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23 Commits

Author SHA1 Message Date
asapersson@webrtc.org
766154aa1d Removed unused code. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 08:35:50 +00:00
asapersson@webrtc.org
7d6bd22019 Propagate estimated RTT from receivers to rtt observer. 
BUG=1613
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 12:14:34 +00:00
stefan@webrtc.org
b2c8a952a7 Improving padding rules and breaking out bw allocation to ViEEncoder. 
BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 13:58:01 +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
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
hclam@chromium.org
2e402ce873 Enqueue packet in pacer if sending fails 
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.

BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1693004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:18:31 +00:00
stefan@webrtc.org
508a84b255 Wire up pacer-based padding. 
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.

Padding will for now only be generated by the first sending RTP module.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1612005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 12:53:37 +00:00
pbos@webrtc.org
a048d7cb0a Include files from webrtc/.. paths in rtp_rtcp/ 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1557004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 14:27:38 +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
77f6b2175e Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." 
> Revert 3933 "Remove traces of deprecated WebRtc_Word types."
> 
> > Remove traces of deprecated WebRtc_Word types.
> > 
> > BUG=314
> > R=tommi@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/1385004
> 
> TBR=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1386004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1397004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 12:02:11 +00:00
pbos@webrtc.org
68e5a68f07 Revert 3933 "Remove traces of deprecated WebRtc_Word types." 
> Remove traces of deprecated WebRtc_Word types.
> 
> BUG=314
> R=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1385004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1386004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 09:30:12 +00:00
pbos@webrtc.org
265a5d298a Remove traces of deprecated WebRtc_Word types. 
BUG=314
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1385004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 09:11:20 +00:00
mflodman@webrtc.org
9f5ebb5251 Adding a payload type for RTX. 
BUG=736
TEST=Modified RTP unittests.

Review URL: https://webrtc-codereview.appspot.com/1278004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 14:55:46 +00:00
stefan@webrtc.org
7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." 
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 14:56:29 +00:00
stefan@webrtc.org
afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. 
We should consider making the same change to the render timestamps generated at the receiver.

BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1283005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:37:40 +00:00
pbos@webrtc.org
2f44673d66 WebRtc_Word32 => int32_t for rtp_rtcp/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:08:41 +00:00
mikhal@webrtc.org
bda7f305c5 Adding RTX on source 
Review URL: https://webrtc-codereview.appspot.com/1190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-15 23:21:52 +00:00
turaj@webrtc.org
b7edd06530 Remove DTMF detection. Talk team has been in the loop and there is no need for 
DTMF detection at the receiver side.

test=voe_auto_test, VoE extended test DTMF
Review URL: https://webrtc-codereview.appspot.com/1168004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 22:27:27 +00:00
stefan@webrtc.org
becf9c897c Fix mismatch between different NACK list lengths and packet buffers. 
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.

BUG=1289

Review URL: https://webrtc-codereview.appspot.com/1065007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 15:09:57 +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
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