Commit graph

13802 commits

Author SHA1 Message Date
Stefan Sauer
f48bb20b4f avi: use new riff api to extract nikon metadata
Fixes #636143
2014-01-04 21:34:38 +01:00
Julien Isorce
70d3ff2f79 rtprtxsend/rtprtxreceive: generate gtk doc 2014-01-03 20:48:30 +01:00
George Kiagiadakis
94e4cd203b test/check: Verify rtprtxsend::ssrc-map property works as expected 2014-01-03 20:48:29 +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
Torrie Fischer
e29b5f8b41 examples: rtp: Add end-to-end rtpbin example with RTX elements
This example demonstrates how to use rtpbin with retransmission (rtx)
elements set in the place of rtpbin's "aux" elements in order to
enable RTP retransmission according to the rules of RFC4588.
2014-01-03 20:48:29 +01:00
Julien Isorce
d2edee4b49 doc: add design-rtpauxiliary.txt to describe how rtpbin deals with auxiliary elements 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
Julien Isorce
5f360f3b13 tests/check: add rtpaux::test_simple_rtpbin_aux
It shows how to use "set-aux-receive" and "set-aux-send"
properties of rtpbin to set rtprtxsend and rtprtxreceive

Build 2 pipelines, one for rtpbin as a sender and one for
rtobin as a receive. Then transmit an audio stream.

It also drops some packets to activate restransmission and
check they are actually retransmited.
2014-01-03 20:48:29 +01:00
Julien Isorce
68149d14e1 tests/check: add rtpcollision::test_rtx_ssrc_collision unit test
check that rtxrtpsend changes its retransmission ssrc when
collision happens
2014-01-03 20:48:28 +01:00
George Kiagiadakis
123bc46b60 tests/check: add rtprtx::test_rtxreceive_data_reconstruction
This unit test verifies that retransmitted rtp packets coming out
of rtprtxreceive are the same as the original ones.
2014-01-03 20:48:28 +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
3e818e218b tests/check: Add unit test for rtxsend's max_size_time property 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
George Kiagiadakis
f7277db9e4 tests/check: Add rtprtx::test_rtxsender_packet_retention
This unit test verifies that the rtxsend element correctly maintains
a buffer of already transmitted rtp packets and that it can
re-transmit all of them correctly on demand. It also verifies
that the limit of this buffer (max-size-packets property) is respected.
2014-01-03 20:48:28 +01:00
Julien Isorce
71bdb5e088 tests/check: add rtprtx::test_drop_multiple_sender unit test
Several senders / one receiver

Similar than test_drop_one_sender but with multiple senders
mixed through the funnel element.
It drops some packets and checks that they are retransmited
correctly.
2014-01-03 20:48:28 +01:00
Julien Isorce
2a2fa7ebc0 tests/check: add rtprtx::test_drop_one_sender unit test
Test for one sender / one receiver

Build the pipeline
videotestsrc ! rtpvrawpay ! rtprtxsend ! rtprtxreceive ! fakesink
and drop some buffers between rtprtxsend and rtprtxreceive
Then it checks that every dropped packet has been re-sent.
It also checks that not too much requests has been sent.
2014-01-03 20:48:27 +01:00
Julien Isorce
2e4ce28443 tests/check: add rtprtx::test_push_forward_seq
add simple unit test that manually push buffers
in rtprtxsend connected to rtprtxreceive.
Drops some buffers and make sure they are retransmisted.
2014-01-03 20:48:27 +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
Julien Isorce
19c0e92031 doc: add design for rtp retransmission
Describe how rtprtxsend and rtprtxreceive generally work
but also how the association algorithm is implemented.
2014-01-03 20:46:14 +01:00
Reynaldo H. Verdejo Pinochet
0e159e3b03 souphttpsrc: use status code macro instead of 407
Rest of the code is using the _PROXY_AUTHENTICATION_REQUIRED
macro too. Easier to understand if you don't recall HTTP
error codes by heart.
2014-01-03 14:15:59 -03:00
Reynaldo H. Verdejo Pinochet
ac7d346355 shout2send: change audio_format field to format
This element and the underlying libshout2 library
can handle video media files too. The code already
handles video/webm so the name gets confusing. Also
add and use DEFAULT_FORMAT macro Instead of hardwiring
SHOUT_FORMAT_VORBIS at init

https://bugzilla.gnome.org/show_bug.cgi?id=721342
2014-01-03 14:15:59 -03:00
Reynaldo H. Verdejo Pinochet
667c803730 shout2send: clarify meaning of the URL prop
https://bugzilla.gnome.org/show_bug.cgi?id=721342
2014-01-03 14:15:59 -03:00
Reynaldo H. Verdejo Pinochet
e6321ecb74 shout2send: docs, add a sample pipeline
And finish adding shout2send to the docs while at it

https://bugzilla.gnome.org/show_bug.cgi?id=721342
2014-01-03 14:15:59 -03:00
Reynaldo H. Verdejo Pinochet
4182f42f7b gdkpixbufoverlay: remove spurious @see_also 2014-01-03 14:15:59 -03:00
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
Sebastian Rasmussen
3f8b423516 v4l2: Handle v4l2_ioctl() errors even in error handling
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721268
2014-01-03 10:59:57 +01:00
Jeremy Huddleston Sequoia
2bc631bcd0 osxvideo: unifdef -DRUN_NS_APP_THREAD 2014-01-02 10:01:54 +01:00
Jeremy Huddleston Sequoia
6fe2115d77 osxvideo: Assume SDK and deployment target are at least Snow Leopard 2014-01-02 10:01:28 +01:00
Jeremy Huddleston Sequoia
aeb3fa72b2 configure: Disable osxvideo on Leopard and earlier
This also moves the "other platforms" check in OS X video to before the
variable is read

https://bugzilla.gnome.org/show_bug.cgi?id=721245
2014-01-02 09:56:14 +01:00
Wim Taymans
c83ed4f61e tests: add AUX receiver unit test 2013-12-31 15:08:49 +01:00
Wim Taymans
b91e0096b7 tests: improve rtpbin test 2013-12-31 15:08:49 +01:00
Wim Taymans
bb2d37b11d rtpbin: add some docs about AUX elements 2013-12-31 15:08:49 +01:00
Wim Taymans
3e83e6a33d tests: add AUX sender unit test 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
841f9ad050 tests: add decoder test 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
Sebastian Dröge
5b1c0a4cfd osx: Make OSX version checks more consistent
And especially also consider update versions, e.g. 10.5 with updates
will be 1051 or similar and thus bigger than MAC_OS_X_VERSION_10_5 but
still won't have the API we want to use.
2013-12-31 10:11:24 +01:00
Jeremy Huddleston
3d163680f0 osxvideosink: Fix build on updated OS X Leopard
https://bugzilla.gnome.org/show_bug.cgi?id=721245
2013-12-31 10:11:24 +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