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gerrit-public.fairphone.software
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platform
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external
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webrtc
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d1c0998730443e4901ec329ee816fd1f115b6223
d1c0998
Adding OrtcFactory, and changing UdpTransport to match current plan.
by deadbeef
· 8 years ago
27edfbc
Roll chromium_revision 10fecf4ab1..0bc260f9e8 (444374:444497)
by buildbot
· 8 years ago
a3c8c90
Roll chromium_revision d9e076c478..10fecf4ab1 (444338:444374)
by buildbot
· 8 years ago
d99a200
Adding some features to proxy.h, and restructuring the macros.
by deadbeef
· 8 years ago
c8ee882
Replace use of ASSERT in test code.
by nisse
· 8 years ago
f20dd00
Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
by philipel
· 8 years ago
6da303d
Reland of Delete rtc::linked_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2579753002/ )
by nisse
· 8 years ago
1b3ce86
Roll chromium_revision 5e5d50d1fe..d9e076c478 (444317:444338)
by buildbot
· 8 years ago
fcc6006
Clear the FrameBuffer in case of a backward jump in the picture id.
by philipel
· 8 years ago
44303ea
Revert of Add experimental simulcast screen content mode (patchset #5 id:80001 of https://codereview.webrtc.org/2636443002/ )
by sprang
· 8 years ago
e8abe3e
Revert of New method StatsObserver::OnCompleteReports, passing ownership. (patchset #2 id:20001 of https://codereview.webrtc.org/2584553002/ )
by nisse
· 8 years ago
2f67b82
Fixing peerconnection reddish video issue
by mbonadei
· 8 years ago
5850a94
Add failure type parameter to onFailure callback.
by sakal
· 8 years ago
2fcd2dd
Update YuvConverter to use GlTextureFrameBuffer.
by sakal
· 8 years ago
a77ce78
Roll chromium_revision 5d804c8487..5e5d50d1fe (444296:444317)
by buildbot
· 8 years ago
4a0c764
Add rtcp::TransportFeedback::GetReceivedPackets()
by danilchap
· 8 years ago
e0e3bdf
Refactor OveruseFrameDetector to use timing in us units
by nisse
· 8 years ago
d3fabe5
Improve computational performance of BWE by switching list to deque.
by terelius
· 8 years ago
1d2d789
Fix race in EndToEndTest.ReceivesFlexfecAndSendsCorrespondingRtcp.
by brandtr
· 8 years ago
a28e971
Add experimental simulcast screen content mode
by sprang
· 8 years ago
b3dc2b7
Move congestion controller processing to the pacer thread.
by nisse
· 8 years ago
c0370ef
Roll chromium_revision 9057f45850..5d804c8487 (444268:444296)
by buildbot
· 8 years ago
f3d5d89
Roll chromium_revision aa99b0c20f..9057f45850 (444210:444268)
by buildbot
· 8 years ago
c7953fa
Remove the IceTransportInternal2.
by zhihuang
· 8 years ago
bad5dad
More minor improvements to BaseChannel/transport code.
by deadbeef
· 8 years ago
b308b03
Roll chromium_revision b09bc11cad..aa99b0c20f (444155:444210)
by buildbot
· 8 years ago
7fa4a72
Increase bitrate adjustment values for VP8 Exynos encoder
by Alex Glaznev
· 8 years ago
6dbbd89
Fix BitrateProber to match the requested bitrate more precisely
by sergeyu
· 8 years ago
f7303fc
Roll chromium_revision 71ee072729..b09bc11cad (444100:444155)
by buildbot
· 8 years ago
2e37a72
Roll chromium_revision 5e608bb301..71ee072729 (444065:444100)
by buildbot
· 8 years ago
9af18bf
Roll chromium_revision e44e863e19..5e608bb301 (444017:444065)
by buildbot
· 8 years ago
e5cbc20
Android: AppRTCMobile: Don't leak CallActivity.
by sakal
· 8 years ago
61c98e0
Remove dependency to Chromium code from WebRTC Java code.
by sakal
· 8 years ago
e08b253
Remove unused lambda capture to unbreak downstream code.
by solenberg
· 8 years ago
0b2d3e2
Revert of Fix flaky WebRtcVideoChannel2BaseTest.GetStats T (patchset #1 id:1 of https://codereview.webrtc.org/2634273002/ )
by perkj
· 8 years ago
2013e29
Disable automatic scaling in tests.
by nisse
· 8 years ago
311a64c
Fix flaky WebRtcVideoChannel2BaseTest.GetStats T
by perkj
· 8 years ago
c08c191
Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
by philipel
· 8 years ago
6c0fd43
Roll chromium_revision c1fcfd706a..e44e863e19 (443985:444017)
by buildbot
· 8 years ago
e8aca24
Move file capturer/renderer tests to the correct location.
by sakal
· 8 years ago
0f0763d
Make the new jitter buffer the default jitter buffer.
by philipel
· 8 years ago
804ab6f
Parse MedianSlopeFilter-parameters to the correct variables.
by terelius
· 8 years ago
7456817
Comparison of videos with reference frame not starting from zero
by mandermo
· 8 years ago
160e4a7
RTCMediaStreamTrackStats.kind added and collected.
by hbos
· 8 years ago
9b96a17
Android GlTextureFrameBuffer: Re-attach texture in setSize
by magjed
· 8 years ago
1fd08c1
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.
by ehmaldonado
· 8 years ago
ece0571
UdpTransport:IsIpAddressValid: Added extra :: check for ipv6
by ossu
· 8 years ago
da5e9d0
Initiate mid-call probing even if estimated bitrate is at max configured bitrate.
by philipel
· 8 years ago
fa5a368
Let FlexfecReceiveStreamImpl send RTCP RRs.
by brandtr
· 8 years ago
9506e12
Reset pendingCameraSwitch as false after failed to post switchCameraOnThread to camera thread.
by hankzhang8945
· 8 years ago
f4caaab
Fix for bwe with overhead on audio only calls.
by michaelt
· 8 years ago
04a057b
Add missing if-clause for residual_echo_likelihood_recent_max
by henrik.lundin
· 8 years ago
d3c3a4e
Roll chromium_revision 9eb76629f8..c1fcfd706a (443880:443985)
by buildbot
· 8 years ago
c117e2e
Do not classify error after stopping the camera as a startup failure.
by sakal
· 8 years ago
2df0734
Add WebRTC.BWE.MidCallProbing.* metrics.
by philipel
· 8 years ago
00c7ad1
Roll chromium_revision 07adb5c7a6..9eb76629f8 (443862:443880)
by buildbot
· 8 years ago
7064d59
RTCTransportStats.dtlsState replaces .activeConnection
by hbos
· 8 years ago
3d200bd
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
by brandtr
· 8 years ago
57c2fff
Periodically update channel parameters and send TargetBitrate message.
by sprang
· 8 years ago
84abeb1
RTC[In/Out]boundRTPStreamStats.mediaTrackId collected.
by hbos
· 8 years ago
93f16d7
delete redundant members in ViEEncoder
by kthelgason
· 8 years ago
e78d266
Make FakeEncoder and FakeH264Encoder thread safe.
by brandtr
· 8 years ago
037b93a
Replace default locale with US locale on Android.
by sakal
· 8 years ago
6deecb2
Refactor TransportFeedback ensuring it's consistency:
by danilchap
· 8 years ago
1f8239c
TrackMediaInfoMap added.
by hbos
· 8 years ago
be02dcd
Roll chromium_revision 4bb402f2c3..07adb5c7a6 (443850:443862)
by buildbot
· 8 years ago
a6069e8
Espresso test case to control loopback call
by mandermo
· 8 years ago
394d460
Delete unused file faketaskrunner.h.
by nisse
· 8 years ago
97b7d5f
Revert of Remove the IceTransportInternal2. (patchset #1 id:1 of https://codereview.webrtc.org/2632563002/ )
by henrikg
· 8 years ago
d04b537
Roll chromium_revision b295b8bfea..4bb402f2c3 (443842:443850)
by buildbot
· 8 years ago
92d2635
Roll chromium_revision d097cd3bbd..b295b8bfea (443824:443842)
by buildbot
· 8 years ago
8aaa511
Remove the IceTransportInternal2.
by zhihuang
· 8 years ago
ab2e044
Roll chromium_revision 95ff5a1ca8..d097cd3bbd (443817:443824)
by buildbot
· 8 years ago
e2173df
Roll chromium_revision 65dfcc360f..95ff5a1ca8 (443816:443817)
by buildbot
· 8 years ago
f0ba1bf
Roll chromium_revision 679dbea4a9..65dfcc360f (443815:443816)
by buildbot
· 8 years ago
4e477a1
Added a new echo likelihood stat that reports the maximum value from a previous time period.
by ivoc
· 8 years ago
cc1e1aa
Roll chromium_revision 87b1dd36c3..679dbea4a9 (443814:443815)
by buildbot
· 8 years ago
5b7747a
Roll chromium_revision 15d49484f9..87b1dd36c3 (443812:443814)
by buildbot
· 8 years ago
5561b4c
Roll chromium_revision b4dbf044c5..15d49484f9 (443809:443812)
by buildbot
· 8 years ago
b42108a
Roll chromium_revision 6c965b0a6c..b4dbf044c5 (443806:443809)
by buildbot
· 8 years ago
fd50d12
Roll chromium_revision 7c0e0135f9..6c965b0a6c (443802:443806)
by buildbot
· 8 years ago
fd1fd60e3
Roll chromium_revision 5f9b3d7676..7c0e0135f9 (443800:443802)
by buildbot
· 8 years ago
1e951ed
Roll chromium_revision 0f3c36d035..5f9b3d7676 (443798:443800)
by buildbot
· 8 years ago
c825a39
Roll chromium_revision 94af7a6d02..0f3c36d035 (443797:443798)
by buildbot
· 8 years ago
510c2ea
Roll chromium_revision 8c47492479..94af7a6d02 (443796:443797)
by buildbot
· 8 years ago
06111a5
Roll chromium_revision 63ec3598c6..8c47492479 (443793:443796)
by buildbot
· 8 years ago
413a1ad
Roll chromium_revision 5671bf116d..63ec3598c6 (443781:443793)
by buildbot
· 8 years ago
e46df31
Roll chromium_revision 9bf0adfb29..5671bf116d (443744:443781)
by buildbot
· 8 years ago
e1f8829
Roll chromium_revision 0bfeb6b121..9bf0adfb29 (443685:443744)
by buildbot
· 8 years ago
f4d4544
Roll chromium_revision 2225560840..0bfeb6b121 (443627:443685)
by buildbot
· 8 years ago
9c342a3
Roll chromium_revision 9d7ae4d5eb..2225560840 (443576:443627)
by buildbot
· 8 years ago
fe0fd41
Stop using deprecated constraints-based version of CreateAudioSource.
by deadbeef
· 8 years ago
8e814d7
Provide better message for when RTCP mux "require" policy is triggered.
by deadbeef
· 8 years ago
482c539
Roll chromium_revision 27032ce525..9d7ae4d5eb (443521:443576)
by buildbot
· 8 years ago
bc5d921
Rename base/analytics/ to base/numerics/
by terelius
· 8 years ago
8313a6f
Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.
by brandtr
· 8 years ago
44b3ef6
Signal target bitrate only for screenshare streams
by sprang
· 8 years ago
36e7d70
Explicitly only add transport-cc RTCP feedback param to default FlexFEC codec.
by brandtr
· 8 years ago
bf08452
Remove use of PCH files in webrtc/sdk/objc
by magjed
· 8 years ago
a9dd4a1
Replace left-over ASSERTs in httpcommon.h and webrtcsession.cc.
by nisse
· 8 years ago
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