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gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
aa445e7b214e654e680158e78cba82376198ee4f
/
modules
/
rtp_rtcp
/
interface
/
rtp_rtcp_defines.h
aa445e7
Make RtpData and RtpFeedback destructors public.
by stefan@webrtc.org
· 11 years ago
d6da239
Added support for sending and receiving RTCP XR packets:
by asapersson@webrtc.org
· 11 years ago
7485573
Add support for multiple report blocks.
by stefan@webrtc.org
· 11 years ago
db74c61
Adds support for combining RTX and FEC/RED.
by stefan@webrtc.org
· 11 years ago
a20e2d4
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ..."""
by stefan@webrtc.org
· 11 years ago
c0976d2
Revert 4582 "Reverts a second set of reverts caused by a bug in ..."
by henrike@webrtc.org
· 11 years ago
efe1f0f
Reverts a second set of reverts caused by a bug in a dependency.
by stefan@webrtc.org
· 11 years ago
7fc75bb
Update talk to 50918584.
by wu@webrtc.org
· 11 years ago
9d71e28
Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
0ba496b
Revert r4301
by tnakamura@webrtc.org
· 11 years ago
b89eed3
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on
by elham@webrtc.org
· 11 years ago
46088d2
Support sending multiple report blocks and keeping track of statistics on several SSRCs.
by stefan@webrtc.org
· 11 years ago
a32d18f
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
by stefan@webrtc.org
· 11 years ago
3b89e10
Proper spacing for end-of-namespace comments.
by pbos@webrtc.org
· 11 years ago
20cfda6
Remove unused multi stream bandwidth estimator.
by solenberg@webrtc.org
· 11 years ago
cbd78ae
Include files from webrtc/.. paths in rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
a0b0025
Add handling of the absolute send time header extension to the rtp_rtcp module.
by solenberg@webrtc.org
· 11 years ago
7bc7e02
Adding a payload type for RTX.
by mflodman@webrtc.org
· 11 years ago
b57da65
WebRtc_Word32 => int32_t for rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
946d240
Adding RTX on source
by mikhal@webrtc.org
· 11 years ago
15a03fd
Remove DTMF detection. Talk team has been in the loop and there is no need for
by turaj@webrtc.org
· 11 years ago
9858fc8
Break out RtpClock to system_wrappers and make it more generic.
by stefan@webrtc.org
· 12 years ago
cd29867
Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.
by phoglund@webrtc.org
· 12 years ago
68950e5
Reformatted RTPReceiver.
by phoglund@webrtc.org
· 12 years ago
7d91c10
Replaced the _audio parameter with a strategy.
by phoglund@webrtc.org
· 12 years ago
e296783
Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled.
by mflodman@webrtc.org
· 12 years ago
b015cbe
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago