Commit graph

974 commits

Author SHA1 Message Date
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
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 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
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
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
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
Andoni Morales Alastruey
4a78a77e65 rtp: fix duplicated symbols with libvpx 2013-05-02 14:03:33 +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
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
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
Sebastian Dröge
b0b0557c48 gst: Add better support for static plugins 2013-04-15 15:54:11 +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
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
Wim Taymans
91a3afc4dc gstpay: use bufferlist to avoid memcpy 2013-04-09 16:53:31 +02:00
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
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
Wim Taymans
ac2bcfa833 theorapay: add delta-unit to output frames 2013-03-31 19:14:04 +02:00
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
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
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
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
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
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
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
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
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
Thijs Vermeir
de41376231 rtp: use appropriate printf format for gsize 2012-12-18 16:02:09 +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
Christian Fredrik Kalager Schaller
485505f323 Fix vp8rtp header names in Makefile 2012-11-07 13:36:33 +01:00
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
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
Sjoerd Simons
a45e7a3fc0 rtpvp8pay: Treat the frame header just like any other partition
When setting up the initial mapping just act as if the global frame
information is another partition. This saves special-casing it later in
the actual packetizing code.
2012-11-01 20:53:47 +00:00
Sjoerd Simons
e9f4e9342f rtpvp8: Add simple payloaders and depayloaders for VP8
Minimal implementation of http://www.webmproject.org/code/specs/rtp/,
version 0.3.2
2012-11-01 20:53:47 +00:00
Wim Taymans
d6fd0ebd04 gstpay: fix for 1.0 events
Caps events are sometimes not followed by a buffer but by an event. Flush any
pending caps before we make a packet with the event.
Chain up to the parent event handler before we attempt to push RTP packets, it
might be a segment event.
2012-11-01 18:42:39 +00:00
Wim Taymans
05232c55a5 gstdepay: fix small leak 2012-11-01 18:42:24 +00:00
Wim Taymans
08e5a197b4 gstdepay: add support for events
Conflicts:
	gst/rtp/gstrtpgstdepay.c
2012-11-01 18:18:19 +00:00
Wim Taymans
54b783b5a3 rtpgstpay: add support for sending events
We currently only send tags and custom events. The other events
might interfere with the receiver timings or are otherwise handled
by RTP.

Conflicts:
	gst/rtp/gstrtpgstpay.c
2012-11-01 18:06:11 +00:00
Wim Taymans
6502d08e43 gstpay: rewrite payloader
Use adapter to assemble the payload and make a flush function to
turn this payload into (fragmented) packets.

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
2012-11-01 17:57:52 +00:00
Wim Taymans
c0713e4b80 gstdepay: check for correct fragment offset
Make sure we only insert the rtp packet in the adapter when the
frag_offset matches. When the first packet of a fragment is dropped,
it avoids putting the remaining packets in the adapter and processing
the partial fragment.

Conflicts:
	gst/rtp/gstrtpgstdepay.c
2012-11-01 12:09:47 +00:00
Wim Taymans
8a402e0c06 gstpay: set C flag on all buffers of the fragment
Set the C flags on all the fragments instead of only those with
caps in them. This makes it easier in the receiver to check if there
is a caps in the assembled fragments just by looking at the last RTP
packet flags.
2012-11-01 12:06:08 +00:00
Wim Taymans
d78ff07f7d gstdepay: use the capsversion
Take the caps from the input caps and store it in the slot given
by capsversion.
2012-11-01 11:37:44 +00:00
Wim Taymans
936c3819b5 gstpay: send caps inline
Place the capsversion on the outgoing caps so that they end up in
an SDP as well. Receivers need to know what capsversion a particular
caps is for to be able to match the caps to the CV in the RTP packets.
Place the caps inside the RTP packet whenever the caps change.

Based on patch by Andrzej Bieniek <andrzej.bieniek@pure.com>

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
2012-11-01 11:34:33 +00:00
Andrzej Bieniek
3b1931a039 gstpay: add debug
Conflicts:
	gst/rtp/gstrtpgstpay.c
2012-11-01 11:28:50 +00:00
Andrzej Bieniek
ee5ecc7773 depay: correctly skip caps header size
Conflicts:
	gst/rtp/gstrtpgstdepay.c
2012-11-01 11:27:13 +00:00
Wim Taymans
e9040e90a5 jpegdepay: store quant tables in zigzag order 2012-10-17 14:23:01 +02:00
Rasmus Rohde
47a8eb7ca8 gstrtpdepay: don't leak input buffer
The rtp buffer is never unmapped in the normal code exit path
of gst_rtp_gst_depay_process(..) resulting in a memory leak.

https://bugzilla.gnome.org/show_bug.cgi?id=685512
2012-10-04 19:44:28 +01:00
Patricia Muscalu
7a863e4d8d rtph264pay: do not push unmapped data
Also do not use a GstBuffer after it has been pushed into the adapter.

https://bugzilla.gnome.org/show_bug.cgi?id=685213
2012-10-04 09:22:50 +01:00
Wim Taymans
dbe941338d rtpvrawdepay: negotiate pool with srcpad caps 2012-09-27 14:15:50 +02:00
Olivier Crête
bc252d29ee rtph264pay: Make sure the caps don't have duplicated sps/pps 2012-09-21 17:36:12 -04:00
Wim Taymans
829c80ce6c fix more caps 2012-09-14 13:30:37 +02:00
Mark Nauwelaerts
8d93246b93 gst: adjust comment style 2012-09-10 14:31:02 +02:00
Mark Nauwelaerts
f24b58d19c rtpamrdepay: unmap rtp buffer
... thereby plugging a memleak.
2012-09-07 15:25:53 +02:00
Mark Nauwelaerts
fa90dfc4df rtph264pay: avoid crashing on NULL access in debug message 2012-09-07 15:25:52 +02:00
Mark Nauwelaerts
8f4bfeb698 rtph263ppay: plug caps leak 2012-09-07 15:25:52 +02:00
Tim-Philipp Müller
9bf90f47cf video/x-xvid -> video/mpeg,mpegversion=4 2012-09-03 02:51:24 +01:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Olivier Crête
264bcf7d6f rtph264pay: Make it actually work after cleanups 2012-08-08 19:49:05 -07:00
Mark Nauwelaerts
1547fdbe5a rtpmparobustdepay: set correct data_size for generated dummy frame
... which prevents getting stuck in a loop if such one is needed.
2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
3e1832f5a4 rtpmparobustdepay: improve and fix debug statement
... so it really informs about next rather than past frame.
2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
31a1cb0a11 rtpmparobustdepay: update available bytewriter space when repositioning
... and add some more assert to catch potential surprises early on.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680558
2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
0bf9d8c6a6 rtpmparobustdepay: modify buffer data rather than buffer itself 2012-07-26 16:34:52 +02:00
Mark Nauwelaerts
c40807f6aa rtpmparobustdepay: avoid leaking bytewriter instance 2012-07-26 16:34:52 +02:00
Wim Taymans
0ed9e07c5d h264depay: small cleanups 2012-07-25 12:49:07 +02:00
Wim Taymans
4b92022120 rtp: always use buffer lists 2012-07-23 16:42:56 +02:00
Patricia Muscalu
3dd99f06f4 rtpmp4vpay: always enable buffer-lists 2012-07-23 16:17:37 +02:00
Patricia Muscalu
15cce2dd26 rtpjpegpay: always enable buffer-lists 2012-07-23 16:15:59 +02:00
Wim Taymans
51371d26ee update for RTP buffer api changes 2012-07-17 16:38:27 +02:00
Patricia Muscalu
d38ac43a27 rtph264pay: use buffer lists
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679994
2012-07-17 10:10:14 +02:00
Tim-Philipp Müller
c6224443a4 rtph264pay: avoid some relocations 2012-07-06 19:11:02 +01:00
Tim-Philipp Müller
3ef35ecdbc rtpmp4vpay: remove deprecated send-config property
Use config-interval instead.
2012-07-06 14:49:18 +01:00
Tim-Philipp Müller
cd1da84bcc rtph264depay: remove deprecated "byte-stream" and "access-unit" properties
These will be picked automatically based on downstream caps now, so
if you want the depayloader to output a specific format, make sure
the element downstream advertises that preference or use a capsfilter
after the depayloader to force it.
2012-07-06 14:46:22 +01:00
Tim-Philipp Müller
cffbf8cfc3 rtph264pay: remove deprecated and non-functional "profile-level-id" property
This is now optionally taken from downstream caps, so can be
specified via a capsfilter after the payloader.
2012-07-06 14:46:22 +01:00
Tim-Philipp Müller
48706beb70 rtph263ppay: accept any h263 input unless downstream forces specific requirements
rtph263ppay should accept any input compatible with its sink template
caps if it just outputs to e.g. udpsink or fakesink.

rtph263ppay ! rtph263pdepay should also work with any compatible input.
This would fail before with not-negotiated errors because the get_caps
function would see the encoding-name in the depayloader's template caps
and default to baseline H.263 because there's no profile/level information
in those caps, which is the right thing to do if downstream has filtercaps
from an SDP, but not if those fields are absent because they can be
anything like with the depayloader's template caps. Makes

  videotestsrc ! avenc_h263p ! rtph263ppay ! rtph263pdepay ! fakesink

work.
2012-07-06 11:57:38 +01:00
Wim Taymans
8eadb9c12c update for query api changes 2012-07-06 11:26:46 +02:00
Javier Jardón
c740490c26 rtp: remove some outdated comments
https://bugzilla.gnome.org/show_bug.cgi?id=679301
2012-07-03 08:58:26 +01:00
Wim Taymans
6d158775bb rtph264pay: cleanups
Use the caps properties for alignment and format.
Remove some old properties, we always want to use bufferlists when we can now.
2012-06-28 12:00:09 +02:00
Wim Taymans
429bda6923 h264pay: prefer AVC, it's easier to parse etc 2012-06-28 11:32:03 +02:00
Wim Taymans
540245894f theoradepay: fix buffer memory
The memory was added to the input buffer instead of the output buffer.
2012-06-14 10:43:56 +02:00
Thiago Santos
78ec03e32f Some printf variable format fixes
The osx compiler complains about those
2012-06-05 17:53:57 -03:00
Edward Hervey
923be8a85b rtpmp2tdepay: Only output integral mpeg-ts packets
From RFC 2250

2. Encapsulation of MPEG System and Transport Streams
...
   For MPEG2 Transport Streams the RTP payload will contain an integral
   number of MPEG transport packets.  To avoid end system
   inefficiencies, data from multiple small MTS packets (normally fixed
   in size at 188 bytes) are aggregated into a single RTP packet.  The
   number of transport packets contained is computed by dividing RTP
   payload length by the length of an MTS packet (188).
....

Since it needs to contain "an integral number of MPEG transport packets", a
simple fix is to check that's the case, and strip off any leftover data.

Fixes #676799

Conflicts:

	gst/rtp/gstrtpmp2tdepay.c
2012-05-26 12:04:54 +02:00
Luis de Bethencourt
c81fff0471 rtp: fix build issue in gstrtph264pay.c 2012-05-24 09:29:25 +01:00
Jonas Holmberg
7bf3a1bf95 rtph264pay: Add unrestricted caps
If there are no profile restrictions downstream, return caps with
profile=constrained-baseline in the first structure and append
unrestricted caps as the last structure.

Fixes bug #672019
2012-05-24 10:01:19 +02:00
Mark Nauwelaerts
182596b3ab rtpmp2tpay: respect mtu and packet boundaries
See #659915.
2012-05-18 12:53:44 +02:00
Youness Alaoui
7703a11073 rtpjpegpay: Allow U and V components to use different quant tables if they contain the same data
This allows some cameras (Logitech C920) that specify different quant
tables but both with the same data, to work.
Bug reported by Robert Krakora
2012-05-16 09:49:08 +02:00
idc-dragon
e0945d0a2d celtdepay: calculate size correctly
The summation was done wrong, causing the de-payloader to exit its loop too
early, before all frames are processed.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674472
2012-04-25 10:29:56 +02:00
Sebastian Dröge
04b70571e5 video: Update for libgstvideo API changes 2012-04-19 12:20:59 +02:00
Edward Hervey
71fc25849e rtp: Use unchecked variant of GstByteWriter where applicable
The size was checked before
2012-04-12 15:50:16 +02:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Sebastian Dröge
aa2cd462da gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:36:38 +02:00
Sebastian Dröge
5cdd49bf25 gst: Update versioning 2012-04-04 14:37:47 +02:00
Wim Taymans
3d61d12e03 update for buffer api change 2012-03-30 18:15:34 +02:00
Wim Taymans
69002aa24f update for buffer changes 2012-03-28 12:53:05 +02:00
Wim Taymans
e310ee8218 caps: improve caps handling
Avoid caps copy and leaks
2012-03-27 16:42:41 +02:00
Mark Nauwelaerts
e5ab3cc0a0 rtph264pay: ensure output caps are set when pushing output data
... even if some SPS/PPS has not passed by yet.
2012-03-26 18:38:34 +02:00
Mark Nauwelaerts
4bbc2a7106 rtpL16(de)pay: fix raw audio format in template caps 2012-03-26 18:38:34 +02:00
Olivier Crête
06f1c1817e rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set
This allows outputting streams in AVC format even if the SPS/PPS are sent inside
the RTP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=654850

Ported from master
2012-03-22 16:18:37 -04:00
Wim Taymans
c44cd8f55b Merge branch 'master' into 0.11
unport gdkpixbuf
not merged: https://bugzilla.gnome.org/show_bug.cgi?id=654850

Conflicts:
	docs/plugins/Makefile.am
	docs/plugins/gst-plugins-good-plugins-docs.sgml
	docs/plugins/gst-plugins-good-plugins-sections.txt
	docs/plugins/gst-plugins-good-plugins.hierarchy
	docs/plugins/inspect/plugin-avi.xml
	docs/plugins/inspect/plugin-png.xml
	ext/flac/gstflacdec.c
	ext/flac/gstflacdec.h
	ext/libpng/gstpngdec.c
	ext/libpng/gstpngenc.c
	ext/speex/gstspeexdec.c
	gst/audioparsers/gstflacparse.c
	gst/flv/gstflvmux.c
	gst/rtp/gstrtpdvdepay.c
	gst/rtp/gstrtph264depay.c
2012-03-22 11:53:24 +01:00
Wim Taymans
ced47580b7 update for bufferpool changes 2012-03-15 22:11:17 +01:00
Wim Taymans
f3a770a20c update for allocation query changes 2012-03-15 20:37:56 +01:00
Olivier Crête
053f33adc8 rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set
This allows outputting streams in AVC format even if the SPS/PPS are sent inside
the RTP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
2012-03-15 14:20:22 -04:00
Wim Taymans
751fcf035b take padding into account 2012-03-14 19:56:56 +01:00
Wim Taymans
734f11e4d3 mp4vpay: we can also handle x-divx 2012-03-14 11:26:35 +01:00
Wim Taymans
fba47d17e8 mp4vdepay: fix buffer handling
Don't always output the payload subbuffer, use a separate variable to
make things clearer and without the error.
2012-03-13 21:31:48 +01:00
Wim Taymans
745210e792 h264depay: unmap on empty packet 2012-03-13 19:26:23 +01:00
Wim Taymans
d65de434f5 rtph264pay: do DTS and PTS correctly 2012-03-13 18:07:18 +01:00
Wim Taymans
e4fed38f49 rtp: fix unmap calls 2012-03-13 17:27:32 +01:00
Wim Taymans
a32d944a38 fix for caps api changes 2012-03-11 19:06:37 +01:00
Sebastian Dröge
78079635a6 dvdepay: Fix 'comparison of unsigned expression >= 0 is always true' compiler warning
This was an actual bug as it could've caused reading from
invalid memory areas when the input is broken.
2012-03-06 14:16:21 +01:00
Wim Taymans
ca9532ccc5 update for new memory api 2012-02-22 02:10:33 +01:00
Olivier Crête
18899cf94d rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-21 10:51:43 +01:00
Olivier Crête
1fe69911a4 rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-20 14:30:55 -05:00
Wim Taymans
225e98d623 Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacenc.c
	ext/jack/gstjackaudioclient.c
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	ext/pulse/plugin.c
	ext/shout2/gstshout2.c
	gst/matroska/matroska-mux.c
	gst/rtp/gstrtph264pay.c
2012-02-10 16:23:14 +01:00
Tim-Philipp Müller
5b25f3737b rtph264pay: add stream-format and alignment to h264 sink caps
We're happy to accept both byte-stream and avc, advertise
that on the sink caps and fix up _get_caps() function to
not just return "video/x-h264".

https://bugzilla.gnome.org/show_bug.cgi?id=606662
2012-02-10 14:08:55 +00:00
Tim-Philipp Müller
6872b40873 rtph264depay: add stream-format and alignment fields to src template caps
Because we can. And so we get a warning if we try to output avc with
nal alignment or somesuch.

https://bugzilla.gnome.org/show_bug.cgi?id=606662
2012-02-10 14:08:55 +00:00
Vincent Penquerc'h
d651baf05a rtpmp2tpay: do not try to flush a packet when no data is available
https://bugzilla.gnome.org/show_bug.cgi?id=668874
2012-01-31 13:12:47 +00:00
Pascal Buhler
c16fed2ad9 rtph264depay: Exclude NALu size from payload length on truncated packets.
https://bugzilla.gnome.org/show_bug.cgi?id=667846
2012-01-30 15:49:07 +00:00
Sebastian Dröge
0b517ce9fb Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-01-25 12:49:34 +01:00
Sebastian Dröge
10554b271f Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacdec.c
	ext/jpeg/gstjpegenc.c
	ext/pulse/pulsesink.c
	sys/v4l2/gstv4l2src.c
2012-01-25 12:49:11 +01:00
Wim Taymans
b4630dd3e0 more memory API porting 2012-01-25 12:30:29 +01:00
Wim Taymans
583d39dd8d update for new memory API 2012-01-25 12:30:28 +01:00
Mark Nauwelaerts
a3ea25bc88 rtpmp4adepay: prevent out-of-bound array access 2012-01-20 17:10:48 +01:00
Mark Nauwelaerts
ed94e01231 rtptheoradepay: remove dead code 2012-01-20 17:10:40 +01:00
Sebastian Dröge
59e08fa503 configure: Remove socket/winsock specific checks
Not necessary anymore.
2012-01-17 16:53:31 +01:00
Vincent Penquerc'h
2a7a38ca07 rtph263ppay: fix caps leak 2012-01-16 15:42:46 +00:00
Sebastian Dröge
4885f34458 rtp: Update for the new audio caps 2012-01-05 10:30:34 +01:00
Tim-Philipp Müller
668e15598b Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	sys/v4l2/gstv4l2object.c
2011-12-08 01:28:26 +00:00
Wim Taymans
b1d771cf8c h263pay: fix invalid return value 2011-12-06 14:23:30 +01:00
Edward Hervey
04520cbe9a rtp: Initialize GstRTPBuffer before usage 2011-12-05 18:39:59 +01:00
Sebastian Rasmussen
c090201ca5 rtpjpegpay: Ceil jpeg dimensions, instead of floor
A JPEG image inside an RTP stream has a preceeding RFC2435 header that
conveys width/height. The dimensions in this header are limited to be
multiples of 8. Since JPEG uses an MCU of 8x8 pixels any image must
already indirectly have image data dimensions that are rounded up in
order to contain enough data to render the image. Therefore this fix
safely rounds the image dimensions in the RFC2435 header up to the
closest multiple of 8.
2011-12-05 10:48:54 +01:00
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:30:27 +00:00
Matej Knopp
1e5dd9e315 Fix printf format compiler warnings on OS X / 64bit
https://bugzilla.gnome.org/show_bug.cgi?id=662615
2011-11-22 01:28:22 +00:00
Wim Taymans
105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Wim Taymans
797523efbd _peer_get_caps() -> _peer_query_caps() 2011-11-15 18:04:44 +01:00
Wim Taymans
75dc9634eb change getcaps to query
Chain up event function in payloaders.
2011-11-15 18:04:44 +01:00
Wim Taymans
af1eec2ece rtp: fix for rtp header changes 2011-11-11 19:21:50 +01:00
Wim Taymans
e84b8dbe94 update for base class rename 2011-11-11 12:32:41 +01:00
Wim Taymans
249d0083cc update for base class rename 2011-11-11 12:25:01 +01:00
Wim Taymans
7e12b58e37 update for adapter api changes 2011-11-10 18:32:58 +01:00
Wim Taymans
fbaf216d25 update for changed base classes 2011-11-10 17:23:47 +01:00
Wim Taymans
85e73e0818 h263ppay: report to 0.11 2011-11-09 12:25:01 +01:00
Wim Taymans
95f3987332 Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacdec.c
	gst/audioparsers/gstflacparse.c
	gst/isomp4/qtdemux.c
2011-11-09 12:18:01 +01:00
Olivier Crête
e15c293f13 rtph263ppay: Return the sink pad template as sink caps, not the src's
https://bugzilla.gnome.org/show_bug.cgi?id=577784
2011-11-08 15:53:39 +01:00
Olivier Crête
4b28d9d44e rtph263ppay: Also implement size/framerate restrictions in getcaps
https://bugzilla.gnome.org/show_bug.cgi?id=577784
2011-11-08 15:53:18 +01:00
Olivier Crête
ff31090671 rtph263ppay: Implement getcaps following RFC 4629, picks the right annexes
https://bugzilla.gnome.org/show_bug.cgi?id=577784
2011-11-08 15:52:57 +01:00
Tim-Philipp Müller
d65490dfad rtp: use GLib's G_BIG_ENDIAN define instead of BIG_ENDIAN
Fixes compiler warning on mingw32
2011-11-03 23:28:31 +00:00
Wim Taymans
b1ef7e8a86 update for meta api change 2011-11-02 09:06:37 +01:00
Wim Taymans
9a8a8e72c8 structure: fix for api update 2011-11-02 09:06:37 +01:00
Wim Taymans
9c14280b1d make some more things compile again 2011-10-27 19:00:52 +02:00
Wim Taymans
fc4684f4c6 fix compilation 2011-10-27 16:03:17 +02:00
Marc Leeman
98075ad70d set colour masks for video/x-raw-rgb in rtpvrawdepay 2011-10-14 09:32:47 +02:00
Wim Taymans
a5cc912140 Merge branch 'master' into 0.11
Conflicts:
	ext/jpeg/gstjpegdec.c
	gst/rtp/gstrtpvrawpay.c
2011-10-13 08:58:06 +02:00
Edward Hervey
1b56d40170 rtpvrawpay: Only use 24 LSB for depth=24 RGB caps
... and indent the masks for clarity
2011-10-12 11:26:50 +02:00
Sjoerd Simons
bf65acf11f gstrtpg722pay: Compensate for clockrate vs. samplerate difference
The RTP clock-rate used for G722 is 8000, even though the samplerate is
16000. Compensate for this by pretending G722 has 8 bits per sample
instead of the 4 bits as if it were a codec that ran at half the speed,
but with twice the number of bits. Fixes #661376
2011-10-10 21:50:28 +01:00
Wim Taymans
87fbd1e784 Merge branch 'master' into 0.11
Conflicts:
	common
	ext/pulse/pulsesink.c
	ext/soup/gstsouphttpclientsink.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtpmanager/gstrtpjitterbuffer.c
	gst/rtpmanager/rtpjitterbuffer.c
	gst/rtsp/gstrtspsrc.c
	sys/ximage/gstximagesrc.c
2011-09-28 12:44:59 +02:00
Mark Nauwelaerts
fd757890eb rtph264depay: improve downstream flow return feedback to upstream
... although basertpdepay does not really make it easy/possible to do so
all the way.
2011-09-20 14:14:39 +02:00
Wim Taymans
83ea243000 Merge branch 'master' into 0.11
Conflicts:
	common
2011-09-06 16:37:03 +02:00
Wim Taymans
33f18b8ea4 Merge branch 'master' into 0.11
Conflicts:
	gst/audioparsers/gstamrparse.c
	gst/isomp4/qtdemux.c
2011-09-06 16:06:25 +02:00
Mark Nauwelaerts
06f8e356a6 rtpmp4gdepay: improve bogus interleaved index compensating
Patch by <gudake@gmail.com>

Fixes #654585.
2011-09-06 13:20:23 +02:00
Olivier Crête
d4778dbe43 rtph263ppay: Set H263-2000 if thats what the other side wants
The static caps states this element supports H263-2000, but setcaps never
sets it, so it was lie.

See https://bugzilla.gnome.org/show_bug.cgi?id=577784
2011-09-05 12:58:55 +02:00
Wim Taymans
24df106272 mp2t: fix encoding name according to RFC3551 2011-08-31 18:45:15 +02:00
Wim Taymans
18065ac823 port to new video flags 2011-08-25 16:41:23 +02:00
Wim Taymans
60f0e44bf6 video: port to new colorimetry info 2011-08-23 19:09:31 +02:00
Wim Taymans
9d6371405e fourcc: remove fourcc from caps 2011-08-22 12:24:15 +02:00
Wim Taymans
77ad0a1363 port more elements to new audio caps and API 2011-08-19 14:01:45 +02:00
Wim Taymans
ee2aa25e04 port to new API 2011-08-03 18:37:27 +02:00
Wim Taymans
4121021bb2 Merge branch 'master' into 0.11
Conflicts:
	ext/pulse/pulsesink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtp/gstrtph264pay.c
	gst/rtpmanager/gstrtpssrcdemux.c
2011-08-03 18:25:30 +02:00
Robert Krakora
f7893b8721 rtpjpegpay: Add support for H.264 payload in MJPEG container
See http://www.quickcamteam.net/uvc-h264/USB_Video_Payload_H.264_0.87.pdf

Fixes bug #655530.
2011-08-03 10:09:42 +02:00
Wim Taymans
5771056ed5 rtpvorbispay: fix porting error 2011-08-02 11:51:45 +02:00
Wim Taymans
49af68ebf4 -good: fix for bufferpool API change 2011-07-29 17:27:07 +02:00
Sjoerd Simons
4c73439ee3 rtph264depay: Cope with FU-A E bit not being set
Some h264 payloaders are unfortunately buggy and don't correctly set the
E bit in FU-A NAL when they have ended. Work around this by assuming
such a fragmentation unit has ended when there was no packet loss and a
new NAL is started
2011-07-27 18:18:13 +01:00
Wim Taymans
3e089bd7a9 rtp: fix compilation 2011-07-26 17:45:01 +02:00
Olivier Crête
2591a882ae rtph264depay: Complete merged AU on marker bit
The marker bit on a RTP packet means the AU has been completed, so push it out
immediately to reduce the latency.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
2011-07-21 17:11:08 +02:00
Olivier Crête
118a7cc36a rtph264pay: Only set the marker bit on the last NALU of a multi-NALU access unit
An access unit could contain multiple NAL units, in that case, only the last
RTP packet of the last NALU should have its marker bit set.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
2011-07-21 17:11:06 +02:00
Mark Nauwelaerts
471904032d rtph264depay: reset upon FLUSH_STOP
... which is particularly needed when merging NAL units, where not resetting
would lead to output of an older (pre-flush) AU (with unintended timestamp).
2011-07-18 14:32:26 +02:00
Wim Taymans
9c087d7d85 Merge branch 'master' into 0.11 2011-07-15 17:06:39 +02:00
Olivier Crête
87c7f303b0 rtppcmApay/depay: Static clock rates on static payloads, dynamic on dynamic
Partially reverts 397dc60b
2011-07-14 20:13:01 -04:00
Olivier Crête
57a832cbb1 rtph264pay: Implement getcaps
Convert profile-level-id from RTP caps into video/x-h264 style caps (with profile and level)
2011-07-13 14:10:35 -04:00
Mark Nauwelaerts
eb82a50bd1 rtp: port remaining to 0.11 2011-07-10 21:50:19 +02:00
Wim Taymans
cc65bff7c1 Merge branch 'master' into 0.11
Conflicts:
	configure.ac
	docs/plugins/inspect/plugin-esdsink.xml
	docs/plugins/inspect/plugin-gconfelements.xml
2011-06-21 18:24:41 +02:00
Mark Nauwelaerts
3daf1ecc21 rtpmp4adepay: fix output buffer timestamps in case of multiple frames 2011-06-21 15:15:33 +02:00
Wim Taymans
3c889415a3 rtp: port some more (de)payloader 2011-06-13 17:14:00 +02:00
Wim Taymans
9a54175e9f rtp: port to 0.11 2011-06-13 16:33:46 +02:00
Wim Taymans
b0fbb1725f rtp: fix for API changes in the base classes 2011-06-13 13:25:49 +02:00
Wim Taymans
0b1bdcf7cb Merge branch 'master' into 0.11
Conflicts:
	sys/ximage/ximageutil.c
2011-06-02 18:51:29 +02:00
Marc Leeman
ff1c05d876 rtpmp4vpay: Deprecated send-config property and replace by config-interval
Fixes bug #622412.
2011-05-26 12:22:52 +02:00
Wim Taymans
d89790d545 Merge branch 'master' into 0.11
Conflicts:
	gst/avi/gstavidemux.c
	gst/rtp/gstrtpac3depay.c
	gst/rtp/gstrtpg726depay.c
	gst/rtp/gstrtpmpvdepay.c
	gst/videofilter/gstgamma.c
2011-05-24 17:34:19 +02:00
Mark Nauwelaerts
397dc60b71 pcmudepay: allow variable sample rate 2011-05-24 13:13:55 +02:00
Mark Nauwelaerts
f335fee99e pcmadepay: allow variable sample rate 2011-05-24 13:13:52 +02:00
Stefan Kost
d122ea0122 rtp: fix static array overruns in a nicer way
Use G_N_ELEMENTS instead of hard-coding the array size.
2011-05-20 10:34:47 +03:00
Stefan Kost
5792d3b9c0 rtp: fix static array overruns
Yes array[10] has elements from 0...9.
2011-05-20 00:53:44 +03:00
Jose Antonio Santos Cadenas
9d32243671 rtp: Fix segmentation fault processing payload buffers
This commit checks if the value returned by
gst_rtp_buffer_get_payload_buffer and
gst_rtp_buffer_get_payload_subbuffer is NULL before using it.
2011-05-18 15:25:24 +02:00
Wim Taymans
31ffc671f2 rtpgstpay: fix buffer leak 2011-04-26 16:04:07 +01:00
Wim Taymans
eb84592cad rtpgstpay: fix buffer leak 2011-04-26 15:58:12 +02:00
Wim Taymans
9a96783abb rtp: port some more elements 2011-04-25 18:14:45 +02:00
Wim Taymans
bf9b4f8362 rtp: port more to 0.11 2011-04-25 17:27:40 +02:00
Wim Taymans
60db07b4bb rtp: port some more (de)payloaders 2011-04-25 13:16:58 +02:00
Wim Taymans
4aa6ca5578 port more plugins to 0.11 2011-04-18 10:54:43 +02:00
Wim Taymans
7555d0949f Merge branch 'master' into 0.11
Conflicts:
	android/apetag.mk
	android/avi.mk
	android/flv.mk
	android/icydemux.mk
	android/id3demux.mk
	android/qtdemux.mk
	android/rtp.mk
	android/rtpmanager.mk
	android/rtsp.mk
	android/soup.mk
	android/udp.mk
	android/wavenc.mk
	android/wavparse.mk
	configure.ac
2011-04-18 10:23:45 +02:00
Tim-Philipp Müller
f325935314 pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.

g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.

Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
2011-04-16 18:15:43 +01:00
Robert Swain
5b18c652fb rtp, rtpmanager: Address unused but set variables
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.

gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
2011-04-16 12:49:16 +01:00
Thibault Saunier
b541208b77 android: Make it ready for androgenizer
Remove the android/ top dir
Fixe the Makefile.am to be androgenized

To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
2011-04-11 01:20:11 +02:00
Haakon Sporsheim
fd545e260d rtpgstpay: declare frag_offset to hold 32bits.
As specified in documenation above and below.

https://bugzilla.gnome.org/show_bug.cgi?id=646954
2011-04-09 23:14:18 +01:00
Alexey Fisher
9b15f9c6a1 rtpspeexpay: Do not transmitt samples with GAP flag
If we get GAP samples, there is no need to transmitt it.
In some situations, microphone is muted, we can drop net traffick
usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s
2011-04-08 13:56:13 +02:00
Wim Taymans
0024300aa2 rtp: port some pay/depayloaders 2011-04-07 19:04:33 +02:00