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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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822d2586fbd6af3ee421b32b86e076088283c6b5
822d258
Move webrtc/build/android -> tools-webrtc/android
by mbonadei
· 8 years ago
81eab61
Count FlexFEC packets in |fec_bitrate_| in RTPSenderVideo.
by brandtr
· 8 years ago
0608ffd
Roll chromium_revision 59592eaa98..319b885718 (445345:445689)
by buildbot
· 8 years ago
365aebd
Make CongestionController::remote_bitrate_estimator_ a non-pointer.
by nisse
· 8 years ago
d2b092f
Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
by johan
· 8 years ago
15389c0
Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor.
by nisse
· 8 years ago
568c9e7
New simulators to test BWE at low bitrates (15-50kbps range).
by terelius
· 8 years ago
a4a7538
Android: Script for building libwebrtc.aar.
by sakal
· 8 years ago
e04064d
Revert of Delete unused class/template ScopedMessageData. (patchset #1 id:1 of https://codereview.webrtc.org/2652663002/ )
by aleloi
· 8 years ago
dc2b3f3
Delete unused class CompositeMediaEngineWithFakeVoiceEngine.
by nisse
· 8 years ago
d83fb92
Delete unused class/template ScopedMessageData.
by nisse
· 8 years ago
c23b0b2
Delete unused classes DesktopId and ScreencastEventCatcher.
by nisse
· 8 years ago
ad45228
Moving get_landmines.py (build/ -> tools-webrtc/)
by mbonadei
· 8 years ago
2b75526
Add linux_memcheck as default trybot.
by Henrik Kjellander
· 8 years ago
914d49d
Revert of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. (patchset #3 id:40001 of https://codereview.webrtc.org/2638933002/ )
by kjellander
· 8 years ago
1b54a5f
Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL.
by deadbeef
· 8 years ago
4c78702
iOS: Add MedianSlopeFilter field trial.
by tkchin
· 8 years ago
5c4f24a
Move implmentation specific constants out of rtp_header_extension.h
by danilchap
· 8 years ago
f53d737
H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
by johan
· 8 years ago
e1405ad
Removed double-special-casing of ISAC in libjingle and WebRtcVoE.
by ossu
· 8 years ago
cb893ee
Removing unused code from webrtc/build
by mbonadei
· 8 years ago
1bed2e4
video_loopback: fall back to fake capturer if we can't open camera.
by sprang
· 8 years ago
435ddf9
Add TransportFeedbackPacketLossTracker.
by minyue
· 8 years ago
ed582f7
Script to start stubbed loopback video test with Espresso
by mandermo
· 8 years ago
0ebdf27
Delete or update left-over ASSERT use and comments.
by nisse
· 8 years ago
da25006
Fixed public_deps for libjingle_peerconnection{,_api}
by ossu
· 8 years ago
50cfe1f
RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector.
by hbos
· 8 years ago
9c3d4c4
Stop leaking FlexfecReceiveStream objects after call shutdown.
by brandtr
· 8 years ago
a067013
Minor style change suggested by internal static analysis tool.
by aleloi
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
f49ff26
GN: Make audio_processing_unittests compile with rtc_enable_protobuf=false
by ehmaldonado
· 8 years ago
fd870db
Add metric for decode time and max decode time in video quality tests.
by philipel
· 8 years ago
0112403
Minor style change suggested by internal static analysis tool.
by aleloi
· 8 years ago
a31cdbc
Roll chromium_revision dcc5978539..59592eaa98 (445328:445345)
by buildbot
· 8 years ago
0b56279
Catch failure to load native dependencies.
by sakal
· 8 years ago
de8ca92
New script to count usage of C++ classes.
by nisse
· 8 years ago
b55bd97
Reland of Creating libwebrtc bundle jar (patchset #1 id:1 of https://codereview.webrtc.org/2640023010/ )
by mbonadei
· 8 years ago
5d0f2e8
Roll chromium_revision 269b6bc66e..dcc5978539 (445317:445328)
by buildbot
· 8 years ago
c152434
Roll chromium_revision 7649e76842..269b6bc66e (445027:445317)
by buildbot
· 8 years ago
3e4faae
Fixing memory leak in FakeTransportController.
by deadbeef
· 8 years ago
8662f94
Only set certificate on DTLS transport if fingerprint is found in SDP.
by deadbeef
· 8 years ago
2197e91
Remove dead code for GtkVideoRenderer.
by pbos
· 8 years ago
f33491e
Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ )
by deadbeef
· 8 years ago
eaa826c
Removing #defines previously used for building without BoringSSL/OpenSSL.
by deadbeef
· 8 years ago
cd3180c
PATENTS: fix reference
by philipp.hancke
· 8 years ago
7bcdb69
Ignore ufrag/password in "a=candidate" lines in SDP.
by deadbeef
· 8 years ago
0fc04b7
Finalize the support for building WebRTC library for iOS with bitcode
by VladimirTechMan
· 8 years ago
f64941f
RTCMediaStreamTrackStats.framesDecoded collected.
by hbos
· 8 years ago
aea1a01
Move webrtc/sdk/DEPS to webrtc/sdk/objc/DEPS
by magjed
· 8 years ago
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
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