Commit graph

7625 commits

Author SHA1 Message Date
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