admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
4,053 Commits 1 Branch 0 Tags
Commit Graph

21 Commits

Author SHA1 Message Date
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
mflodman@webrtc.org
6879c8adad Hooking up first simple CPU adaptation version. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1767004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 11:35:00 +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
mflodman@webrtc.org
1c986e7c89 Removed ViE file API. 
R=asapersson@webrtc.org, niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1723004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 09:12:49 +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
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
andresp@webrtc.org
7707d060bb Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1450008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 10:50:50 +00:00
pwestin@webrtc.org
52b4e8871a Adding trace and changing pacing constants 
BUG=1721,1722
R=mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1380005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 19:02:17 +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
stefan@webrtc.org
bfacda60be Add interface to signal a network down event. 
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:36:01 +00:00
mikhal@webrtc.org
ef9f76a59d Adding a receive side API for buffering mode. 
At the same time, renaming the send side API.

Review URL: https://webrtc-codereview.appspot.com/1104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 23:22:18 +00:00
mikhal@webrtc.org
3d305c64b4 Updates to send side streaming mode: 
1. Disabling frame-droppers from the vie encoder and not the channel.
2. Accounting for qpMax in the VP8 wrapper.

Review URL: https://webrtc-codereview.appspot.com/1101007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-10 18:42:55 +00:00
stefan@webrtc.org
fcd8585874 Enable external encoders with internal picture source. 
CL enables registering of external encoder with internal picture
source on API by adding simple passthrough parameter that is already
supported within video engine.

Review URL: https://webrtc-codereview.appspot.com/1006006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 08:35:40 +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
mflodman@webrtc.org
8dde197788 Only remove encoder state feedback for send channels. 
BUG=1000
TEST=See bug

Review URL: https://webrtc-codereview.appspot.com/938004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-25 13:06:22 +00:00
mflodman@webrtc.org
d6ec386ff5 Revert the revert in r2988 since that wasn't the issue. 
Review URL: https://webrtc-codereview.appspot.com/931005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-25 11:30:29 +00:00
vikasmarwaha@webrtc.org
8239ca5096 Reverse Merged r2884 & r2888 from trunk. 
Review URL: https://webrtc-codereview.appspot.com/929005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-24 22:35:52 +00:00
mikhal@webrtc.org
9fedff7c17 Switching to I420VideoFrame 
Review URL: https://webrtc-codereview.appspot.com/922004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-24 18:33:04 +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