17 Commits

Author SHA1 Message Date
pbos@webrtc.org
dc3fa08331 Remove include_dirs from rtp_rtcp.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4851 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 09:17:13 +00:00
stefan@webrtc.org
7bb8f02274 Adds support for combining RTX and FEC/RED.
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.

Enables retransmissions over RTX by default in the loopback test.

BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:40:11 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:38:54 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
stefan@webrtc.org
f56d612c70 Create gyp target for bwe components.
R=henrikg@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1775004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4311 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-09 12:32:35 +00:00
stefan@webrtc.org
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 14:30:48 +00:00
stefan@webrtc.org
a5cb98cbbd Breaking out RTP header parsing from the RTP module.
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.

Moving bandwidth estimation before the RTP module is also required for RTX.

TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1545004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 12:12:51 +00:00
pbos@webrtc.org
6e788df19e Remove vim/emacs modelines from .gypi files
BUG=1655

Review URL: https://webrtc-codereview.appspot.com/1326005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-16 12:40:34 +00:00
pbos@webrtc.org
8911ce46a4 Generic video-codec support.
Labels frames as key/delta, also marks the first RTP packet of a frame as such,
to allow proper reconstruction even if packets are received out of order.

BUG=1442
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-18 16:39:03 +00:00
andrew@webrtc.org
63e0964039 Fix webrtc compilation errors for Chrome Win64
Mostly disabling warnings in the gyp files.

BUG=1348
BUG=http://crbug.com/166496
BUG=http://crbug.com/167187

Review URL: https://webrtc-codereview.appspot.com/1063011
Patch from Justin Schuh <jschuh@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3423 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-29 06:45:22 +00:00
wjia@webrtc.org
a3c82bf667 Remove '<(library)' in gyp files.
This will remove all usage of '<(library)' in all webrtc gyp files. 
Review URL: https://webrtc-codereview.appspot.com/1049005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18 23:42:21 +00:00
phoglund@webrtc.org
efae5d5901 Extracted rtp receiver payload management to its own class, made video receiver depend on that instead.
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine.

BUG=
TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1022011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 16:10:45 +00:00
phoglund@webrtc.org
c38eef896a Reformatted RTPReceiver.
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)

BUG=
TEST=Trybots, vie_ & voe_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/998007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 10:18:30 +00:00
phoglund@webrtc.org
61f39a3174 Fixed bad header name.
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1001008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3307 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-18 16:02:13 +00:00
phoglund@webrtc.org
07bf43c673 Replaced the _audio parameter with a strategy.
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.

In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.

BUG=
TEST=vie/voe_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1001006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-18 15:40:53 +00:00
pwestin@webrtc.org
571a1c035b Enable paced sender.
Review URL: https://webrtc-codereview.appspot.com/965016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-13 21:12:39 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00