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15 Commits

Author SHA1 Message Date
stefan@webrtc.org
de98478965 Update the remote bitrate estimator before passing the packet to the RTP module. 
This solves the problem of reconstructed packets biasing the bandwidth estimate.

TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1594005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 12:15:40 +00:00
stefan@webrtc.org
08994cc525 Fix a return value mismatch introduced in r4129. 
TBR=mflodman@webrtc.org
TEST=vie_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1584005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 13:28:21 +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
mflodman@webrtc.org
a066cbf37c Don't return an estimated receive BW for channels not receiving video. 
BUG=1834
TEST=ViE RTP autotest
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1572004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 15:00:15 +00:00
solenberg@webrtc.org
561990fd73 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. 
- Changed RemoteBitrateObserver::OnReceivedBitrateChanged() to use a const & instead of non-const *, to avoid unnecessary copying.
- Refactored RemoteBitrateEstimatorTest so it can be instantiated for both single and multi stream BWE (first using a parameterized test, but then as a standard test fixture and a few helper functions).
- Refactored some tests in RemoteBitrateEstimatorTest into a common function CapacityDropTestHelper().

BUG=
R=andresp@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1521004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4086 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 19:04:19 +00:00
pbos@webrtc.org
f5d4cb1958 Include files from webrtc/.. paths in video_engine/ 
BUG=1662
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1492004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 13:44:48 +00:00
pbos@webrtc.org
b238d1210b WebRtc_Word32 -> int32_t in video_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:41:51 +00:00
pwestin@webrtc.org
82dcc9ff11 Remove UDP transport API from ViE 
Review URL: https://webrtc-codereview.appspot.com/1232004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 20:37:14 +00:00
pwestin@webrtc.org
684f0577fb Revert r3667 and r3665 
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. 
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 17:52:42 +00:00
mflodman@webrtc.org
4fd5527ab1 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid 
estimate.

BUG=1377

Review URL: https://webrtc-codereview.appspot.com/1095005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-06 17:46:39 +00:00
stefan@webrtc.org
b586507986 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. 
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.

BUG=1298

Review URL: https://webrtc-codereview.appspot.com/1060005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 14:33:42 +00:00
phoglund@webrtc.org
4cebe6cded Made TickTime immutable, rewrote tick utils to be fakeable. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/798004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 13:37:19 +00:00
stefan@webrtc.org
1a2a6dda26 Fixes a bitrate mismatch between sender and receiver. 
TEST=trybots

BUG=

Review URL: https://webrtc-codereview.appspot.com/928014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-31 12:21:13 +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