|
|
34c5da6b5e
|
Cleaned up logging in video_coding.
Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.
BUG=3153
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-04-11 14:08:35 +00:00 |
|
|
|
71f055fb41
|
Add send frame rate statistics callback
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-12-04 15:09:27 +00:00 |
|
|
|
f1e807c0e5
|
Removing FrameForStorage
R=pwestin@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2142004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-09-05 22:34:41 +00:00 |
|
|
|
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 |
|
|
|
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 |
|
|
|
a4407329d4
|
Include files from webrtc/.. paths in video_coding/.
BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1783006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-07-16 12:32:05 +00:00 |
|
|
|
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 |
|
|
|
7b859cc1e9
|
Webrtc_Word32 => int32_t in video_coding/main/
BUG=
Review URL: https://webrtc-codereview.appspot.com/1279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-04-02 15:54:38 +00:00 |
|
|
|
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 |
|
|
|
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 |
|
