless the spec says there cannot be
two parts with the same MIME type.

https://bugzilla.gnome.org/show_bug.cgi?id=659573
2014-06-11 17:44:56 +01:00
Nicolas Dufresne
9966fdfa75 multipartdemux: Send stream-start event
This event was not sent. Send it before caps, this requires the pad to
be parented. This removes warning like: "Got data flow before
stream-start event".

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731475
2014-06-10 15:43:21 -04:00
Thiago Santos
9fda7b107f qtdemux: avoid looping indefinitely in broken svq3 files
Abort if an atom with size 0 is read from within the svq3 stsd
atoms

https://bugzilla.gnome.org/show_bug.cgi?id=726512
2014-06-10 15:33:33 -03:00
Edward Hervey
f7fc8d74c9 flvdemux: Attempt upstream seek first
If we have an upstream element that can handle the seek (such as
rtmpsrc), try to do that first before attempting it ourself.
2014-06-09 10:04:38 +02:00
Vincent Penquerc'h
40ae581ef2 wavparse: do not include codec_data on raw audio caps
If the wav header contains an extended chunk, we want to keep
the codec_data field, but not for raw audio.

This fixes some elements (such as adder) from failing to intersect
raw audio caps which would otherwise be intersectable.
2014-06-05 10:34:49 +01:00
Edward Hervey
2b9493b5f0 flvdemux: Query duration upstream first
Upstream elements (like rtmpsrc) might be able to provide the duration
more accurately than flvdemux. Especially with index-less vod files
2014-06-05 09:38:29 +02:00
Jan Alexander Steffens (heftig)
303883752e flvdemux: set RESYNC buffer flag when bridging large PTS gaps
So downstream gets notified when this happens.

https://bugzilla.gnome.org/show_bug.cgi?id=725903
2014-06-04 10:28:47 -04:00
Tim-Philipp Müller
341b691b18 matroskademux: don't leak doctype string in error code path
CID 1212145.
2014-06-02 09:57:42 +02:00
Thiago Santos
c25d94b7ef qtdemux: upstream handles seek if fragmented and on time segment
Otherwise we can reject seeks on local files that contain fragmented-like
atoms like 'mvex'. Also improve a message log

https://bugzilla.gnome.org/show_bug.cgi?id=730722
2014-05-30 15:01:50 -03:00
Wim Taymans
a5a7649831 h264depay: 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.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730999
2014-05-30 16:51:37 +02:00
Thiago Santos
fd6b348898 avidemux: remove stream last flow return
GstPad already stores that information

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:51:13 -03:00
Thiago Santos
2b454bf87f qtdemux: remove last flow return from stream struct
It is already stored on GstPad on core

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:51:12 -03:00
Thiago Santos
3b887887be flvdemux: Use GstFlowCombiner
Use the flow combiner to have the standard combination results and avoid
repeating the same code

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:51:07 -03:00
Thiago Santos
c7c25071e3 matroskademux: use GstFlowCombiner
Use the flow combiner to have the standard combination results and avoid
repeating the same code

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:51:02 -03:00
Thiago Santos
da3c031627 avidemux: use GstFlowCombiner
Removes flow return combination code to use the newly added GstFlowCombiner
2014-05-26 15:30:12 -03:00
Thiago Santos
4b0ce7dc30 qtdemux: use GstFlowCombiner
Removes the common code to combining flow returns to let it be
handled by core gstutils' GstFlowCombiner

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 15:30:12 -03:00
Thiago Santos
d423b9f63e qtdemux: parse tkhd transformation matrix and add tags if appropriate
Handle the transformation matrix cases where there are only simple rotations
(90, 180 or 270 degrees) and use a tag for those cases. This is a common scenario
when recording with mobile devices

https://bugzilla.gnome.org/show_bug.cgi?id=679522
2014-05-24 15:38:54 -04:00
Thiago Santos
f0b99d96a9 qtdemux: add tag mappings for _swr, _mak and _mod tags
swr -> Application name
mak -> device manufacturer
mod -> device model
2014-05-23 03:15:42 -03:00
Sebastian Dröge
1cdd3765d6 goom: Use fabs() instead of abs() to calculate the floating point absolute value
tentacle3d.c:268:7: error: using integer absolute value function 'abs' when
      argument is of floating point type [-Werror,-Wabsolute-value]
  if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot + 2.0 * G_PI))) {
      ^
2014-05-19 11:24:06 +02:00
Sebastian Dröge
97fb3655df debugutils: Properly calculate the difference with unsigned types
tests.c:161:16: error: taking the absolute value of unsigned type
      'unsigned long' has no effect [-Werror,-Wabsolute-value]
    t->diff += labs (GST_BUFFER_TIMESTAMP (buffer) - t->expected);
2014-05-19 11:21:36 +02:00
Aleix Conchillo Flaqué
782d65cab1 rtspsrc: always use a random ssrc for the internal session
Use a random SSRC different than 0 for the internal session SSRC.

https://bugzilla.gnome.org/show_bug.cgi?id=730212
2014-05-16 16:58:44 +02:00
Wim Taymans
d004eda79d rtpsession: update last_activity when sending RTP
Also update last_activity when doing something with the internal
source to make sure don't timeout early.

See https://bugzilla.gnome.org/show_bug.cgi?id=730217
2014-05-16 16:55:17 +02:00
Aleix Conchillo Flaqué
a62b280873 rtpbin: update rtp encoder/decoder docs
Use %u in RTP encoder/decoder pads to match other rtpbin pads.

https://bugzilla.gnome.org/show_bug.cgi?id=730146
2014-05-15 15:48:21 +02:00
George Kiagiadakis
7e2138794f rtpsession: remove unused if branch
1) sources that have sent BYE in the past cannot be senders, since
they would have timed out to being receivers in the meantime...
2) sources that have sent BYE are now being removed earlier inside
this function
2014-05-14 16:01:50 +02:00
George Kiagiadakis
85d4c031d4 rtpsession: cleanup sources that have sent BYE 2014-05-14 16:01:50 +02:00
George Kiagiadakis
7d7840cc4a rtpsession: unify nested if clauses 2014-05-14 16:01:50 +02:00
George Kiagiadakis
0e6a31411b rtpsession: timeout internal sources that are inactive for a long time and send BYE 2014-05-14 16:01:50 +02:00
Aleix Conchillo Flaqué
bcd469ff31 rtpjitterbuffer: don't stop looping if event found in the queue
If we are inserting a packet into the jitter queue we need to keep
looping through the items until the right position is found. Currently,
the code stops as soon as an event is found in the queue.

Regarding events, we should only move packets before an event if there
is another packet before the event that has a larger seqnum.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730078
2014-05-14 10:23:28 +02:00
Adrien SCH
8ac30d4c26 matroskamux: fix the memory leak of language attribute
https://bugzilla.gnome.org/show_bug.cgi?id=728418
2014-05-13 19:55:21 -03:00
Edward Hervey
420661bd95 qtdemux: Fix leak of palette_data in error cases
CID #1212151
2014-05-12 16:56:35 +02:00
Edward Hervey
112d948b7e qtmux: Free node_header in error cases
CID #1212134
2014-05-12 16:53:32 +02:00
Edward Hervey
6c4882996f flvdemux: Don't use WARNING for not-linked flow return
Pollutes debug logs for no reason. It's only an error if all pads
return not-linked
2014-05-12 13:46:01 +02:00
Edward Hervey
c09b14c931 flvdemux: Skip unknown tags in push-mode
We add a new mode (SKIP) in push-mode to skip tags that we don't known about

Partially fixes https://bugzilla.gnome.org/show_bug.cgi?id=670712
2014-05-12 13:45:06 +02:00
Wim Taymans
b2e1598e4a rtpjitterbuffer: increment accepted packets after loss
When we detect a lost packet, expect packets with higher
seqnum on the input.

Also update the unit test.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729524
2014-05-09 18:10:32 +02:00
Jason Litzinger
9068e1bb8e Add new test case. 2014-05-09 18:10:32 +02:00
Eric Trousset
bd51aa7aa8 qtdemux: don't respond to a position query in BYTE format with a TIME position
https://bugzilla.gnome.org/show_bug.cgi?id=729553
2014-05-09 16:12:45 +01:00
Tim-Philipp Müller
9872c19491 matroskademux: don't leak doctype string in error code path
CID 1212145.
2014-05-09 14:22:42 +01:00
Tim-Philipp Müller
615f6e55c1 flacparse: skip PICTURE headers without any image data
Fixes warning if the image length is 0.
2014-05-07 00:58:15 +01:00
Guillaume Desmottes
d089f99a39 rtp/README: update pipelines to work with 1.0
- Use gst-libav encoders/decoders instead of gst-ffmpeg
- gstrtpjitterbuffer -> rtpjitterbuffer
- gst-launch-0.10 -> gst-launch-1.0
- Add 'videoconvert' element
- xvimagesink -> autovideosink

https://bugzilla.gnome.org/show_bug.cgi?id=729247
2014-05-05 20:23:56 -04:00
Vincent Penquerc'h
ec38c62563 matroska: rejig test to avoid undefined shift behavior
Coverity 1195121, 1195120
2014-05-05 14:44:57 +01:00
Vincent Penquerc'h
9589c43516 matroskamux: ensure we don't dereference a NULL pointer
while working out the codec ID.

Coverity 1195148
2014-05-05 14:32:06 +01:00
Olivier Crête
b2a52035bf rtprtxreceive: Wait until timeout to clear association requests
If two streams request a retranmission for the same SSRC, ignore the second
one if the first oen is less than one second old, otherwise time out the first
one and ignore the second.
2014-05-04 22:36:59 -04:00
Olivier Crête
0742a5a257 rtpmux: Always let upstream chose the ssrc if it wishes 2014-05-04 19:11:03 -04:00
Mark Nauwelaerts
6c584bc833 rtpjitterbuffer: avoid stall by corrupted seqnum accounting 2014-05-04 13:38:26 +02:00
Olivier Crête
2e54d38dd0 rtpsession: Keep local conflicting addresses in the session
As we now replace the local RTPSource on a conflict, it's no longer possible
to keep local conflicts in the RTPSource, so they instead need to be kept
in the RTPSession.

Also fix the rtpcollision test to generate multiple collisions instead of
one by change the address, as otherwise we detected that it was a single one.
2014-05-03 18:30:20 -04:00
Sebastian Dröge
1d4404d883 Release 1.3.1 2014-05-03 18:02:23 +02:00
Sebastian Dröge
859c751a5d imagefreeze: Set segment position to the stop position of the buffer 2014-05-02 17:14:29 +02:00
Sebastian Dröge
4282d75597 imagefreeze: Properly report errors before stopping the srcpad task 2014-05-02 17:14:29 +02:00
Sebastian Dröge
4933394d35 imagefreeze: Error out if we have no caps yet 2014-05-02 17:14:29 +02:00
Vincent Penquerc'h
218294b9f3 wavparse: avoid dividing by a 0 blockalign
This can be 0. In that case, do not try to cut off the last few
bytes from the last buffer.

Coverity 1146971
2014-05-02 14:49:27 +01:00
Vincent Penquerc'h
590e20cbc9 matroskamux: do not use uinitialized clut on error
If we're missing part of the clut, do not try to use it. It seems
very likely the break was meant to break out of the switch rather
than from the loop.

Coverity 1139878
2014-05-02 14:25:01 +01:00
Vincent Penquerc'h
d917c94037 flxdec: fix integer overflow
Coverity 1139859
2014-05-02 14:18:08 +01:00
Vincent Penquerc'h
60ba2d7aee rtpqdmdepay: remove pointless check
Besides, the pointer was dereferenced earlier anyway.

Coverity 1139853
2014-05-02 14:09:02 +01:00
Vincent Penquerc'h
a846e84349 rtspsrc: remove duplicate test
item was dereference previously.

While there, reorder some test for faster early out.

Coverity 1139844
2014-05-02 14:06:25 +01:00
Vincent Penquerc'h
5c22bcf6e9 matroska: blindly fix writing variable length negative values
Spotted while fixing something else in the area.

Nothing calls this with a negative value.
2014-05-02 13:33:02 +01:00
Vincent Penquerc'h
5b9fa4e63a matroska: do not lose the top bits when writing a > 32 bit value
Coverity 1139806
2014-05-02 13:29:33 +01:00
Vincent Penquerc'h
10663decd9 videoflip: add missing break in switch
Coverity 1139755
2014-05-02 12:10:26 +01:00
Vincent Penquerc'h
a0bc24558e matroska: do not try to call gst_pad_query_default on a NULL pad
gst_matroska_parse_query can be called explicitely with a NULL pad.
If we reach this point with a NULL pad, fail the query.

Coverity 1139715
2014-05-02 11:39:39 +01:00
Vincent Penquerc'h
3915884017 matroska: do not return GST_FLOW_OK if we did not get a buffer
Coverity 1139714 (which will likely come back in another guise,
as the _read_init call can have a failing _map)
2014-05-02 11:28:01 +01:00
Vincent Penquerc'h
f5a9f5e221 matroska: catch failure to map buffer
Avoids dereferencing NULL.

Coverity 1139712
2014-05-02 11:20:33 +01:00
Vincent Penquerc'h
94720fd3a1 avimux: refuse caps with invalid framerate
Coverity 1139701
2014-05-02 10:53:00 +01:00
Vincent Penquerc'h
1be86ebb2a qtmux: handle 0 size packets without dividing by 0
Coverity 1139691
2014-05-02 10:21:09 +01:00
Vincent Penquerc'h
b692539b55 qtdemux: guard against invalid frame size to avoid division by 0
Coverity 1139690
2014-05-02 09:49:32 +01:00
Vincent Penquerc'h
436c8c11a0 qtdemux: trivial typo fix 2014-05-02 09:49:17 +01:00
Vincent Penquerc'h
0253db6d36 mpegaudioparse: remove dead code
A stricer check is already done earlier, and integer overflows
do not seem possible here.

Coverity 1139675
2014-04-30 17:48:53 +01:00
Vincent Penquerc'h
a55b8e9c00 rtpvrawpay: guard against pathological "no space" condition
Even if one woul hope one pixel can fit in a MTU, ensure we do not
overwrite a buffer if this is not the case.

Spotted while looking at Coverity 1208786
2014-04-30 14:50:44 +01:00
Vincent Penquerc'h
dfa2df1c88 rtpjpegdepay: sanity check for NULL qtable
Can happen (at least in crafted stream)

Coverity 1208778
2014-04-30 11:52:10 +01:00
Tim-Philipp Müller
b1473491cf wavparse: pass on tags from upstream if there are any
Don't just ignore upstream tags from e.g. an ID3 tag before
the .wav data, pass them on downstream.

https://bugzilla.gnome.org/show_bug.cgi?id=729223
2014-04-30 01:08:41 +01:00
Wim Taymans
eba3bba524 rtpjitterbuffer: optimize timer update
When we are not doing retransmission, we just need to find the current
seqnum so we can stop when we found it.
2014-04-29 16:26:53 +02:00
Wim Taymans
b2c9646acb rtpjitterbuffer: small optimizations
Small optimizations where we can.
Add some more debug.
2014-04-29 16:21:44 +02:00
Wim Taymans
df04fcbb5d rtpjitterbuffer: signal when next_seqnum changed
Signal the pushing thread when the next_seqnum changed and we might be
able to push a buffer now.
2014-04-29 16:16:17 +02:00
Wim Taymans
3cd0e8ae88 rtpjitterbuffer: only signal event when head changed
After adding a buffer, only signal the pushing thread when the head
buffer changed or else we cause a useless wakeup.
2014-04-29 16:12:29 +02:00
Wim Taymans
18b69419fd rtpjitterbuffer: rework packet insert
Rework the packet queue so that the most common action (insert a packet
at the tail of the queue) goes very fast.

Report if a packet was inserted at the head instead of the tail so that
we can know when to retry _pop or _peek.
2014-04-29 16:02:37 +02:00
Wim Taymans
9994ff2c6c rtpvraw: use plane pointers when needed
Pack/unpack planar formats to/from the first plane.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729058
2014-04-28 14:45:57 +02:00
Nicolas Dufresne
d87cc7bacf goom: Remove french comment saying to prefix functions
All non-static function in this file are already prefixed with goom_.
2014-04-27 21:57:38 -04:00
Tim-Philipp Müller
02436f52c6 goom: fix compilation on ios-arm7-10.9 and osx-x86_64
uint is not a standard type, and the rest of the code uses
Uint which is locally typedefed to unsigned int.

https://bugzilla.gnome.org/show_bug.cgi?id=729067
2014-04-28 00:24:16 +01:00
Luis de Bethencourt
3943c3ec08 goom: fix undefined behaviour of left-shift
Don't left-shift into the sign bit, the result is undefined and potentially
an overflow could flip the sign.
2014-04-27 18:31:48 -04:00
Luis de Bethencourt
5dc2e6bef1 qtdemux: check return from qt_demux_video_caps
Now qtdemux_video_caps() can return NULL. We need to check this return before
using it's value.

https://bugzilla.gnome.org/show_bug.cgi?id=728987
2014-04-26 20:51:36 -04:00
Tim-Philipp Müller
c9597298f9 docs: remove outdated and pointless 'Last reviewed' lines from docs
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
2014-04-26 23:35:17 +01:00
Luis de Bethencourt
c073a6c779 qtdemux: initialize caps pointer to null
Make sure the caps pointer returns initialized when using it in
qtdemux_parse_tree ().

https://bugzilla.gnome.org/show_bug.cgi?id=728987
2014-04-25 18:23:23 -04:00
Jan Schmidt
f2d0ddf113 rtpjitterbuffer: Clear last_pt on flush-stop.
Otherwise, we don't recheck the buffer caps for clock-rate
properly on the next chain.
2014-04-23 18:54:16 +10:00
Sebastian Dröge
25ed0a30a4 deinterlace: Fix compiler warning
gstdeinterlace.c: In function 'gst_deinterlace_output_frame':
gstdeinterlace.c:1537:57: error: 'pattern.length' may be used uninitialized in this function [-Werror=maybe-uninitialized]

This actually is always initialized before it is used there, but
let's just silence gcc here.
2014-04-22 17:29:02 +02:00
Vincent Penquerc'h
f10c3f1a76 rtpmux: fix buffer list drop check
While porting to 0.11, the check was mistakenly made constant,
instead of testing for the return value of process_buffer_locked.

Coverity 1139663
2014-04-21 17:21:20 +01:00
Vincent Penquerc'h
d9eb5f7fde matroska: fix content encoding scope validity check
It's 3 bits, and http://matroska.org/technical/specs/index.html
says it can't be 0.

Coverity 1139660
2014-04-21 17:21:20 +01:00
Vincent Penquerc'h
54c5710adb matroskamux: fix PAR fraction sanity check
It was checking par_num twice, and never par_denum.

Coverity 1139634
2014-04-21 17:21:20 +01:00
Vincent Penquerc'h
25fa88f8aa multiidpsink: warn when setsockopt fails
This doesn't seem to be fatal, but it's good to let the user know
in the logs.

Coverity 1139630
2014-04-21 17:21:20 +01:00
Vincent Penquerc'h
e526412afa interlace: catch failure to create audio info from caps
Coverity 1139627, 1139628
2014-04-21 17:21:20 +01:00
Göran Jönsson
80967c7638 gstrtph264pay: Reset sps pps variable when state change.
Reset last_spspps and sps/pps arrays  when state transition
GST_STATE_CHANGE_PAUSED_TO_READY.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726015
2014-04-21 12:07:20 +02:00
Wim Taymans
3e11ce43b9 jitterbuffer: improve EOS handling
Make a new method to disable the jitterbuffer buffering.
Rework the update_estimated_eos() method. Calculate how much time
there is left to play. If we have less than the delay of the
jitterbuffer, we disabled buffering because we might never be able to
fill the complete jitterbuffer again.
If we receive an EOS event, disable buffering. We will drain the
buffer and eventually push the EOS event out.
When we reach the estimated NPT timeout and we didn't receive an EOS
event, make one and queue it so that it can be pushed.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
2014-04-18 14:07:31 +02:00
Wim Taymans
38a486b374 rtpsession: send reconfigure when internal-ssrc changes
When the internal-ssrc property changes, we want to send a reconfigure
upstream to make payloaders use the new suggested ssrc.
Using the internal-ssrc property to change the SSRC of a stream is not a
good idea and doesn't work when there are multiple senders, we want to
set the SSRC directly on the payloaders. Therefore, deprecate this
property.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725361
2014-04-18 10:21:27 +02:00
Wim Taymans
42cfedde7f jitterbuffer: assume a full buffer when eos
Rework the logic to make buffering messages a little, make sure we
don't make the same message multiple times.
Consider the buffer full when EOS was received.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
2014-04-18 04:27:39 +02:00
Sebastian Dröge
27cf71e209 rtprtxsend: Require clock-rate in the caps and handle no ssrc in the caps properly 2014-04-17 17:58:58 +02:00
Sebastian Dröge
897c02cace rtpjitterbuffer: Unref clock id when waiting for the clock is interrupted 2014-04-17 17:00:37 +02:00
Tim-Philipp Müller
77badda6b9 videomixer: name collectpads object based on videomixer name
Makes it easier to track things in debug logs when there
are multiple mixers and muxers.
2014-04-16 21:40:45 +01:00
Tim-Philipp Müller
f8d15b1e56 videomixer: better logging of incoming events
The pad and parent names are already logged as part of logging
the object. Instead log the full event details.
2014-04-16 21:38:35 +01:00
Sebastian Dröge
b21b46a07a level: Use the correct number of samples to iterate over the input array
Fixes invalid memory accesses and accesses to uninitialised data.
2014-04-16 18:50:50 +02:00
Sebastian Dröge
bd65c36cbb icydemux: Unref dropped events 2014-04-16 18:50:50 +02:00
Vincent Penquerc'h
457712b933 matroska: fix check for amount of data to read
History shows length==0 should set data to NULL and return,
so we do that too instead of trying to read nothing.

Coverity 206205
2014-04-16 17:44:51 +01:00
Vincent Penquerc'h
46a39bdd4f deinterlace: fix sign comparison
history_count is unsigned, so the whole comparison will be made
as unsigned, and fail to reject what it was meant to.

Coverity 206204
2014-04-16 17:44:51 +01:00
Vincent Penquerc'h
c6acd6368b avidemux: remove dead code
sub may not be NULL in this switch, there is a bail out just
before it if so.

Coverity 206098
2014-04-16 17:44:51 +01:00
Vincent Penquerc'h
937269d02e flacparse: remove dead code
The block_size == 0 was shortcut earlier, and the variable is not
modified in the meantime.

Coverity 206097
2014-04-16 17:44:50 +01:00
Vincent Penquerc'h
2e120c9440 videomixer: remove dead code
While it seems to keep a compile time selection, I traced it
to some code copied from videoconvert, where it was removed,
with the following comment:

    Also remove the high-quality I420 to BGRA fast-path as it needs
    the same fix, which causes an additional instruction, which causes
    orc to emit more than 96 variables, which then just crashes.
    This can only be fixed in orc by breaking ABI and allowing more
    variables.

Thus, I remove it here as well.

Coverity 206064
2014-04-16 17:44:50 +01:00
Vincent Penquerc'h
595a9cb5c5 isomp4: fix incorrect masking for multiple tags
Coverity 206058
2014-04-16 17:44:50 +01:00
Vincent Penquerc'h
a5b7c12e35 isomp4: fix wrong atom flags set when adding samples
Coverity 206057
2014-04-16 17:44:50 +01:00
Vincent Penquerc'h
d2b682c271 audiofx: fix comparison of delta time to a threshold
Coverity 206055
2014-04-16 17:44:50 +01:00
Vincent Penquerc'h
7ebfdbeaf8 wavparse: do not rely on call failure keeping return data unmodified
This is clearer this way too.

Coverity 206029
2014-04-16 17:44:50 +01:00
Vincent Penquerc'h
b344b29ff2 isomp4: catch fseek error
Coverity 206028
2014-04-16 17:44:50 +01:00
Vincent Penquerc'h
88eccee88c isomp4: report failures to caller
Coverity 206027
2014-04-16 17:44:50 +01:00
Wim Taymans
783b4ba2c4 rtpjitterbuffer: refuse serialied query when buffering
When we are buffering, we can't block and wait for the serialized query
to complete because the jitterbuffer will not try to forward the query
while buffering. Instead, just refuse the query.
2014-04-16 18:16:33 +02:00
Wim Taymans
233e9e64b8 rtpjitterbuffer: don't free the serialized query
We should never free a serialized query in the queue, it is the upstream
caller that will free it.
2014-04-16 18:16:32 +02:00
Sebastian Dröge
74c23f0f4f videomixer: Create hashtable only when we actually use it
In error cases we previously returned without freeing it.
2014-04-16 17:33:46 +02:00
Sebastian Dröge
d3a2b3c73a videomixer: Chain up to the parent class' dispose function 2014-04-16 17:30:59 +02:00
Marc Leeman
5b4681dfe7 udpsrc: correct LOG msg for -1
Signed-off-by: Marc Leeman <marc.leeman@gmail.com>
2014-04-16 13:54:40 +01:00
Sebastian Dröge
b038fd4eff interleave: Fix negotiation to work at all again
The caps query handling function for the sinkpads was called for
the srcpad, and the sinkpads had none. This commit moves it to the
right pad, but nonetheless the negotiation still looks wrong.

This makes the test pass again after the recent coverity fix
and also allows interleave to work again, but someone should
really review the negotiation code and fix it.
2014-04-15 21:36:30 +02:00
Josep Torra
eaee14aff4 rtph264depay: only guess AU boundaries when aren't indicated by marker
The marker bit isn't mandatory and we had in place code to guess AU
boundaries by detecting a new picture start. This guessing code
didn't work with interlaced content that has proper marker bits
to indicate the AU boundaries. It was leaking the first field buffer
and producing a corrupted output.

fixes: https://bugzilla.gnome.org/show_bug.cgi?id=728041
2014-04-12 04:42:36 +02:00
Jimmy Ohn
ecf188e6cd qtdemux: replace duplicated variable when parsing trex atom
https://bugzilla.gnome.org/show_bug.cgi?id=727878
2014-04-10 09:03:02 +02:00
Sebastian Dröge
d47806320d qtdemux: Properly return stream flags when parsing trex atom
https://bugzilla.gnome.org/show_bug.cgi?id=727867
2014-04-09 08:58:48 +02:00
Edward Hervey
9859515605 interleave: Add missing break in switch statement
The caps query is handled entirely already before.

CID #1139757
2014-04-08 11:31:06 +02:00
Vincent Penquerc'h
31f36d805a avidemux: use frames, not bytes, for position query in VBR streams
Coverity 1139648
2014-04-07 12:58:23 +01:00
Vincent Penquerc'h
42298f65e8 smpte: fix copy/paste error causing unmap on wrong buffer
Coverity 1139647
2014-04-07 12:43:57 +01:00
Vincent Penquerc'h
1d7735b1d6 deinterlace: guard against finding no suitable pattern
The code handles a -1 pattern index, and it seems plausible
that a pattern might be found later, so it seems best to not
send an element error here.

Coverity 1139766
2014-04-07 12:20:12 +01:00
Wim Taymans
5b9945e0a6 rtspsrc: update for new MIKEY API 2014-04-04 17:38:14 +02:00
Wim Taymans
6210cbe1e2 rtspsrc: send sender SSRC in the MIKEY message
Allocate a new SSRC for our RTCP messages back to the server and set
this in the MIKEY message.
2014-04-03 17:40:01 +02:00
Wim Taymans
4f641ef18b rtspsrc: make random number for the CSB
As recommended in the RFC
2014-04-03 17:39:30 +02:00
Wim Taymans
f932da3be6 rtspsrc: don't put spaces in keymgmt header 2014-04-03 12:21:27 +02:00
Wim Taymans
2edd450369 rtspsrc: create and send the RTCP encryption key
Create and make a key for encrypting the RTCP packets back to the server
and wrap this in a MIKEY message that we send as a header in the SETUP
request.
2014-04-03 12:21:27 +02:00
Wim Taymans
a52b7eadfd rtspsrc: free the srtpdec element 2014-04-03 12:18:39 +02:00
Wim Taymans
f0f9451523 rtspsrc: cleanup stream_free function
There is no reason to NULL all fields, we will free the stream anyway.
2014-04-03 12:16:25 +02:00
Wim Taymans
c3de599c4f jitterbuffer: demote warning to debug
For TCP, it is normal that we don't have timestamps so don't WARN on
it.
2014-04-03 12:09:24 +02:00
Thibault Saunier
b95d9cfb21 avidemux: Always set PTS=DTS on raw video streams 2014-03-31 18:38:28 +02:00
Thibault Saunier
511202d50c avidemux: Always set pixel-aspect-ratio on raw video streams
That field is mandatory in caps and if it is not present in the
AVI container, it means square pixels thus 1/1.
2014-03-31 18:38:22 +02:00
Tim-Philipp Müller
821c68822b matroska-mux: add mapping for Opus audio
Might want to consider adding channels/rate
requirement to template caps, but requires
fixing up of encoder and parser first.
2014-03-30 00:35:07 +00:00
Tim-Philipp Müller
b158a1c068 matroska-demux: add mapping for Opus audio codec
https://bugzilla.gnome.org/show_bug.cgi?id=727305
2014-03-30 00:31:11 +00:00
Tim-Philipp Müller
273f389d57 rtpmanager: copy sticky events when exposing pads in more places
https://bugzilla.gnome.org/show_bug.cgi?id=724712
2014-03-29 13:23:02 +00:00
Ognyan Tonchev
2143a6e452 jpegpay: consider header len when calculating payload len
Fixed https://bugzilla.gnome.org/show_bug.cgi?id=726777
2014-03-27 09:45:20 +01:00
Mark Nauwelaerts
3414e3d0b9 matroskademux: segment closing not needed in 1.x
... as sender should keep track of segment base accumulation.
Rather, it may have some adverse effects as a spurious segment event,
e.g. in collectpads.
2014-03-25 21:02:45 +01:00
Mark Nauwelaerts
9a30726226 matroskademux: early sending pending codec-data for all streams
... at least before syncing across all streams might cause some gap
activity on any of those streams, notably sparse streams.

See also #712134
2014-03-25 21:02:45 +01:00
Mark Nauwelaerts
1e135a38cc matroskamux: handle both sticky and non-sticky custom event 2014-03-25 21:02:45 +01:00
Wim Taymans
e7c8fa1127 rtspsrc: only expose streams on dataflow
Only probe on buffers, we don't want to expose the streams on events.
2014-03-25 11:44:27 +01:00
Wim Taymans
3b497bf7d5 rtspsrc: copy sticky events to ghostpad
When we expose internal pads as ghostpads, first copy the sticky events
so that we have the caps and segment etc.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724712
2014-03-25 11:36:40 +01:00
Wim Taymans
67f3113759 rtspsrc: srtp handling 2014-03-25 10:23:24 +01:00
Wim Taymans
4846be1491 rtspsrc: set SSRC on caps if known 2014-03-25 10:23:00 +01:00
Wim Taymans
5ec8c96966 rtspsrc: put caps on udpsrc instead of using the signals
Try to avoid using the request-pt-map to get caps but set them directly
on the udpsrc element. That way, the caps get nicely transformed as they
pass through the different elements in the rtpbin, including the AUX and
decoder/encoder elements.
2014-03-24 17:07:06 +01:00
Wim Taymans
2b59828e0b rtspsrc: use profile to set rtcp caps
Use the negotiated profile to set x-rtcp or x-srtcp caps
2014-03-24 15:35:09 +01:00
Wim Taymans
a7b55d7687 rtspsrc: set udpsrc to READY
READY is enough to allocate ports now
2014-03-24 15:34:26 +01:00
Wim Taymans
d3c736c50f udpsrc: improve caps handling
Protect caps with the lock.
Don't push the caps event from the set_property function but mark the
pad for reconfiguration so that it will renegotiate and push the new
caps event in the streaming thread.
2014-03-24 15:22:04 +01:00
Wim Taymans
5e44fa3e31 udpsrc: open/close socket in NULL<->READY state
We should open the socket when going to NULL<->READY and not in the
start/stop vemthod, which is called in READY<->PAUSED. This makes it
possible to allocate a socket without going to PAUSED (and starting the
negotiation).
2014-03-24 15:15:34 +01:00
Wim Taymans
a4f6f963ec rtspsrc: free caps in ptmap array
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726696
2014-03-24 14:35:01 +01:00
Wim Taymans
d6c5fbc87c rtspsrc: handle NULL rtpmap and parse error better 2014-03-20 11:12:51 +01:00
Mathieu Duponchelle
6cf0f19c14 videomixer: Port to new collectpads API
See: https://bugzilla.gnome.org/show_bug.cgi?id=724705
2014-03-16 17:44:40 +01:00
Per x Johansson
2a362c6fb1 matroskademux: fix assert on fps lower than 1
Fixes assert caused by gst_duration_to_fraction calling
gst_util_uint64_scale_int with a denominator of 0 when fps is less
than 1.

https://bugzilla.gnome.org/show_bug.cgi?id=726106
2014-03-12 09:08:31 +01:00
Thiago Santos
373eceef7c videomixer2: store video info with buffers to keep it in sync
Instead the queued buffer might have an old caps while the pad
is already storing the information for a new caps. Mixing those
while handling buffers will often lead to issues

https://bugzilla.gnome.org/show_bug.cgi?id=725948
2014-03-11 00:49:19 -03:00
Olivier Crête
15d276058e rtp: Remove caps restrictions from RTP depayloader sink caps
Remove caps restrictions that correspond to the default and are not
required in SDP. With the new usage of having pads require a subset
of the caps, they will make the negotiation fail.
2014-03-06 12:06:43 -05:00
Olivier Crête
5a9b988b85 rtpspeexdepay: Remove caps restrictions for depayloader
The "encoding-params" is optional in the SDP, because we now require
a subset of the caps, it would fail caps negotiatioin if it wasn't present.
So removed it from the template caps.
2014-03-06 11:03:04 -05:00
Wim Taymans
224239096d rtspsrc: skip streams with same control url
Keep track of what streams we did the SETUP for. We only need to
configure caps, wait for pads and push events on setup streams. We can
remove the disabled state of the stream and simplify some checks.
After we setup a stream, skip the other streams that have the same
control url. Use a skipped flag to mark streams that should be skipped.
2014-03-06 12:30:54 +01:00
Wim Taymans
3b27fc2f0f rtspsrc: remove obsolete code 2014-03-06 12:30:54 +01:00
Wim Taymans
27d883fe64 rtspsrc: just use the SDP index as the stream id
Use the index of the media stream in the SDP as the stream id instead of
keeping a separate counter.
2014-03-06 12:30:54 +01:00
Wim Taymans
99a9d2873c rtspsrc: handle NULL control urls better 2014-03-05 15:44:25 +01:00
Wim Taymans
d2f93e3afc session: small cleanups
It's nicer to explicitly check for NULL on pointer types to make it
clear that it's a pointer and not a boolean.
2014-03-05 14:28:26 +01:00
Wim Taymans
5818a0de1a session: handle unknown SSRC in FIR
https://bugzilla.gnome.org/show_bug.cgi?id=725712
2014-03-05 14:27:47 +01:00
Alessandro Decina
c4bf6e8b7e rtspsrc: fix seeking
Call gst_rtspsrc_connection_flush (src, FALSE) to reset connections as
non-flushing before sending PAUSE and PLAY with the new npt range. Without this
patch, those commands would fail with EINTR as the connections were still
flushing.
2014-03-05 11:39:09 +01:00
Thiago Santos
fd12ff4c29 avidemux: expose xsub as a subtitle instead of as a video
It is placed inside a 'vids' struct, so it was being exposed on
a pad named video_%d. XSUB are subtitles and this patch adds
an special case for it to be exposed in a subpicture_%d pad
2014-03-04 20:29:45 -03:00
Thiago Santos
dee861630a avidemux: do not try to add a tag with tag_name set to NULL
This can happen if there are subtitles in the stream, leading to
an assertion
2014-03-04 20:29:45 -03:00
Wim Taymans
70de0e4e99 rtspsrc: Add support for multiple payload types
A media stream can have multiple payload types. Parse all the payload
types and collect the caps information. We then have to store the
pt<->caps mapping instead of 1 pt and 1 caps.
Parse the profile from the SDP and use that to negotiate the transport
instead of always using AVP.
Rework how we do some tweaks for ASF and Realmedia.
2014-03-04 16:40:34 +01:00
Wim Taymans
dbe92c9147 rtspsrc: refactor payload handling 2014-03-04 11:34:39 +01:00
Wim Taymans
b4caf09011 jitterbuffer: fix buffer level with invalid DTS
It is possible that the DTS is invalid (when we receive RTP packets from
TCP, for example). As a fallback, use the reconstructed PTS value to
calculate the buffer level.
2014-03-03 11:34:00 +01:00
Thiago Santos
0443c2593a Revert "aacparse: put codec data on caps for loas format"
This reverts commit e459cf3e01.

This was pushed by accident, the bug should likely be fixed in
libav https://bugzilla.libav.org/show_bug.cgi?id=644
2014-02-27 23:15:04 -03:00
Thiago Santos
e459cf3e01 aacparse: put codec data on caps for loas format
gst-libav audio decoder also needs codec data for LOAS format, otherwise
it will complain about not having a decoder config and skip all packets

https://bugzilla.gnome.org/show_bug.cgi?id=596772
2014-02-27 17:10:03 -03:00
Tim-Philipp Müller
f3163fb45f matroskademux: align raw audio memory to powers of two
https://bugzilla.gnome.org/show_bug.cgi?id=725008
2014-02-27 00:46:39 +00:00
Tim-Philipp Müller
c3dc53e551 matroskademux: calculate alignment properly for audio depths not a multiple of 8 2014-02-27 00:46:39 +00:00
Matej Knopp
d33b4dce63 matroskademux: fix crash with 24-bit raw audio
Do not try to align audio buffers to odd numbers,
which will get us a NULL buffer which we then
crash on.

https://bugzilla.gnome.org/show_bug.cgi?id=725008
2014-02-27 00:46:28 +00:00
Tim-Philipp Müller
5bad2d8b70 rtpmanager: re-enable -Werror 2014-02-27 00:12:13 +00:00
Tim-Philipp Müller
1d7f5c7a83 rtpjitterbuffer: fix compiler warning
gstrtpjitterbuffer.c: In function 'gst_rtp_jitter_buffer_loop':
gstrtpjitterbuffer.c:2978:3: error: 'result' may be used uninitialized in this function
   while (result == GST_FLOW_OK);
   ^
2014-02-27 00:11:11 +00:00
Sebastian Dröge
d4bdf5a1b1 rtpjitterbuffer: Fix uninitialized variable compiler warning 2014-02-26 21:11:23 +01:00
Jake Foytik
6dd9142592 rtpjitterbuffer: Remove raw comparisons of RTP sequence numbers
Several conditional statements perform comparison on RTP sequence
numbers without taking the sequence number rollover into account.
Instead, use the gst_rtp_buffer_compare_seqnum function to perform the
comparison.

https://bugzilla.gnome.org/show_bug.cgi?id=725159
2014-02-26 21:11:21 +01:00
Göran Jönsson
53ffd9e1ca rtph264pay: only update last_spspps time if all sps/pps got sent successfully
This fixes an issue with gst-rtsp-server where no sps and pps are
sent for the first intra frame, because the payloader starts working
already when receiving DESCRIBE but there is no transports so it tries
to send sps and pps, but that fails with a FLUSHING flow. But the time
for last sent sps and pps would still be set, so when PLAY arrives and
the first intra frame is to be sent there is no sps and pps sent due to
that time since last sps pps is less than spspps_interval.

https://bugzilla.gnome.org/show_bug.cgi?id=724213
2014-02-25 10:48:24 +00:00
Santiago Carot-Nemesio
b9a953161f rtspsrc: Fix deadlock when task creation is no successful
https://bugzilla.gnome.org/show_bug.cgi?id=725124
2014-02-25 10:10:31 +01:00
Stefan Sauer
fdb5d460de autodetect: demote candidate error to warning and plug fake{sink,src}
In the case where we have no suitable candidate we post a warning and plug a
fake-element. Do the same when non of the candidate work.

This is more consistent and plugin the fakesink as a fallback is probably
helpful for running unit tests without requiring hardware src/sink elements.

Fixes #722981
2014-02-23 20:34:43 +01:00
Darryl Gamroth
7a65277119 audiofxbaseiirfilter: check if coefficients are provided inside filter lock
https://bugzilla.gnome.org/show_bug.cgi?id=719524
2014-02-22 20:01:41 +01:00
Reynaldo H. Verdejo Pinochet
0898de65c8 aacparse: be more strict at ADTS header parsing
Adds two extra checks:

- Sampling frequency on header can't be 15.
- Frame size should be at least 9 or 7, depending
  on whether CRC protection is present.

https://bugzilla.gnome.org/show_bug.cgi?id=724638
2014-02-21 15:04:11 -03:00
Reynaldo H. Verdejo Pinochet
c3a4bb1657 aacparse: make sure we have enough ADTS data
We need at least 6 bytes to pass over to _get_frame_len()
but we were just checking for a minimum of 2 bytes for the
syncword.

https://bugzilla.gnome.org/show_bug.cgi?id=724638
2014-02-21 15:04:11 -03:00
Stefan Sauer
0566ea06e5 autodetect: check if the kid has a sync property
previously autovideosrc did not have a sync property and v4l2src has none either.
2014-02-20 22:52:57 +01:00
Stefan Sauer
bf6a2f9afd autodetect: use a common baseclass
This makes the actual elements super simple. We're using the ELEMENT_FLAG to
configure source/sink and a string for the Audio/Video type.
2014-02-20 21:28:43 +01:00
Aleix Conchillo Flaqué
62f5a27416 rtspsrc: add tls-database property
Add support for a new property: tls-database. If the property is set,
the certificate database will be given to the rtsp connection if TLS
protocol is being used. If the server certificate can't be verified with
the default database, this additional database will be used.

https://bugzilla.gnome.org/show_bug.cgi?id=724396
2014-02-20 20:03:40 +01:00
Stefan Sauer
c0fd8e0c39 autodetect: extract common helper code
The function to generate the pretty names is basically the same. Use one and add
a parameter.
2014-02-19 21:27:17 +01:00
Stefan Sauer
a4fd0f9351 docs: use docbook markup for xi:include
It turns out that the change in gtk-doc-1.20 which wraps the |[]| content in
CDATA break xi:inlcude examples. As in a whole jhbuild checkout these where
the only 4, we're fixing them instead.
2014-02-18 22:54:45 +01:00
Stefan Sauer
9d9ffba17e isomp4mux: fix copy and paste
This fixes doc warnings.
2014-02-18 22:35:45 +01:00
Stefan Sauer
35da463618 docs: use the gtk-doc syntax to link to properties
Don't use docbook unless needed. Also stip other docbook tags in the the files we fix.
2014-02-18 22:35:00 +01:00
William Jon McCann
577d873009 docs: fix mismatched para tags
newer gtkdoc is more sensitive to mismatched docbook tags.
This fixes the build in master.
2014-02-14 22:26:08 +01:00
Wim Taymans
353e510f94 rtpjitterbuffer: add support for serialized queries
See https://bugzilla.gnome.org/show_bug.cgi?id=723850
2014-02-14 15:59:46 +01:00
Wim Taymans
bbe6d9a258 rtpsession: proxy caps and allocation on RTP pads
recv_rtp_sink: allow proxying of the allocation query.
send_rtp_sink: allow proxying of caps and allocation. This allows us to
query caps downstream as well as get an allocator from downstream.
send_rtp_src: allow proxy of caps, this makes the caps query do
upstream.

See https://bugzilla.gnome.org/show_bug.cgi?id=723850
2014-02-14 12:05:55 +01:00
Thiago Santos
7f1d51ba90 qtdemux: handle tags in mac encoding
Check the charset from (C)*** tags and set the charset
to convert from MAC encoding if suitable.

https://bugzilla.gnome.org/show_bug.cgi?id=723166
2014-02-13 12:37:03 -03:00
divhaere
19a307930a matroska: add support for GRAY8, BGR and RGB video colourspaces in V_UNCOMPRESSED codec
https://bugzilla.gnome.org/show_bug.cgi?id=723849
2014-02-11 21:22:33 +01:00
Sebastian Dröge
4ecccb6ff6 goom: Remove unused functions 2014-02-09 23:38:44 +01:00
Sebastian Dröge
aafcbbb2fe matroskaparse: Comment out some unused functions used only from the commented out pull-mode code 2014-02-09 23:21:20 +01:00
Sebastian Dröge
3bc53f0840 rtprtxsend: Fix unitialized variable compiler warning
variable 'rtx_ssrc' is used uninitialized whenever
'if' condition is false [-Werror,-Wsometimes-uninitialized]
2014-02-08 17:24:06 +01:00
Sebastian Dröge
3d8f078b61 rtpac3depay: Remove unused variable 2014-02-08 17:21:19 +01:00
Sebastian Dröge
29ea0db5a3 flx: Fix typo in header include guard
error: '__GST_FLX_FMT__H__' is used as a header guard here,
followed by #define of a different macro [-Werror,-Wheader-guard]
2014-02-08 17:19:39 +01:00
Thiago Santos
f5f27f7d0d qtmux: remove have_dts flag from pads
It was used in the past in 0.10 when there was no explicit DTS
field in buffers, now we have it in 1.x series and we can
check it directly with GST_BUFFER_DTS_IS_VALID
2014-02-07 13:10:25 -03:00
Thiago Santos
f89ba82f29 qtmux: improve support for sparse streams
Do not try to use subsequent buffer timestamps to calculate
sparse streams durations because the stream is sparse and
the buffers might not be 'time adjacent'. So rely on the
duration and give the option to the pad to provide
custom 'empty' buffers to represent the gaps in the
stream, this can vary on how the data is represented.

Right now, the only sparse stream supported is tx3g subtitles.
2014-02-07 13:10:24 -03:00
Thiago Santos
99e966e2e1 qtmux: add support for text/x-raw subtitles
Adds it to mp4mux, qtmux and gppmux.

Buffers need to be prefixed with 2 bytes for the text length before
being muxed.

https://bugzilla.gnome.org/show_bug.cgi?id=581295
2014-02-07 13:10:24 -03:00
Thiago Santos
d644cda79b qtmux: add support for the TX3G atoms
Adds functions for creating and setting values related to the
tx3g atom for raw text subtitle support.

QTFF spec has information on those atoms

https://bugzilla.gnome.org/show_bug.cgi?id=581295
2014-02-07 13:10:24 -03:00
Thiago Santos
2ae1897273 qtmux: add subtitle support to qtmuxmap structures
adds basic stubs for subtitle support around the qtmux and
qtmuxmap structures. Still no real subtitle implemented, but
basic functions in place

https://bugzilla.gnome.org/show_bug.cgi?id=581295
2014-02-07 13:10:24 -03:00
Reynaldo H. Verdejo Pinochet
2f8a1aa870 matroska: factor out read context init/reset
While at this, move _track_reset() to track-ids
so it can be called from the common read context
reset routine.

https://bugzilla.gnome.org/show_bug.cgi?id=722705
2014-02-06 13:25:12 -03:00
Wim Taymans
575332d127 effectv: fix doc section of revtv element 2014-02-06 12:21:07 +01:00
Matthieu Bouron
200eb7498d deinterlace: do not try set deinterlace method if passthrough is enabled
Fixes an issue with progressive content and unsupported video formats
for the deinterlace method.

https://bugzilla.gnome.org/show_bug.cgi?id=719636
2014-02-04 21:44:35 +01:00
Rafał Mużyło
ac4df5e2c5 gst: Don't use endianness-specific S8 audio format
It does not exist.

https://bugzilla.gnome.org/show_bug.cgi?id=723331
2014-02-04 13:44:29 +01:00
Per x Johansson
46bc1677a4 matroskamux: Fix constantly growing used uid list
Moves the used uid list to the class to avoid having it grow forever.

https://bugzilla.gnome.org/show_bug.cgi?id=723269
2014-01-30 11:59:28 -03:00
Mike Sheldon
659939f0f0 wavparse: Ignore Broadcast Wave Format (BWF) tags when searching for 'fmt' chunk
https://bugzilla.gnome.org/show_bug.cgi?id=723125
2014-01-29 20:16:48 +01:00
Mark Nauwelaerts
d25a183ccc ac3parse: custom get_sink_caps handling for private stream caps
... now that those are transformed rather than parsed, some transforming
of caps is required as well to make auto-plugging succeed.
2014-01-27 20:07:41 +01:00
Sebastian Dröge
8054cd5df3 Revert "rtspsrc: Proxy rtpjitterbuffer do-retransmission property"
This reverts commit 9f7b1128b1.

This should be handled automatically be rtspsrc if the AVPF profile
is used, and manual enabling of it can be done with the new-manager
signal.
2014-01-24 12:37:39 +01:00
Wim Taymans
43feb82feb rtspsrc: add signal to notify of new manager
So that you can configure and connect to signals on the rtpbin.

See https://bugzilla.gnome.org/show_bug.cgi?id=722866
2014-01-24 10:22:59 +01:00
Aleix Conchillo Flaqué
9f7b1128b1 rtspsrc: Proxy rtpjitterbuffer do-retransmission property
https://bugzilla.gnome.org/show_bug.cgi?id=722866
2014-01-24 09:14:59 +01:00
Wim Taymans
204bd715d2 rtpjitterbuffer: handle expected packet being an RTX packet
If the expected packet (do_next_seqnum is TRUE) is the one we requested
for retranmission earlier, do the logic to update the retransmission
statistics as well before setting up the timers for the next expected
packet.
Also reset the retransmission counter if the timer is reused for another
seqnum.
2014-01-21 17:52:44 +01:00
Wim Taymans
ddb0b9c422 rtpbin: add a caps accumulator for the request-pt-map signal
Add an accumulator that stops the signal emission as soon as a caps has
been retrieved. Otherwise the default handler would continue emitting
the signal and possibly overwrite the result with NULL again.
2014-01-21 15:48:20 +01:00
Wim Taymans
ef20dfe031 rtxreceive: copy flags and timestamps from original buffer 2014-01-21 15:29:27 +01:00
Wim Taymans
9a3d4d7cbe rtpjitterbuffer: ignore invalid timestamps in rtt calculation
When the input buffer does not have a valid timestamp, don't try to
calculate the round-trip-time.
2014-01-21 15:29:26 +01:00
Reynaldo H. Verdejo Pinochet
cf0c780138 matroskaparse: better default caps when none set
Uses information gathered during EBML parsing to
forge a more suitable set of caps instead of blindly
assuming everything is video/x-matroska.

For consistency, stream type reset was added to
matroska-demux too.

https://bugzilla.gnome.org/show_bug.cgi?id=722311
2014-01-21 11:11:46 -03:00
George Kiagiadakis
1a300eb509 rtprtxsend: ensure that no rtx buffers are sent after EOS
To do that, enqueue the EOS event to be sent from the srcpad task
thread and flush the queue right afterwards, so that no more rtx
buffers can be sent, even if there are more requests coming in.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722370
2014-01-21 15:00:37 +01:00
George Kiagiadakis
133913a11a rtprtxsend: run a new GstTask on the src pad
The reason behind this is to minimize the retransmission delay.
Previously, when a NACK was received, rtprtxsend would put a
retransmission packet in a queue and it would send it from chain(),
i.e. only after a new buffer would arrive.

This unfortunately was causing big delays, in the order of 60-100 ms,
which can be critical for the receiver side.

By having a separate GstTask for pushing buffers out of rtxsend,
we can push buffers out right after receiving the event, without
waiting for chain() to get called.
2014-01-21 14:54:01 +01:00
Sebastian Dröge
e178cf60ae rtpvp8pay: Don't leak input buffers
https://bugzilla.gnome.org/show_bug.cgi?id=722414
2014-01-20 10:13:19 +01:00
Mark Nauwelaerts
829cec51c7 avimux: reset some more audio pad data when needed 2014-01-19 17:53:45 +01:00
Mark Nauwelaerts
3ea338ce27 avimux: write correct blockalign for vbr audio
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720659
2014-01-19 17:53:45 +01:00
Aleix Conchillo Flaqué
cdbb2ba6b8 rtpjitterbuffer: do not drop serialized events when latency is set
Serialized events are now queued in the jitter buffer, so we don't
want to drop them even latency is set.

https://bugzilla.gnome.org/show_bug.cgi?id=722372
2014-01-18 10:38:46 +01:00
Michael Olbrich
447556fe6b avimux: don't make the buffer writable unless absolutely necessary
https://bugzilla.gnome.org/show_bug.cgi?id=722396
2014-01-17 19:25:15 -03:00
Sebastian Dröge
809d105982 matroskademux: Don't skip all video frames until the first keyframe
Instead do it like all other demuxers and let parsers and decoders
handle that. The keyframe information inside the container might
be completely wrong like in the sample file of the bug report,
and if it is correct and we push no keyframes, then the parsers
and decoders will handle that properly anyway.

https://bugzilla.gnome.org/show_bug.cgi?id=682276
2014-01-15 22:49:58 +01:00
Thiago Santos
52fc078310 qtdemux: remove elst_offset variables
They are not used anymore
2014-01-15 15:33:45 -03:00
Thiago Santos
5fe1b3eb28 qtdemux: remember reverse playback when verifying the segment end
Check if the rate is positive or negative to correctly compare the current
position with the segment to make reverse playback work
2014-01-15 15:33:45 -03:00
Thiago Santos
90a5565229 qtdemux: do not ignore empty segments
Make sure empty segments are used and pushed with a gap event
to represent its data (or lack of it)

Each QtSegment is mapped into a GstSegment with the corresponding
media range. For empty QtSegments a gap event is pushed instead
of GstBuffers and it advances to the next QtSegment.

To make this work with seeks, need to keep track of the starting
'base' to make sure it remains consistently increasing when
pushing new segment events.
For example: if a seek makes qtdemux start from 5s, the first
segment will have a base=0. When the next segment is activated,
its base time will be QtSegment.time - qtdemux.segment_base so
that it doesn't include the first 5s that weren't played and
shouldn't be accounted on the running time

This purposedly will remove the fix made for
https://bugzilla.gnome.org/show_bug.cgi?id=700264, at this
point it was decided to respect the gaps, even if they cause
a delay on playback, because that's the way the file was crafted.

https://bugzilla.gnome.org/show_bug.cgi?id=345830
2014-01-15 15:33:45 -03:00
George Kiagiadakis
397c4ed7a0 rtprtxsend: remove wrong check for payload type not having been set
1) pt can be lower than 96
2) there is no point in checking that because rtprtxsend will not
   even store buffers for payload types that it doesn't know about,
   so this case will never be reached
2014-01-15 10:13:12 +01:00
George Kiagiadakis
55746eaa4c rtprtxsend: fix data locking when creating rtx packets
This patch moves the creation of rtx packets to be done early,
in the src_event() function, when they are requested. The purpose
is to run gst_rtp_rtx_buffer_new() with the object locked to
protect internal data, because if it is done at the pushing stage,
we would have to lock and unlock multiple times in a row while we
are pushing the rtx buffers.

Previously there was no locking at all, which was terribly wrong.
2014-01-15 10:13:11 +01:00
George Kiagiadakis
3d9ca102c9 rtprtxsend: lock access to internal data in sink_event() function 2014-01-15 10:13:11 +01:00
George Kiagiadakis
ee8ae3000e rtprtxsend: remove unnecessary call to reset() from finalize()
...and use _free_full() on the pending buffers queue now that
reset() is not being called
2014-01-15 10:13:11 +01:00
George Kiagiadakis
f9f7e6e721 rtprtxsend: remove unused parameter from the internal reset() method 2014-01-15 10:13:11 +01:00
George Kiagiadakis
6d588ad6bb rtprtxsend: Use g_slice_* for allocating internal structures 2014-01-15 10:13:11 +01:00
George Kiagiadakis
75859ae924 rtprtxreceive: remove stupid mutex unlock in the middle of chain() 2014-01-15 10:13:11 +01:00
George Kiagiadakis
bf347dc50c rtprtxreceive: use GST_DEBUG_OBJECT / GST_WARNING_OBJECT instead of GST_DEBUG / g_warning 2014-01-15 10:13:11 +01:00
George Kiagiadakis
47788929d3 rtprtxreceive: fix integer format specifiers in GST_DEBUG
seqnum in this function is 32-bit, so G_GUINT16_FORMAT would
produce undefined output on big endian systems
2014-01-15 10:13:11 +01:00
George Kiagiadakis
252dfc34c8 rtprtxsend: change the rtx_pt_map directly in set_property() instead of delaying it for chain()
The same lock is held, so there is no point in complicating it...
2014-01-15 10:13:11 +01:00
George Kiagiadakis
8a0ae00ea8 rtprtxreceive: change the rtx_pt_map directly in set_property() instead of delaying it for chain()
The same lock is held, so there is no point in complicating it...
2014-01-15 10:13:11 +01:00
George Kiagiadakis
513ffc45b5 rtprtxreceive: simplify the code of finalize() 2014-01-15 10:13:11 +01:00
George Kiagiadakis
0fdae5f2f7 rtprtxreceive: use the GstObject lock instead of a new one 2014-01-15 10:13:11 +01:00
George Kiagiadakis
c945200ff2 rtprtxsend: use the GstObject lock instead of a new one 2014-01-15 10:13:11 +01:00
Vincent Penquerc'h
2ad1f20e7b Revert "aacparse: relax the detection of ADTS"
This was pushed by mistake along with the V4L2 fix.

This reverts commit 8eb4b032be.
2014-01-14 09:43:56 +00:00
Justin Joy
70be4fa24a rtpg726pay: don't leak encoding_name string
https://bugzilla.gnome.org/show_bug.cgi?id=722159
2014-01-14 10:29:47 +01:00
Akihiro Tsukada
8eb4b032be aacparse: relax the detection of ADTS
According to ISO/IEC 13818-7, "channel_config" field in ADTS header
may have value of 0, as in the case of frame with PCE.
gst_aac_parse_detect_streams() returned FALSE for those frames
and discarded them.
2014-01-13 09:08:50 +00:00
Tim-Philipp Müller
88ac735af3 matroskademux: don't leak TOC chapter list 2014-01-10 16:50:11 +00:00
Vincent Penquerc'h
f8158baa93 matroskamux: remove obsolete write-dummy-and-overwrite-on-eos code
The need for rewriting apparently is obsolete 0.10 leftover.
We now have caps for subtitles when we create the headers,
so we always write the correct data in the first place.
2014-01-10 08:54:04 +00:00
Tim-Philipp Müller
335b619cd5 rtprtxsend: remove duplicate assignment
Coverity CID 1151680
2014-01-09 23:55:16 +00:00
Vincent Penquerc'h
1c6ee3fba4 matroskamux: write subtitle codec ID and data at start when known
This avoids issues with writing dummy data first, then having
to come back and write correct data later. Doing so prevents
the muxed stream from being actually streamable.

https://bugzilla.gnome.org/show_bug.cgi?id=712134
2014-01-09 18:29:32 +00:00
Thiago Santos
5adedf9f5a qtmux: respect the HDLR box string format for mov and isomedia
Mov spec says it uses a pascal style string, while isomedia uses
a null terminated one. Store the current atoms flavor into the HDLR
to be able to generate the correct output.

https://bugzilla.gnome.org/show_bug.cgi?id=705982
2014-01-09 11:58:46 -03:00
Wim Taymans
7f8c4dceb4 Revert "matroskamux: Use the running time for container timestamps, not buffer timestamps"
This reverts commit b3aa8755fe.

We are already using the running-time because they were placed on the
buffers with gst_collect_pads_clip_running_time(). Arguably it would be
better to not modify the incomming buffers but collectpads seems to want
to use absolute timestamps from the buffers for finding the best buffer
(this can be changed with a custom compare function..).
2014-01-08 11:32:54 +01:00
Aleix Conchillo Flaqué
441f286e28 rtpbin: remove unused list of decoders
remove list of decoders, which are already handled by the list of elements.

https://bugzilla.gnome.org/show_bug.cgi?id=719938
2014-01-08 10:23:52 +01:00
Sebastian Dröge
2cddf3a0a9 matroskamux: Error out if ADPCM caps don't contain the layout field 2014-01-08 09:57:48 +01:00
Nicola Murino
bbb5a2853e matroskamux: Add support for g726 ADPCM
https://bugzilla.gnome.org/show_bug.cgi?id=720995
2014-01-08 09:57:48 +01:00
Wim Taymans
2e9e80badf rtspsrc: use new method to get media-type
Use the new method to get the media type of a transport.
2014-01-07 15:04:02 +01:00
Sebastian Dröge
5506dc3076 matroskamux: Add HEVC / h265 support 2014-01-06 14:55:36 +01:00
Sebastian Dröge
77745289c4 matroskademux: Add HEVC / h265 support 2014-01-06 14:55:36 +01:00
Stefan Sauer
73fe1d1f6f wavparse: remove ifdef'ed code
We do have adtl and cue parse as part of toc handling alreday. The fmt code is a left over from <0.10 times.
2014-01-06 13:55:36 +01:00
Stefan Sauer
9dde5e29da avidemux, waveparse: more logging for unhandled chunks
Always print a warning with the tag and if possible do a memdump.
2014-01-06 13:55:36 +01:00
Stefan Sauer
addf5c79a2 avidemux: expose 'strn' - stream name - as title tag 2014-01-05 22:47:42 +01:00
Stefan Sauer
5384da2a1f avidemux: parse fuji strd
We can get maker, model and capture date from this chunk.
Fixes #636143
2014-01-05 22:42:10 +01:00
Stefan Sauer
1be2922802 avidemux: ... and use the local api both times 2014-01-05 21:47:00 +01:00
Stefan Sauer
9a203fceeb avidemux: copy the riff api for ncdt into the element
This chunk is avi specific, no need to expose this as public api.
2014-01-05 21:40:21 +01:00
Sebastian Dröge
a4a7dafc32 matroskamux: Add missing semicolon from last commit 2014-01-05 10:28:34 +01:00
Sebastian Dröge
b3aa8755fe matroskamux: Use the running time for container timestamps, not buffer timestamps
Buffer timestamps have no real meaning here, and for selecting the next
buffer we already use the running time anyway.
2014-01-05 10:23:44 +01:00
Stefan Sauer
f48bb20b4f avi: use new riff api to extract nikon metadata
Fixes #636143
2014-01-04 21:34:38 +01:00
George Kiagiadakis
9226091235 rtprtxreceive: modify to use a payload-type map like rtprtxsend 2014-01-03 20:48:29 +01:00
George Kiagiadakis
c8a04bc7b2 rtprtxsend: do not keep history of packets with an unknown payload type
This allows to disable retransmission per payload type by not putting
a certain payload type in the map.
2014-01-03 20:48:29 +01:00
Wim Taymans
130ad1b1fa rtprtxsend: Allow SSRC-multiplexing and multiple payload types in the original stream
Conflicts:
	tests/examples/rtp/server-rtpaux.c
2014-01-03 20:48:29 +01:00
George Kiagiadakis
41285697ac rtprtxsend: Add an rtx-ssrc property to allow external control of the ssrc
This is useful when one needs to know the SSRC beforehands, so that it can
be used for SRTP for example.
2014-01-03 20:48:29 +01:00
Wim Taymans
679b5a8682 session: also push EOS event to RTCP srcpad 2014-01-03 20:48:29 +01:00
Wim Taymans
03e4a180da session: place SSRC in Retransmission event 2014-01-03 20:48:29 +01:00
George Kiagiadakis
0a8b149e9e rtprtxsend: use a realistic limit for the value of max-size-packets
G_MAXINT16 is chosen because if the queue contains more than
G_MAXINT16 packets, seqnum comparison will not work properly.
2014-01-03 20:48:28 +01:00
George Kiagiadakis
51edc07127 rtprtxsend: use a GSequence to implement the buffer queue
This has the advantage that searching the queue to find the
buffer with the requested seqnum is done with binary search.
2014-01-03 20:48:28 +01:00
George Kiagiadakis
487fa8c989 rtprtxsend: retransmit packets in the same order as the rtx requests 2014-01-03 20:48:28 +01:00
George Kiagiadakis
7d530ab59f rtprtxsend: Handle the max_size_time property
This property allows you to specify the amount of buffers
to keep in the retransmission queue expressed as time (ms)
instead of buffer count (which is the max_size_buffers property).
2014-01-03 20:48:28 +01:00
George Kiagiadakis
920a55532c rtprtxsend: keep important buffer information in a private structure
This is to avoid mapping a buffer every time we need to read a seqnum
or a timestamp.
2014-01-03 20:48:28 +01:00
Julien Isorce
5a1aa75961 rtpmanager: add new rtprtxsend / rtprtxreceive elements
The purpose of the sender RTX object is to keep a history
of RTP packets up to a configurable limit (in time). It will
listen for custom retransmission events from downstream. When
it receives a request for retransmission, it will look up the
requested seqnum in its list of stored packets. If the packet
is available, it will create a RTX packet according to RFC 4588
and send this as an auxiliary stream.

The receiver will listen to the custom retransmission events
from the downstream jitterbuffer and will remember the SSRC1
of the stream and seqnum that was requested. When it sees a
packet with one of the stored seqnum, it associates the SSRC2
of the stream with the SSRC1 of the master stream. From then
on it knows that SSRC2 is the retransmission stream of SSRC1.
This algorithm is stated in RFC 4588. For this algorithm to
work, RFC4588 also states that no two pending retransmission
requests can exist for the same seqnum and different SSRCs or
else it would be impossible to associate the retransmission with
the original requester SSRC.
When the RTX receiver has associated the retransmission packets,
it can depayload and forward them to the source pad of the element.

RTX is SSRC-multiplexed

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084
2014-01-03 20:47:59 +01:00
Matthieu Bouron
0bbdb9bb1d deinterlace: support any video formats and any caps features if deinterlace mode allows it
https://bugzilla.gnome.org/show_bug.cgi?id=719636
2014-01-03 11:22:01 +01:00
Wim Taymans
bb2d37b11d rtpbin: add some docs about AUX elements 2013-12-31 15:08:49 +01:00
Wim Taymans
d08e05b4ef rtpbin: add support for AUX sender and receiver
AUX elements are elements that can be inserted into the rtpbin
pipeline right before or after 1 or more session elements.

The AUX elements are essential for implementing functionality such
as error correction (FEC) and retransmission (RTX).

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087
2013-12-31 15:08:48 +01:00
Wim Taymans
ae22c95881 rtpbin: make request_element method internally
We can use the same method to create encoder and decoder elements, they
are just internal elements that we create.
2013-12-31 15:08:48 +01:00
Stéphane Cerveau
e7912641c3 wavparse: Skip id3 tag
Skip id3 tag during wav parse.

https://bugzilla.gnome.org/show_bug.cgi?id=721241
2013-12-31 10:39:21 +01:00
Edward Hervey
711c73290c avimux: Add missing break
I guess no-one noticed we no longer could mux WMV3 ...

COVERITY CID 1139759
2013-12-30 17:23:22 +01:00
Edward Hervey
91c5b09fb4 rtpvrawpay: Add missing break
COVERITY CID 1139762
2013-12-30 17:20:37 +01:00
Wim Taymans
ee7f41ba2e rtpsession: internal-ssrc is no longer deprecated 2013-12-30 17:00:45 +01:00
Wim Taymans
e721d26c68 rtpbin: add Since tags 2013-12-30 16:59:20 +01:00
Wim Taymans
5a2bc1405e rtpbin: add signal for new jitterbuffer
Emit a signal when a new jitterbuffer is created so that the app can
have a chance to configure it.
2013-12-30 16:52:28 +01:00
Wim Taymans
3f3b2d0886 rtpbin: handle multiple encoder instances
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
2013-12-30 16:28:57 +01:00
Wim Taymans
05c8edc174 rtpbin: fix memory leaks 2013-12-30 15:17:05 +01:00
Wim Taymans
9345c2280a rtpbin: expect the pads on the encoders
Don't use request pads for the encoder elements, the signal handler
should request the pads and make sure they are available with the right
name.
2013-12-30 15:17:05 +01:00
Wim Taymans
cbc80d10dd rtpbin: request-rtp-encoder are no action signals
The request-rtp-encoder signals are not action signals so mark them
correctly and use an accumulator to collect the result value.
2013-12-30 15:17:05 +01:00
Stefan Sauer
2e277bb341 wavparse: emit midi-base-note tag from data in 'smpl' chunk
Add parsing of the 'smpl' chunk. Right now we only grab the midi-base-note and
emit it as a tag.
2013-12-30 14:41:47 +01:00
George Kiagiadakis
5ddf6a5e32 gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision
When a collision is found on the internal ssrc, we have to change it.
Ideally, we want also the payloader upstream to follow this change and use
the new internal ssrc. Ideally we want this condition to be always met:
if there is one payloader sending on this session, its ssrc should match the
internal ssrc.
2013-12-30 14:03:05 +01:00
George Kiagiadakis
17517ca491 rtpsession: allow setting internal-ssrc again 2013-12-30 14:03:05 +01:00
Edward Hervey
e732b86b8e y4mencode: Remove dead code
set/get property isn't used
2013-12-30 13:50:35 +01:00
Edward Hervey
ac40045d0d rtpqcelpdepay: Remove uneeded variable 2013-12-30 13:50:35 +01:00
Aleix Conchillo Flaqué
47c65fc269 rtpbin: allow dynamic RTP/RTCP encoders/decoders
* gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
  added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
  and request-rtcp-decoder). The user will be able to provide encoders
  or decoders dynamically. The encoders must follow the srtpenc API and
  the decoders the srtpdec API. Having separate signals for RTP and RTCP
  allows the user to use different encoders/decoders or provide the same
  one (e.g. that would be the case for srtpenc).

  Also, rtpbin now allows application/x-srtp in its pads.

  https://bugzilla.gnome.org/show_bug.cgi?id=719938
2013-12-30 11:24:00 +01:00
Wim Taymans
f48bbabafc rtpjitterbuffer: dynamically recalculate RTX parameters
Use the round-trip-time and average jitter to dynamically calculate the
retransmission interval and expected packet arrival time.

Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412
2013-12-30 11:18:51 +01:00
Wim Taymans
416bd9a2c3 rtpjitterbuffer: calculate average jitter 2013-12-30 11:18:51 +01:00
Wim Taymans
7181a21ca9 rtpsession: use RTT from the Retransmission event
Place the estimated RTT in the Retransmission event and let the session
manager use that instead of the hardcoded value.
2013-12-30 11:18:50 +01:00
Wim Taymans
e996f73d0c jitterbuffer: take more accurate running-time for NACK
Don't use the current time calculated from the tmieout loop for when we
last scheduled the NACK because it might be unscheduled because of a max
packet misorder and then we don't accurately calculate the current time.
Instead, take the current element running time using the clock.
2013-12-30 11:18:50 +01:00
Thiago Santos
c1cd2f81f9 qtdemux: improve mss_mode/fragmented special handling
Make it clear what should be handled purely by mss mode:
1) Expose the streams on the first moof as there are no moov atoms
2) Properly cleanup streams on flushes

Add a note about the meaning of upstream_newsegment and mss_mode
for future reference.

Make all other special fragment handling shared for both dash
and mss streams.
2013-12-27 12:04:49 -03:00
Thiago Santos
a82f3418fd qtdemux: drain the adapter before pushing EOS
In a fragmented scenario, qtdemux is operating in push mode
and it gets a fragmented buffer. While processing its data
downstream gets unlinked (or a input-selector changes its
active pad and returns not-linked). Qtdemux stops processing
this fragment and returns not-linked upstream, leaving the
remaining data in its adapter.

When it gets an EOS it should make sure that all the data it
had received is pushed before pushing EOS.
2013-12-27 12:00:27 -03:00
Wim Taymans
bf878d75d1 rtspsrc: use aggregate control for PLAY/PAUSE/TEARDOWN
Use the aggregate control instead of the original request url to perform
PAUSE/PLAY and TEARDOWN.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721003
2013-12-26 11:27:30 +01:00
Sebastian Dröge
2f07b570f7 rndbuffersize: Proxy CAPS, ALLOCATION, SCHEDULING and srcpad events properly 2013-12-24 14:40:25 +01:00
Nicola Murino
5b1108dd5f matroskamux: adpcm max block align is 8192 2013-12-24 10:00:16 +01:00
Sebastian Dröge
4baf8080f2 matroskamux: Use correct codec id for ADPCM/DVI 2013-12-23 15:46:48 +01:00
Sebastian Dröge
7cae8922cb matroskademux: Check for the correct size of codec_data in the ACM case 2013-12-23 15:46:43 +01:00
Nicola Murino
00ea1cb003 matroskamux: basic adpcm support
https://bugzilla.gnome.org/show_bug.cgi?id=664339
2013-12-23 15:31:04 +01:00
Sebastian Dröge
371482a90c qtdemux: Fix calcuation of descriptor length
https://bugzilla.gnome.org/show_bug.cgi?id=720813
2013-12-23 15:09:49 +01:00
Tim-Philipp Müller
9c9efffd8c udpsrc: on receive error only unmap and unref buffer if one was alloced and mapped
coverity CID 1139866.
2013-12-19 20:35:03 +00:00
Tim-Philipp Müller
627109ce4d multiudpsink: fix misleading comment
Those are not allocated on the stack.
2013-12-19 12:47:22 +00:00
Todd Agulnick
8bab119af9 Some compiler warning fixes to satisfy XCode compiler
https://bugzilla.gnome.org/show_bug.cgi?id=720513
2013-12-16 16:52:40 +01:00
Sebastian Dröge
2927805749 wavpackparse: Post AUDIO_CODEC tag 2013-12-16 10:03:06 +01:00
Sebastian Dröge
753d3c23a2 sbcparse: Post AUDIO_CODEC tag 2013-12-16 10:03:06 +01:00
Sebastian Dröge
05e196cbb6 flacparse: Post AUDIO_CODEC tag
https://bugzilla.gnome.org/show_bug.cgi?id=720512
2013-12-16 10:03:06 +01:00
Sebastian Dröge
29f2cae129 dcaparse: Post AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Sebastian Dröge
d2ab5199bc amrparse: Post AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Sebastian Dröge
6f89b430ea ac3parse: Post AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Sebastian Dröge
b3abbe3f5e aacparse: Post AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Sebastian Dröge
c07424a534 mpegaudioparse: Use pbutils functionality to create the AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Olivier Crête
ada6ea668b rtpsession: Add error message if the app tries to set the internal-ssrc 2013-12-13 17:36:36 -05:00
Olivier Crête
d715010d78 rtpsession: Only count nacks when a nack packet is received
Not when any RTCP feedback packet is.
2013-12-13 16:08:35 -05:00
Olivier Crête
7af9fdbca6 rtpsession: Process PSFB FIR requests which lack the media ssrc
According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
So in that case, we ignore the retained feedback and just let it through
to the rtp_session_process_fir() function which will check for the actual
SSRC inside the FCI.

Fixes a regression introduced by commit 57c27ec3
2013-12-13 16:01:07 -05:00
George Kiagiadakis
6a2de911fa rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders
Previously, when the session had multiple internal sender SSRCs, it would
issue SR reports with RB blocks only on the first RTCP timeout and afterwards
SR reports would be sent empty. This was because the "generation" number
in RTPSource would increase more than once during the same cycle and afterwards
it would always be greater than the session's generation, which would cause
it to be skipped from being included in RBs.

This commit fixes this problem by:
1) Increasing the RTPSource generation only at the end of each cycle,
which essentially fixes the problem but only when the internal senders
are less than GST_RTCP_MAX_RB_COUNT.
2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
SR the given RTPSource has been reported in, which also fixes the problem
when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
necessary because of the fact that any RTPSource is marked as reported
in itself's SR and makes it impossible to know if it has been reported
in other SRs too or not, and which.
2013-12-12 16:44:27 +01:00
George Kiagiadakis
c78a115154 rtpsession: keep extra stats for scheduling BYE
Keep an extra stats structure for scheduling the BYE packets. When we
decide to schedule BYE, make a copy of the current stats into the
bye_stats. Then while we schedule the BYE, update and use only the
bye_stats. When we finished scheduling the BYE packet, we use the
regular stats again.
2013-12-12 10:38:43 +01:00
George Kiagiadakis
282028e753 rtpsession: when we schedule BYE, only deal with BYE sources
When we are doing the RTCP timeout to schedule BYE packets, don't
generate RTCP for all sources but only for the sources marked as BYE.
2013-12-12 10:34:38 +01:00
George Kiagiadakis
6a421c3d81 rtpsession: reset state after scheduling BYE
After we do RTCP, we are not scheduling bye anymore.
2013-12-12 10:32:48 +01:00
George Kiagiadakis
0a0ff100ef rtpsession: also count NACKS when no signal was pending 2013-12-12 10:31:38 +01:00
George Kiagiadakis
bec9c04ea0 session: ignore RTCP packets for the BYE sources
When we are scheduling BYE packets, ignore all RTCP for the sources that
are scheduling a BYE packet. Other sources that are not scheduling BYE
should continue receiving RTCP packets as usual.
2013-12-12 10:09:25 +01:00
Julien Isorce
33b398e345 rtpsession: determine if the session is doing point-to-point
In this case T_dither_max is set to 0 according to RFC 4585
2013-12-10 16:57:56 +01:00
Wim Taymans
eee515cb2c rtpjitterbuffer: serialize events in the buffer
Serialize events into the jitterbuffer by inserting them with a -1
seqnum.
Update unit test to expect events from the streaming thread.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986
2013-12-10 11:57:37 +01:00
Wim Taymans
36e78bc5ca rtpjitterbuffer: detect -1 seqnum
Keep the seqnum as a full guint so that we can check for -1 entries and
deal with them correctly.
Immediately try to push -1 seqnum.
2013-12-10 11:04:06 +01:00
Wim Taymans
4a2e0f4ff4 rtpjitterbuffer: reorganize jitterbuffer items
Keep the oldest item at the head and the newest items on the tail. This
makes it easier to deal with -1 seqnums.
2013-12-10 11:01:03 +01:00
Wim Taymans
ea2a222cef jitterbuffer: correctly check for invalid values
Check for -1 on the guint from the buffer item instead of on the guint16
or guint32.
Also insert -1 seqnum at the head of the jitterbuffer.
2013-12-09 23:34:10 +01:00
Sebastian Dröge
f3c3dee148 mulawdec: Require caps to be set before accepting any data 2013-12-05 12:15:29 +01:00
Sebastian Dröge
d585bd7bbd rtptheorapay: Don't send headers twice if we got them from the caps already 2013-12-04 21:58:29 +01:00
Sebastian Dröge
d105de6e0f rtptheorapay: Don't leak config data when receiving a second CAPS event 2013-12-04 21:58:29 +01:00
Sebastian Dröge
0915d696c7 rtpvorbispay: Don't send headers twice if we got them from the caps already 2013-12-04 21:58:29 +01:00
Sebastian Dröge
967280df42 rtpvorbispay: Don't leak config data when receiving a second CAPS event 2013-12-04 21:58:29 +01:00
Sebastian Dröge
d87f6cf483 rtpstreamdepay: Add RFC4571 RTP stream depayloading element
https://bugzilla.gnome.org/show_bug.cgi?id=719829
2013-12-04 21:58:29 +01:00
Sebastian Dröge
c5284dc047 rtpstreampay: Add RFC4571 RTP stream payloading element
https://bugzilla.gnome.org/show_bug.cgi?id=719829
2013-12-04 21:58:29 +01:00
Thiago Santos
1fd094d96b qtdemux: improve fragment-start tracking
Some buffers can have multiple moov atoms inside and the strategy
of using the gst_adapter_prev_pts timestamp to get the base timestamp
for the media of the fragment would fail as it would reuse the same
base timestamp for all moofs in the buffer instead of accumulating
the durations for all of them.

Heres a better explanation of the issue:
qtdemux receives a buffer where PTS(buf) = X
buf -> moofA | moofB | moofC

The problem was that PTS(buf) was used as the base timestamp for
all 3 moofs, causing all buffers to be X based. In this case we want
only moofA to be X based as it is what the PTS on buf means, and the
other moofB and moofC just use the accumulated timestamp from the
previous moofs durations.

To solve this, this patch uses gst_adapter_prev_pts distance
result, this allows qtdemux to calculate if it should use the
resulting pts or just accumulate the samples as it can identify
if the moofs belong to the same upstream buffer or not.

https://bugzilla.gnome.org/show_bug.cgi?id=719783
2013-12-04 10:36:38 -03:00
Wim Taymans
0d55724a2b audioparsers: don't leak template caps 2013-12-04 09:12:07 +01:00
Wim Taymans
e0a5c07e8d audioparsers: use ACCEPT_INTERSECT flag
The parser can accept input that is not completely specified. Use the
ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
check for intersection only. This allows us to proxy downstream
constraints while still allowing non-subset caps as input.
We can then also remove the appended template caps workaround.
Make a unit-test to check the new feature.

This reverts commit 26040ee38c

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024
2013-12-03 22:26:44 +01:00
Wim Taymans
e3f393f7e6 audioparsers: remove fields from filter
We need to remove the fields from the filter when we can convert
between them.
2013-12-03 21:39:57 +01:00
Wim Taymans
e8313a1e70 audioparsers: refactor code to remove caps fields 2013-12-03 21:29:13 +01:00
Tim-Philipp Müller
a424fb289b deinterlace: microoptimisation: avoid some unnecessary GValue copies 2013-12-02 00:10:43 +00:00
Tim-Philipp Müller
63b0e84add deinterlace: fix off-by-one crash when downstream caps contain a list of framerates
https://bugzilla.gnome.org/show_bug.cgi?id=719544
2013-12-01 23:33:04 +00:00
Thiago Santos
079dde49ed qtdemux: Use the timestamp of the moof as the base fragment start
In SmoothStreaming fragmented scenario, the timestamps are calculated
starting from the fragment buffer timestamp. When there is a not-linked
return from downstream, qtdemux will return upstream and will keep the
non-pushed data into its adapter.

On a new fragment buffer pushed to qtdemux, the new buffer timestamp
would overwrite the previous one that should be used on the still
to be pushed buffers. Because of this, this patch will also
update the fragment_start timestamp from the adapter last pts
to make sure the moof and timestamps are in sync and will result
in correct timestamps for all fragments.
2013-11-29 17:28:48 -03:00
Thiago Santos
45c16599ff qtdemux: avoid re-reading the same moov and entering into loop
In the scenario of "mdat | moov (with fragmented artifacts)" qtdemux
could read the moov again after the mdat because it was considering the
media as a fragmented one.

To avoid this loop this patch makes it store
the last processed moov_offset to avoid parsing it again.
And it also checks if there are any samples to play before
resturning to the mdat, so that it knows there is new data to be played.

https://bugzilla.gnome.org/show_bug.cgi?id=691570
2013-11-29 17:28:48 -03:00
Thiago Santos
fcc78aa3bd qtdemux: do not free streams if they were not created locally
When parsing a trak only free streams on failures if those streams
were created locally. They could have been created from a previous
fragment, in this case we they have valid info from the other fragment.
Including pads.

https://bugzilla.gnome.org/show_bug.cgi?id=691570
2013-11-29 17:28:48 -03:00
Sebastian Dröge
220a947dc7 videomixer: Simplify NV12/21 blending code macros 2013-11-29 19:57:46 +01:00
Sebastian Dröge
b0529e0fe8 videomixer: Fix segfault when filling the background of a UYVY frame
https://bugzilla.gnome.org/show_bug.cgi?id=712401
2013-11-29 19:52:34 +01:00
Tim-Philipp Müller
4278ab18ff qtdemux: fix compilation with gst debuging disabled
qtdemux.c:9452:1: error: label at end of compound statement
2013-11-29 09:21:52 +00:00
Jonas Holmberg
0ab0421759 rtph264pay: Map inbuffer once only
Do not call gst_buffer_extract() twice since each call will map and
unmap the biffer.

https://bugzilla.gnome.org/show_bug.cgi?id=719434
2013-11-28 16:08:40 -05:00
Tim-Philipp Müller
b8f689a9d9 videoflip: don't crash on tag events without orientation tag
Would crash in g_free() trying to free an uninitialised pointer.

https://bugzilla.gnome.org/show_bug.cgi?id=719497
2013-11-28 16:09:04 +00:00
Wim Taymans
e8edecc56e rtpsession: don't unref buffer twice
Cleaning the packet info will already unref the buffer.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078
2013-11-28 16:51:13 +01:00
Jan Schmidt
b3b89dfec1 qtdemux: Add HydrogenAudio ReplayGain tags
Identical to the itunes (tm) version, but labelled with
org.hydrogenaudio.replaygain as the producer.
2013-11-28 22:36:44 +11:00
Mathieu Duponchelle
532598e360 videomixer: explicitly fail when alpha information would have been lost. 2013-11-27 16:35:46 +01:00
Sebastian Dröge
fb14f66696 matroska-demux: Allow a bit more variation when detecting common framerates
Instead of +/- 1ns we allow 2ns now. Due to rounding errors there are
some Matroska files out there with 33.333331ms per frame for 30fps.
2013-11-26 11:17:42 +01:00
Sebastian Dröge
20ad174679 matroska-demux: Use gst_util_double_to_fraction() instead of GValue magic 2013-11-26 10:21:04 +01:00
Nicolas Dufresne
c42bc9efa0 videoflip: Set default method at contruction
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712333
2013-11-25 14:03:21 -05:00
Wim Taymans
710d1f3a2a rtpjitterbuffer: improve clear-pt-map handling
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
2013-11-25 15:52:22 +01:00
Jan Schmidt
fdfc6a2a86 qtdemux: Discard 2 byte subpicture packets
As for text subtitles and as suggested in #712643, throw
away the 2 byte terminator packets that some encoders insert.

This will make things better when remuxing and causes generation
of gap events.
2013-11-25 12:24:22 +11:00
Tim-Philipp Müller
901ec63462 rtpjitterbuffer: fix wake-up when new buffers come in after running empty
Spotted by 'gratias' on IRC. Probably introduced in recent refactoring.

https://bugzilla.gnome.org/show_bug.cgi?id=715039
2013-11-25 00:37:50 +00:00
Mark Nauwelaerts
643e6fdc36 matroskamux: correctly handle negative relative timestamps
... rather than scaling these as unsigned.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712744

Based on patch by Krzysztof Kotlenga <pocek@users.sf.net>
2013-11-23 12:25:05 +01:00
MathieuDuponchelle
83f8ee1d41 videomixer2: Merge tag events to send them in collected.
Otherwise there were race conditions where we would send tags
on a flushing srcpad.

We have a test for that in GES, but this should be tested
systematically with harness in the future as I believe it
is useful for exactly that kind of cases.

https://bugzilla.gnome.org/show_bug.cgi?id=708165
2013-11-22 18:54:35 -03:00
Thibault Saunier
a45d470236 qtdemux: Use GstVideoInfo helper to create caps for raw video
This way we do not miss mandatory fields in caps.
At the same time use the gst_pb_utils_get_codec_description
helper to get codec description.

https://bugzilla.gnome.org/show_bug.cgi?id=712335
2013-11-22 18:52:54 -03:00
Thibault Saunier
6ff7522ba2 matroskademux: Use GstVideoInfo helper to create caps for raw video
This way we do not miss mandatory fields in caps.
At the same time use the gst_pb_utils_get_codec_description helper to
get codec description.

https://bugzilla.gnome.org/show_bug.cgi?id=712328
2013-11-22 18:52:54 -03:00
Thibault Saunier
1fc591238b multifilesrc: Implement seeking in case of multiple images
https://bugzilla.gnome.org/show_bug.cgi?id=712254
2013-11-22 18:52:54 -03:00
Wim Taymans
4c9474905b rtpjitterbuffer: pass downstream flowreturn to upstream
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712722
2013-11-22 12:27:31 +01:00
Tim-Philipp Müller
d9c2914c90 g_memmove() is deprecated
Just use plain memmove(), g_memmove() is deprecated in
recent GLib versions.

https://bugzilla.gnome.org/show_bug.cgi?id=712811
2013-11-21 15:30:34 +00:00
Wim Taymans
3a1199c2f7 rtpvorbisdepay: handle packets > 0xffff
Handle input packet sizes larger than 16 bits in the depayloader.
Remove size restrictions on the payloader.
2013-11-21 11:32:15 +01:00
Wim Taymans
43e9b56122 rtptheoradepay: handle packets > 0xffff
Reorganize some things in the depayloader so that it can handle packets larger
than 16 bits.
Remove the size restriction on the payloader.
2013-11-21 11:30:28 +01:00
Jan Schmidt
81e2c8130a isomp4: Handle mp4s subpicture streams better.
Clean up the handling of mp4s streams. Use the generic esds
descriptor function to extract the palette, instead of hard coding
a wrong magic offset.

Add some more size safety checks when parsing ES descriptors, and
replace magic numbers with the descriptive constants that are already
defined.

Enhance dump output for stsd atoms.

Streams from both bug 712643 and historic bug 568278 now both work
correctly.

Fixes: #712643
2013-11-21 02:28:27 +11:00
Jan Schmidt
217d2d8deb qtdemux: Sort fourcc declarations and remove duplicates 2013-11-20 22:08:25 +11:00
Jan Schmidt
b6f581eecc qtdemux: Merge all the fourcc headers into one
Remove qtdemux_fourcc.h and ftypcc.h and put it all in fourcc.h
2013-11-20 21:48:03 +11:00
Wim Taymans
0c6f4efe4a rtpjitterbuffer: avoid mapping the buffer
Reuse the parsed structure to get the timestamps.
2013-11-19 10:12:00 +01:00
Tim-Philipp Müller
28f524a551 rtspsrc: fix 'make check'
Fix generic/states check. Also, g_return_if_fail() is
not for internal state checking.
2013-11-18 17:13:49 +00:00
Tim-Philipp Müller
d506409af5 docs: get rid of 'Since: 0.10.x' markers
And some gtk-doc markup fixes.
2013-11-18 14:47:35 +00:00
Tim-Philipp Müller
548e756e0a rtpmanager: fix Since markers
Should be next stable release series version
2013-11-16 12:15:14 +00:00
George Kiagiadakis
387e3b918a rtpjitterbuffer: Fix stats property field names and documentation 2013-11-15 16:23:34 +02:00
Torrie Fischer
acf74435e3 gstrtpsession: Implement a number of feedback packet statistics
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693
2013-11-15 15:21:19 +01:00
Thiago Santos
cfdadd4114 qtdemux: remove math operation from loop
The elst_offset doesn't change inside the loop, so compute it
outside
2013-11-14 18:15:20 -03:00
Stefan Sauer
1a4e7338d9 qtmux: fix playback regression
In ae1150e85c flipping a condition misaligned the
else branch, where for there condition that was change there is none.
Fixes #712303
2013-11-14 20:56:36 +01:00
Wim Taymans
b450d31503 rtpjitterbuffer: rename property to 'stats'
This makes the unit test work.
We can later also add more stats, not specific to retransmission.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711411
2013-11-14 09:24:26 +01:00
Torrie Fischer
22ceb80ba9 rtpjitterbuffer: implement rtx statistics 2013-11-14 09:24:26 +01:00
Wim Taymans
2e5b462ae3 jitterbuffer: advance expected seqnum after dropping
After dropping a buffer, move our expected seqnum

Conflicts:
	gst/rtpmanager/gstrtpjitterbuffer.c
2013-11-13 12:02:57 +01:00
Wim Taymans
a065b4fcde gstpay: only send one caps
Only send one caps in a packet. Two caps can happen when setcaps is called and
the config-interval expires at the same time.
2013-11-13 12:02:57 +01:00
Sebastian Dröge
9ae6981578 rtspsrc: Use the synced buffer mode in auto mode if a clock provider is in the SDP 2013-11-13 10:54:19 +01:00
Wim Taymans
e4bc81d7d2 rtpsession: remove collision reconfigure event
Remove bogus reconfigure event on collision, we don't want to send the event on
the receiving RTP pad and the collision event is now handling this
case.

See https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-11-11 15:27:18 +01:00
Julien Isorce
b32fc6f416 gstrtpsession: send custom upstream event "GstRTPCollision" on send_rtp_sink pad
See https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-11-11 15:25:52 +01:00
Mark Nauwelaerts
49d52a64d6 ac3parse: correctly handle timestamps when parsing x-private1-ac3
... the way it has always worked fine in a52dec.
2013-11-11 13:35:29 +01:00
George Kiagiadakis
b81b2efa3e rtpjitterbuffer: fix crash when do-retransmission=true and a lot of buffers are lost
The problem here was that the jitterbuffer lock was unlocked to push
the event, but that caused another thread to remove the timer currently
being processed, probably because the amount of rtx events
(and therefore timers) was getting too high. The solution is to
unlock and push the event only after timer processing has finished.

fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131
2013-11-11 11:51:45 +01:00
Per x Johansson
b3e0b1dbca matroskademux: Avoid division by zero assert in gst_matroska_demux_search_pos
https://bugzilla.gnome.org/show_bug.cgi?id=711829
2013-11-11 11:30:54 +01:00
Philippe Normand
0ee332378b wavenc: generate a non-empty data header
Restore the behavior of the element to the state before commit
db29522a43. A non-empty header is
generated and when the EOS event is received the header is generated
again, this time with the correct size.

https://bugzilla.gnome.org/show_bug.cgi?id=711699
2013-11-09 11:22:12 +01:00
Wim Taymans
c8db05d610 rtpsource: update receiver stats for sender
An internal sender in a session is also a receiver of its own packets so update
the receiver stats. Other senders in the session will use this info to generate
correct RB blocks in their SR reports.
2013-11-07 16:24:30 +01:00
Wim Taymans
268a75e705 rtpsource: refactor receiver stats update 2013-11-07 16:24:30 +01:00
Thiago Santos
33ebda8ecf qtdemux: handle fragmented files with mdat before moofs
Assume a file with atoms in the following order: moov, mdat, moof,
mdat, moof ...

The first moov usually doesn't contain any sample entries atoms (or
they are all set to 0 length), because the real samples are signaled
at the moofs. In push mode, qtdemux parses the moov and then finds the mdat,
but then it has 0 entries and assumes it is EOS.

This patch makes it continue parsing in case it is a fragmented file so that
it might find the moofs and play the media.

https://bugzilla.gnome.org/show_bug.cgi?id=710623
2013-11-07 11:22:04 -03:00
Thiago Santos
0e78ffc9d6 qtdemux: When using a buffered mdat, store all received data for later use
In push mode, when qtdemux can't use a seek to skip the mdat buffer it has
to buffer it for later use.

The issue is that after parsing the next moov/moof, there might be some
trailing bytes from the next atom in the file. This data was being discarded
along with the already parsed moov/moof and playback would fail to continue
after the contents of this moov/moof are played.

This is particularly bad on fragmented files that have the mdat before the
corresponding moof. So you'd get:

mdat|moof|mdat|moof ...

When a moof was received, it usually came with some extra bytes that would
belong to the next mdat (because upstream doesn't care about atoms alignment).
So those bytes were being discarded and playback would fail.

This patch makes qtdemux store those extra bytes to reuse them later after the
mdat is emptied.

https://bugzilla.gnome.org/show_bug.cgi?id=710623
2013-11-07 11:22:03 -03:00
Sebastian Dröge
fd89e36c8a multiudpsink: Also use the bind-port property if no bind-address was given 2013-11-07 09:50:39 +01:00
Sebastian Dröge
111982de28 rtpvp8pay: Make Picture ID mode configurable and default to no picture ID
Some implementations (linphone) only support no picture at all in the
stream and will fail if one is provided.

https://bugzilla.gnome.org/show_bug.cgi?id=711497
2013-11-05 17:26:49 +01:00
Paul HENRYS
8eceb8f327 Add call to gst_rtp_h264_pay_clear_sps_pps() when receiving a STREAM_START event
https://bugzilla.gnome.org/show_bug.cgi?id=692787
2013-11-04 14:36:28 -05:00
Rico Tzschichholz
b137f79581 rtsp: Add missing gio-2.0 deps and includes 2013-11-02 23:12:13 +01:00
Sebastian Dröge
f180f3d1ba audioiirfilter: Fix initialization coefficient handling
Broke unit test.
2013-11-01 18:31:36 +01:00
Aleix Conchillo Flaque
82b8374af8 rtspsrc: allow setting tls certificate validation flags
Added a new property "tls-validation-flags". If the url transport is
TLS, the validation flags will be set to the rtsp connection.

https://bugzilla.gnome.org/show_bug.cgi?id=711230
2013-11-01 16:47:36 +01:00
Sebastian Dröge
2559557ff1 audioiirfilter: Don't crash if no filter coefficients are provided
...and by default use a identity filter.

https://bugzilla.gnome.org/show_bug.cgi?id=710215
2013-10-31 22:43:49 +01:00
Wim Taymans
e96f8f519c rtspsrc: proxy new buffer mode 2013-10-31 10:38:35 +01:00
Wim Taymans
43645d5981 jitterbuffer: add new timestamp mode
Add a new timestamp mode that assumes the local and remote clock are
synchronized. It takes the first timestamp as a base time and then uses the RTP
timestamps for the output PTS.
2013-10-31 10:15:25 +01:00
Sebastian Dröge
4a8082856a matroska-demux: Fix compiler warning
matroska-demux.c: In function 'gst_matroska_demux_add_stream':
matroska-demux.c:1379:7: error: format '%u' expects argument of type 'unsigned int', but argument 4 has type 'guint64' [-Werror=format=]
       "%03u", context->uid);
       ^
2013-10-30 22:13:06 +01:00
Matthieu Bouron
52d0588c21 videomixer: remove unneeded guint comparaison
https://bugzilla.gnome.org/show_bug.cgi?id=711010
2013-10-29 16:38:26 +00:00
Matthieu Bouron
ec8c141d6a y4menc: fix uninitialized variable warning
https://bugzilla.gnome.org/show_bug.cgi?id=711011
2013-10-28 14:20:13 +00:00
Thiago Santos
2eec7909aa qtdemux: check if the end_time is defined before using it
Avoids sending EOS too soon because of overflow. Can happen on
fragmented mp4 playback.
2013-10-25 11:30:36 -03:00
Thiago Santos
673301ef48 qtdemux: use correct unref function
Events aren't GstObjects, but GstMiniObjects
2013-10-23 13:38:56 -03:00
Stefan Sauer
ae1150e85c qtdemux: rename chunks_are_chunks to chunks_are_samples and flip the logic
As the variable name suggests, sometimes chunks are chunks. Rename the variable
to tell what they are when they are not chunks.
2013-10-15 09:53:30 +02:00
Stefan Sauer
6789ba1ece qtdemux: fix typos and add more logging for unhandled parts 2013-10-15 09:53:30 +02:00
Ognyan Tonchev
c81ce6b152 multiudpsink: Fix memory leak
Unmap all GstMemory of the current buffer when flushing.

https://bugzilla.gnome.org/show_bug.cgi?id=710110
2013-10-14 18:21:54 +02:00
Tim-Philipp Müller
771ffe5609 flvmux: fix broken sample pipeline
which was muxing raw audio and video into flvmux, which won't work,
even if there were converters.
2013-10-12 20:44:31 +01:00
Tim-Philipp Müller
29effb522a flvmux: require stream-format=raw for mpeg-2 too, but don't require framed field
raw implies that it's framed already. Fixes .. ! faac ! flvmux
2013-10-12 20:37:41 +01:00
Sebastian Dröge
b8f9e966d5 wavenc: A-Law and Mu-Law don't have width/depth/signed caps fields
https://bugzilla.gnome.org/show_bug.cgi?id=709614
2013-10-08 11:28:04 +02:00
Sebastian Dröge
a5bf9f24c9 deinterlace: Fix handling of planar video formats in greedyh method
https://bugzilla.gnome.org/show_bug.cgi?id=709507
2013-10-07 12:54:11 +02:00
Reynaldo H. Verdejo Pinochet
38c5e5efdc matroska: Trivial grammar fix on debug msg 2013-10-06 10:02:09 -07:00
Reynaldo H. Verdejo Pinochet
1cb31eeacc matroskamux: Add context flag for WebM
WebM has a couple of specific requirements we need to handle.
Idea is to set this flag once and just rely on mux->is_webm
at run time instead of repeatedly figuring this out from
GST_MATROSKA_DOCTYPE_WEBM (which requires a strcmp()).
2013-10-06 09:54:28 -07:00
Reynaldo H. Verdejo Pinochet
edeed575ae matroska: Do not write SegmentUID for WebM mux
WebM spec states SegmentUID is Unsupported. Files produced
with gstreamer without this change will spit an error like
this when passed to mkvalidator:

ERR201: Invalid 'SegmentUID' for profile 'webm' in Info at 192
2013-10-06 08:12:50 -07:00
Matej Knopp
cf12017ef8 matroskademux: make dvd palette change event sticky
So they don't get lost.

https://bugzilla.gnome.org/show_bug.cgi?id=709454
2013-10-05 10:55:03 +01:00
Nicolas Dufresne
ed77b22f2b videoflip: Add automatic flip mode driven by image-orientation tag
https://bugzilla.gnome.org/show_bug.cgi?id=709312
2013-10-04 14:52:57 -04:00
Wim Taymans
d4892859d4 jitterbuffer: fix race in flush-start/flush-stop
When flush-stop arrives before we process the result of the _push() in the
loop function, we might pause even though we are not flushing anymore. Fix this
race by waiting for the srcpad loop function to completely pause after doing the
flush-start.
2013-10-04 12:35:18 +02:00
Mathieu Duponchelle
ef548c2b28 videomixer: Update videoconvert copy
https://bugzilla.gnome.org/show_bug.cgi?id=709390
2013-10-04 10:57:36 +02:00
Mathieu Duponchelle
3d780c5c6d videomixer: Check if the pad needs reconfiguration in collected
https://bugzilla.gnome.org/show_bug.cgi?id=709384
2013-10-04 10:53:26 +02:00
Sebastian Dröge
21947f9d13 qtdemux: Add support for the mp2v fourcc for MPEG-2 video
https://bugzilla.gnome.org/show_bug.cgi?id=709270
2013-10-03 11:59:25 +02:00
Ognyan Tonchev
30f62a2eec matroskademux: Fix memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=709266
2013-10-02 16:17:33 +02:00
Sreerenj Balachandran
e779b6587b qtdemux: Add HEVC support
https://bugzilla.gnome.org/show_bug.cgi?id=709093
2013-10-02 11:54:24 +02:00
Ognyan Tonchev
93d5e182d2 rtpgstpay: Fix memory leak
We were leaking the GList nodes of the pending buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=709079
2013-10-02 11:07:16 +02:00
Wim Taymans
00056965e8 rtpjitterbuffer: fix race when updating the next_seqnum
If we were not waiting for the missing seqnum when we insert the lost packet
event in the jitterbuffer, we end up not updating the next_seqnum and wait
forever for the lost packets to arrive. Instead, keep track of the amount of
packets contained by the jitterbuffer item and update the next expected
seqnum only after pushing the buffer/event. This makes sure we correctly handle
GAPS in the sequence numbers.
2013-09-30 12:31:00 +02:00
Wim Taymans
fde438791e rtpjitterbuffer: small debug improvement 2013-09-30 12:30:23 +02:00
Wim Taymans
6e7d547be4 rtpjitterbuffer: reset skew does not reset clock-rate
Don't reset the clock-rate when we reset the skew correction algorithm.
Reset the skew correction algorithm when we change the clock-rate.
2013-09-30 11:53:08 +02:00
Wim Taymans
03d520eb69 rtpjitterbuffer: pause timer when PAUSED
Also pause the timer when we go to the PAUSED state. It is possible that we
don't have a clock or base-time in PAUSED to perform the timeouts.
2013-09-30 11:16:32 +02:00
Wim Taymans
4a31aec513 rtpjitterbuffer: improve debug 2013-09-30 11:15:25 +02:00
Hans Månsson
041946423a mp4mux: Do not require framerate in peer video caps
Remove the framerate restriction on the caps.

Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708864
2013-09-28 13:02:11 +02:00
Wim Taymans
8c5ce0dbdc rtspsrc: also go into the loop function after connect
When we have opened the stream, go into the loop function so that we can
receive messages from the server.
2013-09-27 15:08:31 +02:00
Matej Knopp
40c0586c17 matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8
https://bugzilla.gnome.org/show_bug.cgi?id=707933
2013-09-27 14:38:19 +02:00
Wim Taymans
d4b4b4e924 rtpjitterbuffer: don't calculate skew without rtptime
Skip trying to calculate the skew when we don't have an rtptime.
It causes problems when lost packet events are placed in the jitterbuffer.
2013-09-26 16:21:33 +02:00
Wim Taymans
6095e2e859 rtspsrc: disable checks when linking pads
We know the pad links will work (and we don't check the return value
anyway).
2013-09-25 17:42:02 +02:00
Wim Taymans
2efd58fc84 rtpbin: avoid some pad link checks
Link pads without checks, we know it will work.
2013-09-25 17:38:31 +02:00
Sebastian Dröge
4a91a93d4e qtmux: Don't error out if downstream is not seekable for non-fragmented variants
Doing so would be a regression over 1.0 and breaks the unit test.
However the result will be most likely unusable, so let's post
a warning message on the bus.
2013-09-25 13:25:34 +02:00
Wim Taymans
97f4674655 rtpjitterbuffer: calculate some stats 2013-09-25 10:50:05 +02:00
Wim Taymans
b1d29483bb rtpjitterbuffer: move send_lost_event function
Move the send_lost_event function to the do_lost_event handling, there is no
need to have a separate function.
2013-09-25 10:50:05 +02:00
Thiago Santos
dc02d91c14 qtdemux: add code to parse creation time earlier than 1970
Use g_date_time seconds manipulation to allow to cover the quicktime
spec for creation_time. It uses seconds since 1904.

Both paths could be done using the generic approach of seconds since
1904 with GDateTime handling, but the first path using seconds from
1970 should be more commonly found and avoids a few objects creation and
ref/unref, so keep it there for performance.

Additionally, the code for handling seconds since 1970 changed from >
to >= because having 0 seconds since 1970 is also a valid case for that
path to handle.

https://bugzilla.gnome.org/show_bug.cgi?id=707975
2013-09-24 15:16:54 -07:00
Matej Knopp
a1a493dae4 matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily
https://bugzilla.gnome.org/show_bug.cgi?id=708505
2013-09-24 15:12:44 -07:00
Wim Taymans
adf5d96044 rtpmanager: update docs 2013-09-23 16:34:15 +02:00
Wim Taymans
e5019de80d docs: update docs with 1.0 element names 2013-09-23 15:36:47 +02:00
Wim Taymans
8ce674da87 rtpjitterbuffer: always store lost event in jitterbuffer
Always prepare a lost event in the jitterbuffer, it is to wake up and make the
pushing thread continue. We drop the event when we are not supposed to push lost
events downstream.
2013-09-23 14:45:27 +02:00