Commit graph

41 commits

Author SHA1 Message Date
Miguel Paris
bae3636197 rtp: fix gst_rtp_buffer_ext_timestamp taking into account backwards
If timestamp goes forwards more than allowed, we consider that the
timestamp belongs to the previous counting, so the extended timestamp
is unwrapped.

https://bugzilla.gnome.org/show_bug.cgi?id=783443
2017-12-21 17:27:42 -05:00
Tim-Philipp Müller
778589cd5b test: fix indentation 2016-03-30 22:41:54 +01:00
Stian Selnes
93196092d1 rtcpbuffer: Add API for APP packets
https://bugzilla.gnome.org/show_bug.cgi?id=761944
2016-03-24 14:24:11 +02:00
Haakon Sporsheim
d8e9a711a0 rtcpbuffer: Add profile-specific extension API.
https://bugzilla.gnome.org/show_bug.cgi?id=761950
2016-03-24 14:22:54 +02:00
Tim-Philipp Müller
e5fb7275c2 tests: fix indentation of various unit tests 2016-01-06 01:12:13 +00:00
Stian Selnes
0a668c1866 rtpbuffer: Add map flag to skip padding
Encrypted RTP buffers may contain encrypted padding, hence it's
necessary to have an option to relax the validation in order to
successfully map the buffer.

When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
gst_rtp_buffer_map() will map the buffer like if padding is not
present.

https://bugzilla.gnome.org/show_bug.cgi?id=752705
2015-10-15 22:42:37 +03:00
Stian Selnes
1586981b1b rtcpbuffer: Fix validation of packets with padding
The padding (if any) is included in the length of the last packet, see
RFC 3550.

Section 6.4.1:
   padding (P): 1 bit
      If the padding bit is set, this individual RTCP packet contains
      some additional padding octets at the end which are not part of
      the control information but are included in the length field. The
      last octet of the padding is a count of how many padding octets
      should be ignored, including itself (it will be a multiple of
      four).

Section A.2:
   *  The padding bit (P) should be zero for the first packet of a
      compound RTCP packet because padding should only be applied, if it
      is needed, to the last packet.

   *  The length fields of the individual RTCP packets must add up to
      the overall length of the compound RTCP packet as received.

https://bugzilla.gnome.org/show_bug.cgi?id=751883
2015-07-06 12:06:47 +03:00
Jose Antonio Santos Cadenas
9931bef8ca rtcpbuffer: Update package validation to support reduced size rtcp packets
According to this section of the rfc.
https://tools.ietf.org/html/rfc5506#section-3.4.2
The validation should be updated to accept more types of RTCP
packages, with this mask change feedback packages will be also
accepted.

Change-Id: If5ead59e03c7c60bbe45a9b09f3ff680e7fa4868
2015-06-05 10:18:21 +02:00
Edward Hervey
ce16635f46 check: Use GST_CHECK_MAIN () macro everywhere
Makes source code smaller, and ensures we go through common initialization
path (like the one that sets up XML unit test output ...)
2015-06-02 16:14:39 +02:00
Sebastian Dröge
29758da255 rtp: Fix GBytes memory leak in test 2014-04-15 10:29:12 +02:00
Wim Taymans
c6915ea785 check: add some more checks
Add header and payload length check in case of CSRCs.

See https://bugzilla.gnome.org/show_bug.cgi?id=723196
2014-02-04 14:32:48 +01:00
Sebastian Rasmussen
c734f9fba8 rtpbuffer: Allow subbuffering of empty buffers
See https://bugzilla.gnome.org/show_bug.cgi?id=720162
2013-12-10 12:38:56 +01:00
Branko Subasic
4dd5c5b808 rtpbuffer: add gst_rtp_buffer_get_payload_bytes
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.

The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.

https://bugzilla.gnome.org/show_bug.cgi?id=698562
2013-06-18 11:23:40 +02:00
Wim Taymans
02b6e58eef tests: add NTP64 and ntp56 header extension checks 2012-11-06 09:18:54 +01:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-03 23:05:09 +00:00
Wim Taymans
1968127650 rtp: Fix extension data support
Allocate header, payload and padding in separate memory blocks in
gst_rtp_buffer_allocate().
don't use part of the payload data as storage for the extension data but store
it in a separate memory block that can be enlarged when needed.
Rework the one and two-byte header extension to make it reserve space for the
extra extension first.
Fix RTP unit test. Don't map the complete buffer or make assumptions on the
memory layout of the underlaying implementation. We can now always add extension
data because we have a separate memory block for it.
2012-08-22 09:56:39 +02:00
Wim Taymans
11a494d5c9 rtp: Add support for multiple memory blocks in RTP
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
2012-07-17 16:41:36 +02:00
Olivier Crête
cb044668d3 rtcpbuffer: Set the map.size to the current size of the RTCP packet
maxsize is the maximum size
2012-01-27 19:01:55 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Wim Taymans
ea9ef0ee63 tests: fix some tests 2012-01-19 15:32:52 +01:00
Wim Taymans
f9967e4aac Merge branch 'master' into 0.11
Conflicts:
	gst-libs/gst/video/video.h
	gst/playback/gstplaysinkaudioconvert.c
	gst/playback/gstplaysinkvideoconvert.c
	tests/check/libs/rtp.c
2011-06-02 12:18:13 +02:00
Tim-Philipp Müller
4f6da2bcbc tests: fix some more unused-but-set-variable warnings with gcc 4.6 2011-05-29 13:32:04 +01:00
Wim Taymans
e33b73f9df tests: fix RTP and RTCP unit tests 2011-03-28 18:42:09 +02:00
Wim Taymans
40dc12da3a tests: work on porting the unit tests 2011-03-28 14:12:24 +02:00
Tim-Philipp Müller
0d39e2896e tests: fix invalid free and buffer list leak in rtp library unit test 2010-11-02 12:29:05 +00:00
Thiago Santos
8818ea08bd tests: rtp: No need to unref buffer from bufferlist
Buffers obtained from buffer list iterators don't need to
be unreffed.

Test was failing due to this.
2010-10-06 16:19:49 -03:00
Wim Taymans
77c78a6a9d check: fix rtp checks
Fix the checks for the extension support in RTP.
2010-10-05 17:13:09 +02:00
Olivier Crête
96aa439867 tests: Test the manipulations of bufferlists containing RFC 5285 header extensions 2010-10-05 16:19:14 +02:00
Olivier Crête
7612c137bd tests: Add test for RTP header extension functions 2010-10-05 16:19:14 +02:00
Wim Taymans
66c388a0e0 rtp: add bufferlist support 2009-06-18 18:51:04 +02:00
Edward Hervey
83fe624025 tests: Fix indentation 2009-02-22 13:43:35 +01:00
Wim Taymans
9ce042e2a7 Avoid overflows in the padding checks by doing the check slightly
differently.
Add a unit test to check for correct behaviour.
2009-01-21 13:09:29 +01:00
Edward Hervey
c5ae184910 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_event):
Remove erroneous gst_buffer_ref().
* tests/check/libs/rtp.c: (GST_START_TEST):
Don't forget to unref the buffer once you're done with it.
2008-12-12 19:41:28 +00:00
Olivier Crete
3c9df39c15 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes #563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
2008-12-08 12:08:32 +00:00
Wim Taymans
86ab51207b gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
2008-05-14 20:28:02 +00:00
Bernard B
d06df554a9 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes #533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
2008-05-14 13:43:12 +00:00
Wim Taymans
5c1a648f83 tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
Original commit message from CVS:
* tests/check/libs/rtp.c: (GST_START_TEST):
Add check for RTP buffer defaults, padding and marker bit API.
2008-02-27 10:55:03 +00:00
Thijs Vermeir
b8d39bc200 Add gst_rtp_buffer_set_extension_data()
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data):
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add gst_rtp_buffer_set_extension_data()
Add a unit test for this addition. Fixes #511478.
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
2008-02-01 11:09:16 +00:00
Jan Schmidt
f0ca338866 tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
Original commit message from CVS:
* tests/check/Makefile.am:
Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
* tests/check/elements/audiorate.c: (do_perfect_stream_test):
* tests/check/elements/playbin.c:
* tests/check/libs/mixer.c: (test_element_interface_supported),
(gst_implements_interface_init):
* tests/check/libs/rtp.c: (GST_START_TEST):
Fix various assignment type mismatches.
2008-01-12 23:24:02 +00:00
Tim-Philipp Müller
03992b8779 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
2007-09-07 17:37:03 +00:00
Haakon Sporsheim
b2948f2453 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
Original commit message from CVS:
Based on patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix up GstRTPHeader helper struct so that compilers will not under
any circumstances add padding in between our fields, as currently
happens with MSVC on win32, because that would lead to us sending
out RTP payloads with broken RTP headers (#471194).
Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/rtp.c:
Add some simple unit tests for GstRTPBuffer. Some are disabled
because the code tested still needs fixing (set_csrc() does not work).
2007-09-07 16:46:05 +00:00