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21 Commits

Author SHA1 Message Date
phoglund@webrtc.org
a22a9bd9ca Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional. 
The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.

BUG=
TEST=vie & voe_auto_test full runs

Review URL: https://webrtc-codereview.appspot.com/1014006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-14 10:01:55 +00:00
andrew@webrtc.org
f908011eb4 Remove extra line. 
TBR=elham

Review URL: https://webrtc-codereview.appspot.com/1024008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-11 22:39:55 +00:00
mflodman@webrtc.org
2f225cadde Add logs when no RTCP RR has been received for three regular RTCP intervals. 
BUG=1267
TEST=Unittest added.

Review URL: https://webrtc-codereview.appspot.com/1019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 13:54:43 +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
pwestin@webrtc.org
1b6da28047 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. 
Landing of 573005 On behalf of an1kumar@gmail.com

TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1002008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-21 17:46:24 +00:00
phoglund@webrtc.org
ad0ed582b5 Fixed a missed initialization (found by valgrind FYI bot). 
http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/327/steps/memory%20test%3A%20memcheck_voe_auto_test/logs/stdio

BUG=
TEST=Reproduced valgrind error, verified gone after fixing.

Review URL: https://webrtc-codereview.appspot.com/1008005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 09:14:36 +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
phoglund@webrtc.org
7659d914bb Decoupled video rtp receiver from rtp receiver. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/995005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 09:57:37 +00:00
phoglund@webrtc.org
92bb417cb1 Decoupled RTP audio processor from RTP receiver. 
BUG=
TEST=Ran vie_auto_test, rtp_rtcp_unittests, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-13 10:48:24 +00:00
stefan@webrtc.org
8d0cd07d0c Add test to verify that padding only frames are passing through the RTP module. 
Review URL: https://webrtc-codereview.appspot.com/934023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 14:01:46 +00:00
phoglund@webrtc.org
ef90c3227e Will now correctly identify the first-ever received packet as the first packet in its frame. 
We used to flag the _second_ packet in the first frame as the first. Subsequent frames worked as intended.

BUG=1103
TEST=vie_auto_test --automated, rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/964020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 16:30:40 +00:00
mflodman@webrtc.org
7c894b7cc7 Wire up CallStats to provide modules with correct RTT. 
BUG=769
TEST=Manual test since there is no ViE APi to get RTT for receive channels.

Review URL: https://webrtc-codereview.appspot.com/937027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 12:40:15 +00:00
andrew@webrtc.org
418443c531 Remove operator overloading from RTPFragmentationHeader. 
Instead supply a CopyFrom() method.

TEST=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/972004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-23 19:17:23 +00:00
mflodman@webrtc.org
1c61196095 Removed not used include. 
TEST=Compiles.

Review URL: https://webrtc-codereview.appspot.com/966025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-22 09:37:27 +00:00
stefan@webrtc.org
4100b0402e Move SSRC list to RemoteBitrateEstimator. 
BUG=1105

Review URL: https://webrtc-codereview.appspot.com/965027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-19 10:09:20 +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
asapersson@webrtc.org
1726661ca2 Update parsed non ref frame info. 
Review URL: https://webrtc-codereview.appspot.com/932015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 13:39:51 +00:00
pwestin@webrtc.org
c66e8b3f31 pre-factor cleanup pre-work. 
Review URL: https://webrtc-codereview.appspot.com/938010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 17:01:04 +00:00
asapersson@webrtc.org
e5b49a0472 Update timestamp offset for re-transmitted packets. 
BUG=1059
Review URL: https://webrtc-codereview.appspot.com/930011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-06 13:09:39 +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