minyue@webrtc.org
|
c1a40a7b68
|
This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate.
This CL is going to be combined with another CL in ACM, which is to be landed.
TEST=passed_try_bots
BUG=
R=stefan@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6262 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-05-28 09:52:06 +00:00 |
|
solenberg@webrtc.org
|
a07923339b
|
Remove external encryption API for VoE.
BUG=
R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-02-18 11:27:22 +00:00 |
|
pbos@webrtc.org
|
9aca5b34e1
|
Remove #pragma once
BUG=1830
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1568004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4130 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-05-29 13:19:09 +00:00 |
|
pbos@webrtc.org
|
956aa7e087
|
Include files from webrtc/.. paths in voice_engine/
BUG=1662
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1434005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-05-21 13:52:32 +00:00 |
|
pbos@webrtc.org
|
9213521ea9
|
Remove const for plain data types in voice_engine/
BUG=1644
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1463004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-05-14 08:31:39 +00:00 |
|
pbos@webrtc.org
|
6141e13873
|
WebRtc_Word32 -> int32_t in voice_engine/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1305004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-04-09 10:09:10 +00:00 |
|
solenberg@webrtc.org
|
a442d4d983
|
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
Today I had to figure out this code was legacy. Now next person doesn't have to.
BUG=
Review URL: https://webrtc-codereview.appspot.com/1247004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2013-03-28 09:14:36 +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 |
|