Commit graph

7849 commits

Author SHA1 Message Date
Tim-Philipp Müller
69ec922c16 icydemux: does not need to link against zlib 2014-11-23 16:24:06 +00:00
Miguel París Díaz
6daa57868f rtpjitterbuffer: ensure rtx_retry_period >= 0
https://bugzilla.gnome.org/show_bug.cgi?id=739344
2014-11-22 14:48:57 +00:00
Arun Raghavan
45e716e75d rtpbin: Fix up new_jitterbuffer signal prototype 2014-11-20 22:42:59 +05:30
Arun Raghavan
56436ccced rtpbin: Document how to control per-SSRC retransmission 2014-11-20 20:24:42 +05:30
Wim Taymans
3d7b0f30d7 rtpgstpay: put 0-byte at the end of events
Put a 0-byte at the end of the event string. Does not break ABI because
old depayloaders will skip the 0 byte (which is included in the length).
Expect a 0-byte at the end of the event string or a ; for old
payloaders.

See https://bugzilla.gnome.org/show_bug.cgi?id=737591
2014-11-20 13:14:14 +01:00
Wim Taymans
9d2902d978 rtpgstdepay: avoid buffer overread.
Check that a caps event string is 0 terminated and the event string is
terminated with a ; to avoid buffer overreads.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737591
2014-11-20 12:44:26 +01:00
Tim-Philipp Müller
488d0b93cd qtmux: don't limit max video resolution to 4096x4096
MAX isn't entirely correct as upper limit either,
it should really be MAXUINT32, but it's unlikely
to be a problem in the near future.

https://bugzilla.gnome.org/show_bug.cgi?id=740407
2014-11-20 10:45:53 +00:00
Aleix Conchillo Flaqué
00ca83629b rtspsrc: fix leak for mikey base64 decoded key-mgmt
https://bugzilla.gnome.org/show_bug.cgi?id=740392
2014-11-20 09:15:56 +01:00
Wim Taymans
e95da8410f videobalance: fix unhandled format in passthrough
In passthrough we can handle all formats.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740387
2014-11-20 09:02:36 +01:00
Jan Alexander Steffens (heftig)
bf73d834b2 flvdemux: Restrict resyncing to TS regressions
The behavior of resyncing video and audio indepen-
dently can cause A/V desyncs. Lets restrict resyncs
to jumps backward for now.

https://bugzilla.gnome.org/show_bug.cgi?id=736397
2014-11-19 11:58:19 -05:00
Matthew Waters
0053ad0847 videomixer: fix up QoS handling for live sources
Only attempt adaptive drop when we are not live

https://bugzilla.gnome.org/show_bug.cgi?id=739996
2014-11-17 23:16:03 +11:00
Arun Raghavan
1c3b233fef rtpmanager: Trivial typo fix 2014-11-10 13:16:50 +05:30
Sebastian Dröge
7a909917b5 matroska-mux: Use G_DEFINE_TYPE() to register the pad instead of manually registering it 2014-11-09 11:04:33 +01:00
Göran Jönsson
ec05d3b6d8 matroskamux: make GstMatroskamuxPad get_type() function thread-safe
https://bugzilla.gnome.org/show_bug.cgi?id=739722
2014-11-07 21:20:31 +00:00
Josep Torra
038cc7b004 rtsp: fix build in gst-uninstalled setup 2014-11-06 21:38:43 +01:00
Thibault Saunier
99bbc2bbe4 imagefreeze: Handle seqnums
https://bugzilla.gnome.org/show_bug.cgi?id=739366
2014-11-06 12:20:25 +01:00
Wim Taymans
26d682d23f videomixer2: reverse order of params for converter 2014-11-03 15:26:06 +01:00
Tim-Philipp Müller
c756fd6a55 goom2k1: post QoS messages when dropping frames due to QoS 2014-11-02 19:42:03 +00:00
Tim-Philipp Müller
b03056eede goom: post QoS messages when dropping frames due to QoS 2014-11-02 19:31:01 +00:00
Tim-Philipp Müller
85c3c36712 matroskamux: tweak writing app tag string a little 2014-11-02 19:02:35 +00:00
Tim-Philipp Müller
3956f5addc Sprinkle some G_PARAM_DEPRECATED and #ifndef GST_REMOVE_DEPRECATED 2014-11-02 16:58:30 +00:00
Tim-Philipp Müller
d940c21b78 rtpjitterbuffer: implement get/set for new rtx-min-retry-timeout property
Properties are so much more useful if you can actually set
and get their values.
2014-11-02 13:06:33 +00:00
Nicolas Dufresne
0f4f948f5f rtpvp8: Use VP8 encoding name
Both Firefox and Chrome uses VP8 as the encoding in their SDP.
Adding this now defacto standard name removes the need for special
case in SDP parsing code.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2014-11-01 11:26:26 -04:00
Tim-Philipp Müller
92c1d289b8 rtpmp2tpay: fix up template caps so we can output the default pt 33
Add fixed payload type for mp2t to template caps as well, so
our output caps match the advertised default pt. Fixes a
regression from 1.2.

There's still something wrong with caps negotiation though,
rtpmp2tpay payload=96 ! fakesink will not output caps with
payload=96.
2014-11-01 12:40:07 +00:00
Aleix Conchillo Flaqué
d15ebcbf62 rtspsrc: mikey related memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739430
2014-10-31 10:03:47 +00:00
Sebastian Dröge
4aac09e708 aacparse: Always set profile/level on the caps
We have the information already, so why not use it?
2014-10-26 11:47:25 +01:00
Tim-Philipp Müller
b02d73a0ed rtpjitterbuffer: fix crash on some 32-bit systems
Make sure to pass right number of bits to gst_structure_new()
which is a vararg function.

Fixes elements/rtpaux unit test on ppc32.
2014-10-25 12:45:31 +01:00
Tim-Philipp Müller
401782c19d interleave: intersect result with filter caps in caps query
Fixes crash in audiotestsrc because of an unsupported format
getting negotiated on big-endian systems with
audiotestsrc ! interleave ! audioconvert ! wavenc
2014-10-25 11:08:48 +01:00
Wim Taymans
bd09dc96e9 rtpjitterbuffer: limit the retry frequency
When the RTT and jitter are very low (such as on a local network), the
calculated retransmission timeout is very small. Set some sensible lower
boundary to the timeout by adding a new property. We use the packet
spacing as a lower boundary by default.
2014-10-22 15:04:24 +02:00
Miguel París Díaz
4b5243c43d gstrtpjitterbuffer: add "rtx-min-delay" property
This property is useful to set a min time to wait before sending a
retransmission event.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=735378
2014-10-22 15:00:27 +02:00
Wim Taymans
0b81b316b5 jitterbuffer: Refactor code
Refactor some code dealing with calculating various timeouts.

See https://bugzilla.gnome.org/show_bug.cgi?id=735378
2014-10-22 14:59:57 +02:00
Miguel París Díaz
e6504e3a65 rtpsession: fix Early Feedback Transmission
In early retransmission we are allowed to schedule 1 regular RTCP packet
at an earlier time. When we do that, we need to set allow_early to FALSE
and ignore/drop (or merge) all future requests for early transmission.
We now first check if we can schedule an early RTCP and if we can,
actually prepare the data for the next RTCP interval.

After we send the next regular RTCP after the early RTCP, we set
allow_early to TRUE again to allow more early requests.

Remove the condition for the immediate feedback for now.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
2014-10-22 13:13:47 +02:00
Wim Taymans
09f179139d rtpjitterbuffer: make debug line less confusing 2014-10-21 13:10:53 +02:00
Wim Taymans
2e7f5c08cf jitterbuffer: rework resync handling
Add a need-resync state, this is when we need to try to lock on to a
time/RTPtime pair.
Always check the RTP timestamps and if they go backwards, mark ourselves
as need-resync.
Only resync when need-resync is TRUE and we have a valid time. Otherwise
we keep the old values. This avoids locking on to an invalid time and
causing us to timestamp everything with -1.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
2014-10-21 11:57:34 +02:00
Aleix Conchillo Flaqué
bd392d72ee rtspsrc: set full stream caps on internal src TCP pads
Set the complete stream caps on the TCP internal src pads. Otherwise,
ptdemux will not properly detect the caps change.

https://bugzilla.gnome.org/show_bug.cgi?id=737868
2014-10-21 11:33:01 +02:00
Sjoerd Simons
0ee384b251 rtpmux: Don't set PROXY_CAPS flag on the src pad
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.

https://bugzilla.gnome.org/show_bug.cgi?id=738722
2014-10-21 10:52:00 +02:00
Vineeth T M
1131db8c1f videobox: critical error when element properties set as max/min
left, right, top, bottom can be set from range of -2147483648 to 2147483647
when i launch the videobox element with that values, it gives a critical error

(gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed
This happens because min cannot be equal to max.

https://bugzilla.gnome.org/show_bug.cgi?id=738838
2014-10-20 12:53:51 +02:00
Tim-Philipp Müller
f3fec86bc9 Revert "rtp: add h265 RTP payloader + depayloader"
This reverts commit d06ba9051f.

This breaks the build, as it depends on parser API in -bad.
2014-10-15 17:48:46 +01:00
Jurgen Slowack
d06ba9051f rtp: add h265 RTP payloader + depayloader 2014-10-15 17:34:50 +02:00
Peter G. Baum
b5e46c05d7 wavenc: Support RF64 format
https://bugzilla.gnome.org/show_bug.cgi?id=725145
2014-10-14 10:24:50 +02:00
David Sansome
8154c90c9b equalizer: Don't call iirequalizer's transform_ip in passthrough mode
It tries to map the read-only buffer with GST_MAP_READWRITE and crashes.

https://bugzilla.gnome.org/show_bug.cgi?id=737886
2014-10-13 08:30:03 +02:00
Olivier Crête
51a8bedced rtpsource: Rename seqnum-base to seqnum-offset in caps
This was modified back in 1.0 in GstRtpBasePayload
2014-10-10 18:33:34 -04:00
Olivier Crête
155ed569c3 rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-10-10 18:12:32 -04:00
Olivier Crête
b3069634bd rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-10-10 18:12:23 -04:00
Luis de Bethencourt
cff880401d goom2k1: removing block of code that does nothing
The loop in zoomFilterSetResolution is meant to change the values in the
zf->firedec[] array. Each iteration writes the value of decc onto the arrya,
but no conditions that change the value of decc are ever met and the array is
filled with zero for each element. Which is the initial state of the
array before the loop begins.

The loop does nothing.

https://bugzilla.gnome.org/show_bug.cgi?id=728353
2014-10-08 14:07:56 +01:00
Stefan Sauer
98222a67ff rtpjitterbuffer: don't log all clock_rate changes as warnings.
We never initialize clock_rate explicitly, therefore it is 0 by default. The
parameter is a uint32 and the only caller ensure that it is >0, therefore it
won't become -1 ever.
2014-10-04 17:17:13 +02:00
Matej Knopp
e1d275cfec aacparse: fix memory leak when prepending ADTS headers
https://bugzilla.gnome.org/show_bug.cgi?id=737761
2014-10-02 10:41:28 +03:00
Antonio Ospite
7ae7f657fa interleave: interleave samples following the Default Channel Ordering
In order to have a full mapping between channel positions in the audio
stream and loudspeaker positions, the channel-mask alone is not enough:
the channels must be interleaved following some Default Channel Ordering
as mentioned in the WAVEFORMATEXTENSIBLE[1] specification.

As a Default Channel Ordering use the one implied by
GstAudioChannelPosition which follows the ordering defined in SMPTE
2036-2-2008[2].

NOTE that the relative order in the Top Layer is not exactly the same as
the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users
using so may channels are already aware of such discrepancies.

[1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx
[2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf

Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127
2014-10-02 10:21:26 +03:00
Sebastian Dröge
7729f4ce81 wavenc: Send CAPS event after the pad was activated
Otherwise the CAPS event will be dropped and we never configure any caps at
all, leading to weird behaviour in many situations. Especially header
rewriting is not going to work if a capsfilter is after wavenc.

https://bugzilla.gnome.org/show_bug.cgi?id=737735
2014-10-02 10:10:11 +03:00
Sebastian Dröge
1a2adf5123 videomixer: Actually use the correct GstVideoInfo for conversion 2014-10-01 17:29:29 +03:00