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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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3c9151b9532b372a35f0b81cc7c6b34d820506e1
3c9151b
Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )
by mbonadei
· 8 years ago
a62a82b
Creating libwebrtc bundle jar
by mbonadei
· 8 years ago
fefe076
RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector.
by hbos
· 8 years ago
2d4d653
Fix msan flake in rtcstats_integrationtest.cc.
by hbos
· 8 years ago
c854ac3
Stop camera onStop instead of onPause.
by sakal
· 8 years ago
42f6d2f
RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector.
by hbos
· 8 years ago
7319f26
Roll chromium_revision 780d18a4ff..7649e76842 (445004:445027)
by buildbot
· 8 years ago
30fe5e0
Prevent downstream linter warnings.
by sakal
· 8 years ago
3556406
Camera1Session: Fix camera sometimes getting stopped twice.
by sakal
· 8 years ago
9e30274
RTCMediaStreamTrackStats collected on a per-attachment basis.
by hbos
· 8 years ago
fd6c94d
Allow more config changes for CallActivity.
by sakal
· 8 years ago
3e92290
Load library dependencies in AppRTCMobile.
by sakal
· 8 years ago
be850e1
Clear out cached codecs when calculating new codec lists.
by noahric
· 8 years ago
204030a
Roll chromium_revision bdeae63b37..780d18a4ff (444971:445004)
by buildbot
· 8 years ago
a01d2f5
Roll chromium_revision 34215edf2e..bdeae63b37 (444898:444971)
by buildbot
· 8 years ago
888874f
Allow GCC 4.9 to compile Chromium
by floppymaster
· 8 years ago
8944ab3
Roll chromium_revision 1a7fcf6220..34215edf2e (444851:444898)
by buildbot
· 8 years ago
b2cdd93
Remove the dependency of TransportChannel and TransportChannelImpl.
by zhihuang
· 8 years ago
9d643e8
Roll chromium_revision 113278e435..1a7fcf6220 (444801:444851)
by buildbot
· 8 years ago
537798b
Roll chromium_revision d50ce8a895..113278e435 (444743:444801)
by buildbot
· 8 years ago
9410b51
GN: Add audio_conference_mixer_unittests to modules_unittests.
by ehmaldonado
· 8 years ago
d748863
Fix PseudoTcp to handle incoming packets with invalid SEQ field
by sergeyu
· 8 years ago
3cd896c
Roll chromium_revision 6d8c754784..d50ce8a895 (444712:444743)
by buildbot
· 8 years ago
eef94d9
Video collected by VideoFileRenderer is first saved on the native heap, then saved to disk during release.
by mandermo
· 8 years ago
3626865
GN: Refactor modules_unittests to eliminate package boundary violations.
by ehmaldonado
· 8 years ago
d32bf75
Pass SdpAudioFormat through Channel, without converting to CodecInst
by kwiberg
· 8 years ago
093dac1
Roll chromium_revision 0f65d3f753..6d8c754784 (444698:444712)
by buildbot
· 8 years ago
b935984
Revert of Move congestion controller processing to the pacer thread. (patchset #5 id:80001 of https://codereview.webrtc.org/2637783003/ )
by nisse
· 8 years ago
6ce9259
Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ )
by zhihuang
· 8 years ago
daeffb2
Roll chromium_revision be0566f991..0f65d3f753 (444668:444698)
by buildbot
· 8 years ago
5aed06c
make the DtlsTransportWrapper inherit form DtlsTransportInternal
by zhihuang
· 8 years ago
04926b8
Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
by kjellander
· 8 years ago
f847178
Roll chromium_revision adc103db18..be0566f991 (444630:444668)
by buildbot
· 8 years ago
4b96466
Roll chromium_revision e9762587b9..adc103db18 (444575:444630)
by buildbot
· 8 years ago
f15825f
Roll chromium_revision 0bc260f9e8..e9762587b9 (444497:444575)
by buildbot
· 8 years ago
3078b55
Reduce the log verbosity in sslstreamadapter_unittest
by skvlad
· 8 years ago
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
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