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Author SHA1 Message Date
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
Wim Taymans
9f3345fcc2 rtpjitterbuffer: schedule lost event differently 
Schedule the lost event by placing it inside the jitterbuffer with the seqnum
that was lost so that the pushing thread can interleave and push it properly.
 2013-09-23 14:45:27 +02:00
Wim Taymans
ae389aeb0c rtpjitterbuffer: remove list debug 2013-09-23 14:45:26 +02:00
Wim Taymans
28641e3145 rtpjitterbuffer: add type to the item 
So that the upper layer can know what data is contained in the item.
 2013-09-23 14:45:26 +02:00
Wim Taymans
479c7642fd rtpjitterbuffer: fix flush 
Pass function to flush to properly free the queue items.
 2013-09-23 14:45:25 +02:00
Wim Taymans
0cc887eb98 rtpjitterbuffer: append seqnum -1 packets 2013-09-23 14:45:25 +02:00
Wim Taymans
39a2ba669d rtpjitterbuffer: use structure to hold packet information 
Make the jitterbuffer operate on a structure containing all the packet
information. This avoids mapping the buffer multiple times just to get the RTP
information. It will also make it possible to store other miniobjects such as
events later.
 2013-09-23 14:45:25 +02:00
Wim Taymans
1760817005 rtpjitterbuffer: update expected timer when possible 
When we receive a packet and we have some missing packets, we can update their
estimated arrival times based on the timestamp difference.
 2013-09-23 14:45:25 +02:00
Wim Taymans
fdc1ed1680 rtpjitterbuffer: fix order of timeout events 
Improve the order of the timeout events, if there are timers with the same
timeout, we want to trigger the lowest seqnum first. For this we need to loop
over the complete array of timers to find the best one before triggering the
timeout.
 2013-09-23 14:45:25 +02:00
Wim Taymans
0b1a7edfea rtpjitterbuffer: send lost event before signaling next buffer 
First send the lost event, then update the next_seqnum counter and then
send the signal to the pushing thread that it can retry to push a buffer. This
avoids pushing out buffers before the lost event is pushed.
 2013-09-23 14:45:25 +02:00
Wim Taymans
5051f51f0a jitterbuffer: configure clock-rate on jitterbuffer 
Add a get and setter to configure the clock-rate in the jitterbuffer instead of
passing it as an argument to the insert method.
 2013-09-23 14:45:24 +02:00
Wim Taymans
3c421e7e48 rtpjitterbuffer: add option to reset retransmission timers 2013-09-23 14:45:24 +02:00
Wim Taymans
6f4deab298 rtpjitterbuffer: stop the timer thread 
The timeout code could release the lock so we need to check if we are allowed to
wait for the clock some more.
 2013-09-23 14:45:24 +02:00
Wim Taymans
cba4e6a707 rtpjitterbuffer: unlock only once 2013-09-23 14:45:24 +02:00
Wim Taymans
5dc207948c rtpjitterbuffer: improve flush and shutdown 
There is no need to unschedule the timer in flush-start, flush-stop will remove
the timers and unschedule.
Unschedule the current timer before attempting to join the timer thread.
 2013-09-23 14:45:23 +02:00
Wim Taymans
a512cc2d3c rtpjitterbuffer: set correct expected time 
When we already have a timer for a packet, skip it but don't forget to adjust
the dts to the expected dts of the next packet.
 2013-09-23 14:45:23 +02:00
Wim Taymans
517ea0f4d9 jitterbuffer: improve debug 2013-09-23 14:45:23 +02:00
Wim Taymans
c395bf62dd alpha: use POFFSET instead of OFFSET 
Use the more correct POFFSET macro to get the offset of a component in its
plane. The offset macro gives the offset of the component relative to the start
of the frame.
 2013-09-23 14:45:23 +02:00
Sebastian Dröge
94ad6724ba goom: Fix MMX assembly compilation with clang 
clang does not want or need a clobber list for emms:
error: clobbers must be last on the x87 stack

Patch taken from the FreeBSD ports, provided by
Dan McGregor <dan.mcgregor@usask.ca>
 2013-09-21 18:48:19 +02:00
Sebastian Dröge
d8841b4832 matroska-demux: Make sure that subtitle buffers are \0-terminated 
https://bugzilla.gnome.org/show_bug.cgi?id=707933
 2013-09-20 10:22:40 +02:00
Andoni Morales Alastruey
cfefdaebb6 qtmux: handle issues correctly when downstream is not seekable 
The streamable property only make sense for fragmented formats.
For regular MP4, when downstream is not seekable we can't rewrite
the headers, so qtmux can only work with fast-start=TRUE, where
the headers are written finishing the file.
For fragmented MP4, when streamable is not seekable and the streamable
property is FALSE, we must enforce streamable=TRUE warning the user
about this change

https://bugzilla.gnome.org/show_bug.cgi?id=707242
 2013-09-20 10:09:48 +02:00
Andoni Morales Alastruey
9ae5082204 qtmux: make "streamable" TRUE as default 
The most common use case for fragmented MP4 (Dash and Smooth Streaming)
is producing streamable content (even for VOD). streamable=FALSE would only
be used to generate fragmented MP4 with and index of MOOF's that could
be reproduced without a playlist/manifest
https://bugzilla.gnome.org/show_bug.cgi?id=707242
 2013-09-20 10:09:48 +02:00
Andoni Morales Alastruey
5732684e18 qtmux: deprecate the streamable property for non-fragmented MP4 
The streamable property only makes sense for fragmented MP4.
https://bugzilla.gnome.org/show_bug.cgi?id=707242
 2013-09-20 10:09:48 +02:00
Wim Taymans
926e2fa93b alpha: don't assume planar formats have just 1 block 
Don't assume planar formats have just one memory block with the data but use the
macros to access the right memory block where a component can be found.
 2013-09-19 16:50:44 +02:00
Wim Taymans
fd6c57cf10 rtpjitterbuffer: keep delay as a separate variable in timer 
Keep a separate delay in the timer so that we still know the original timestamp
of the packet that this timer refers to. We can then place the correct
running-time in the Retransmission event.
 2013-09-19 14:32:48 +02:00
Wim Taymans
d34184dd03 rtpjitterbuffer: fix writability of properties 2013-09-19 14:32:48 +02:00
Wim Taymans
6bb2626498 rtpjitterbuffer: reevaluate the current timer after timeout 
When we trigger the timeout logic of a timer, reevaluate it because it is
possible that it still has the lowest timeout.
 2013-09-18 16:33:40 +02:00
Wim Taymans
8d021b6ede rtpjitterbuffer: don't update time when unscheduled 
Don't try to estimate the current time when we got unscheduled.
 2013-09-18 16:31:26 +02:00
Wim Taymans
65606a25bf rtpjitterbuffer: init packet spacing on first buffer 
Already init the packet spacing variables on the first buffer so that we can
calculate the spacing on the second buffer already.
 2013-09-18 16:29:37 +02:00
Wim Taymans
f2efdf28f5 rtpjitterbuffer: push the lost event from the timer thread 
Instead of pushing the lost event from the chain function, schedule a timeout
that will push the lost event from the timer thread. This avoid blocking the
upstream thread while we push and sync the event.
 2013-09-18 14:57:09 +02:00
Wim Taymans
5d5fc03e04 rtpjitterbuffer: round gap duration to multiple of duration 
Make sure the gap duration in the lost event is a multiple of the packet
duration.
Enable another test.
 2013-09-18 14:12:47 +02:00
Wim Taymans
6e4a051d40 rtpjitterbuffer: keep track of duration 
Keep track of the estimated duration of missing packets and use it in the lost
event.
Enable another unit test
 2013-09-18 12:29:38 +02:00
Wim Taymans
ac3bb3acf6 rtpjitterbuffer: handle large gaps with one lost event 
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
 2013-09-18 11:59:28 +02:00
Wim Taymans
26402e1c21 rtpjitterbuffer: refactor lost event sending 
Also make sure we only increment the expected seqnum and last
output timestamp.
 2013-09-18 11:57:06 +02:00
Wim Taymans
f49981a597 jitterbuffer: refactor timeout triggers 2013-09-17 23:29:56 +02:00
Wim Taymans
047021c443 jitterbuffer: simplify the timeout code 
Keep track of the current time in the timeout loop.
Loop over all timers and trigger all the expired ones, we can do this in the
same loop that selects the new best timer.
 2013-09-17 23:29:56 +02:00
Wim Taymans
fa1ef3328b jitterbuffer: rearrange timer update code 
Also update the timers when retransmission is disabled. We need to
do this because when we added LOST timers when we detected missing packets and
we need to remove those timers when the packet finally arrives.
 2013-09-17 23:29:56 +02:00
Tim-Philipp Müller
7a76595b22 videomixer: link to libm for maths stuff 
Fixes undefined references to rint and pow on ubuntu
build bot.
 2013-09-17 22:02:04 +01:00
Wim Taymans
232fdd8b56 jitterbuffer: release lock on shutdown 2013-09-17 15:19:42 +02:00
Matej Knopp
b2982bb749 qtmux: remove MAX_TOLERATED_LATENESS 
https://bugzilla.gnome.org/show_bug.cgi?id=707411
 2013-09-16 11:11:12 -03:00
Wim Taymans
4de919a17a jitterbuffer: use separate thread for timeouts 
Use a separate thread for scheduling the timeouts instead of using the
downstream streaming thread that might block at any time.
 2013-09-16 15:55:55 +02:00
Matej Knopp
b363832c2c qtmux: set first_ts to DTS for streams that have DTS 
https://bugzilla.gnome.org/show_bug.cgi?id=707340
 2013-09-16 12:14:00 +02:00
Matej Knopp
39f7e52266 qtmux: make sure duration is a valid number for last buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=707340
 2013-09-16 12:14:00 +02:00
Matej Knopp
4e3c13c87c qtmux: use segment.start or last buffer end time in case of missing DTS 
https://bugzilla.gnome.org/show_bug.cgi?id=707340
 2013-09-16 12:14:00 +02:00
Matej Knopp
85728c04c4 Revert qtmux: Use buffer PTS if DTS is not set" 
This reverts commit f72c3cf71fde622067f41f31a53978ba4c94469d.

https://bugzilla.gnome.org/show_bug.cgi?id=707340
 2013-09-16 12:13:54 +02:00
Sebastian Dröge
d646a34681 videomixer: Update orc generated files 
https://bugzilla.gnome.org/show_bug.cgi?id=708131
 2013-09-16 11:03:06 +02:00
Olivier Crête
b9ceafe5af rtpsession: Demux RTCP buffers from the RTP stream 
If there are RTCP buffers in the RTP stream, process them as
RTCP. This way, we want receive streams following RFC 5761

https://bugzilla.gnome.org/show_bug.cgi?id=687657
 2013-09-13 16:25:49 +02:00
Jan Schmidt
299d3f5c42 rtp: Remove bogus extra caps from L24 template. 
The extra caps entry in the template was making it sometimes
get plugged for any dynamically allocated payload type.
 2013-09-13 23:27:49 +10:00
Wim Taymans
28e5f90988 rtpbin: use PacketInfo for the sender 
Avoid mapping the packet multiple times when sending RTP.
 2013-09-13 14:34:28 +02:00
Wim Taymans
a02c9473d8 rtpbin: store more in the PacketInfo 
Store all info in the PacketInfo so that we can avoid mapping the packet
multiple times.
 2013-09-13 14:34:28 +02:00
Wim Taymans
e5c789abd6 session: store more in the PacketInfo structure 2013-09-13 14:34:28 +02:00
Wim Taymans
47662f9ca4 rtpbin: RTPArrivalStats -> RTPPacketInfo 
Rename a structure because we are also going to use this for the sender
bits.
 2013-09-13 14:34:28 +02:00
Wim Taymans
c795b72988 source: small cleanups 2013-09-13 14:34:27 +02:00
Thiago Santos
566b0dce40 qtdemux: only update stop position if seek requests it 
Check for GST_SEEK_TYPE_NONE for stop poistion and only update
the stop time if it is requested. Otherwise just maintain whatever
was stored at the segment

https://bugzilla.gnome.org/show_bug.cgi?id=707530
 2013-09-13 09:21:12 -03:00
Rico Tzschichholz
8ed1ff6821 rtp: Add missing headers tp fix make dist 
In addition to a956a6ceb2
 2013-09-13 14:06:13 +02:00
Sebastian Dröge
b95ddd55cd flacparse: Make sure we have enough data to read image tags 
Thanks to iputinei for reporting this on IRC.
 2013-09-12 15:39:51 +02:00
Wim Taymans
9f9ba21404 jitterbuffer: handle segments with non-0 start 
We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to
transform it back to a buffer timestamp before pushing out the buffer.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931
 2013-09-12 15:04:30 +02:00
Seán de Búrca
9d3dbd6581 matroskademux: Fix off-by-one in validation of UTF-8 
https://bugzilla.gnome.org/show_bug.cgi?id=707933
 2013-09-12 09:19:15 +02:00
Thibault Saunier
9f4a8ccdf4 videomixer: Do not check if caps are empty when they are NULL 
In the case the caps are actually NULL, we should just concider it the
same way as empty caps in that case.
 2013-09-11 14:33:31 -03:00
Seán de Búrca
268058eb37 videomixer: fix build if orc is not installed 
https://bugzilla.gnome.org/show_bug.cgi?id=707886
 2013-09-11 00:17:44 +01:00
Thiago Santos
193ce9110e matroskademux: Preserve seqnum when pushing seek upstream 
After converting a seek from time to bytes, use the same seqnum
on the event that goes upstream
 2013-09-10 17:57:49 -03:00
Thiago Santos
be0eeae491 qtdemux: track streams that are EOS on push mode to finish earlier 
When the segment has a defined stop position, qtdemux should check
when streams reach this position and mark those as EOS. When all
streams are EOS it will return GST_FLOW_EOS to upstream to allow
the pipeline to finish instead of continuously consume buffers
from upstream that are not useful for the segment.

https://bugzilla.gnome.org/show_bug.cgi?id=707530
 2013-09-10 16:43:17 -03:00
Thiago Santos
33cf8b679d qtdemux: preserve stop of segment when doing seeks in push mode 
When handling seeks in push mode, qtdemux converts the seek to bytes
and pushes upstream. It needs to keep track of the seek and the
subsequent segment to be able to map them back to the requested
seek time and properly preserve the segment stop of the seek.

This is done by using the start offset in bytes of the seek,
that should be the same of the segment from upstream. And this
is also backwards compatible with what qtdemux already was using.

https://bugzilla.gnome.org/show_bug.cgi?id=707530
 2013-09-10 16:42:36 -03:00
Mathieu Duponchelle
8db40a8c7f videomixer: Add colorspace conversion 
https://bugzilla.gnome.org/show_bug.cgi?id=704950
 2013-09-10 10:37:23 +02:00
Mathieu Duponchelle
707e39fe7a videomixer: Don't send reconfigure event when formats or PAR are different 
It is racy with multiple pads.

https://bugzilla.gnome.org/show_bug.cgi?id=704950
 2013-09-10 10:36:48 +02:00
Mathieu Duponchelle
8db3648544 videomixer: Bundle private copies of videoconvert code 
Ideally, this would be part of libgstvideo.
Prefixes videoconvert symbols with videomixer_.

https://bugzilla.gnome.org/show_bug.cgi?id=704950
 2013-09-10 10:36:30 +02:00
Wim Taymans
9f9bcbc405 rtspsrc: only wait if we flushed 
Only wait for the STREAM_LOCK when we flushed something when sending
a command for PAUSED or PLAYING.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611
 2013-09-09 15:13:46 +02:00
Wim Taymans
7b2e002879 rtspsrc: return when a flush was issued 
Make gst_rtspsrc_loop_send_cmd() return TRUE when the current
action has been flushed
 2013-09-09 15:13:46 +02:00
David Holroyd
a956a6ceb2 rtp: add L24 pay and depayloader 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734
 2013-09-09 15:13:46 +02:00
Matej Knopp
a5ceab82dd matroskademux: fix leaking buffer and caps 
https://bugzilla.gnome.org/show_bug.cgi?id=707688
 2013-09-07 15:50:36 +01:00
Tim-Philipp Müller
60e72b0254 udpsrc: fix build on win32 
gstudpsrc.c:855:15: error: #if with no expression
 2013-09-05 19:46:37 +01:00
Wim Taymans
5d2ff288b3 avidemux: handle unseekable streams 
Handle streams that we can't seek in and ignore them in the
seek logic.
 2013-09-04 15:53:05 +02:00
Wim Taymans
6f0e8a8b87 avidemux: only check video compression for video streams 
Or else we might deref a stream with a NULL strf.vids and segfault
 2013-09-04 15:53:05 +02:00
Alex Ashley
a965185dee qtdemux: Add support for the avc3 sample entry format of the AVC file format 
Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
structure for fragmented MP4 called "avc3". The principal difference
between AVC1 and AVC3 is the location of the codec initialisation
data (e.g. SPS, PPS). In AVC1 this data is placed in the initial
MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data
goes in the first sample of every fragment (i.e. the first sample in
each mdat box).  The principal reason for avc3 is to make it easier
for client implementations, because it removes the requirement to
insert the SPS+PPS in to the decoder pipeline every time there is a
representation change.

This commit adds support for the "avc3" atom, which is almost identical
to the "avc1" atom, except it does not contain any SPS or PPS data.

https://bugzilla.gnome.org/show_bug.cgi?id=702004
 2013-09-04 13:33:22 +02:00
Mathieu Duponchelle
b68f419b6f videomixer: Don't set EOS to FALSE when the collectpad *is* EOS 
https://bugzilla.gnome.org/show_bug.cgi?id=707238
 2013-09-04 11:09:04 +02:00
Matej Knopp
349afc633a flacparse: cleanup on error after state change 
https://bugzilla.gnome.org/show_bug.cgi?id=707229
 2013-09-03 18:06:18 +02:00
Sebastian Dröge
7f59436979 udpsrc: Bind to multicast addresses on non-Windows systems 
On Windows it's not possible to bind to a multicast address
but the OS will make sure to filter out all packets that
arrive not for the multicast address the socket joined.

On Linux and others it is necessary to bind to a multicast
address to let the OS filter out all packets that are received
on the same port but for different addresses than the multicast
address

And deprecate the multicast-group property and replace it with the
address property.

https://bugzilla.gnome.org/show_bug.cgi?id=707042
 2013-09-03 11:23:24 +02:00
Matej Knopp
73751dbbe7 flacparse: Free GstBaseParseFrame if pushing a header failed 2013-09-03 10:10:49 +02:00
Sebastian Dröge
edf6d28765 udpsrc: Refactor address resolval into its own function 2013-09-03 10:10:49 +02:00
Tim-Philipp Müller
966f848edb replaygain: fix taglist leak in rganalysis 
And add some FIXMEs.
 2013-09-02 23:00:29 +01:00
Sebastian Dröge
1971c43279 flacparse: Properly propagate downstream flow returns upstream 
https://bugzilla.gnome.org/show_bug.cgi?id=707229
 2013-09-02 11:56:33 +02:00
Tim-Philipp Müller
1dfc1f2686 Don't use setlocale in plugins() 
Only apps should call setlocale(), not libraries.
 2013-09-01 21:18:38 +01:00
Wim Taymans
d851b8a8b4 rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay 
RTP buffer allocation should not be done with padding for the specific MPEG2
header as the padding is done at the end of the buffer and the last byte is
the size of the padding.

https://bugzilla.gnome.org/show_bug.cgi?id=706970
 2013-08-29 13:15:15 +02:00
Bernhard Miller
f7528d274b autovideosink: add sync property 
https://bugzilla.gnome.org/show_bug.cgi?id=706955
 2013-08-29 12:23:24 +02:00
Bernhard Miller
2fa68fce07 autoaudiosink: introduce sync property 
https://bugzilla.gnome.org/show_bug.cgi?id=706955
 2013-08-29 12:23:23 +02:00
Thiago Santos
9549289a18 qtdemux: push buffers after segment stop until reaching a keyframe 
This should make decoders able to precisely push buffers until the stop
time in case they need the next keyframe to do it.

Also, according to gst_segment_clip, it should only push a buffer that
the starting ts is strictly smaller than the segment stop, so we change
the min < comparison for <=
 2013-08-28 12:58:56 -03:00
Sebastian Dröge
76293efd72 Release 1.1.4 2013-08-28 12:52:25 +02:00
Wim Taymans
2a8566ddec matroska-mux: remove framerate restriction 
Remove the framerate restriction on the caps.
 2013-08-27 15:25:16 +02:00
Wim Taymans
f1106cde66 session: only update next check time when reconsidering 
Don't update the next RTCP check time in all cases but only when we
reconsidered. This avoids delaying sending a full RTCP packet when we
are doing early feedback.
 2013-08-27 09:55:52 +02:00
Wim Taymans
47065db0b6 session: add more debug 2013-08-27 09:55:52 +02:00
Wim Taymans
454d75951e jitterbuffer: fix types of the retransmission event 2013-08-27 09:55:52 +02:00
Wim Taymans
dd4af0d11c jitterbuffer: only timeout EXPECTED timers on gap 
Only timeout the EXPECTED timers when we detect a large seqnum gap.
 2013-08-27 09:44:18 +02:00
Wim Taymans
4b7bcc2ec1 rtsession: fix locking 
We need to take the session lock when getting and manipulating the
source.
 2013-08-26 11:50:27 +02:00
Wim Taymans
3f46527f75 rtpsession: add some more debug 2013-08-26 11:50:13 +02:00
Mathieu Duponchelle
5d21f8f2e3 videomixer: don't send flush_stop twice. 
If we get flush start and a seek we need to only send flush_stop once.

More info at #706441
 2013-08-23 20:17:11 -04:00
Tim-Philipp Müller
9b0bcc01a0 multipartdemux: propagate discont 2013-08-23 15:57:46 +01:00
Tim-Philipp Müller
c3af414cbf multipartdemux: remove dynamic sourcpads when going from PAUSED to READY 2013-08-23 15:57:46 +01:00
Tim-Philipp Müller
7d78a68c8d multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last 
https://bugzilla.gnome.org/show_bug.cgi?id=637754
 2013-08-23 15:57:46 +01:00
Wim Taymans
54e7e7547a rtxqueue: add property to configure queue size 2013-08-23 15:47:25 +02:00
Wim Taymans
84833bed11 rtpbin: proxy jitterbuffer do-retransmission property 2013-08-23 12:10:19 +02:00
Michael Olbrich
23d4044e2c avimux: unmap the correct buffer 
The audio buffer was mapped so unmap it and not the video buffer

https://bugzilla.gnome.org/show_bug.cgi?id=706642
 2013-08-23 11:32:52 +02:00
Wim Taymans
89b9019e3e rtx: various improvements 
Use locking
Don't push from the event handler, collected packets in a queue and push from
the chain function.
Clear queues on shutdown.
 2013-08-21 17:02:27 +02:00
Wim Taymans
ee15bc9284 session: generate events correctly 
Do correct shifting of the bitmask for lost packets.
 2013-08-21 17:02:27 +02:00
Wim Taymans
67523d3ecb rtp: register rtx element better 2013-08-21 17:02:26 +02:00
Wim Taymans
f626e29897 jpegdepay: add some more debug 2013-08-21 12:56:35 +02:00
Wim Taymans
77ed44a88a rtpgstdepay: only push events when they changed 
Keep track of the STREAM_START and TAG events and only push them
when they changed.
 2013-08-21 12:10:00 +02:00
Wim Taymans
b144809b7c rtpgstpay: taglists should not be merged in 1.0 2013-08-21 10:52:59 +02:00
Wim Taymans
69b0dcd7df rtpgstdepay: flush on FLUSH_STOP event 2013-08-21 10:28:50 +02:00
Wim Taymans
5ff9093843 rtpgstpay: reset on state change 
Do full reset on state change to READY
 2013-08-21 10:03:52 +02:00
Wim Taymans
ae9239aac7 rtpgstpay: reset on FLUSH_STOP 
Clear the adapter and pending buffer list on FLUSH_STOP.
 2013-08-21 09:55:20 +02:00
Wim Taymans
2e8955df39 rtpgstpay: don't use clock for config interval 
We can't use the clock to time our config-interval because we are not
live (or there might not be a clock or the clock might not be running).
Instead just simply take the timestamp diff.
 2013-08-21 09:39:30 +02:00
Wim Taymans
182f96ff79 rtpgstay: don't use // comments 2013-08-21 09:33:04 +02:00
Youness Alaoui
e22f7e91c4 rtspsrc: Fix response argument in handle-request signal 2013-08-21 09:06:02 +02:00
Youness Alaoui
6636efd31a rtspsrc: Add sdes property and proxy it to rtpbin 2013-08-21 09:06:02 +02:00
Youness Alaoui
62a6f58697 Send a stream-start whenever we send tags 
This is to make sure tags are cleared on the client if the
stream-start was previously lost, otherwise, the client may end
up with a merged taglist of multiple songs
 2013-08-21 09:06:01 +02:00
Youness Alaoui
05bcfee5a3 rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval 
This is useful in case the packet containing the inlined caps was lost
or if new client joins an already running RTP stream and they missed
the previous tag events.
This also makes the payloader keep a list of merged tags so the retransmitted
tag event contains all previously received. A STREAM_START event will
flush the list of tags.
 2013-08-21 09:06:01 +02:00
Youness Alaoui
1f4ca28868 rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time 2013-08-21 09:06:01 +02:00
Youness Alaoui
9257409613 rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps 2013-08-21 09:06:01 +02:00
Youness Alaoui
2d53289b6b rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList 
This is necessary to fix event/caps sending. If we send a STREAM_START
packet, it will cause an error because the stream didn't receive its
caps and new-segment events, so we must wait for the first buffer before
sending the stream-start event buffer. However, the caps will be sent
at the same time and so the 'inline caps' will be set for the event.
We need to be able to payload individual packets (data, caps or events)
and only send them when we call flush.
 2013-08-21 09:06:01 +02:00
Youness Alaoui
0070ba76f2 rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START 2013-08-21 09:06:01 +02:00
Youness Alaoui
6155b27971 rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3 2013-08-21 09:06:01 +02:00
Wim Taymans
587dc055e9 jitterbuffer: handle EOS 
When the queue is empty, and we received EOS, pause and push an EOS
event downstream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387
 2013-08-20 14:36:59 +02:00
Wim Taymans
533f26fc99 jitterbuffer: update docs 2013-08-20 10:26:15 +02:00
Wim Taymans
c7f9ef8012 jitterbuffer: update all timers 
Keep looping over all registered timers so that we can mark them lost instead of
stopping as soon as we find the timer for the current seqnum.
 2013-08-20 10:25:17 +02:00
Wim Taymans
5debda9ca1 jitterbuffer: remove unused variables 2013-08-20 08:55:50 +02:00
Wim Taymans
a88db5fa2c jitterbuffer: reorganize timer handling 
Restructure handling of incomming packet and the gap with the expected seqnum
and register all timers from the _chain function.
Convert a timer to a LOST packet timer when the max amount of retransmission
requests has been reached.
 2013-08-19 22:04:51 +02:00
Wim Taymans
d9d6eac4bb jitterbuffer: refactor packet spacing calculation 2013-08-19 22:04:50 +02:00
Wim Taymans
c4dc159656 jitterbuffer: keep track of last seqnum and dts 2013-08-19 22:04:50 +02:00
Wim Taymans
652ce95ca6 jitterbuffer: small cleanups 2013-08-19 22:04:50 +02:00
Wim Taymans
b4a35bbe82 jitterbuffer: reset retransmission timers in add/reschedule 
Reset the retransmission timers when adding and rescheduling a timer.
 2013-08-19 22:04:50 +02:00
Wim Taymans
cf8a0652f3 jitterbuffer: rename variables for packet spacing 2013-08-19 22:04:50 +02:00
Wim Taymans
ec82e4ab7c jitterbuffer: remove lost timer when we get the packet 
When we receive a packet, also remove the LOST timer for it.
 2013-08-19 22:04:50 +02:00
Wim Taymans
2f03b43b21 jitterbuffer: expected seqnum must increase 
Only update the expected seqnum when it is bigger than the previous expected
seqnum.
 2013-08-19 22:04:50 +02:00
Wim Taymans
c5bf376bb5 jitterbuffer: add more debug 2013-08-19 22:04:50 +02:00
Wim Taymans
ff825a2919 rtxqueue: add retransmission queue element 2013-08-19 22:04:50 +02:00
Wim Taymans
5fe18ee432 session: add some docs 2013-08-19 22:04:49 +02:00
Wim Taymans
482dacfb54 session: handle NACK feedback and generate events 
Handle and parse the feedback NACK packets and generate a Retransmission
event for each NACKed packet
 2013-08-19 22:04:49 +02:00
Thibault Saunier
e47ffb203b videomixer: Do not send flush_stop ourself after a flush_start 
When we receive a flush_start, we should wait for the next flush_stop
and foward it, not create a flush_stop ourself.
 2013-08-17 11:40:27 +02:00
Wim Taymans
db90f6e68d h264depay: init debug category early 
Init the debug variable when we register the element because it is also used by
the payloader element when it calls the add_sps_pps method.
 2013-08-16 17:12:19 +02:00
Chris Bass
3e9dea3f8c qtdemux: check denominator isn't zero before scaling duration. 
When gst_qtdemux_configure_stream sets fps_d, check that n_samples is
non-zero before using it as a denominator to scale the stream duration.

https://bugzilla.gnome.org/show_bug.cgi?id=706076
 2013-08-16 10:14:30 +02:00
Wim Taymans
f11c2c9b3b jitterbuffer: forward flush before stopping dataflow 
First forward the flush event and then stop our loop function.
 2013-08-14 16:19:32 +02:00
Olivier Crête
4c6e636720 rtph264pay: Use the SPS/PPS handling function from the depayloader 
Remove duplicated copies

https://bugzilla.gnome.org/show_bug.cgi?id=705553
 2013-08-13 10:38:23 -04:00
Olivier Crête
742b90747d rtph264depay: Make the SPS/PPS deduplication function generic 
Make it not touch any internals of the depayloader

https://bugzilla.gnome.org/show_bug.cgi?id=705553
 2013-08-13 10:38:23 -04:00
Chris Bass
b40bf67526 aacparse: allow conversion from raw AAC to ADTS 
This patch will prepend ADTS headers to raw AAC audio frames, allowing
upstream elements to link to decoders that only support AAC in ADTS format.

Note that no error correction bits are added to ADTS frames in this code.

https://bugzilla.gnome.org/show_bug.cgi?id=615740
 2013-08-13 15:58:23 +02:00
Sebastian Dröge
282afae244 rtspsrc: Only free GCheckSum after its last usage 
https://bugzilla.gnome.org/show_bug.cgi?id=705760
 2013-08-13 12:44:11 +02:00
Matej Knopp
2269ac8f28 qtdemux: elst should offset samples instead of buffers 
The current approach where buffers are offset is not ideal, as during seek
and loop current time is compared to sample times.

https://bugzilla.gnome.org/show_bug.cgi?id=700264
 2013-08-12 13:48:04 +02:00
Thibault Saunier
6c349d6ec3 videomixer: Send EOS if buf_end >= segment.stop 
That means the whole segment is already played, and we are sure we
are EOS at that point.

Also handle segment seeks, and do not send EOS in that case.
 2013-08-11 19:05:18 +02:00
Matej Knopp
96afba915a avidemux: send proper stream_start event 
https://bugzilla.gnome.org//show_bug.cgi?id=705449
 2013-08-08 11:57:32 +02:00
Sebastian Dröge
9863e08839 matroskademux: Don't print warnings during flushing and stop as soon as possible 
https://bugzilla.gnome.org//show_bug.cgi?id=705442
 2013-08-08 11:53:15 +02:00
Tim-Philipp Müller
957c8e3e61 rtpvp8depay: mark key frames and delta frames properly 
https://bugzilla.gnome.org/show_bug.cgi?id=705550
 2013-08-07 11:14:38 +01:00
Wim Taymans
48174164eb session: add NACK feedback in RTCP 2013-08-06 15:50:19 +02:00
Wim Taymans
4379ed1dee source: add methods to register NACK 
Add a method to register a missing packet for an ssrc along with
methods to get the missing packets and clear them.
 2013-08-06 15:50:19 +02:00
Wim Taymans
50638b8106 session: handle Retransmission event and schedule NACK 
Handle the retransmission event from downstream and use it to schedule a NACK
request.
 2013-08-06 15:50:19 +02:00
Wim Taymans
0bddbd682d session: pass data to remove func 
Pass the data to the remove function because we are going to deref it when there
is pli or fir.
 2013-08-06 15:50:19 +02:00
Thibault Saunier
38946bd9f4 qtdemux: Fix compilation 2013-08-06 15:31:38 +02:00
Thibault Saunier
593a31f2b4 qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE 2013-08-06 15:17:44 +02:00
Thibault Saunier
c5fa4666b7 videomixer: Make sure to send EOS if the buffer end time equals the segment end time 
Otherwize EOS never gets sent in that particular case.
 2013-08-06 12:21:33 +02:00
Sjoerd Simons
d14d4c436c goom: Ensure src caps are writable 
In some cases the src caps determined by goom weren't writable, causing
a bunch of assertion failures and failed caps. Fixed by always
explicitely making the caps writable

https://bugzilla.gnome.org/show_bug.cgi?id=705475
 2013-08-05 15:33:39 +02:00
Wim Taymans
3c82de59f9 session: use common send_rtcp method 
Reuse the send_rtcp method that already asks for the current time when
requesting a keyframe.
 2013-08-05 15:02:59 +02:00
Wim Taymans
3c14c6021c session: Don't use ClockTimeDiff for unsigned delays 2013-08-05 15:02:59 +02:00
Edward Hervey
4f4f6432cc qtmux: Use buffer PTS if DTS is not set 
Avoids ending up with completely bogus scaled duration/pts when new
buffers have invalid DTS.
 2013-08-04 17:15:38 +02:00
Tim-Philipp Müller
7272dec5fe rtpdec: use generic marshaller 2013-08-04 11:20:41 +01:00
Tim-Philipp Müller
fe098e3aff udp: remove unused marshal and enumtypes files 2013-08-04 11:03:07 +01:00
Tim-Philipp Müller
7469cd3a4c rtpmanager: use generic marshaller 2013-08-04 11:03:07 +01:00
Wim Taymans
7584f91f31 jitterbuffer: send event in right direction 2013-08-04 00:24:36 +02:00
Wim Taymans
9613e481ad session: add FIR and PLI like other RTCP packets 
Add the FIR and PLI packets like the other RTCP packet instead of from the
on-sending-rtcp default signal handler.
 2013-08-03 00:33:24 +02:00
Wim Taymans
743e1b1191 jitterbuffer: fix property ranges 2013-08-02 17:22:55 +02:00
Wim Taymans
cd0164f4cc jitterbuffer: push retransmission events 2013-08-02 16:43:59 +02:00
Wim Taymans
9a13267e85 jitterbuffer: add support for retransmission retry 
When we didn't receive a packet after requesting retransmission, retry
asking for retransmission for a certain period.
 2013-08-02 14:54:56 +02:00
Wim Taymans
e9ad5126db jitterbuffer: add properties 
Add properties to control retransmission parameters
 2013-08-02 14:47:56 +02:00
Wim Taymans
a8c7ff7489 jitterbuffer: use corrected timeout when rescheduling 
When we recalculate the timeout, use the corrected timeout value depending on
the timer type.
 2013-08-02 12:44:58 +02:00
Wim Taymans
9c7e3e3455 jitterbuffer: update timers after queueing 
Else we might update the timer needlessly for duplicates.
 2013-08-02 12:43:00 +02:00
Wim Taymans
ebd6b8f8ab jitterbuffer: move method up 2013-08-02 12:42:08 +02:00
Wim Taymans
f6b6797874 jitterbuffer: small cleanup 2013-08-02 06:28:32 +02:00
Wim Taymans
0e41414926 jitterbuffer: unschedule old expected packets 
When we receive a new packet, unschedule old outstanding packets when their
seqnum is too far away.
 2013-08-01 23:36:07 +02:00
Wim Taymans
70695466ed jitterbuffer: refactor timer update 2013-08-01 23:32:00 +02:00
Wim Taymans
4ab3f5d3da jitterbuffer: update timers when removing 
Update the timers when we remove a timer.
Handle canceled timers, make them unschedule the current timer and
trigger the timeout code.
 2013-08-01 23:24:29 +02:00
Wim Taymans
b983cf675b jitterbuffer: fix typo 2013-08-01 23:22:02 +02:00
Wim Taymans
f3c658cbe6 jitterbuffer: improve timeout management 
If we change the seqnum of an existing timer and we were waiting for
that timer, unschedule it. If we change the timeout of an existing timer and we
were waiting on it, only unschedule when the new time is smaller.
 2013-08-01 15:40:52 +02:00
Wim Taymans
77e5d320ab jitterbuffer: install timer for expected arrival 
Install a timer that is triggered when the expected arrival time of a packet
expired.
 2013-08-01 15:11:13 +02:00
Wim Taymans
f08d98404e jitterbuffer: improve unschedule of timers 
Conflicts:
	gst/rtpmanager/gstrtpjitterbuffer.c
 2013-08-01 14:57:11 +02:00
Wim Taymans
9d3b824e2a jitterbuffer: move code around 2013-08-01 12:21:53 +02:00
Wim Taymans
fe32e80c92 jitterbuffer: estimate inter packet spacing 
When we see two packets with consecutive seqnums and a different RTP time, use
the DTS difference as the inter packet spacing estimate.
 2013-08-01 12:07:11 +02:00
Wim Taymans
255b7106f5 jitterbuffer: keep track of current timeout 2013-08-01 12:01:15 +02:00
Wim Taymans
7e43dba19b jitterbuffer: cleanup timer handling 2013-08-01 11:49:10 +02:00
Wim Taymans
9d88ac9cbb jitterbuffer: reset is only possible with a GAP 2013-08-01 11:40:41 +02:00
Wim Taymans
f864131227 jitterbuffer: operate on DTS 
Make the jitterbuffer schedule the timeouts based on the DTS instead
of the PTS. This makes it all smoother with reordered frames and gives
the decoder time to reorder the frames in time.
 2013-08-01 11:36:56 +02:00
Wim Taymans
80c5934290 jitterbuffer: rename timout variable 2013-08-01 11:14:12 +02:00
Wim Taymans
aa951433ee jitterbuffer: small cleanup 2013-07-31 17:08:58 +02:00
Wim Taymans
69c78f72d5 jitterbuffer: block output in paused or buffering 2013-07-31 16:59:58 +02:00
Wim Taymans
4fbbc53a49 jitterbuffer: store pts in timer 
Only store the pts in the timer so that we can both do timeouts with timings on
the input and output of the jitterbuffer.
 2013-07-31 16:59:09 +02:00
Wim Taymans
77846d35c6 rtpjitterbuffer: refactor jitterbuffer 
Refactor the jitterbuffer code. Make separate function for peeking a buffer,
pushing the next buffer, waiting for timeouts and handling the timeouts.

The main loop now tries to push as many buffers as it can until it runs out of
buffers or when it detects a seqnum discont. Then it will wait for some event to
happen before attempting to push more buffers.

Make methods to register timeouts in an array. These timeouts are registered
when we detect a missing packet, sync for the first packet or when we find an
estimation for the end-of-stream.

This greatly simplifies and clarifies the code and also makes it possible to
register more complicated timeout schemes later.
 2013-07-30 23:24:23 +02:00
Wim Taymans
ea931d4f57 rtpjitterbuffer: use NULL to ignore percent 
If we pass NULL to pop and push we ignore the percent result.
 2013-07-30 23:24:23 +02:00
Wim Taymans
b3e8a85a54 jitterbuffer: refactor 
Move eos estimation into separate function
 2013-07-30 23:24:22 +02:00
Tim-Philipp Müller
a5532b4510 flvdemux: don't leak stream_id string 
https://bugzilla.gnome.org/show_bug.cgi?id=705142
 2013-07-30 14:28:19 +01:00
Sebastian Dröge
2e35b36aab gst: Don't swap start/stop for negative rates in the SEGMENT query 2013-07-29 12:12:41 +02:00
Matej Knopp
47ed79fb1c qtdemux: Check for data size when parsing h264 codec data from strf atom 2013-07-29 11:53:07 +02:00
Sebastian Dröge
722ef42196 matroskademux: Implement SEGMENT query 2013-07-29 10:53:54 +02:00
Sebastian Dröge
d135373beb flvdemux: Implement SEGMENT query 2013-07-29 10:53:47 +02:00
Sebastian Dröge
4e78974c87 avidemux: Implement SEGMENT query 2013-07-29 10:50:59 +02:00
Matej Knopp
2dcdfe07f7 qtdemux: Support H264 fourcc 
https://bugzilla.gnome.org/show_bug.cgi?id=704996
 2013-07-29 09:11:39 +02:00
Sebastian Dröge
1fbb6d30a6 avidemux: Fix duration reporting in push mode 
https://bugzilla.gnome.org/show_bug.cgi?id=700933
 2013-07-28 17:38:56 +02:00
Sebastian Dröge
89a3dc2ecd avidemux: Don't forget unmapping and unreffing buffer 2013-07-28 17:32:59 +02:00
Matej Knopp
1947587784 avidemux: unmap buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=704951
 2013-07-28 17:32:59 +02:00
Wim Taymans
02359f9219 session: don't make buffer writable prematurely 
There is no reason to make the SR buffer writable at this point. This is better
delayed until needed.
 2013-07-26 22:31:41 +02:00
Wim Taymans
0261199fc4 session: ignore RTCP for inactive sources 2013-07-26 22:31:23 +02:00
Wim Taymans
a4b4ca53c0 session: small cleanup 2013-07-26 22:25:17 +02:00
Wim Taymans
e0abd2e9b5 session: handle partial RTCP report blocks 
When we have more SSRCs to report than what fit in an RTCP packet, use a
generation counter to make sure all of them end up in a packet eventually.
 2013-07-26 17:29:10 +02:00
Wim Taymans
6cce6fb04c session: create SSRC before doing session cleanup 
Make the internal source before we do session cleanup
 2013-07-26 17:29:10 +02:00
Wim Taymans
5b0298c63e session: reorganize the report block code 2013-07-26 17:29:10 +02:00
Matej Knopp
7335b81c47 matroskademux: fix memory leak in check_subtitle_buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=704921
 2013-07-26 17:11:31 +02:00
Wim Taymans
3c44cd7c83 session: refactor active and sender checks 2013-07-26 14:21:40 +02:00
Wim Taymans
e952f7ba43 session: remove internal sources on timeout 
When an internal source times out and becomes a receiver, remove it.
 2013-07-26 12:18:01 +02:00
Wim Taymans
e9e2fe3950 session: create an internal source for RTCP 
When we need to do RTCP and we don't have an internal source yet,
make one.
 2013-07-26 12:18:01 +02:00
Wim Taymans
bd0709c15c session: remove old code to change SSRC 
Remove code used to change the SSRC after a collision. We now send
a RECONFIGURE event upstream to make the upstream element change the SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
88f5a5f355 source: don't update packet SSRC 
Remove the code to update the SSRC in packets, it can never be called now that
we always use a source with matching packet SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
abc90da1dc session: delay allocation of internal source 
Allocate the internal source when we receive a caps with the SSRC or when we see
a buffer with the SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
e0a1ce1291 session: generate reconfigure on collision 
When we detect a collision, change the SSRC that we suggest upstream
and trigger RECONFIGURE. This should make upstream select a new SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
495d43c089 session: produce RTCP for all internal sources 
Loop over all the internal sources and produce RTCP. We also need
to queue the RTCP packets and send them when we are finished.
 2013-07-26 12:18:00 +02:00
Wim Taymans
9505fd4150 session: deprecate internal source and ssrc properties 
Deprecate the internal source and internal ssrc properties. There might
be more than one internal source.
 2013-07-26 12:17:59 +02:00
Wim Taymans
3d6ee1fb5e session: internal sources don't use probation 2013-07-26 12:17:59 +02:00
Wim Taymans
0e53e9109e session: give caps to session 
Let the session parse the caps and update its SSRC when needed.
 2013-07-26 12:17:59 +02:00
Wim Taymans
c06482a2cb session: make method to suggest available SSRC 
Make a method to suggest the best available SSRC. This is the SSRC of the last
created internal source and is used to instruct upstream to produce this
SSRC.
 2013-07-26 12:17:59 +02:00
Wim Taymans
33ce50e8b1 session: keep SDES and set on new internal sources 
Keep track of the SDES ourselves and set it on all newly created
internal sources.
 2013-07-26 12:17:59 +02:00
Wim Taymans
5652f02b76 session: make method to make internal sources 
Add a method to obtain an internal source and use it to create
our internal source
 2013-07-26 12:17:59 +02:00
Wim Taymans
7f83927c95 session: count internal sources and how many are senders 2013-07-26 12:17:58 +02:00
Wim Taymans
719343c206 rtpsession: separate BYE marking and scheduling 
First mark sources with BYE and then schedule the BYE RTCP message.
 2013-07-26 12:17:58 +02:00
Wim Taymans
391943ba82 session: get SSRC from RTCP packet itself 
Get the SSRC from the RTCP packet instead.
 2013-07-26 12:17:57 +02:00
Wim Taymans
a3f75a17ef session: fix bandwidth calculation 
We iterate over all sources and the internal one is also in the
hashtable so avoid adding it twice.
 2013-07-26 12:17:57 +02:00
Wim Taymans
9eaef9d332 session: add some docs 2013-07-26 12:17:56 +02:00
Wim Taymans
2163355a47 session: Rearrange RTCP reporting a little 
Make a function to generate an RTCP packet for a source, pass the source as a
parameter.
Move timeout of collisions to session cleanup phase.
 2013-07-26 12:17:56 +02:00
Wim Taymans
a3bf374351 session: move check for is_early around 
Move the check for the early RTCP to where it is needed and used.
 2013-07-26 12:17:56 +02:00
Wim Taymans
b069db6a2e session: parse packet outside of the session lock 2013-07-26 12:17:56 +02:00
Wim Taymans
57c27ec319 session: do nicer checks for internal sources 2013-07-26 12:17:56 +02:00
Wim Taymans
93d07298ff session: let source keep track if it sent BYE 2013-07-26 12:17:56 +02:00
Wim Taymans
0c9c1434a8 source: reset more 2013-07-26 12:17:56 +02:00
Wim Taymans
1d02496d15 source: also use the source for bye_reason 
Store the BYE reason in our internal source object. Rename the methods on the
source object a little because now the BYE can be received in RTCP or
set when the session wants to send BYE.
 2013-07-26 12:17:56 +02:00
Wim Taymans
ddd071e54c session: configure sdes with structure only 
Remove code to configure the SDES with methods and types, only
allow configuration with GstStructure
 2013-07-26 12:17:55 +02:00
Wim Taymans
0060e1d45d session: refactor add and find source 
Make functions to find and add a source to the hashtable.
 2013-07-26 12:17:55 +02:00
Wim Taymans
adb0d68c07 session: remove source from sync_rtcp 
We don't need to know the sender source of the session in the
callback, the SR packet is for all participants in the session.
 2013-07-26 12:17:55 +02:00
Wim Taymans
bf7d8173b3 jitterbuffer: add some more debug 2013-07-26 12:17:55 +02:00
Vincent Penquerc'h
91d4abceaa aacparse: allow conversion from ADTS to raw AAC 
Some muxers (eg, qtmux) only support raw AAC, so this allows linking
an encoder that outputs ADTS only to those muxers.

The conversion is simple (omit the first 7 or 9 bytes of the frame),
but has to be done in pre_push instead of handle_frame as 1.0 does
not seem to allow skipping bytes there as 0.10 used to.

Other conversions are not supported (yet).
 2013-07-26 09:44:11 +01:00
Vincent Penquerc'h
55e9338846 aacparse: fix object_type parsing off-by-one in ADTS frame 
According to http://wiki.multimedia.cx/index.php?title=ADTS,
the value stored in ADTS headers is one less than the object
type of the AAC stream.

A look at ffmpeg shows it also adds 1 to the value read off
the ADTS header.

Note that this might break other things that happen to have
an inverse off by one to match the existing code.
 2013-07-26 09:44:10 +01:00
Thiago Santos
7eac4c7c03 avidemux: fix seqnum handling for seeks 
Use the same seqnum as the seek for flushes/segments that are
caused by the seek. Also do the same for segment events

Fixes #676242
 2013-07-25 15:24:31 -03:00
Thiago Santos
8bd12e12b3 matroskademux: fix seqnum handling for seeks 
Use the same seqnum as the seek for flushes/segments that are
caused by the seek. Also do the same for segment events

Fixes #676242
 2013-07-25 15:24:31 -03:00
Thiago Santos
e49b6e7c35 qtdemux: correctly handle seqnum for seeks and segments 
Use the same seqnum on messages and events for derived events.
Fixed for flushes / stream-start / segment after a seek, and segment
after a segment.

Fixes #676242
 2013-07-25 15:24:31 -03:00
Wim Taymans
c44a29bd53 bin: fix compilation 2013-07-24 14:17:45 +02:00
Wim Taymans
cc92ef1db2 vrawdepay: fix UYVP format 2013-07-24 12:42:31 +02:00
Wim Taymans
8191b6fcd2 vrawpay: fix UYVP format 2013-07-24 12:41:58 +02:00
Wim Taymans
37af93c361 vrawpay: fix caps 2013-07-24 12:41:44 +02:00
Wim Taymans
f87875e35b rtpjitterbuffer: fix locking 
Take the lock earlier so that we do things that follow with the right
locking.
 2013-07-24 10:49:03 +02:00
Wim Taymans
dece8413ef rtpsession: don't use invalid times in RTCP timeouts 
An invalid timeout can be calculated when we disabled RTCP by setting the
bandwidth to 0. Make sure all code can handle this case.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626
 2013-07-23 17:41:48 +02:00
Wim Taymans
25e0f0d6b6 rtpsession: lock session when changing bandwidth 
Take the session lock when changing the bandwidth properties so that we don't
end up with inconsistent behaviour.
 2013-07-23 17:41:48 +02:00
Wim Taymans
c337265ee4 session: reset some RTCP variables 
The early_send time was set to 0 and always triggering an early RTCP packet.
 2013-07-23 17:41:48 +02:00
Edward Hervey
3d48d25756 qtdemux: Add all the mpeg XDCAM variants 
This should cover all known XDCAM variants (which are all mpeg2 video)

Fixes #672227
 2013-07-23 15:03:31 +02:00
Carlos Rafael Giani
95429f1d4b rtpbin: added custom downstream sync event 
rtpbin can now send a custom in-band downstream event which informs
downstream that the bin has received an RTCP SR packet. This is useful
for applications which want to drop the initial unsynchronized received
RTP packets.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
 2013-07-23 06:25:20 +02:00
Tim-Philipp Müller
f18b1f7e80 deinterlace: fix on-the-fly changing of "mode" and "fields" properties 
We call setcaps() to reconfigure ourselves, but we need to pass
the current *sink* caps, not the source caps then. Also fix a
caps leak.

https://bugzilla.gnome.org/show_bug.cgi?id=641599
 2013-07-22 18:00:16 +01:00
Sebastian Dröge
0c2ff91a5c wavparse: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
169b490664 rtspsrc: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
5a9f4a3cbc rtpsession: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
57dd1189d5 matroskademux: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
1a0278ed64 qtdemux: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
1122698491 flvdemux: Add support for group-id in the stream-start event 2013-07-22 15:30:12 +02:00
Sebastian Dröge
6cc16da531 avidemux: Add support for group-id in the stream-start event 2013-07-22 15:30:12 +02:00
Mathieu Duponchelle
d67a671bfb videomixer: use gst_util_uint64_scale*_round. 
There could be a case where:
      1) you do a new set_caps after buffers have been processed.
      2) ts_offset gets set to a different value, eg 0.033333333
      3) your pads get EOS, but the check dor that doesn't work
         because you use ts_offset + a truncated value < segment.stop
      4) so in the next collected, you end up comparing for example:
      0.9999999999 > 1., which is false and means you don't send EOS.

Also adds scale_round in two other places where it potentially could
have caused problems.
 2013-07-21 19:21:57 -04:00
Olivier Crête
96a8fb92e2 qtdemux: Add WRLE support 2013-07-19 14:58:30 -04:00
Tim-Philipp Müller
aa7d597120 qtdemux: make files from Vivotek camera play 
Skip tracks of 'vivo' subtype with empty stsd instead of
erroring out saying that the file is broken.

https://bugzilla.gnome.org/show_bug.cgi?id=699791
 2013-07-19 19:38:30 +01:00
Tim-Philipp Müller
ce52b319ff qtmux: when streaming don't try to seek when stopping 
It might cause errors in sinks that are not seekable and
have reported this (like e.g. fdsink)

https://bugzilla.gnome.org/show_bug.cgi?id=696228
 2013-07-19 17:31:38 +01:00
Wim Taymans
bdd3c31902 qtdemux: simplify some helpers 
Some helper functions are not needed anymore or can be simplified.
 2013-07-19 17:26:54 +02:00
Wim Taymans
61a8937ced qtdemux: for non-raw video, move palette in caps 
We only need to append the palette to raw video buffers, non-raw video has the
palette in the caps still.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292
 2013-07-19 17:14:46 +02:00
Arnaud Vrac
40ab78825c qtdemux: nitpicking in esds parsing 2013-07-19 14:26:18 +02:00
Arnaud Vrac
d0d25a5e1f qtdemux: set proper caps for mpeg-1 audio 
Remove AAC specific fields from mpeg-1 audio caps, remove assumption
that the mpeg1 audio layer is 3, and set `parsed' field.

https://bugzilla.gnome.org/show_bug.cgi?id=704548
 2013-07-19 14:26:08 +02:00
Arnaud Vrac
5def061d20 qtdemux: remove chapter stream 
Remove all streams that are actually table of contents, since we will
never need the data after parsing them.
 2013-07-18 11:48:12 +02:00
Arnaud Vrac
ae67c13416 qtdemux: send gap event for sparse streams in push mode 
This allows to pre-roll at least if the next subtitle buffer
is far away.
 2013-07-18 11:48:11 +02:00
Arnaud Vrac
1237898351 qtdemux: do not use indexes from sparse stream when seeking in push mode 
This makes seeking more accurate in push mode, since the previous
keyframe on a sparse stream might be far away.
 2013-07-18 11:48:11 +02:00
Arnaud Vrac
e561d12655 qtdemux: advertise subtitle streams as sparse 2013-07-18 11:48:11 +02:00
Arnaud Vrac
6e26f1d067 mastrokademux: do not push discont buffers if they aren't discont 
Unset the discont flag instead of posssibly pushing a buffer with
a flag that's still set.

https://bugzilla.gnome.org/show_bug.cgi?id=682110
 2013-07-17 18:10:11 +01:00
Wim Taymans
4c97701650 qtdemux: extract the palette from stsd 
Sometimes a palette is inside the stsd, extract it instead of always using
the default one
 2013-07-17 15:17:19 +02:00
Sebastian Dröge
9f73447229 goom2k1: Fix event handling and negotiate as soon as possible 2013-07-17 14:30:16 +02:00
Sebastian Dröge
78c7c16e9e goom: Fix event handling and negotiate as soon as possible 2013-07-17 14:28:43 +02:00
Wim Taymans
6b82c89562 qtdemux: add support for WRAW 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292
 2013-07-17 09:57:17 +02:00
Wim Taymans
f698483bb3 qtdemux: palette is appended to buffers, not in caps 
Fix the palette handling, in 1.0 we append the palette to the buffer instead of
placing it on the caps.

See also https://bugzilla.gnome.org/show_bug.cgi?id=704292
 2013-07-17 09:57:16 +02:00
Olivier Crête
54c5a7f690 rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders 2013-07-16 15:37:49 -04:00
Arnaud Vrac
54bba4f60c qtdemux: reset segment on flush stop 
cca2f555d1 introduces a regression, where the demux segment is not
reset on flush stop, so the next upstream segment event will calculate
an invalid base time on the new segment to be sent downstream.

https://bugzilla.gnome.org/show_bug.cgi?id=704255
 2013-07-16 10:47:20 +02:00
Matej Knopp
ca32442f86 qtdemux: offset samples according to edit list 
https://bugzilla.gnome.org/show_bug.cgi?id=700264
 2013-07-15 09:59:23 +02:00
Matej Knopp
ae92ea21a1 aacparse: be less verbose when parsing LOAS streams 
https://bugzilla.gnome.org/show_bug.cgi?id=704162
 2013-07-15 07:55:08 +02:00
Matej Knopp
3111161e8a qtdemux: unselect instead of ignoring disabled track, detect chapter track 
https://bugzilla.gnome.org/show_bug.cgi?id=704007
 2013-07-12 11:45:33 +02:00
Kyosuke Nekomura
4d517e94ef audioecho: Fix handling of delay property in PLAYING/PAUSED state 
https://bugzilla.gnome.org/show_bug.cgi?id=703901
 2013-07-12 09:36:16 +02:00
Olivier Crête
3aa20e7c8d rtpmux: Enable proxy caps on the src pads 2013-07-11 17:21:22 -04:00
Matej Knopp
7b69f427f1 qtdemux: correct argument order in gst_util_uint64_scale_int_round 
https://bugzilla.gnome.org/show_bug.cgi?id=703350
 2013-07-10 09:20:17 +02:00
Olivier Crête
1997acc8b2 rtpmux: Keep caps order from the peer or the filter 2013-07-09 17:43:31 -04:00
Sebastian Dröge
3d0988f46f videomixer: Fix handling of buffers without a duration 
We'll have to pop buffer from collectpads and store it
internally only to get the timestamp of the next buffer.
If we continue to keep it in collectpads, no new buffer
to calculate the end time will ever arrive.

https://bugzilla.gnome.org/show_bug.cgi?id=703743
 2013-07-09 12:42:17 +02:00
Sebastian Dröge
9e9d2ce098 videomixer: Fix negotiation with 0/1 framerates 
https://bugzilla.gnome.org/show_bug.cgi?id=703743
 2013-07-09 11:53:28 +02:00
Jonas Holmberg
beebe2b7af matroskademux: Unlock stream lock after use 
Stream lock of sink pad was not unlocked after non-updating seek.
 2013-07-09 11:25:14 +02:00
Ognyan Tonchev
aa2d96c46b multipartmux: Re-set need_segment flag after FLUSH_STOP 
https://bugzilla.gnome.org/show_bug.cgi?id=703182
 2013-07-09 09:16:20 +02:00
Sebastian Dröge
0cc77d8e30 rtph263ppay: Don't pass upstream filter caps to downstream 
Downstream usually can't accept video/x-h263 but only application/x-rtp,
so we would always get an empty intersection here.

https://bugzilla.gnome.org/show_bug.cgi?id=702632
 2013-07-08 14:10:44 +02:00
Wim Taymans
ab24598443 rtspsrc: avoid some strdup 2013-07-02 11:13:25 +02:00
Wim Taymans
7c950ef3f2 rtspsrc: add select-stream signal 
Add a signal to let the app select what streams will be selected.

See https://bugzilla.gnome.org/show_bug.cgi?id=634419
 2013-07-02 10:40:35 +02:00
Wim Taymans
2d276e1bcb rtspsrc: avoid strdup 2013-07-02 10:40:35 +02:00
J. Rick Ramstetter
f01b751e52 rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin 
https://bugzilla.gnome.org/show_bug.cgi?id=703426
 2013-07-02 10:12:17 +02:00
Wim Taymans
1db7e62060 rtspsrc: add signal to notify of the SDP 
This way, the app can look and modify the SDP.
 2013-07-01 17:31:30 +02:00
Matej Knopp
4053e1d6ac qtdemux: compute framerate from average sample duration 
https://bugzilla.gnome.org/show_bug.cgi?id=703350
 2013-07-01 12:53:17 +02:00
Alban Browaeys
97015d3c93 flvdemux: Add flvversion 1 to the flash-video caps 
This allows using avdec_flv which requires this field to be
present in the caps. FLV only supports flash-video version 1
right now.

https://bugzilla.gnome.org/show_bug.cgi?id=703076
 2013-07-01 11:43:46 +02:00
Sebastian Dröge
5f6469fe2a deinterleave: Don't hold object lock while sending events downstream 
Based on a patch by Kishore Arepalli <kishore.arepalli@gmail.com>

https://bugzilla.gnome.org/show_bug.cgi?id=703114
 2013-07-01 11:37:00 +02:00
Sebastian Dröge
75b5a54f17 matroskademux: Add MPEG4 video profile/level to the caps 2013-07-01 11:01:13 +02:00
Sebastian Dröge
423bddac6a matroskademux: Add AAC profile/level to the caps 
https://bugzilla.gnome.org/show_bug.cgi?id=703312
 2013-07-01 11:01:13 +02:00
Wim Taymans
c469434ea8 vorbispay: add support for config-interval 
Align code with the theora payloader and add support for the config-interval to
periodically send out the config headers.
 2013-06-28 15:21:56 +02:00
Wim Taymans
006562c9f4 theorapay: small cleanups 2013-06-28 15:21:12 +02:00
Wim Taymans
cdc66462ce theorapay: handle streamheaders as well 2013-06-28 12:08:19 +02:00
Wim Taymans
3169432ed4 vorbispay: always collect headers on data 
When we see a data packet, always check if we need to collect any previous
headers.
 2013-06-28 12:07:58 +02:00
Wim Taymans
6c716dfc25 vorbispay: handle streamheader as well 
Take config strings from the streamheader when we can

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=664312
 2013-06-28 11:43:17 +02:00
David Svensson Fors
692206d3a7 rtph264pay: avoid double buffer unmap on error 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703171
 2013-06-27 17:14:11 +02:00
Wim Taymans
3289a2963b rtspsrc: reset-sync before play 
Call reset-sync on the rtpbin before we go to playing. This makes us require SR
packets for all streams again before we attempt to sync them. If we don't reset,
it might be that we combine SR packets from before and after the PAUSE/PLAYING
state change and end up with huge bogus offsets.
 2013-06-27 17:02:14 +02:00
Wim Taymans
519305d14d jitterbuffer: improve sync on first packets 
Don't throw away the first RTCP packet if it arrives before the first
RTP packet but remember and use it to signal sync once we get the
RTP packet.

See https://bugzilla.gnome.org/show_bug.cgi?id=691400
 2013-06-27 16:23:20 +02:00
Wim Taymans
8969f00661 jitterbuffer: only signal loop when active 
Only signal the loop function when it is active.
 2013-06-27 16:15:45 +02:00
Wim Taymans
4bd2ffb26e jitterbuffer: signal timestamp discont 
We can now use the RESYNC buffer flag to mark a timestamp discont when we update
the ts-offset property.
 2013-06-27 16:13:37 +02:00
Wim Taymans
4258ddcc36 jpegpay: turn some errors into warnings 
Turn some errors into warnings, we can continue processing so this should
not be fatal.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=657079
 2013-06-26 20:49:41 +02:00
Wim Taymans
bb9d42b976 rtspsrc: avoid some flushes 2013-06-26 14:58:53 +02:00
Wim Taymans
f39ef2ab68 rtspsrc: handle data message when waiting for reply 
When we are waiting for a server reply, handle data messages instead of
ignoring them.
 2013-06-26 14:41:36 +02:00
Wim Taymans
61219dc6ed rtspsrc: handle data messages in separate method 
Refactor and make a method to handle a data message.
 2013-06-26 14:41:36 +02:00
Wim Taymans
a4be0c6de3 rtspsrc: add some more docs to handle-request signal 
See https://bugzilla.gnome.org/show_bug.cgi?id=702705
 2013-06-25 20:36:18 +02:00
Youness Alaoui
52e440c91b Send a clock_provide message on the bus when we get a netclock 2013-06-25 14:50:47 +02:00
Youness Alaoui
547df8e14f rtspsrc: Expose use-pipeline-clock property 2013-06-25 14:50:33 +02:00
Wim Taymans
35f6e79b94 udpsink: bind to the given interface 
Actually call BINDTODEVICE to bind to the interface as given by the
property.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702819
 2013-06-24 17:13:05 +02:00
Sebastian Dröge
3c9aba91dc matroska: Add initial VP9 support 2013-06-21 18:22:13 +02:00
Youness Alaoui
95906b8f1c rtsp: go back into the loop after doing pause 
After we do a pause request, go back to loop mode so that we can listen
for server messages again.

See https://bugzilla.gnome.org/show_bug.cgi?id=702705
 2013-06-21 10:42:20 +02:00
Olivier Crête
2cd6f53e24 rtpptdemux: Wait after the caps to forward the other events 
First forward the stream-start, then the caps, then the rest
 2013-06-20 23:16:59 -04:00
Wim Taymans
b96d931bf4 rtspsrc: fix race in state change to paused 
When we go to paused, we first flush the connection and then send the pause
command. As a result of the flushing, the scheduled paused command can get
lost. Wait until the connection is completely flushed and the rtsp task is
waiting before issuing the paused or playing request.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702705
 2013-06-20 14:43:47 +02:00
Wim Taymans
8428423c04 qtdemux: handle SEGMENT query 2013-06-20 11:31:22 +02:00
Kishore Arepalli
5b32891ae1 avidemux: duration query returns zero for DV video in avi 
https://bugzilla.gnome.org/show_bug.cgi?id=702625
 2013-06-19 11:17:22 +02:00
Sebastian Dröge
b001da2926 qtdemux: Disable usage of allocation queries 
This can only reliably work if demuxers have a
separate streaming thread per srcpad. This should be
done in a demuxer base class, which integrates parts
of multiqueue

https://bugzilla.gnome.org/show_bug.cgi?id=701856
 2013-06-19 11:07:48 +02:00
Alex Ashley
46a137c810 Avoid skipping moov atoms for fragmented MP4 files. 
bug #700505

Following a representation change that causes a resolution change,
the video decoder fails to decode correctly. Dashdemux detects the
representation change and pushes a new caps event and an
initialization segment (a new moov atom) to the downstream qtdemux,
but it doesn't handle this new moov yet, it will only parse the
first one it receives.

This commit changes qtdemux to accept a new moov in a dash bitstream
switching scenario.
 2013-06-19 01:44:22 -03:00
Thiago Santos
384e8f6c34 qtdemux: send stream-start only once for each stream 
Do not send stream start again when reconfiguring a pad for new caps.
That is common for adaptive streams
 2013-06-19 00:55:30 -03:00
Jens Georg
745be945ce rtpmp2tdepay: accept mislabelled streams from GStreamer 0.10 as well 
The mp2t payloader in 0.10 mislabelled the streams as MP2T-ES
instead of MP2T, so accept that as well for compatibility reasons.

https://bugzilla.gnome.org/show_bug.cgi?id=702457
 2013-06-17 15:39:17 +01:00
Wim Taymans
d9bc48edc9 rtspsrc: manage element state ourselves 
Lock the state of the all our elements and manage their states
outselves. Because we are working async, we can't rely on the state
change function to set the state at the right time or to return the
right return value from the state change function.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702046
 2013-06-16 05:40:13 +02:00
Bruno Gonzalez
e89a48616b matroskademux: Don't unlock stream lock without locking it first 
https://bugzilla.gnome.org/show_bug.cgi?id=702167
 2013-06-14 14:10:13 +02:00
Wim Taymans
51c9f7989f rtpsession: Use the right hashtable to calculate bandwidth 
Don't use an unused hashtable to iterate source to calculate bandwidth.
Remove unused code.
 2013-06-13 16:02:19 +02:00
Sebastian Dröge
01cc493944 Revert "videomixer: When all sinkpads are eos, update output segment stop and forward it" 
This reverts commit 2d3910fc79.

It's not solving any problem and instead causes code to fall apart.

https://bugzilla.gnome.org/show_bug.cgi?id=701519
 2013-06-12 18:25:59 +02:00
Tim-Philipp Müller
213cd2777b matroskademux: mark subtitle streams as sparse in stream-start event 
And also mark the streams that should be selected by default if
marked so in the headers.

https://bugzilla.gnome.org/show_bug.cgi?id=600648
 2013-06-12 15:31:22 +01:00
Stefan Sauer
39c4c5f251 audiopanorama: add prebuilt files 2013-06-11 22:14:33 +02:00
Stefan Sauer
349a60e164 audiopanorama: cleanup of transform() 
Only map input if we are reading it. Cleanup the logging and the comments a bit.
 2013-06-11 21:48:18 +02:00
Stefan Sauer
1dc06932a2 audiopanorama: use orc to speedup processing 
Use special variants for the case when we don't change the panorama (pan=0.0).
Simplify the processing functions by passing the panorama value directy instead
of the instance. Use orc for clearing buffers too.
 2013-06-11 21:48:18 +02:00
Mathieu Duponchelle
6e23f1fec4 videomixer: check last end_time after conversion to running segment 
The last end_time was saved after conversion, so the comparison
had to be made after conversion for it to make sense.

https://bugzilla.gnome.org/show_bug.cgi?id=701385
 2013-06-11 21:03:35 +02:00
Mathieu Duponchelle
4243714301 videomixer: add mix->segment.start to output_end_time 
When the segment start is not 0, this created a situation where
the output_end_time is inferior to output_start_time, and the duration
of the next buffer ended up underflowing.

https://bugzilla.gnome.org/show_bug.cgi?id=701385
 2013-06-11 21:03:03 +02:00
Sebastian Dröge
e2b46a776f matroskademux: Send stream headers after the segment event 
https://bugzilla.gnome.org/show_bug.cgi?id=700799
 2013-06-11 13:54:53 +02:00
Sebastian Dröge
adc9f0bd10 qtdemux: Do allocation query after exposing all pads and no-more-pads 
Also configure video streams as early as possible.

Related https://bugzilla.gnome.org/show_bug.cgi?id=701856
but not fixing that.
 2013-06-11 12:27:19 +02:00
Sebastian Dröge
ab275b62a8 flvdemux: Don't forward CAPS events from upstream 
Just use the default pad event handler.

https://bugzilla.gnome.org/show_bug.cgi?id=701976
 2013-06-11 12:27:19 +02:00
Stefan Sauer
4ef27eb0f9 audiopanorama: move the enum to the header and use instead of gint 
Move the enum for the processing method to the header so that we can use the
type for the instance struct.
 2013-06-09 20:39:48 +02:00
Sebastian Dröge
1ba08e331c wavenc: Link with libgstbase for GstByteWriter 2013-06-07 15:15:15 +02:00
Sebastian Dröge
db1c2a28a6 wavparse: Push stream-start event in pull mode before anything else 2013-06-07 13:27:07 +02:00
Sebastian Dröge
048866f1b1 Release 1.1.1 2013-06-05 18:31:40 +02:00
Sebastian Dröge
ea75b890dc wavenc: Fix taglist ref handling that made the unit test fail 2013-06-05 15:50:04 +02:00
Wim Taymans
0d27829a6b udpsink: avoid leaking the host 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701586
 2013-06-05 12:14:01 +02:00
Thiago Santos
7c12435f9b qtdemux: make sure taglist is writable before adding tags 
Avoids assertions
 2013-06-02 15:37:06 -03:00
Thiago Santos
78dfdee2aa qtdemux: effectively skip tracks that weren't listed on the 1st moov 
Without this, stream is NULL and the code will try to access it, leading
to segfaults.
 2013-06-02 13:06:15 -03:00
Thiago Santos
70fca21c28 qtdemux: skip redundant check 
!got_moov is already checked the line above
 2013-06-02 13:06:15 -03:00
Stefan Sauer
bcf1bba689 level: remove unused variables in instance struct 2013-06-01 21:34:37 +02:00
Anton Belka
db29522a43 wavenc: add tags & toc support 
Write tags as LIST INFO chunk. Format the toc as cue + LIST adtl chunk. Remove
old #ifdef'ed code.
 2013-06-01 21:34:37 +02:00
Wim Taymans
1f0600ee6f Revert "rtph264pay: Restructuring to allow for adding optional caps" 
This reverts commit 61666898cf.

This commit changes what the set_sps_pps() function does, not it doesn't
set caps anymore (and should have been renamed). The main problem is that
not all call sites are updated and thus leak the string.
 2013-05-31 15:18:48 +02:00
Wim Taymans
1516c14881 Revert "rtph264pay/depay: Add frame dimensions a payloaded caps" 
This reverts commit 3dca756a5d.

The H264 RTP spec has no attributes for width and height.
 2013-05-31 15:11:12 +02:00
Wim Taymans
b79d217396 Revert "rtph264pay/depay: Add optional framerate caps for use in SDP" 
This reverts commit d8825e2a5c.

There is no framerate attribute in the h264 RTP spec.
 2013-05-31 15:09:51 +02:00
Wim Taymans
190b3d6688 Revert "rtpjpegpay/depay: Replace framesize caps with width/height" 
This reverts commit 0075d111b4.

Extra application/x-rtp are SDP fields, which are strings.
 2013-05-31 15:08:16 +02:00
Wim Taymans
f870cef8bc Revert "rtpjpegpay/depay: Replace framerate caps field with fraction" 
This reverts commit 9fd25a810b.

We deal with sdp attributes in application/sdp, which are always strings.
 2013-05-31 15:05:51 +02:00
Wim Taymans
25082a50b9 rtspsrc: add extra TLS url protocols 
We also support TLS protocols now.
 2013-05-31 12:34:22 +02:00
Sebastian Dröge
e2e1d1a158 videomixer: Add FIXME comment about the DURATION query from adder 
Currently the code just takes with maximum upstream duration, which
is wrong. It should be the maximum upstream duration in running time.
 2013-05-30 23:56:38 +02:00
Mathieu Duponchelle
5223868caa videomixer: Set a reference to mix->current_caps as the QUERY_CAPS result. 2013-05-30 15:36:48 -04:00
Stefan Sauer
6feaf69bec level: misc cleanups 
Fix some oudated comments. Sort out some confusion of interval_frames and num_frames.
 2013-05-30 17:38:55 +02:00
Stefan Sauer
52282b5faa level: fix discontinuities in timestamps 2013-05-28 19:09:12 +02:00
Wim Taymans
80850df711 rtspsrc: create and push stream-start in TCP mode 2013-05-28 15:45:49 +02:00
Wim Taymans
4fc1f3088b rtspsrc: remove some obsolete code 
It is not needed to do a state change from the _play() function on
ourselves. The state change function already did that and we don't want to
interfere with that (or use hacks to avoid interference).
 2013-05-28 15:10:07 +02:00
Wim Taymans
e6f850996b rtspsrc: set RTCP caps on the RTCP pads 2013-05-28 12:26:25 +02:00
Wim Taymans
63f0ecbbe7 rtpsession: send stream-start and segment events 
Also send stream-start and segment event on the RTCP pad.
We don't need to send anything on the sync_src pad because we
already forwarded all incomming events.
 2013-05-28 12:26:25 +02:00
Wim Taymans
779bcc093c rtspsrc: add signal to handle server requests 
Add a signal to be notified of a server request. The signal handler can then
construct the response message for the server.

See https://bugzilla.gnome.org/show_bug.cgi?id=632207
 2013-05-28 12:26:24 +02:00
Nicolas Dufresne
cd30a81ee3 videomixer: Maintain z-order when new pad are added 
https://bugzilla.gnome.org/show_bug.cgi?id=701109
 2013-05-27 22:43:25 -04:00
Thibault Saunier
7a3df1ab31 videomixer: Always handle flush_stop_pending atomically 
It is not protected with the COLLECT_PADS_STREAM_LOCK anymore
 2013-05-25 12:20:08 -04:00
Thibault Saunier
608bd3e2db videomixer: Do not take COLLECT_PADS_STREAM_LOCK when unnecessary 
Collectpad takes the lock itself when receiving serialized events
and we should not take it for not serialized ones
 2013-05-25 11:03:31 -04:00
Sebastian Dröge
1b5a8ac41c flxdec: Properly skip non-frame chunks 2013-05-24 19:34:05 +02:00
Sebastian Dröge
ae3ee32f42 flxdec: Flush data from adapter after reading it 
Otherwise we're going in an infinite loop, reading the same data
over and over again.
 2013-05-24 19:31:14 +02:00
Andoni Morales Alastruey
a62af107ae goom2k1: fix more duplicated symbols 2013-05-24 09:29:23 +02:00
Sebastian Rasmussen
9fd25a810b rtpjpegpay/depay: Replace framerate caps field with fraction 
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
 2013-05-23 21:05:49 +02:00
Sebastian Rasmussen
0075d111b4 rtpjpegpay/depay: Replace framesize caps with width/height 
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.

Keep parsing a-framerate, x-framerate and x-dimensions in rtpjpegdepay
to be backwards compatible with previous payloaders.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
 2013-05-23 21:05:43 +02:00
Sebastian Rasmussen
d8825e2a5c rtph264pay/depay: Add optional framerate caps for use in SDP 
This allows for applications to format SDP attributes and still do SDP
offer/answer based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
 2013-05-23 21:04:17 +02:00
Sebastian Rasmussen
3dca756a5d rtph264pay/depay: Add frame dimensions a payloaded caps 
This allows for applications to format SDP attributes and still do SDP
offer/answer based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
 2013-05-23 21:04:11 +02:00
Sebastian Rasmussen
61666898cf rtph264pay: Restructuring to allow for adding optional caps 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
 2013-05-23 21:04:00 +02:00
Sebastian Dröge
e26b8c2832 (dyn|multi)udpsink: Add properties to specify the bind address and port 
By default we use the any addresses and a random port for binding the socket.
 2013-05-23 18:42:09 +02:00
Sebastian Dröge
5b79b8ff3c (dyn|multi)udpsink: Bind socket before using it 
https://bugzilla.gnome.org/show_bug.cgi?id=700878
 2013-05-23 18:05:07 +02:00
Sebastian Dröge
1ed7f7a6a8 (multi)udpsink: Add missing getters for socket-v6 and used-socket-v6 properties 2013-05-23 17:26:31 +02:00
Nicolas Dufresne
d8c5e31657 videomixer: Don't hold stream-lock while pushing non-serialized events 
https://bugzilla.gnome.org/show_bug.cgi?id=700868
 2013-05-23 09:20:04 -04:00
Nicolas Dufresne
a7e0f251ca videomixer: Don't hold object lock while sending events 
https://bugzilla.gnome.org/show_bug.cgi?id=700868
 2013-05-23 09:20:04 -04:00
Sebastian Dröge
ecc6c607ff deinterlace: The return value of gst_pad_set_caps() is not relevant anymore 
Caps can fail to be set because the pad is not linked yet for example.
 2013-05-22 17:34:07 +02:00
David Schleef
318cd39c3e qtdemux: Add error if file has playready drm 2013-05-21 18:21:49 -07:00
Thibault Saunier
18ef4f18d0 videomixer: Send a reconfigure event upstream if sinkpad caps are not usable 
https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2013-05-21 12:15:36 -04:00
Alexander Schrab
a1df050356 mulawdec: Handle NULL buffers in handle_frame 
https://bugzilla.gnome.org/show_bug.cgi?id=698894
 2013-05-21 15:18:04 +02:00
Sebastian Rasmussen
2361567bae rtpjpegpay/depay: Add framesize caps for use in SDP 
The format of the value adheres to RFC6064 and it is meant to be parsed
and included in the SDP sent by gst-rtsp-server to its clients.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
 2013-05-21 09:09:03 +02:00
Sebastian Rasmussen
919eed0787 rtpjpegpay: Add optional framerate caps for use in SDP 
The format of the value adheres to RFC4566 and it is meant to be parsed
and included in the SDP sent by gst-rtsp-server to its clients.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
 2013-05-21 09:08:21 +02:00
Mathieu Duponchelle
2d3910fc79 videomixer: When all sinkpads are eos, update output segment stop and forward it 
https://bugzilla.gnome.org/show_bug.cgi?id=699793
 2013-05-20 21:06:56 +02:00
Mathieu Duponchelle
521c9a7b5d videomixer: Don't reset the output segment on flush stop 
Only init it when getting from READY to PAUSED, and change it on seek events.

https://bugzilla.gnome.org/show_bug.cgi?id=699793
 2013-05-20 21:03:03 +02:00
Thibault Saunier
86b106091c videomixer: Send caps event from the streaming thread 
This way we avoid races in caps negotiation and we make sure
that the caps are sent after stream-start.

https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2013-05-19 09:28:04 -04:00
Thibault Saunier
718f9004d0 videomixer: Do not send flush_stop when receiving a seek 
There is no reason to send a flush-stop when receiving a seek event.
In the case of a flushing seek, we could eventually want to, but in
the code path were we check if the seek is "flushing", we have the
following comment that makes sense:

"we can't send FLUSH_STOP here since upstream could start pushing data
after we unlock mix->collect.
We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
forwarding the seek upstream or from gst_videomixer_collected,
whichever happens first."

https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2013-05-19 09:28:04 -04:00
Thibault Saunier
85b6852deb videomixer2: Protect flush_stop_pending with the collectpad stream lock 
And make sure to expect a flush-stop after a flush-start

https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2013-05-19 09:28:04 -04:00
Michael Olbrich
d1c56376d6 rtpmp4apay: clear config buffer before using it 
This is necessary because parts of the memory are only modified with "|="

https://bugzilla.gnome.org/show_bug.cgi?id=700514
 2013-05-18 10:57:56 +01:00
Thiago Santos
55caa99ccd qtdemux: Do not expect EOS after a segment event if upstream is mss 
In case qtdemux is handling a mss stream, do not mark the stream to wait
for EOS after a segment. Even if it seems to be the last one according to
the current streams information.

MSS handling is different here because there is another demuxer driving
the pipeline
 2013-05-16 16:50:49 -03:00
Thiago Santos
5517e352ab qtdemux: only set channels and rate if qtdemux knows it 
Setting both of those to 0 is pointless and means that qtdemux
doesn't know the real value. Avoid setting it in this case.
 2013-05-16 16:50:49 -03:00
Arnaud Vrac
6edcc564ba qtdemux: set alac caps using info from codec buffer 
The samplerate field in the STSD atom is not right for some ALAC files
(usually when audio is 96kHz/24bits), so the audio caps must be
extracted from the codec data.

https://bugzilla.gnome.org/show_bug.cgi?id=700382
 2013-05-15 18:42:11 +01:00
Arnaud Vrac
8ed611cdbc avidemux: do not push discont buffers if they aren't discont 
https://bugzilla.gnome.org/show_bug.cgi?id=682110
 2013-05-15 13:16:11 +01:00
Joshua M. Doe
837dcfb363 videocrop: Add support for GRAY16_LE/GRAY16_BE 
https://bugzilla.gnome.org/show_bug.cgi?id=700331
 2013-05-15 09:29:30 +02:00
Sebastian Dröge
41e1af3751 rgvolume: Send all events through the proxypads instead of just sending to the target 
Otherwise the sticky events are missing on the proxypads.
 2013-05-14 17:29:58 +02:00
Sebastian Dröge
4fdbf88a65 matroskaparse: Make sure to send a segment event before dataflow 2013-05-14 13:52:18 +02:00
Sebastian Dröge
5c8bb90262 deinterlace: Improve handling of min/max buffer numbers of the buffer pool 2013-05-14 09:45:12 +02:00
Matej Knopp
30c00f4fb7 deinterlace: set caps for buffer pool config 2013-05-14 09:38:24 +02:00
Olivier Crête
4f0fdabf10 multifilesink: Let the base class do get_times 
This will make sync=TRUE work, the default is still sync=FALSE
 2013-05-13 13:34:22 -04:00
Nicolas Dufresne
f67c227878 interleave: Send stream-start before caps event 2013-05-13 15:37:38 +02:00
Nicolas Dufresne
04c9f43567 rtpmux: Send stream-start before caps 2013-05-13 15:37:05 +02:00
Sebastian Dröge
6dee7d3a06 icydemux: Fix sticky event handling 2013-05-13 15:19:25 +02:00
Sebastian Dröge
9ac456bd43 flvmux: Push sticky events in the right order 2013-05-13 15:06:03 +02:00
Sebastian Dröge
0ab23ef5c9 deinterleave: Fix sticky event handling 2013-05-13 14:54:35 +02:00
Sebastian Dröge
c94fbf6206 deinterleave: Code style fixes 2013-05-13 13:55:44 +02:00
Sebastian Dröge
f28ab45f3e rtpgstpay: First let baseclass handle events, then put them into the stream 
Fixes handling of sticky events.

https://bugzilla.gnome.org/show_bug.cgi?id=700213
 2013-05-13 13:44:35 +02:00
Tim-Philipp Müller
8359b6bff1 multipartdemux: fix example pipeline 
Need jpegparse.
 2013-05-10 14:01:14 +01:00
Nicolas Dufresne
0b737fba0d shapewipe: Can't map twice the same buffer for writing 
I took the opportunity to simplify that code a bit. We now use
gst_buffer_make_writable() to make the buffer writable and map twice the
same buffer, with first map being read/write, and second read only. This
get rid of the critical:

GStreamer-CRITICAL **: gst_structure_set_name: assertion `IS_MUTABLE

https://bugzilla.gnome.org/show_bug.cgi?id=700044
 2013-05-10 09:27:02 +02:00
Nicolas Dufresne
13a5d0304d shapewipe: Ensure caps are writable 
The exist one case where that we endup with original caps in ret, in which
case we are not guaratied to have writable caps. Simply ensure this is the
caps are writable before entering the loop.

https://bugzilla.gnome.org/show_bug.cgi?id=700044
 2013-05-10 09:26:07 +02:00
Nicolas Dufresne
59c2f459de shapewipe: Fix sample pipeline in documentation 
https://bugzilla.gnome.org/show_bug.cgi?id=700044
 2013-05-10 09:26:00 +02:00
Sebastian Dröge
3110b7cc31 Revert "videomixer2: Take into account new segments" 
This reverts commit 84ae670ab4.

Actually this is not how it is supposed to work. videomixer
creates a [0,-1] segment and then puts frames of the different
streams there based on their running times in their own segments.
 2013-05-09 16:26:19 +02:00
Mathieu Duponchelle
84ae670ab4 videomixer2: Take into account new segments 
Also forward the event downstream on the next opportunity.

https://bugzilla.gnome.org/show_bug.cgi?id=699793
 2013-05-09 16:18:54 +02:00
Tim-Philipp Müller
643450c9b8 Revert "gstrtspsrc: set buffer-size for multicast buffers" 
This reverts commit 2481e95d03.

This is already done five lines above, it was added a year
ago in commit 561b131e.
 2013-05-09 09:09:59 +01:00
Nicolas Dufresne
2d53229a86 audiowsinclimit: Frequence property renamed cutoff 
Updating the documentation to reflect this change.

See: https://bugzilla.gnome.org/show_bug.cgi?id=699964
 2013-05-09 08:46:04 +02:00
Aha Unsworth
2481e95d03 gstrtspsrc: set buffer-size for multicast buffers 
For receiving video data via RTSP when the video is sent via
multicast there is no way to specify the udpsrc buffer-size.

On windows the native network buffer is not large and with video
i-frames being huge the buffer is to small and you get i-frame corruption,
it looks terrible, and there is no (easy) way to set the udpsrc buffer-size.

https://bugs.freedesktop.org/show_bug.cgi?id=52264
 2013-05-08 16:57:53 -03:00
Sebastian Dröge
1588cda9a1 videomixer2: Send stream-start before caps event 
https://bugzilla.gnome.org/show_bug.cgi?id=699895
 2013-05-08 16:02:46 +02:00
Thiago Santos
a0e934e72e qtdemux: push new caps events when caps change 
Whenever the demuxer has a new caps on a stream, it should set the
new_caps variable to true and a new caps event will be pushed before
the next buffer
 2013-05-07 19:29:17 -03:00
Thiago Santos
725faab590 qtdemux: do not push discont buffers if they aren't discont 
qtdemux takes its buffers from a GstAdapter. Those buffers are created
from the larger buffer that it obtained from upstream and they carry
the same flags, including DISCONT if it is set. In these cases, all
buffers that qtdemux is going to push would be marked as DISCONT.

This scenario can make parsers/decoders flush on every buffer leading
to no decoding at all hapenning. This patch prevents this by unsetting
the flag if it shouldn't be set.
 2013-05-07 19:29:17 -03:00
Thiago Santos
4d073beeee qtdemux: some code cleanup for mss handling code 
* Explicitly init variables for fragmented formats at init
* Do not use GstClockTime type if the variable isn't a timestamp
* Fix a style/readability issue at an if block
* Group 2 mss mode conditional blocks together to improve readability

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:29:17 -03:00
Thiago Santos
d1b91c755c qtdemux: avoid storing non-time newsegments to push later 
This can confuse downstream when they get a byte segment after receiving
the natural time segment from qtdemux that it sends when starting to
push buffers. This is specially the case with parsers that try to
convert the position from byte to time format and might miss the
correct position for playback to start.
 2013-05-07 19:29:17 -03:00
Thiago Santos
895525b5cb qtdemux: avoid setting fields to non-writable caps 2013-05-07 19:29:17 -03:00
Wim Taymans
544d926732 qtdemux: don't send so many segment events 
Only send one segment event in the beginning of the stream, not
after each moov and moof atom.

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:29:17 -03:00
Wim Taymans
d9cd4fcc17 qtdemux: place incomming timestamps on output 
Place the incomming timestamp (if any) directly onto the outgoing buffers
and interpollate other timestamps.

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:29:17 -03:00
Thiago Santos
cca2f555d1 qtdemux: improve reset of internal status 
Reset different variables on state changes to ready and when
handling a flush-stop. For handling flush stops we should check
if there is an upstream adaptive demuxer driving the pipeline as this
means that qtdemux will get a new moov atom. For 'standard' isomedia
streams this isn't true and qtdemux should keep the previous moov
information around.

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:29:17 -03:00
Thiago Santos
6c69e59677 qtdemux: prepare qtdemux to accept multiple dash moovs in a row 
Whenever dashdemux switches bitrates it sends a new moov with the
new stream configuration. qtdemux should now handle this by splitting
the exposing and configuration of streams into separate functions. When
the stream is new it is configured and exposed, when it is a new bitrate
of an existing stream it is only reconfigured.

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:25:30 -03:00
Andre Moreira Magalhaes (andrunko)
2a7d3d1598 qtdemux: Move FLUSH_STOP/PAUSED_TO_READY handling to a reset method. 
Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:18:03 -03:00
Louis-Francis Ratté-Boulianne
d499b461da qtdemux: Remove old pads when exposing streams and other general fixes. 
Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:18:03 -03:00
Thiago Santos
a3c19eeea1 qtdemux: handle mss streams 
smoothstreaming streams should be handled as a special kind of
fragmented isomedia. In MSS the fragments will not contain a
'moov' atom with the media descriptions, this has to be extracted
from the caps.

Additionally, there should be another demuxer upstream that is likely
going to be the one to answer/act on queries and events, so qtdemux has
to forward those upstream.
 2013-05-07 19:18:03 -03:00
Sebastian Rasmussen
9532b04947 rtpgstpay: fix invalid memory access in event handler 
First process event in payloader, then hand it to the
base class which takes ownership of the event.

https://bugzilla.gnome.org/show_bug.cgi?id=699637
 2013-05-04 10:49:23 +01:00
Tim-Philipp Müller
68ac392e8f ac3parse, dcaparse: check buffer size before trimming 
and unref old buffer as soon as possible.
 2013-05-04 10:08:47 +01:00
Andoni Morales Alastruey
3462282b83 dcaparse: add support for "audio/x-private1-dts" 2013-05-03 13:44:23 +02:00
Andoni Morales Alastruey
4531381541 ac3parse: add support for "audio/x-private1-ac3" 2013-05-03 13:44:23 +02:00
Andoni Morales Alastruey
4a78a77e65 rtp: fix duplicated symbols with libvpx 2013-05-02 14:03:33 +02:00
Andoni Morales Alastruey
584fdbad84 goom2k1: fix duplicated symbols with goom 2013-05-02 14:03:26 +02:00
Sebastian Dröge
ae05ed4f05 rtph264pay: If the adapter is empty on EOS don't try to map its content 
https://bugzilla.gnome.org/show_bug.cgi?id=699314
 2013-05-01 15:49:45 +02:00
Ognyan Tonchev
0584d5c4c9 matroskademux: add stream-format=raw to aac caps 
https://bugzilla.gnome.org/show_bug.cgi?id=699303
 2013-05-01 15:47:15 +02:00
Tim-Philipp Müller
7ccb387e85 udp: log WARNING debug message if UDP multicast is likely to be broken 2013-04-27 11:25:12 +01:00
Tim-Philipp Müller
4273eccace udpsrc: add includes to get socklen_t defined on Windows 
https://bugzilla.gnome.org/show_bug.cgi?id=692400
 2013-04-27 11:16:54 +01:00
Yury Delendik
4bc06859d1 qtdemux: add support for VP6F VP6 flash codec 
https://bugzilla.gnome.org/show_bug.cgi?id=699010
 2013-04-27 09:39:45 +01:00
Edward Hervey
3e5ad52c0d monoscope: Fix debug statement 2013-04-26 12:16:49 +02:00
Alexander Schrab
3ec9673dfc mulawdec: change base class to GstAudioDecoder 
https://bugzilla.gnome.org/show_bug.cgi?id=698894
 2013-04-26 08:46:34 +02:00
Mathieu Duponchelle
6b153ce385 videomixer: send stream-start event. 2013-04-25 16:09:34 -03:00
Wim Taymans
1df2e623b5 docs: add some pay/depayloaders 
See https://bugzilla.gnome.org/show_bug.cgi?id=551631
 2013-04-25 14:05:55 +02:00
Sebastian Dröge
fb0384fa0d mulaw: Some minor memleak fixes and cleanup 2013-04-25 12:44:15 +02:00
Alexander Schrab
f0edb5fb70 mulawenc: change to gstaudioencoder base, added bitrate tags 2013-04-25 12:36:15 +02:00
Sebastian Dröge
b1af93f791 (multi)udpsink: Use separate sockets for IPv4 and IPv6 
https://bugzilla.gnome.org/show_bug.cgi?id=534243
 2013-04-25 12:12:23 +02:00
Sebastian Dröge
0b552150ce dynudpsink: Use separate sockets for IPv4 and IPv6 
https://bugzilla.gnome.org/show_bug.cgi?id=534243
 2013-04-25 12:09:27 +02:00
Sebastian Dröge
ed8ea46424 udp: Don't include removed gstudp.h in noinst_HEADERS 2013-04-25 10:43:56 +02:00
Sebastian Dröge
afb284e3a9 udp: Remove unused enum type 2013-04-25 09:16:14 +02:00
Sebastian Dröge
a957457cc1 udp: Use the generic marshaller instead of generating marshallers 2013-04-25 09:13:51 +02:00
Sebastian Dröge
07d3363436 udpsrc: Rename instance variable from host to multi_group 
This is more consistent as it's used for the multicast-group property.
 2013-04-25 09:07:41 +02:00
Sebastian Dröge
427673d283 udpsrc: Add bind-address property 
This is equivalent to multicast-group currently for backwards compatibility.
In 2.0 this should be handled separately, the former only being the multicast
group and the latter always being the address the socket is bound to, even if
a multicast group is given.
 2013-04-25 09:05:12 +02:00
Wim Taymans
5ba3fd3c63 vrawdepay: return output buffer from process 
Return the output buffer from the process function instead of pushing
it ourselves. This way, the subclass can actually deal with the return
value of the push.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693727
 2013-04-24 16:24:25 +02:00
Wim Taymans
eac9efb92e rtp: a marker bit should translate to RESYNC 
A marker bit on an audio packet does not mean a DISCONT (in the GStreamer sense
of missing data) but it means that the packet is the end of a talkspurt and thus
a good opportunity to resync to the clock. Use the RESYNC buffer flag to note
this.
Real discontinuities are marked with DISCONT still when the seqnum has a GAP or
when the input buffer has the DISCONT flag set.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=627204
 2013-04-24 15:42:45 +02:00
Sebastian Dröge
fdb667ae00 rtpjpegdepay: Drop frame if it's less than 2 bytes large 
https://bugzilla.gnome.org/show_bug.cgi?id=677560
 2013-04-22 10:19:29 +02:00
Sreerenj Balachandran
504360fe36 autodetect: use _plugin_feature_rank_compare API instead of duplicating the code. 2013-04-18 14:00:06 +02:00
Olivier Crête
24bb263d54 videomixer: Don't unref query, we don't own it 
Fixes double-unref bug. Bug found by Youness Alaoui
 2013-04-16 19:29:48 -04:00
Sebastian Dröge
b0b0557c48 gst: Add better support for static plugins 2013-04-15 15:54:11 +02:00
Andoni Morales Alastruey
2ea9a66dd5 goom2k1: fix duplicated symbol with goom 2013-04-15 08:43:05 +02:00
Wim Taymans
9d7519f66e rtp: register tag image types 
The rtpgstdepay needs the type to be available in order to deserialize the
event.
 2013-04-12 16:18:42 +01:00
Wim Taymans
b1f4587d75 rtpgstdepay: handle event parse failures better 2013-04-12 16:18:42 +01:00
Anton Belka
b959d827be wavenc: add TOC setter support 2013-04-12 14:35:47 +02:00
Stefan Sauer
f4577ff492 wavenc: small cleanups for toc handling 
Don't add empty labl/note chunks. Always pass instance as the first param. Add more logging.
 2013-04-12 14:35:47 +02:00
Sebastian Dröge
b17750ed9e rtspsrc: Proxy the ntp-sync property of rtpbin 2013-04-12 12:58:50 +02:00
Sebastian Dröge
53dae1585e rtspsrc: Give the manager always the name "manager" 
This allows to use the GstChildProxy interface to adjust
properties on it.
 2013-04-12 12:51:05 +02:00
Anton Belka
bda2703e88 wavenc: add 'note' chunk support 2013-04-11 20:47:18 +02:00
Wim Taymans
f8013487c9 rtspsrc: add support for NetClientClock 
When the server suggests a GstNetTimeProvider in the SDP, set up a
GstNetClientClock that slaves to the remote clock and suggest this clock in
provide_clock.
 2013-04-11 15:00:05 +01:00
Wim Taymans
f96aa414e1 udpsink: avoid alloc and free in render function 
Avoid doing alloc and free in the render function for each buffer. Instead,
allocate the needed arrays in _init and use those.
 2013-04-11 14:57:11 +01:00
Stefan Sauer
48b9919e31 waveparse: remove superfluous g_list_first() calls 
The variables already point to the start of the list.
 2013-04-10 14:25:24 +02:00
Andreas Fenkart
20d3ec8810 rtpsbcdepay: fix sbc frame length calculation for mono and stereo modes 
https://bugzilla.gnome.org/show_bug.cgi?id=697463
 2013-04-09 23:17:57 +01:00
Anton Belka
5ae92ce770 wavparse: add 'note' chunk support 
Add 'note' chunk support in TOC as GST_TAG_COMMENT

https://bugzilla.gnome.org/show_bug.cgi?id=696549
 2013-04-09 22:58:27 +02:00
David Schleef
a55ccff854 qtdemux: check value inside enda to set endianness 2013-04-09 13:30:17 -07:00
Wim Taymans
ece73b786a icydemux: avoid copy when we can 2013-04-09 17:34:12 +02:00
Wim Taymans
91a3afc4dc gstpay: use bufferlist to avoid memcpy 2013-04-09 16:53:31 +02:00
Wim Taymans
3d7d757521 udpsink: improve debug 2013-04-09 16:53:31 +02:00
Alexander Schrab
79d5a7d03c wavparse: error out if we receive eos before any valid data 
https://bugzilla.gnome.org/show_bug.cgi?id=696684
 2013-04-09 00:27:31 +01:00
Matej Knopp
67c2219687 deinterlace: force deinterlacing in "interlaced" mode 
https://bugzilla.gnome.org/show_bug.cgi?id=697467
 2013-04-07 20:48:21 +01:00
Nicola Murino
c41c16424d rtpsbcdepay: fix printf format compiler warnings 
https://bugzilla.gnome.org/show_bug.cgi?id=697343
 2013-04-05 13:50:19 +01:00
Stefan Sauer
b79f667ef4 level: resync on discont 
Drop pending data on discont and start a new cycle with a new base timestamp.
Cleanup some variables.
 2013-04-04 22:49:49 +02:00
Olivier Crête
f8831c0cd2 rtpsbcdepay: Rank as secondary 
This way, it will be selected by decodebin
Bug reported by andreas.fenkart@streamunlimited.com

https://bugzilla.gnome.org/show_bug.cgi?id=697227
 2013-04-03 18:25:36 -04:00
Stefan Sauer
2e56032031 level: subdivide buffers for sample accurate interval handling 
Previously we would skip level message when processing buffers > the requested
interval. Also the message frequency would contain quite some jitter due to only
considering them at the end of buffers.

Cleanup the tests while we're at it.
 2013-04-03 21:40:17 +02:00
Stefan Sauer
b062171dda spectrum: remove old since comment 2013-04-03 20:30:08 +02:00
Sebastian Dröge
d80ff8e7f3 rtspsrc: Proxy the multicast-iface property of udpsrc 2013-04-03 17:53:13 +02:00
Olivier Crête
6f3734c305 rtpssrcdemux: Only forward stick events while holding the sinkpad stream lock 
Otherwise we get a race where if the RTCP packet comes in first and while
it is added the pads, the segment event arrives on the RTP stream, the event
may be lost completely and never forwarded.
 2013-04-02 23:42:42 -04:00
Olivier Crête
76679f9ae9 rtpssrcdemux: No need to explicitely forward the caps 
They are forwarded with the other events
 2013-04-02 23:42:41 -04:00
Olivier Crête
4ad8693f3c rtpssrcdemux: Remove unused GstSegment 2013-04-02 23:42:41 -04:00
Olivier Crête
7293b0eff7 rtpssrcdemux: Simplify event forwarding 
Use the gst_pad_forward() mechanic, this way we won't miss pads that are
added while we are pushing
 2013-04-02 23:42:41 -04:00
Olivier Crête
f4c3aef13a rtpssrcdemux: Don't cross the internal links 
We had the wrong condition to check for the internal links, so RTP and RTCP
pads got crossed!
 2013-04-02 23:42:41 -04:00
Tim-Philipp Müller
078ff16abe matroskademux: fix some debug messages 2013-04-03 00:49:37 +01:00
Arnaud Vrac
00b46b4744 matroskademux: handle TrueHD audio codec id 
https://bugzilla.gnome.org/show_bug.cgi?id=697113
 2013-04-02 22:47:54 +01:00
Wim Taymans
ac2bcfa833 theorapay: add delta-unit to output frames 2013-03-31 19:14:04 +02:00
Matej Knopp
5686512b77 qtmux: use timestamp delta as duration if possible 
https://bugzilla.gnome.org/show_bug.cgi?id=696437
 2013-03-30 15:18:45 -07:00
Josep Torra
509631f60b rtp: fixes debug message printf related compiler warnings in SBC depayloader 2013-03-30 09:44:41 +01:00
Arun Raghavan
87bdad4bfc rtp: Add an rtpsbcdepay element 
Pretty straightforward - takes SBC encapsulated in RTP, depayloads, and
pushes out SBC buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-03-28 17:22:33 +00:00
Tim-Philipp Müller
477cc51fe7 rtp: fix SBC payloader 
Init RTP buffer on stack correctly, so mapping it works
without criticals and the payloader actually works.
 2013-03-27 22:18:34 +00:00
David Schleef
53f8b05b08 Use %03u for format in gst_pad_create_stream_id_printf() 2013-03-25 18:57:08 -07:00
Sebastian Dröge
56062768af capssetter: Prevent unneeded caps copying and allocation 2013-03-25 10:12:03 +01:00
Dirk Van Haerenborgh
766c5b22ed capssetter: Pass any or filter caps upstream 
capsetter accepts anything and just forwards different caps,
as such it should return ANY caps on the sinkpad.

https://bugzilla.gnome.org/show_bug.cgi?id=693005
 2013-03-25 10:11:32 +01:00
Tim-Philipp Müller
35769f7c5d wavparse: expose CUE sheet items as tracks not chapter entries in TOC 
https://bugzilla.gnome.org/show_bug.cgi?id=677306
 2013-03-24 17:55:55 +00:00
Tim-Philipp Müller
163a7afa1a wavenc: add some example pipelines 2013-03-23 12:59:26 +00:00
Anton Belka
e808173483 wavenc: add TOC support 
https://bugzilla.gnome.org/show_bug.cgi?id=680998
 2013-03-23 12:55:08 +00:00
Matej Knopp
f29e62c131 qtdemux: make empty subtitle buffer recognition more robust 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-23 11:24:23 +00:00
David Schleef
c0443a17c4 qtmux: Fix test regression with one buffer streams 2013-03-22 15:14:15 -07:00
David Schleef
5bd2864101 qtdemux: split large raw audio samples 
In order to deal with a file that has samples that are 24 seconds
long.  Seeking still doesn't work with such files.
 2013-03-22 14:14:05 -07:00
David Schleef
364433c105 qtmux: Remove documentation for dts-method 2013-03-22 14:14:04 -07:00
David Schleef
6571e388be qtmux: deprecate dts-method property 2013-03-22 14:14:04 -07:00
David Schleef
ee56a7cb99 qtmux: Fix problems causing bad durations in file 
- Fix up out-of-order incoming DTS values.
- Fix duration of initial sample.
 2013-03-22 14:14:04 -07:00
David Schleef
816e186029 qtmux: fix all timestamps once first_ts is determined 2013-03-22 14:14:04 -07:00
David Schleef
258c40c6dd qtmux: Use PTS/DTS from incoming buffers 
Remove old DTS guessing code.
 2013-03-22 14:14:04 -07:00
Nicola Murino
709f05234f qtmux: expose mulaw caps 
https://bugzilla.gnome.org/show_bug.cgi?id=696052
 2013-03-22 20:08:06 +00:00
Rodolfo Schulz de Lima
874808fd2c qtdemux: fix sample leak when processing private qt tags 
https://bugzilla.gnome.org/show_bug.cgi?id=696355
 2013-03-22 08:47:17 +00:00
Matej Knopp
d8ac666137 qtmux: set stream language code from tag 
https://bugzilla.gnome.org/show_bug.cgi?id=696358
 2013-03-22 08:40:26 +00:00
Matej Knopp
49d9050e9a qtdemux: send GAP events for subtitle streams 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:37 +00:00
Matej Knopp
516a0b8acb qtdemux: ignore empty subtitle buffers 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:34 +00:00
Matej Knopp
f494635126 qtdemux: recognize SBTL subtype for subtitles 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:14 +00:00
Anton Belka
0f97b6f978 flacparse: add support for the toc-select event 
Select tracks from the CUE sheet by sending a toc-select
event based on the uid in the TOC.

https://bugzilla.gnome.org/show_bug.cgi?id=540891
 2013-03-21 00:38:48 +00:00
Michael Smith
b85c5f236b mp4mux: in faststart mode, don't output up to 4 kB of garbage at the end. 2013-03-19 18:09:31 -07:00
Tim-Philipp Müller
5240b7453c sbcparse: pack multiple frames into one output buffer 
Don't output a single buffer for every tiny SBC frame
 2013-03-20 00:35:17 +00:00
Kishore Arepalli
288e05c99d deinterlace: fix infinite loop on EOS with non-default methods or fields 
Fixes problem of infinite loop in gst_deinterlace_reset_history.
Last field in the history was never deinterlaced because idx becomes negative.

Happens e.g. with method=scalerbob fields=bottom or
method=greedyl fields=top

https://bugzilla.gnome.org/show_bug.cgi?id=695644
https://bugzilla.gnome.org/show_bug.cgi?id=693173
 2013-03-17 14:47:26 +00:00
Tim-Philipp Müller
dfde4179e8 avimux: change raw video caps order so that GRAY8 is last 
People like colours.

https://bugzilla.gnome.org/show_bug.cgi?id=695543
 2013-03-12 00:16:18 +00:00
Ognyan Tonchev
3f8ad30cee rtph264pay: Don't use upstream caps with peer_query_caps () 
Calling gst_pad_peer_query_caps () on the src pad with the caps
upstream can produce as a filter from gst_rtp_h264_pay_getcaps ()
is wrong and makes caps negotiation fail if upstream caps are not
NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=695629
 2013-03-11 16:55:13 -04:00
Dirk Van Haerenborgh
065bdf5925 avimux: support raw BGR 
https://bugzilla.gnome.org/show_bug.cgi?id=695543
 2013-03-11 14:51:00 +01:00
Dirk Van Haerenborgh
d7743cf7c6 avidemux: support raw video with negative height 
https://bugzilla.gnome.org/show_bug.cgi?id=695541
 2013-03-11 14:23:46 +01:00
Tim-Philipp Müller
694dbcc5a0 dtmf: move dtmf plugin from -bad to -good 
https://bugzilla.gnome.org/show_bug.cgi?id=687416
 2013-03-09 01:18:30 +00:00
Tim-Philipp Müller
a4c5aa38ec Merge branch 'dtmf-moved-from-bad' 
https://bugzilla.gnome.org/show_bug.cgi?id=687416
 2013-03-09 00:30:38 +00:00
Sebastian Dröge
539126c097 matroska: Include config.h, it's needed for _stdint.h 2013-03-03 11:59:31 +01:00
Sebastian Dröge
1810786083 flacparse: Fix (wrong) use of uninitialized variable compiler warning 2013-03-03 11:53:04 +01:00
Tim-Philipp Müller
677bfecc6f qtdemux: add variant field to H.263 caps 
avdec_h263 won't get plugged otherwise.
 2013-03-02 13:59:52 +00:00
Arnaud Vrac
1cff6427f1 qtdemux: skip disabled tracks 
ISO/IEC 14496-12 specifies disabled tracks should be completely
ignored, so just do it.

Avoids deadlock during prerolling for some files.

Also prevents 'chapter' subtitle tracks from showing up.

https://bugzilla.gnome.org/show_bug.cgi?id=693993
https://bugzilla.gnome.org/show_bug.cgi?id=628790
 2013-03-02 13:54:23 +00:00
Stefan Sauer
15a81baea5 spectrum: remove the since doc-comment from 0.10 2013-02-28 09:43:12 +01:00
Stefan Sauer
b62cb3edcd level: add a "post-messages" property and deprecate "message" 
In spectrum this was changed from 0.10 to 1.0, lets do this here too.
 2013-02-28 09:43:12 +01:00
Olivier Crête
df5ca83baf rtpmp4gdepay: streamtype is not put by all RTSP server, not make it optional 
Specific case here is Wowza 3.5.0
 2013-02-26 14:19:10 -05:00
Thomas Vander Stichele
df8f5f2f83 level: put back deprecation warnings 2013-02-25 00:35:58 +01:00
Thomas Vander Stichele
52b7aab711 level: send last message on EOS 2013-02-25 00:19:22 +01:00
Mark Nauwelaerts
56e2767c20 avidemux: push mode: handle some more 0-size buffer cases 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684944
 2013-02-24 19:28:07 +01:00
Tim-Philipp Müller
8004ae0369 matroskamux: fix up example pipeline in docs 2013-02-23 18:50:52 +00:00
Paul HENRYS
10802cae73 rtpsession: Fix wrong code organisation in case of collision 
change_ssrc field of RTPSession should be set before calling
rtp_session_schedule_bye_locked () as this function will call reconsider function
that will wake up rtcp_thread which will call rtp_session_on_timeout () that will
check change_ssrc to change the ssrc.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184
 2013-02-22 09:28:07 +02:00
Jean-François Fortin Tam
f5cb19e287 alpha: improve descriptions of chroma keying-related properties and enums 
https://bugzilla.gnome.org/show_bug.cgi?id=694374
 2013-02-22 00:09:56 +00:00
Youness Alaoui
a65fd146f8 alpha: Do not override the method with custom r/g/b values 
Depending on the order g_object_set() calls aare made, the
target r/g/b settings will override the method if set to
green/blue. Change that so we do not use the target-r/g/b values
unless the method is set to custom.

https://bugzilla.gnome.org/show_bug.cgi?id=694374
 2013-02-22 00:04:51 +00:00
Ognyan Tonchev
42d8b96f2d auparse: do not leak src_caps 
https://bugzilla.gnome.org/show_bug.cgi?id=694275
 2013-02-21 19:31:59 +00:00
Wim Taymans
a61055809f rtpsession: only delay RTCP when we are a sender 
Only delay the RTCP thread when we are a sender, which we can know because we
have a send_rtp_src pad. Otherwise we might delay the RTCP thread if we
are only a receiver and then there is no code path that wakes up the
RTCP thread and we end up without RTCP packets.
 2013-02-20 21:07:41 +02:00
Tim-Philipp Müller
5b19be933b qtdemux: fix up dodgy code that tries to fix up a broken moov atom 
After gst_buffer_new_and_alloc() gst_buffer_copy_into() will likely
append to the already-existing memory instead of filling it.
 2013-02-18 20:04:05 +00:00
Tim-Philipp Müller
34b81f7c93 qtdemux: fix potential crash on short MOOV atom 
Don't unmap short MOOV atom buffer twice, which happened
in the case where we don't fix up the MOOV atom.

Fixes crashes when thumbnailing partial mp4 file where
the MOOV atom is still incomplete.

https://bugzilla.gnome.org/show_bug.cgi?id=694010
 2013-02-18 16:35:08 +00:00
Stefan Sauer
99f84b8c4c audiopanorama: remove channel-mask from caps 
The channel-mask is only needed for channels>2 which we don't do.
 2013-02-15 21:30:15 +01:00
Tim-Philipp Müller
01c6512d5f udpsrc: use g_socket_set_option() to set buffer size with newer GLib versions 
So we have to worry less about portability.

https://bugzilla.gnome.org/show_bug.cgi?id=692400
 2013-02-15 14:11:36 +00:00
Sebastian Dröge
a7ddbc03fe rtp-payloading: Fix unit test caps and AMR depayloader sink template caps 
Fields were missing from the actual caps, or too many fields
existed in the template caps.
 2013-02-13 12:02:46 +01:00
Michael Smith
e3430b0d07 qtdemux: extract codec_data for ProRes 2013-02-12 13:19:53 -08:00
Tim 'mithro' Ansell
c499a81848 avimux: Fixing buffer leak in gst_avi_mux_do_buffer 
gst_avi_mux_do_buffer was leaking data from gst_collect_pads_pop.
 2013-02-12 10:09:05 +01:00
Mark Nauwelaerts
bf81dce432 avidemux: correct duration for audio VBR buffers in pull mode 2013-02-10 15:10:32 +01:00
Mark Nauwelaerts
f0645b79c5 avidemux: proper position reporting and push mode timestamping 
... and align current_total semantics in push and pull mode,
which tracks bytes for CBR and blocks for VBR.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691481
 2013-02-08 21:41:55 +01:00
Wim Taymans
2d5319c1fa rtpsession: delay RTCP until first RTP packet 
Delay sending the first RTCP packet until we have sent the first RTP packet.
Otherwise we will send out a Receiver Report instead of a sender report.

See https://bugzilla.gnome.org/show_bug.cgi?id=691400
 2013-02-08 17:05:27 +01:00
Wim Taymans
2971ed44ee rtpsession: remove dead code 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355
 2013-02-07 15:06:40 +01:00
Paul HENRYS
0e91c949d8 rtpptdemux: forward sticky events and then set caps 
When a new src pad is added, first forward the sticky events and then
set the caps on the src pad

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786
 2013-02-07 14:38:20 +01:00
Markovtsev Vadim
7cebe2fc41 rtpjitterbuffer: improve debug output 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688935
 2013-02-07 14:32:26 +01:00
Wim Taymans
978cc9f538 rtpbin: rework cleanup of streams 
Move the work of cleaning up the client streams in the free_stream
function. This allows us to properly clean up the client streams when we
remove an RTP stream as well.

Based on patch by Sujay <sdatar@cisco.com>

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156
 2013-02-07 13:02:34 +01:00
Tim 'mithro' Ansell
3a5d17e852 videomixer2: avoid caps leak 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693307
 2013-02-07 11:40:35 +01:00
Wim Taymans
c3077012c0 jitterbuffer: do skew estimation only for new timestamps 
Only run the skew estimation code when we have a new RTP timestamp. If we have
the same RTP timestamp, we simply use the previous estimation. This works
because the new observation with the same RTP timestamp has to have a bigger
receiver time and is thus not going to influence the estimation except for
causing more jitter.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023
 2013-02-06 17:15:11 +01:00
Wim Taymans
640de61740 rtspsrc: only EOS when our source sends BYE 
Only EOS when we receive a BYE event from the SSRC of our stream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453
 2013-02-06 14:01:16 +01:00
Wim Taymans
0540492ab2 rtspsrc: save the stream SSRC 
Conflicts:
	gst/rtsp/gstrtspsrc.c
 2013-02-06 14:00:56 +01:00
Wim Taymans
c8fb1c720c rtspsrc: flush connection when stopping 
When we stop, we can flush all pending commands so that we can stop and
join the task.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924
 2013-02-06 13:18:18 +01:00
Stefan Sauer
96f8775a0d spectrum: remove outdates readme 
Lets remove the readme from pre-0.1.0 that is completely irrelevant now.
 2013-02-05 22:02:13 +01:00
Stefan Sauer
86ae581928 audiopanorama: add more debug logging 2013-02-05 18:51:27 +01:00
Rico Tzschichholz
682e49a752 audioparsers: fix typo in noinst_headers 2013-02-04 18:38:41 +00:00
Stefan Sauer
1f1fe47cb6 audiopanorama: further port to 1.0 
Transformcaps is not called with caps containing single structures anymore. Also add missing filter handling. Still does not negotiate though.
 2013-02-04 11:08:23 +01:00
Stefan Sauer
d187b96ee2 audiopanorama: fix caps 
We don't turn float into 32bit pcm. Looks like a typo from updating the caps.
 2013-02-03 22:45:52 +01:00
Olivier Crête
fe3e535853 level: Add missing coma between formats 2013-02-03 13:14:50 +01:00
Matthew Waters
b9151a9c28 videomixer: fix eos timestamp check 
fixes hang in videotestsrc num-buffers=20 ! videomixer ! fakesink

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692935
 2013-01-31 16:45:38 +01:00
Dirk Van Haerenborgh
18ff57d6b3 avimux: add support for raw monochrome 8-bit video 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692932
 2013-01-31 13:00:17 +01:00
Wim Taymans
747447d298 rtpsession: avoid '...is used uninitialized' 2013-01-29 10:32:51 +01:00
Youness Alaoui
f6a00ad6e9 qtdemux: set interleaved layout correctly for LPCM audio 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:44:01 +00:00
Youness Alaoui
a76524ea08 qtdemux: add support for LPCM fourcc (uncompressed audio in Quicktime7) 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:57 +00:00
Youness Alaoui
69b814546a qtdemux: print all debug for sound sample description v2 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:49 +00:00
Youness Alaoui
92ff8a9b09 qtdemux: sound sample description v2 doesn't override samples_per_packet 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:42 +00:00
Youness Alaoui
ee3d9cbd98 qtdemux: pass stsd data to qtdemux_audio_caps() 
We will need that later for LPCM format support. Disable
QDM2 parsing of stsd data which dead code before as well
because data was always NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:38 +00:00
Youness Alaoui
6d3ff78575 qtdemux: add len check for sound sample descriptions v1 and v2 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:28 +00:00
Tim-Philipp Müller
629772f735 rtpmanager: use C89-style comments 2013-01-28 23:07:34 +00:00
Olivier Crête
451217c437 gstrtpsession: Fix double-declared variable 2013-01-28 18:06:15 -05:00
Olivier Crête
7300d489fe rtp: Fix compilation errors in previous patches 2013-01-28 17:58:20 -05:00
Haakon Sporsheim
86c13ceae6 rtpsession: Ensure MT safe event handling and plug event leak. 
https://bugzilla.gnome.org/show_bug.cgi?id=667826
 2013-01-28 17:44:31 -05:00
Idar Tollefsen
268c998a32 rtpsession: mt-safe event-push 
By taking a ref of the sink-pad under lock, it won't dissappear
while the push is taking place

https://bugzilla.gnome.org/show_bug.cgi?id=667816
 2013-01-28 17:34:50 -05:00
Pascal Buhler
f459fe2673 rtpssrcdemux: Safely push on pads that might be removed due to a RTCP BYE 
https://bugzilla.gnome.org/show_bug.cgi?id=667815
 2013-01-28 17:01:27 -05:00
Tim-Philipp Müller
721dd1ab26 sbcparse: init some variables to avoid bogus compiler warnings 2013-01-28 11:58:50 +00:00
Wim Taymans
4397c8ffbf rtpdepay: remove payload type restrictions 
Remove the pt restrictions for all the depayloaders that have an
encoding-name. We can use this to autoplug decoders.
Remove the encoding-name for all the payloaders with a fixed payload
type.
We now either have an encoding-name or a pt in the sinkpad caps of
a depayloader.

See https://bugzilla.gnome.org/show_bug.cgi?id=639292
 2013-01-28 12:41:04 +01:00
Marc Leeman
bab2f3c92b rtp: remove payload requirements from selected depayloaders 
encoding name is required in the caps and is a better fit for autoplugging than
the pt value. Hardware manufacturers have a bad habit of skimming through RFCs
and in this case; use unassigned numbers for encoders instead of dynamic
numbers.

In essence, this patch will add support for a lot of Bosch hardware encoders
without breaking autoplugging.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292
 2013-01-28 12:23:41 +01:00
Mark Nauwelaerts
a1a579afeb qtdemux: push mode: only parse moov 1 once 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691570
 2013-01-27 12:54:20 +01:00
Tim-Philipp Müller
47fccbe635 rtpdtmfsrc: fix compiler warning 
gstrtpdtmfsrc.c: In function 'gst_dtmf_src_prepare_message.isra.1':
gstrtpdtmfsrc.c:669:3: error: 's' may be used uninitialized in this function
 2013-01-26 22:58:29 +00:00
Olivier Crête
db5c3f4048 rtpdtmfdepay: Fix missing work in doc 2013-01-25 21:06:05 -05:00
Olivier Crête
92f9a9d9ff rtpdtmfsrc: Post the messages after the clock wait 
This way, the messages will be closer in time to when the packets are sent out
 2013-01-25 20:45:43 -05:00
Olivier Crête
0d316b4f43 rtpdtmfsrc: Only set the duration when starting to send 
The duration depends on the clock rate, which could change due to renegotiation
 2013-01-25 20:45:43 -05:00
Olivier Crête
90497aa3cd rtpdtmfsrc: remove "ssrc" from caps 
ssrc is uint and we don't have a uint range type
 2013-01-25 20:45:43 -05:00
Tim-Philipp Müller
d62019fff2 qtmux: set language to 'undefined' instead of English by default 2013-01-24 21:08:51 +00:00
Mark Nauwelaerts
0777a600e3 audioparsers: sbc: fix bogus compiler warning 
gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame':
gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i
 2013-01-22 19:26:09 +01:00
Thijs Vermeir
16128f0234 autoparsers: use appropriate printf format for gsize 2013-01-16 14:32:56 +01:00
Tim-Philipp Müller
9455a3aee1 rtpsbcpay: update some fields in the caps to their new name 
and to match the parser. "mode" got renamed to "channel-mode"
and "allocation" to "allocation-method".
 2013-01-16 10:19:36 +00:00
Tim-Philipp Müller
9f7a949773 audioparsers: add SBC audio parser 
From-scratch rewrite, the bluez one was useless and broken.

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-01-15 17:45:30 +00:00
Tim-Philipp Müller
39ef892938 rtp: import rtpsbcpay from bluez and port to 1.0 
Compiles, but not tested yet (sbc elements still need to be ported).

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-01-10 12:43:50 +00:00
Olivier Crête
c6dea5d09c dtmf/spandsp: Move dtmfdetect to use libspandsp 
Remove our copy of the tone_detect.c file and use the original
from libspandsp. Also move the element to the spandsp plugin.
 2013-01-09 20:05:16 -05:00
Marcel Holtmann
4196feb659 rtpsbcpay: Remove workaround for compiler warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
fe79c60d74 rtpsbcpay: Add pragma based workaround for GStreamer warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
08e95e7249 rtpsbcpay: Update copyright information 2013-01-10 00:15:36 +00:00
Marcel Holtmann
7fa03c0076 rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin 2013-01-10 00:15:35 +00:00
Marcel Holtmann
27a6b0abfe rtpsbcpay: Update copyright information 2013-01-10 00:15:35 +00:00
Marcel Holtmann
f890079aae rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup) 2013-01-10 00:15:35 +00:00
Johan Hedberg
7d4f846112 rtpsbcpay: More coding style fixes 2013-01-10 00:15:35 +00:00
Luiz Augusto von Dentz
151ad9b28d rtpsbcpay: Remove possible extra memcpy for gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
69c8374b7c rtpsbcpay: Fix bug sending empty packages and remove a buffer copy. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
7b3e4356ea rtpsbcpay: Fix runtime warnings of gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
f74f061f3b rtpsbcpay: Update gstreamer plugin to use new sbc API. 2013-01-10 00:13:14 +00:00
Marcel Holtmann
b9be04f07b rtpsbcpay: Update copyright information 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
687400ecf4 rtpsbcpay: Fixes gstreamer caps and code cleanup. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
a4f9624261 rtpsbcpay: Fix gtreamer payloader sending fragmented frames. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
41e2f4f544 rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
96971cd323 rtpsbcpay: Make a2dpsink to act like a bin and split the payloader. 2013-01-10 00:13:14 +00:00
Wim Taymans
72402cc649 rtp: small improvements 2013-01-08 16:27:42 +01:00
Wim Taymans
af055d9574 jitterbuffer: refactor handle sync code 
Move the code that combines the last SR packet and the current jitterbuffer sync
values into a sync structure, into its own function. We want to reuse this bit
later.
 2013-01-07 15:50:33 +01:00
Wim Taymans
87f7d6b9bf rtp: include downstream latency in SR calculations 
When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
 2013-01-07 15:45:10 +01:00
Wim Taymans
c631ed3300 rtpsession: don't cast event functions 
There is no need to cast the event functions and only causes problems later when
we change the signature later and things silently compiles wrong code.
 2013-01-07 14:25:14 +01:00
Wim Taymans
8dcde8b3ea rtp: more debug 2013-01-07 14:23:34 +01:00
Wim Taymans
6b7d05ac57 rtpsession: improve debug 2013-01-07 14:22:48 +01:00
Tim-Philipp Müller
cf1f6aff0d udpsrc: sanity check size of available packet data for reading to avoid memory waste 
On Windows and OS/X, _get_available_bytes() may not return the size
of the next pending packet, but the size of all pending packets in
the kernel-side buffer, which might be rather large depending on
configuration. Sanity-check the size returned by _get_available_bytes()
to make sure we never allocate more memory than the max. size for
a packet, if it's an IPv4 socket.

https://bugzilla.gnome.org/show_bug.cgi?id=610364
 2013-01-04 14:00:55 +00:00
Tim-Philipp Müller
95a37196b3 rtspsrc: add "proxy-id" and "proxy-pw" properties 
to match souphttpsrc. user/password passed via the URI
will still take precedence though.

https://bugzilla.gnome.org/show_bug.cgi?id=395427
 2012-12-31 00:22:27 +00:00
Wim Taymans
8cfec6a88d rtspsrc: fix cmd comparison 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690476
 2012-12-20 17:12:30 +01:00
Wim Taymans
75616fac9a rtspsrc: add some more debug 2012-12-20 17:12:20 +01:00
Jonas Holmberg
e12457f138 rtpjpegpay: handle width and height > 2040 
If width or height is greater than 2040 set width and height to zero in
the rtp header and add x-dimensions to outcaps.

Solves #684955
 2012-12-20 15:40:49 +01:00
Wim Taymans
dcb0e0af93 avidemux: cleanup in flag define 2012-12-20 13:04:52 +01:00
Wim Taymans
0e522bc69a avidemux: improve debug 2012-12-20 13:04:52 +01:00
Thijs Vermeir
de41376231 rtp: use appropriate printf format for gsize 2012-12-18 16:02:09 +01:00
Thijs Vermeir
df88341ffb deinterlace: use appropriate printf format for gsize 2012-12-18 16:02:09 +01:00
Philippe Normand
2bd77e1c8a interleave: set src pad caps upon last sink pad CAPS event 
Gather caps on all sink pads before setting the src pad caps. This is
specially needed when the audio channel mapping is set on the sink
pads and the element needs to preserve it on its src pad.

https://bugzilla.gnome.org/show_bug.cgi?id=690267
 2012-12-18 12:58:43 +01:00
Tim-Philipp Müller
f4cb0c4315 matroskademux: skip empty tags 
instead of trying to add tags with empty strings, which
causes criticals at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=690358
 2012-12-17 22:55:12 +00:00
Sebastian Dröge
c49dede772 audioparsers: Make sure the caps are actually writable before changing them 2012-12-17 15:17:12 +01:00
Sebastian Dröge
26040ee38c audioparsers: Use the peer caps for restrictions instead of the srcpad allowed caps 
Otherwise we will intersect with the srcpad template caps and add all the caps fields
that the parser will ever set, no matter if downstream restricts this field or not.
This requires upstream to set this field on the caps to successfully negotiate.

https://bugzilla.gnome.org/show_bug.cgi?id=690184
 2012-12-17 15:01:02 +01:00
Alexey Fisher
7e47e3b92d matroskamux: set appropriate block header flag for VP8 invisible frames 
Useful for debugging mostly.

https://bugzilla.gnome.org/show_bug.cgi?id=654259
 2012-12-16 23:30:13 +00:00
Tim-Philipp Müller
8a3b116d1f docs: add rtpmux and rtpdtmfmux to plugin docs 
https://bugzilla.gnome.org/show_bug.cgi?id=629117
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
3295b5d791 rtpmanager: move rtpmux and rtpdtmfmux elements from -bad 
https://bugzilla.gnome.org/show_bug.cgi?id=629117
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
de204ba754 rtpmux: Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
2778a1757f rtpmux: Use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-12-16 16:36:39 +00:00
Olivier Crête
15dfdc58d4 rtpmux: Misc fix for 0.11 
Convert the incoming caps before proxying them
Clear the last_pad when going to ready

tests: Implement accept_caps, don't leak event
 2012-12-16 16:36:38 +00:00
Wim Taymans
83262be703 rtpmux: update for RTP buffer api changes 2012-12-16 16:36:38 +00:00
Sebastian Dröge
f17064a8ea rtpmux: Update for GST_PLUGIN_DEFINE() API changes 2012-12-16 16:36:34 +00:00
Wim Taymans
c86156ad8f rtpmux: fix compilation 2012-12-16 16:35:36 +00:00
Wim Taymans
6826bbb6da rtpmux: fix for caps api changes 2012-12-16 16:35:33 +00:00
Matej Knopp
bb345a584d rtpmux: Fix compiler warnings 2012-12-16 16:35:29 +00:00
Olivier Crête
af4e999c59 rtpmux: Unref non-forwarded events 
Also, don't unref forwarded ones
 2012-12-16 16:35:29 +00:00
Olivier Crête
a8789d1df1 rtpmux: resync iterator on resync 2012-12-16 16:35:29 +00:00
Olivier Crête
0c54079af5 rtpmux: Re-push sticky events on input pad change 2012-12-16 16:35:29 +00:00
Olivier Crête
21831b430f rtpmux: Don't leak gvalue from iterator 2012-12-16 16:35:29 +00:00
Wim Taymans
ccc4b960fc rtpmux: more porting 2012-12-16 16:35:26 +00:00
Olivier Crête
f20a6b1d16 rtpmux: port to 0.11 2012-12-16 16:35:26 +00:00
Wim Taymans
35b6668fb6 rtpmux: make request pads take _%u 2012-12-16 16:35:22 +00:00
Olivier Crête
aa3607ef5c rtpdtmfmux: Add last-stop to dtmf-event upstream events 
Add the running time of the last outputted buffer to the
upstream "dtmf-event" events so that the dtmf source does not
leave a gap.
 2012-12-16 16:35:22 +00:00
Edward Hervey
d137482fe5 rtpmux: Remove dead assignments 2012-12-16 16:35:22 +00:00
Stefan Kost
55aae6bfab rtpmux: add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2012-12-16 16:35:15 +00:00
Olivier Crête
9674d5cc23 rtpmux: Improve documentation 
Add an example pipeline, and try to explain a bit more what it does.
 2012-12-16 16:35:15 +00:00
Stefan Kost
ca27a279ba rtpdtmfmux: remove unused variable 2012-12-16 16:35:15 +00:00
Stefan Kost
c85dceeacb rtpdtmfmux: remove unused signal boilerplate 2012-12-16 16:35:15 +00:00
Stefan Kost
2353f8d852 rtpmux: no need to ref pad in _chain() 2012-12-16 16:35:15 +00:00
Youness Alaoui
e42d2eebcb rtpmux: Unlock the right mutex 
The mutex locked is for the 'mux' object, but we unlock the
pad, which means that if the rtpmux gets a flush, then the
object lock will stay locked forever, causing it to freeze
the next time it tries to take it.

Fixes bug #627991
 2012-12-16 16:35:15 +00:00
Olivier Crête
78d1ebac9e rtpmux: Add support for GstBufferList 
Factor out most of the buffer handling and implement a chain_list
function. Also, the DTMF muxer has been modified to just have a
function to accept or reject a buffer instead of having to subclass
both chain and chain_list.
 2012-12-16 16:35:15 +00:00
Olivier Crête
c00f14419b rtpmux: Don't leak invalid buffers 2012-12-16 16:35:15 +00:00
Tim-Philipp Müller
a45429d81d rtpmux: fix missing debug log message argument 2012-12-16 16:35:15 +00:00
Olivier Crête
4a8d0243b5 rtpdtmfmux: Add some debug messages 2012-12-16 16:35:14 +00:00
Olivier Crête
423ce98666 rtpdtmfmux: Remove stream-lock event handling 2012-12-16 16:35:14 +00:00
Olivier Crête
a4500c0e74 rtpdtmfmux: Update doc for simplification 2012-12-16 16:35:14 +00:00
Olivier Crête
70097866de rtpdtmfmux: Drop buffers on non-priority sinks when something is incoming on the priority sink 2012-12-16 16:35:14 +00:00
Olivier Crête
f6548fe9b6 rtpdtmfmux: Add priority sink pads 2012-12-16 16:35:14 +00:00
Olivier Crête
2bcea1537b rtpdtmfmux: Cleanup event function 2012-12-16 16:35:14 +00:00
Olivier Crête
8e58646f5c rtpmux: Aggregate incoming segments 2012-12-16 16:35:14 +00:00
Olivier Crête
7be57cac3a rtpdtmfmux: Update documentation 2012-12-16 16:35:14 +00:00
Olivier Crête
e590fc1f32 rtpmux: Simplify request pad creation 2012-12-16 16:35:14 +00:00
Benjamin Otte
2867e00225 rtpmux: gst_element_class_set_details => gst_element_class_set_details_simple 2012-12-16 16:35:10 +00:00
unknown
fb7266884d rtpmux: update the current_ssrc from the caps 
Fixes #604101
 2012-12-16 16:33:47 +00:00
Håvard Graff
eab65e84ca rtpmux: release pads when disposing 
Because of an allocated priv (GstRTPMuxPadPrivate), the element will
leak memory if not gst_rtp_mux_release_pad() is called. This would
previously only happen if release_request_pad() was called explicitly,
somthing that should not be neccesary.

Fixes #604099
 2012-12-16 16:33:46 +00:00
Wim Taymans
0d54122804 dtmfmux: method name cleanups 2012-12-16 16:33:46 +00:00
Olivier Crête
3841cc74cf rtpmux: Don't ignore requested pad name 2012-12-16 16:33:46 +00:00
Olivier Crête
d93295ff9d rtpmux: Remove empty finalize 2012-12-16 16:33:46 +00:00
Olivier Crête
5e90a4e86b rtpmux: Free the pad private data on pad release 
Free the pad private data on pad release instead of using a weak ref,
which is not thread safe. Also, lock the content of the pad private using the element's
object lock.
 2012-12-16 16:33:46 +00:00
Olivier Crête
4be63c9add rtpmux: Reject wrong caps 2012-12-16 16:33:46 +00:00
Olivier Crête
0111bafb3a rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr> 2012-12-16 16:33:46 +00:00
Olivier Crête
fcc1522d2e rtpmux: Fix leak 
Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>
 2012-12-16 16:33:46 +00:00
Olivier Crête
ff6686f1c7 rtpmux: Fix warning 2012-12-16 16:33:46 +00:00
Olivier Crête
00791f930b rtpmux: Set different caps depending on the input 2012-12-16 16:33:46 +00:00
Olivier Crête
ed0b407038 rtpmux: Only free pad private when pad is disposed 2012-12-16 16:33:45 +00:00
Olivier Crête
92bb5199ac rtpmux: Remove useless caps mangling 2012-12-16 16:33:45 +00:00
Olivier Crête
3ccf3217fe rtpmux: Rename variable for more clarity 2012-12-16 16:33:45 +00:00
Olivier Crête
4b958f6d8d rtpmux: Use GST_BOILERPLATE 2012-12-16 16:33:45 +00:00
Olivier Crête
abe57be248 rtpmux: Do the includes locally 2012-12-16 16:33:45 +00:00
Olivier Crête
05844c89e9 rtpmux: Add GST_DEBUG_FUNCPTRs 2012-12-16 16:33:45 +00:00
Olivier Crête
fd102b95ab rtpdtmfmux: Release locked pad on release_pad 
Release the special pad if the pad is removed from the muxer.
 2012-12-16 16:33:45 +00:00
Laurent Glayal
00f8bab712 rtpdtmfmux: Release special on pad dispose 
Fixes #577690
 2012-12-16 16:33:45 +00:00
Stefan Kost
a4a22454dc docs: various doc fixes 
No short-desc as we have them in the element details.
Also keep things (Makefile.am and sections.txt) sorted.
Reword ambigous returns. No text after since please.
 2012-12-16 16:33:41 +00:00
Olivier Crête
7d4395a910 rtpmux: Move rtpmux from gst-plugins-farsight to -bad 2012-12-16 16:33:27 +00:00
Olivier Crête
68215752f4 rtpmux: Re-indent to Gst style 2012-12-16 16:33:24 +00:00
Olivier Crête
c7d0809434 rtpmux: Document rtp muxer a bit 2012-12-16 16:33:20 +00:00
Laurent Glayal
47c7a93df2 rtpmux: Add signals before stream lock and after unlocking 2012-12-16 16:33:17 +00:00
Olivier Crête
f1656ed8b0 rtpmux: Let ssrc through getcaps 2012-12-16 16:33:14 +00:00
Olivier Crête
1529dffaf9 rtpmux: Rename have_base to have_ts_base 2012-12-16 16:33:11 +00:00
Olivier Crête
57563517bd rtpmux: Protect the seqnum with object lock in rtpmux 2012-12-16 16:33:08 +00:00
Olivier Crête
d3237eaf95 rtpmux: Remove unused sink_ts_base 2012-12-16 16:33:04 +00:00
Olivier Crête
cc23958183 rtpmux: Have getcaps to force the same clockrate on all pads 2012-12-16 16:33:01 +00:00
Olivier Crête
dc36590d0c rtpmux: Validate RTP data in RTP Mux 2012-12-16 16:32:57 +00:00
Olivier Crête
360c8d4f1d rtpmux: Remove unused clock-rate property 2012-12-16 16:32:54 +00:00
Olivier Crête
b86232d0dc rtpmux: Clarify locking in rtpdtmfmux 2012-12-16 16:32:50 +00:00
Laurent Glayal
4b607cdda5 rtpmux: Missing format parameter 2012-12-16 16:32:47 +00:00
Håvard Graff
b313c80367 rtpmux: Update seqnum base in rtp muxer 
With help from Wim
 2012-12-16 16:32:43 +00:00
Håvard Graff
c479f90274 rtpmux: Fix some more leaks 2012-12-16 16:32:40 +00:00
Håvard Graff
1b5e769e0b rtpmux: Fix leak 2012-12-16 16:32:37 +00:00
Olivier Crête
5cbb0de823 rtpmux: Don't unref caps we don't know (thanks Wim) 2012-12-16 16:32:32 +00:00
Olivier Crête
cebf506949 rtpmux: Put per-buffer debug at level LOG 2012-12-16 16:32:29 +00:00
Olivier Crête
3c12a423b7 rtpmux: Make debug print accurate 2012-12-16 16:32:25 +00:00
Olivier Crête
c49f4c87c6 rtpmux: Set our caps on the buffers 2012-12-16 16:32:22 +00:00
Olivier Crête
ec63da9366 rtpmux: Take the clock-base stored from the last setcaps 2012-12-16 16:32:18 +00:00
Olivier Crête
674c074114 rtpmux: Store the clock-base on setcaps 2012-12-16 16:32:15 +00:00
Olivier Crête
90264b9686 rtpmux: Add padprivate to the request pads 2012-12-16 16:32:11 +00:00
Olivier Crête
15d661ba3e rtpmux: Make indentation more correct 2012-12-16 16:31:56 +00:00
Olivier Crête
3a7d09a749 rtpmux: Fix typo 2012-12-16 16:31:53 +00:00
Olivier Crête
91aef3ec5e rtpmux: Set seqnum-base and clock-base in caps from rtpmuxer 2012-12-16 16:31:50 +00:00
Zeeshan Ali
6ea5ca354d rtpmux: more debug 
20070815135038-f3f1e-9c7a5490a525c6e8753cb1b8c03354df99132b5c.gz
 2012-12-16 16:31:46 +00:00
Youness Alaoui
f0e209b638 rtpmux: missing comment 
20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz
 2012-12-16 16:30:33 +00:00
Olivier Crete
3ed5590da6 rtpmux: Make buffer writable before writing into it 
20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz
 2012-12-16 16:30:31 +00:00
Olivier Crete
dd13f7c8ef rtpmux: Set pads active when adding them to a potentially running element 
20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz
 2012-12-16 16:30:27 +00:00
Olivier Crete
1c5075f927 rtpmux: Fix multiple ref leaks (patches by SP GLE) 
20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz
 2012-12-16 16:30:23 +00:00
Zeeshan Ali
42f455e902 rtpmux: send event to all src pads 
20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz
 2012-12-16 16:30:18 +00:00
Zeeshan Ali
dba101bb0f rtpmux: print a warning if receive an error iterating sinkpads 
20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz
 2012-12-16 16:30:15 +00:00
Zeeshan Ali
baa48dc6bc rtpmux: deal with all the gst_iterator_next() return values 
20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz
 2012-12-16 16:30:12 +00:00
Zeeshan Ali
de40874670 rtpmux: Return correct value from the event handler 
20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz
 2012-12-16 16:30:08 +00:00
Zeeshan Ali
ed76f67e96 rtpmux: Ville's original patch to fix the traversal of dtmf event 
20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz
 2012-12-16 16:30:05 +00:00
zeeshan.ali@nokia.com
94ebe07862 rtpmux: Set the correct ts-offset on the get_prop value 
20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz
 2012-12-16 16:30:01 +00:00
zeeshan.ali@nokia.com
1ee542c378 rtpmux: Refactorize state_change 
20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz
 2012-12-16 16:29:58 +00:00
zeeshan.ali@nokia.com
2498ba671a rtpmux: set SSRC on the packets 
20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz
 2012-12-16 16:29:55 +00:00
zeeshan.ali@nokia.com
ee69c2690d rtpmux: Code clean-up and more debug output 
20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz
 2012-12-16 16:29:52 +00:00
zeeshan.ali@nokia.com
1c799ce964 rtpmux: Use own clock-base 
20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz
 2012-12-16 16:29:48 +00:00
zeeshan.ali@nokia.com
b04630d7a2 rtpmux: Only accept RTP streams that have the same clock-rate 
20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz
 2012-12-16 16:29:45 +00:00
zeeshan.ali@nokia.com
6fe1e02efd rtpmux: Some more code-cleanups 
20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz
 2012-12-16 16:29:42 +00:00
zeeshan.ali@nokia.com
1603223ee5 rtpmux: return newpad instead of NULL and warn if failed to create a pad 
20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz
 2012-12-16 16:29:38 +00:00
zeeshan.ali@nokia.com
23d3ed5c5f rtpmux: Refactorize the RTPMux code 
20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz
 2012-12-16 16:29:35 +00:00
zeeshan.ali@nokia.com
21e6e951f6 rtpmux: Some more doc fixing 
20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz
 2012-12-16 16:29:32 +00:00
zeeshan.ali@nokia.com
0de7fb6f37 rtpmux: More Refactoring 
20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz
 2012-12-16 16:29:29 +00:00
zeeshan.ali@nokia.com
0f755657ce rtpmux: More documentation 
20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz
 2012-12-16 16:29:26 +00:00
zeeshan.ali@nokia.com
5483c78ac0 rtpmux: Refactor the event handler function 
20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz
 2012-12-16 16:29:23 +00:00
zeeshan.ali@nokia.com
db1523ae60 rtpmux: Add RTPDTMFMux element 
20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz
 2012-12-16 16:29:19 +00:00
zeeshan.ali@nokia.com
97ff54dce7 rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable 
20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz
 2012-12-16 16:29:16 +00:00
zeeshan.ali@nokia.com
1a227ac7e5 rtpmux: Put more helpful description 
20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz
 2012-12-16 16:29:13 +00:00
zeeshan.ali@nokia.com
d876c0d8cc rtpmux: remove the (commented-out) code for blocking the pads 
20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz
 2012-12-16 16:29:10 +00:00
zeeshan.ali@nokia.com
209228c44d rtpmux: Drop buffers instead of blocking the sinkpads 
20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz
 2012-12-16 16:29:05 +00:00
zeeshan.ali@nokia.com
795822ffa5 rtpmux: Implement stream locking, needed for DTMF 
20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz
 2012-12-16 16:29:02 +00:00
zeeshan.ali@nokia.com
fd209faa56 rtpmux: use GST_*_OBJECT instead of g_* 
20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz
 2012-12-16 16:28:58 +00:00
zeeshan.ali@nokia.com
b0208cb0a6 rtpmux: No need to manage pads, parent does that for us 
20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz
 2012-12-16 16:28:55 +00:00
zeenix@gmail.com
74e9071dad rtpmux: Fix copyright header 
20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz
 2012-12-16 16:28:51 +00:00
zeeshan.ali@nokia.com
3c4cdf1541 rtpmux: The first implementation of RTP muxer 
20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz
 2012-12-16 16:28:41 +00:00
Tim-Philipp Müller
b19122bac8 scaletempo: no need for a private struct 2012-12-15 21:27:01 +00:00
Tim-Philipp Müller
61913ab7b4 audiofx: move scaletempo element from -bad 
https://bugzilla.gnome.org/show_bug.cgi?id=687262
 2012-12-14 13:16:17 +00:00
Sebastian Dröge
314765c294 scaletempo: Fix event leak 2012-12-14 13:16:17 +00:00
Sebastian Dröge
490e408991 scaletempo: Fix timestamp tracking 2012-12-14 13:16:17 +00:00
Sebastian Dröge
502eb8d1b7 scaletempo: Implement LATENCY query 2012-12-14 13:16:17 +00:00
Sebastian Dröge
c7589817cb scaletempo: Store instance private data in the instance struct 
Getting it over and over again via G_TYPE_INSTANCE_GET_PRIVATE()
is really slow.
 2012-12-14 13:16:17 +00:00
Tim-Philipp Müller
e552bd484f scaletempo: use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
d2dd91ac47 scaletempo: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-12-14 13:16:17 +00:00
Wim Taymans
cb1743d578 scaletempo: ffmpegcolorspace is no more 2012-12-14 13:16:17 +00:00
Sebastian Dröge
93e1091d7f scaletempo: Update for GST_PLUGIN_DEFINE() API changes 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
3286cdd542 scaletempo: port to 0.11 2012-12-14 13:16:16 +00:00
Stefan Kost
62d780cd51 scaletempo: improve the docs 
Fix the syntax, add more explanation and xref the properties.
 2012-12-14 13:16:16 +00:00
Chris E Jones
caf2b6cb5c scaletempo: Correctly handle newsegment events with stop==-1 
Fixes bug #645420.
 2012-12-14 13:16:16 +00:00
Stefan Kost
6d54058982 scaletempo: add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2012-12-14 13:16:16 +00:00
Benjamin Otte
38bc2dfb4a scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple 2012-12-14 13:16:16 +00:00
Thiago Santos
2d72ec153a scaletempo: properly update new segments 
Scaletempo was missing an update of 'stop' in
new segment parameters when pushing it downstream,
which caused files to end earlier when rate < 1.

Fixes #599903

Based on patch by: Bastian Hecht <hechtb@gmail.com>
 2012-12-14 13:16:16 +00:00
Maximilian Högner
2fe7a97f1c scaletempo: Explicitely cast to signed integers to fix a segfault 
Fixes bug #585660.
 2012-12-14 13:16:16 +00:00
Michael Smith
1b1f6f56d6 scaletempo: Do not use void pointer arithmetic. 2012-12-14 13:16:16 +00:00
Stefan Kost
9284c85b33 scaletempo: Return the result of parent_class->event() 
Original commit message from CVS:
* gst/audiofx/gstscaletempo.c:
Return the result of parent_class->event().
 2012-12-14 13:16:16 +00:00
Rov Juvano
43e79f7769 Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r... 
Original commit message from CVS:
Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-scaletempo.xml:
* examples/scaletempo/Makefile.am:
* examples/scaletempo/demo-gui.c: (pop_status_bar),
(status_bar_printf), (demo_gui_seek_bar_format), (update_position),
(demo_gui_seek_bar_change), (demo_gui_do_change_rate),
(demo_gui_do_set_rate), (demo_gui_do_rate_entered),
(demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled),
(demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause),
(demo_gui_do_play_pause), (demo_gui_do_open_file),
(demo_gui_do_playlist_prev), (demo_gui_do_playlist_next),
(demo_gui_do_about_dialog), (demo_gui_do_quit),
(demo_gui_request_set_stride), (demo_gui_request_set_overlap),
(demo_gui_request_set_search), (demo_gui_rate_changed),
(demo_gui_playing_started), (demo_gui_playing_paused),
(demo_gui_playing_ended), (demo_gui_player_errored),
(demo_gui_stride_changed), (demo_gui_overlap_changed),
(demo_gui_search_changed), (demo_gui_set_player_func),
(demo_gui_set_playlist_func), (build_gvalue_array),
(create_action), (demo_gui_show_func), (demo_gui_set_player),
(demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property),
(demo_gui_set_property), (demo_gui_init), (demo_gui_class_init),
(demo_gui_get_type):
* examples/scaletempo/demo-gui.h:
* examples/scaletempo/demo-main.c: (handle_error_message),
(handle_quit), (main):
* examples/scaletempo/demo-player.c: (no_pipeline),
(demo_player_event_listener), (demo_player_state_changed_cb),
(demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate),
(demo_player_scale_rate_func), (demo_player_set_rate_func),
(_set_state_and_wait), (demo_player_load_uri_func),
(demo_player_play_func), (demo_player_pause_func), (_seek_to),
(demo_player_seek_by_func), (demo_player_seek_to_func),
(demo_player_get_position_func), (demo_player_get_duration_func),
(demo_player_scale_rate), (demo_player_set_rate),
(demo_player_load_uri), (demo_player_play), (demo_player_pause),
(demo_player_seek_by), (demo_player_seek_to),
(demo_player_get_position), (demo_player_get_duration),
(demo_player_get_property), (demo_player_set_property),
(demo_player_init), (demo_player_class_init),
(demo_player_get_type):
* examples/scaletempo/demo-player.h:
* gst/audiofx/Makefile.am:
* gst/audiofx/gstscaletempo.c: (best_overlap_offset_float),
(best_overlap_offset_s16), (output_overlap_float),
(output_overlap_s16), (fill_queue), (reinit_buffers),
(gst_scaletempo_transform), (gst_scaletempo_transform_size),
(gst_scaletempo_sink_event), (gst_scaletempo_set_caps),
(gst_scaletempo_get_property), (gst_scaletempo_set_property),
(gst_scaletempo_base_init), (gst_scaletempo_class_init),
(gst_scaletempo_init):
* gst/audiofx/gstscaletempo.h:
* gst/audiofx/gstscaletempoplugin.c: (plugin_init):
Add scaletempo plugin, which allows to scale the speed of audio without
changing the pitch by handling seeks with a rate!=1.0.
Integrate it into the docs and add the example application for it.
Fixes bug #537700.
 2012-12-14 13:16:15 +00:00
Havard Graff
9c94f1187c jitterbuffer: bundle together late lost-events 
The scenario where you have a gap in a steady flow of packets of
say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer
will idle up until it receives the first buffer after the gap, but will
then go on to produce 499 lost-events, to "cover up" the gap.

Now this is obviously wrong, since the last possible time for the earliest
lost-events to be played out has obviously expired, but the fact that
the jitterbuffer has a "length", represented with its own latency combined
with the total latency downstream, allows for covering up at least some
of this gap.

So in the case of the "length" being 200ms, while having received packet
500, the jitterbuffer should still create a timeout for packet 491, which
will have its time expire at 10,02 seconds, specially since it might
actually arrive in time! But obviously, waiting for packet 100, that had
its time expire at 2 seconds, (remembering that the current time is 10)
is useless...

The patch will create one "big" lost-event for the first 490 packets,
and then go on to create single ones if they can reach their
playout deadline.

See https://bugzilla.gnome.org/show_bug.cgi?id=667838
 2012-12-13 12:00:43 +01:00
Wim Taymans
a858bf46db rtspsrc: fix TCP reconnect 
Ignore other commands when reconnecting, otherwise the loop function would pause
and the reconnection would not happen. Continue looping after doing a reconnect
so that we have a chance to actually read the new data.
 2012-12-13 09:30:59 +01:00
Philippe Normand
a8fa9f2b47 deinterleave: properly set srcpad channel position 
The src pad caps always describe a single audio channel so only the
first position matters if deinterleave is configured to keep channel
positions in its src pads.
 2012-12-12 11:20:56 +00:00
Wim Taymans
b1dc816772 rtspsrc: timeout on udpsrc is in nanoseconds 2012-12-12 11:09:42 +01:00
Wim Taymans
32bd981303 udpsrc: improve timeouts 
Make it possible to set the timeout after we went to the READY state by using
the timeout when checking the condition. This also makes it possible to set the
timeout with a higher granularity than seconds.
 2012-12-12 11:08:13 +01:00
Wim Taymans
abd7e33db6 deinterlace: add support for strides 
Implement stride support correctly by taking it from the GstVideoFrame.
Propose a bufferpool upstream when not operating in passthrough.
 2012-12-11 13:00:46 +01:00
Aleix Conchillo Flaque
3503aef946 rtspsrc: do not change state to PLAYING if currently chaning state 
* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be
  happening in the application thread, so we don't change the state to
  PLAYING in the gstrtspsrc thread unless it is safe.

  A specific case is when chaning the state to NULL from the application
  thread. This will synchronously try to stop the task (with the element
  state lock acquired), but we will try a gst_element_set_state from
  gstrtspsrc thread which will block on the element state lock causing a
  deadlock.

  https://bugzilla.gnome.org/show_bug.cgi?id=684312
 2012-12-10 15:13:22 +01:00
Tim-Philipp Müller
672ab8fb5b webmux: fix linking with shout2send element 
Shout2send only accepts webm format, not matroska, but due
to a bug in matroskamux, webmmux's source pad is also created
with the matroska source pad template as pad template, which
makes the link function think it can't link webmmux to shout2send.

Also add unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=689336
 2012-11-30 17:22:34 +00:00
Wim Taymans
64cdbb77a9 rtspsrc: use new option parser function 2012-11-27 11:13:37 +01:00
Tim-Philipp Müller
5dee61a8d5 law: fix accidental file permissions change 
https://bugzilla.gnome.org/show_bug.cgi?id=687469
 2012-11-26 15:17:13 +00:00
Tim-Philipp Müller
314efb684b qtdemux: avoid criticals if unknown fourcc has space at beginning or end 
https://bugzilla.gnome.org/show_bug.cgi?id=682936
 2012-11-25 14:16:09 +00:00
Tim-Philipp Müller
efaa80fbc6 videobox: fix border filling for planar YUV formats 
We would get a green border instead of a black one, for
example.

https://bugzilla.gnome.org/show_bug.cgi?id=684991
 2012-11-24 19:32:51 +00:00
Tim-Philipp Müller
ef6c16a32e mulaw: const-ify some arrays 2012-11-24 14:27:33 +00:00
Roland Krikava
3be45f7022 mulawdec: fix integer overrun 
There might be more than 65535 samples in a chunk of data.

https://bugzilla.gnome.org/show_bug.cgi?id=687469
 2012-11-24 14:24:41 +00:00
Wim Taymans
5d0507c09e rtspsrc: pause the task instead of spinning 
Actually pause the loop task instead of spinning forever.
 2012-11-22 11:34:31 +01:00
Joshua M. Doe
fe9fb8d8a7 videoflip: Add gray 8/16 support 2012-11-20 12:49:49 +01:00
Wim Taymans
c28bfa8902 rtspsrc: handle segment event 
Make a segment event when we send a new range header to a client (first PLAY
request or after a seek). Send the segment event in interleaved mode.
Clean the segment event on cleanup

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382
 2012-11-16 15:38:29 +01:00
Wim Taymans
bd91bd3193 rtspsrc: fix check for active streams 
A stream can be active without a srcpad yet and we want to send
events on those streams as well.
 2012-11-16 15:22:46 +01:00
Wim Taymans
11cf4d4fd3 rtspsrc: create and add pads outside of lock 
Create and add the ghostpad for the new stream outside of the lock because it
is not needed and causes deadlocks.
 2012-11-16 13:33:44 +01:00
Aleix Conchillo Flaque
6c855edf03 rtspsrc: allow client to disable reconnection 
* gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before,
  rtspsrc always tried to reconnect to the server when the RTSP
  connection was closed by the server. This property lets the user
  decide whether it wants rtspsrc to reconnect or not.

  https://bugzilla.gnome.org/show_bug.cgi?id=683912
 2012-11-16 12:55:10 +01:00
Wim Taymans
e2a4d28c1f rtspsrc: clear variables before retrying 
Else we might unref an old udpsrc twice in cleanup.
 2012-11-16 12:17:37 +01:00
Wim Taymans
cc9cb26be1 rtspsrc: propose ports in multicast 
When the user configured a port-range, propose ports from this range
as the multicast ports. The server is free to ignore this request but if it
honours it, increment our ports so that we suggest the next port pair for the
next stream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420
 2012-11-16 12:17:37 +01:00
Wim Taymans
5025b3f1b3 rtspsrc: add more debug 2012-11-16 12:17:37 +01:00
Tim-Philipp Müller
6f1aa3e4d5 multifilesink: post messages in max-size mode as well 
No reason not to really.
 2012-11-16 09:13:22 +00:00
Wim Taymans
c33507f186 udpsrc: post error before stopping 2012-11-15 14:48:59 +01:00
Tim-Philipp Müller
bdf3c77828 gst_adapter_prev_timestamp -> gst_adapter_prev_pts 
https://bugzilla.gnome.org/show_bug.cgi?id=675598
 2012-11-14 00:13:36 +00:00
Nicolas Dufresne
673d2d24b8 videoflip: Add NV12/NV21 support 
https://bugzilla.gnome.org/show_bug.cgi?id=688225
 2012-11-13 14:25:04 +01:00
Wim Taymans
c755af0cb0 rtpsource: protect against invalid RTP packets 2012-11-12 11:18:30 +01:00
Tim-Philipp Müller
35fafae241 videocrop: add support for YV12 
We can do I420, so we can do YV12 as well.
 2012-11-10 18:21:28 +00:00
Alessandro Decina
b916d2b398 multifilesink: don't write stream headers with key-unit-event 
Don't write stream headers, let upstream elements insert them in the stream if
all_headers=true is set in key unit events.
 2012-11-10 12:41:33 +01:00
Nicolas Dufresne
e111068f7b videocrop: Add NV12/NV21 support 
https://bugzilla.gnome.org/show_bug.cgi?id=687964
 2012-11-10 01:52:44 +01:00
Sebastian Dröge
c70ba7765a udpsrc: Also clear GError 2012-11-09 11:22:30 +01:00
Sebastian Dröge
b86d20e45b udpsrc: Don't error out if we get an ICMP destination-unreachable message when trying to read packets 
See bug #529454 and #687782 and commit
751f2bb364
 2012-11-09 11:20:27 +01:00
Christian Fredrik Kalager Schaller
485505f323 Fix vp8rtp header names in Makefile 2012-11-07 13:36:33 +01:00
Nicolas Dufresne
1ad8ebac44 videocrop: Add support for automatic cropping 
This change enable automatic cropping using -1 set to left, top, right or
bottom property. In the case both side are set to automatic cropping, the
croping will be done equally on both side (in the odd case, right and
bottom cropping will be 1 pixel more).

https://bugzilla.gnome.org/show_bug.cgi?id=687761
 2012-11-07 11:20:24 +01:00
Marc Leeman
7cbca3dcd1 rtsp: the RTCP port number is inclusive 
The configured port number pair has its upper bound set to the maximum
allowed RTCP port, inclusive.

See https://bugzilla.gnome.org/show_bug.cgi?id=639420
 2012-11-06 13:22:58 +01:00
Tim-Philipp Müller
beb3c9c9be Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:09:59 +00:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Wim Taymans
9857e6af4d vrawdepay: don't access rtp buffer after unmap 
Read the marker bit before we unmap the rtp packet.
 2012-11-02 18:48:17 +00:00
Douglas Bagnall
0b898ab911 videoconvert: Compare y offset with height, not width, when testing for overlap 
This could have prevented images showing that should have when the
source height is greater than its width.

When width exceeds height, as is common, it probably only caused a
miniscule amount of unnecessary work.  I haven't tested.
 2012-11-02 09:29:30 +01:00
Tim-Philipp Müller
5ac789408b rtpvp8: include config.h and minor style fixes 2012-11-01 21:10:21 +00:00
Tim-Philipp Müller
4a849d6690 rtp: fix tabs/space mess in Makefile.am 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
321acd14dc rtp: move VP8 payloader and depayloader from -bad 
Spec is still in draft state, but should hopefully not
change much now. Besides, we announce things as VP8-DRAFT-IETF-01
in our caps, so even if things change in incompatible ways it
should not break anything.

https://bugzilla.gnome.org/show_bug.cgi?id=687263
 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
44efab8e3d rtpvp8: use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
bc7dbbbd4f rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-11-01 20:53:48 +00:00
Sebastian Dröge
4853001547 rtpvp8: update for GST_PLUGIN_DEFINE() API changes 2012-11-01 20:53:48 +00:00
Wim Taymans
fccfca38d4 rtpvp8: update for buffer changes 2012-11-01 20:53:48 +00:00
Danilo Cesar Lemes de Paula
3edffb13e3 rtpvp8; fix compatibility with the third draft 
https://bugzilla.gnome.org/show_bug.cgi?id=671073
 2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
d9581832a0 rtpvp8: port some more to new memory API 2012-11-01 20:53:47 +00:00
Olivier Crête
c6761daa27 rtpvp8: port to 0.11 2012-11-01 20:53:47 +00:00
Sebastian Dröge
2c5ea76bdc rtpvp8pay: Fix typo 2012-11-01 20:53:47 +00:00
Youness Alaoui
1cf155d70d rtpvp8: Update the pay/depay to the ietf-draft-01 spec 2012-11-01 20:53:47 +00:00
Vincent Penquerc'h
88aade4150 rtpvp8: fix bitstream parsing using the wrong kind of bitreader 
VP8 uses a probabilistic bool coder, not a straight bit coder.
This fixes parsing when error-resilient is set.

This commit includes a copy of libvpx's bool coder, BSD licensed.

https://bugzilla.gnome.org/show_bug.cgi?id=652694
 2012-11-01 20:53:47 +00:00
Olivier Crête
97c3f3617c rtpvp8: Reject unknown bitstream versions 2012-11-01 20:53:47 +00:00
Edward Hervey
74a1a704bf rtpvp8: Fix unitialized variable 
Makes macosx compiler happy.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
6ed6318076 rtpvp8depay: Accept packets with only one byte of data 
When fragmenting partions it can happen that an RTP packet only caries 1
byte of RTP data.
 2012-11-01 20:53:47 +00:00