Commit graph

13717 commits

Author SHA1 Message Date
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
Stefan Sauer
d1223ebd10 wavparse: split the test
This way one failure won't shadow the other test and also if one fails we get
better disgnostics through the test-name.
2014-01-06 21:13:37 +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
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