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25 Commits

Author SHA1 Message Date
wu@webrtc.org
6c75c98964 Propagate capture ntp timestamp from rtp to renderer. 
Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl.

TEST=new tests and try bots
BUG=3111
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 17:46:33 +00:00
solenberg@webrtc.org
b1f5010075 VoE changes to allow forwarding of packets from VoE to ViE BWE. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 10:38:25 +00:00
wu@webrtc.org
ebdb0e3ad0 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. 
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.

This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.

BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 23:49:08 +00:00
pbos@webrtc.org
5ab756703e Revert r5294 to re-roll r5293. 
To fix races in test each stream now owns its own encoder/decoder.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 12:24:44 +00:00
turaj@webrtc.org
41e2615e02 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." 
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
> 
> BUG=
> R=mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5409004

TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-15 18:42:32 +00:00
solenberg@webrtc.org
341e91441a Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 23:57:54 +00:00
pbos@webrtc.org
57eb858698 Remove ".." from include_dirs in build/common. 
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2332004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 10:20:27 +00:00
xians@webrtc.org
c94abd313e Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 18:15:09 +00:00
xians@webrtc.org
0729460acb Added a "interleaved_" flag to webrtc::AudioFrame. 
And also did some format refactoring on the AudioFrame class, no change on the functionalities on those format refactoring code.

BUG=
TEST=compile
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5032 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 12:50:46 +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
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents. 
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 16:22:53 +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
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
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +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
solenberg@webrtc.org
7ebbea14a9 Add handling of the absolute send time header extension to the rtp_rtcp module. 
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1480004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 11:10:31 +00:00
stefan@webrtc.org
7bc465bd21 Fix issues with incorrect wrap checks when having big buffers and high bitrate. 
Introduces shared functions for timestamp and sequence number wrap checks.

BUG=1607
TESTS=trybots

Review URL: https://webrtc-codereview.appspot.com/1291005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 17:48:02 +00:00
pbos@webrtc.org
ab9202b673 Removing remaining WebRtc_Word32 not in typedefs.h 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1306006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 17:59:17 +00:00
stefan@webrtc.org
03e3117d87 Removed redundant VP8 width/height and made sure the generic width/height is set. 
Review URL: https://webrtc-codereview.appspot.com/1158005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 09:59:27 +00:00
fischman@webrtc.org
aea96d36e3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver 
to avoid ODR violations with peerconnectioninterface.h in libjingle.

Conflict introduced in
https://webrtc-codereview.appspot.com/1060005/diff/14010/webrtc/modules/interface/module_common_types.h#newcode326

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1105011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-19 22:09:36 +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
andrew@webrtc.org
ae1a58bba4 Replace AudioFrame's operator= with CopyFrom(). 
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders.

Review URL: https://webrtc-codereview.appspot.com/1031007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 04:44:30 +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
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