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gerrit-public.fairphone.software
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webrtc
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0077272a476c38da24d2d5825bca038d289bcda6
0077272
Roll chromium_revision 4101b15..fbc4ecf (376664:376680)
by kjellander
· 9 years ago
da1d656
Roll chromium_revision 789f25d..4101b15 (376663:376664)
by kjellander
· 9 years ago
b9f943d
Roll chromium_revision 1120bd3..789f25d (376660:376663)
by kjellander
· 9 years ago
a18f638
Include "sharedexclusivelock.cc" in Chromium GN build.
by jbauch
· 9 years ago
fa830dc
Roll chromium_revision 9dc1788..1120bd3 (376655:376660)
by kjellander
· 9 years ago
bf81175
Roll chromium_revision 5618e25..9dc1788 (376642:376655)
by kjellander
· 9 years ago
330d3d8
Roll chromium_revision fa5d546..5618e25 (376142:376642)
by kjellander
· 9 years ago
b9dd7c5
Remove GetTransport() from TransportChannelImpl
by mikescarlett
· 9 years ago
9bf5cde
Update build_ios_libs.sh script to build new Objective-C API and gather header files.
by hjon
· 9 years ago
91fe304
vp9: Adjust parameter for a test in videoprocessor_integrationtest.cc
by Marco
· 9 years ago
a9d0892
Add initial bitrate and frame resolution parameters to quality scaler.
by Alex Glaznev
· 9 years ago
0013dcc
Simplify SSRC usage inside ViEEncoder.
by Peter Boström
· 9 years ago
7254890
Nuke SetSenderBufferingMode.
by Peter Boström
· 9 years ago
da9ae0c
Revert of CQ: Change Android trybots to not run device tests. (patchset #1 id:1 of https://codereview.webrtc.org/1715643002/ )
by kjellander
· 9 years ago
e2d83d6
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
by sprang
· 9 years ago
45c44f0
Simplify EncoderStateFeedback.
by Peter Boström
· 9 years ago
9674d7c
Revert of Prevent data race in MessageQueue. (patchset #3 id:40001 of https://codereview.webrtc.org/1675923002/ )
by jbauch
· 9 years ago
fc968a2
Fix sequence-number replay race for padding.
by Peter Boström
· 9 years ago
88788ad
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/
by kwiberg
· 9 years ago
df88460
Prevent data race in MessageQueue.
by jbauch
· 9 years ago
1e80ce4
webrtc::RtpPacket name freed for better RtpPacket
by Danil Chapovalov
· 9 years ago
c51d694
CQ: Disable linux_baremetal pending installation fix.
by kjellander@webrtc.org
· 9 years ago
728012e
Changed the semantics of Buffer::Clear to not alter the capacity
by ossu
· 9 years ago
ecdeb4c
CQ: Change Android trybots to not run device tests.
by kjellander@webrtc.org
· 9 years ago
c4e3ead
Blacklist "build/c++11" cpplint filter.
by jbauch
· 9 years ago
4458d09
Drop support for playing output through aplay in intelligibility_proc
by Alejandro Luebs
· 9 years ago
b3fb71c
Add RTCAudioSession proxy class.
by Zeke Chin
· 9 years ago
9ac4df1
iOS: Enable modules_unittests and common_audio_unittests
by kjellander
· 9 years ago
235aaa7
Fix Linux 32-bit compilation after sysroot switch.
by Henrik Kjellander
· 9 years ago
66a9928
Roll chromium_revision 1d144ca..fa5d546 (375480:376142)
by kjellander@webrtc.org
· 9 years ago
0e2e50c
Always append the BYE packet type at the end
by aleungbroadsoft
· 9 years ago
452df1c
Suppress UBSan errors in common_audio
by henrik.lundin
· 9 years ago
f45381e
VideoCapturerAndroid: Report onFirstFrameAvailable() for textures as well
by Magnus Jedvert
· 9 years ago
5199c74
AndroidVideoCapturer getSupportedFormats(): Change interface from JSON string to List/vector
by Magnus Jedvert
· 9 years ago
347c0bb
Android GLShader: Check return value of glCreateShader()
by magjed
· 9 years ago
3ee73a5
Make RemoteBitrateEstimator::GetStats() virtual.
by Stefan Holmer
· 9 years ago
fd22e6c
Change PeerConnectionFactory.setVideoHwAccelerationOptions to be able to replace Egl context.
by Per
· 9 years ago
74db777
Revert of Remove GetTransport() from TransportChannelImpl (patchset #3 id:40001 of https://codereview.webrtc.org/1691673002/ )
by guidou
· 9 years ago
59c634b
Re-add RemoteBitrateEstimator::GetStats.
by Stefan Holmer
· 9 years ago
3234819
Fix and simplify the power estimation in the IntelligibilityEnhancer
by Alejandro Luebs
· 9 years ago
ee18220
Remove GetTransport() from TransportChannelImpl
by mikescarlett
· 9 years ago
ee75c7a
Compile rtc_base_objc for Mac.
by tkchin
· 9 years ago
e3c6c82
When doing continual gathering, remove the local ports when a corresponding network is dropped.
by honghaiz
· 9 years ago
a08bb0d
Disabled the test EndToEndTest RestartingSendStreamPreservesRtpState due to the test being flaky.
by peah
· 9 years ago
b7f89d6
Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
by kwiberg
· 9 years ago
dabf07f
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/vad/
by kwiberg
· 9 years ago
a293ef0
Apply VideoOptions per stream.
by nisse
· 9 years ago
789ba92
Simplify CongestionController.
by Stefan Holmer
· 9 years ago
bad7804
Remove unused VideoSendStream TransportAdapter.
by Peter Boström
· 9 years ago
62eaacf
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/
by kwiberg
· 9 years ago
28c99bc
iOS: Include legacy objc API in all.gyp + fix H264 libyuv dependency
by kjellander
· 9 years ago
4b4dc86
Remove conference_mode flag from AudioOptions and VideoOptions.
by nisse
· 9 years ago
22785c7
Exclude legacy objc API tests properly.
by kjellander
· 9 years ago
69e59e6
[rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
by danilchap
· 9 years ago
67680c1
Ignore padding-only RTX packets in test.
by Peter Boström
· 9 years ago
a332e2d
Added boilerplate code for being able to test the upcoming AEC functionality.
by peah
· 9 years ago
0206000
iOS: Add resource files for tests and implement OutputPath
by kjellander
· 9 years ago
85d8bb0
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/transient/
by kwiberg
· 9 years ago
9d3584c
Implementing unified plan encoding of msid.
by deadbeef
· 9 years ago
25d6a0f
Adding TSan suppressions temporarily to fix some flaky unit tests.
by deadbeef
· 9 years ago
e1a0c94
Add network cost as part of the connection ranking.
by honghaiz
· 9 years ago
2c38c20
Fix out-of-buffer write in iLBC
by henrik.lundin
· 9 years ago
44c65e9
Enable adaptive threshold experiment by default.
by Stefan Holmer
· 9 years ago
9d0c432
Remove video-codec max bitrate from TMMBN.
by Peter Boström
· 9 years ago
d20327c
Increase the allowed number of probe packets in test to please msan.
by Stefan Holmer
· 9 years ago
ee31f0a
Fix out-of-buffer read in iLBC
by henrik.lundin
· 9 years ago
62a5ccd
Update bitrate only when we have incoming packet.
by Stefan Holmer
· 9 years ago
58cf5f1
Changed order of events when synthesizing a call.
by peah
· 9 years ago
0453ef8
Prevent busy-looping PacedSender on small packets.
by Peter Boström
· 9 years ago
1794b26
Extract ViESyncModule outside ViEChannel.
by Peter Boström
· 9 years ago
a3dc79e
Move SSLIdentity Generate() implementations from .h to .cc file.
by Torbjorn Granlund
· 9 years ago
71e92dc
Avoid overflow in WebRtcSpl_Sqrt
by henrik.lundin
· 9 years ago
092c951
Roll chromium_revision aefd358..1d144ca (375443:375480)
by kjellander
· 9 years ago
e8dc081
Implement certificate lifetime parameter as required by WebRTC RFC.
by torbjorng
· 9 years ago
b1ae3a4
Stop decoders in VideoReceiveStream destructor.
by Peter Boström
· 9 years ago
461121c
Replaced eglbase_jni with with holding a EglBase in PeerConnectionFactory.
by perkj
· 9 years ago
8259c2d
Roll chromium_revision 8d1f312..aefd358 (375401:375443)
by kjellander
· 9 years ago
8110482
Rename gtest_exclude for rtc_pc_unittests
by kjellander@webrtc.org
· 9 years ago
56e6269
Rename gtest_exclude for rtc_media_unittests.
by Peter Boström
· 9 years ago
88c52a7
Disable VerifyHistogramStatsWithRed on DrMemory.
by Peter Boström
· 9 years ago
16c5a96
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/
by kwiberg
· 9 years ago
3959397
Reland of Don't send FEC for H.264 with NACK enabled. (patchset #1 id:1 of https://codereview.webrtc.org/1692783005/ )
by Peter Boström
· 9 years ago
cde5d6b
removed five redundant avsync tests to make webrtc_perf_test faster
by Danil Chapovalov
· 9 years ago
e829f58
Rename libjingle_p2p_unittest -> rtc_pc_unittests
by kjellander@webrtc.org
· 9 years ago
3747838
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/test/
by kwiberg
· 9 years ago
1d9ada7
Roll chromium_revision 99c33e8..8d1f312 (375382:375401)
by kjellander
· 9 years ago
2d0c332
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/
by kwiberg
· 9 years ago
04af839
Move refcount.h and scoped_ref_ptr.h to rtc_base_approved. BUG=
by tommi
· 9 years ago
290ab41
Roll chromium_revision ee3223b..99c33e8 (375377:375382)
by kjellander
· 9 years ago
91d9756
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/
by kwiberg
· 9 years ago
3f5f5aa
Roll chromium_revision e1bc525..ee3223b (375361:375377)
by kjellander
· 9 years ago
be61562
Moved the GainControlForNewAGC class to be a separate file.
by peah
· 9 years ago
46e2cb8
Roll chromium_revision 96c72eb..e1bc525 (375334:375361)
by kjellander
· 9 years ago
88ec91d
Roll chromium_revision c9db86b..96c72eb (374913:375334)
by kjellander
· 9 years ago
88b0a22
Add VP9 to full stack tests.
by asapersson
· 9 years ago
29ffdc1
Revert of Don't send FEC for H.264 with NACK enabled. (patchset #5 id:80001 of https://codereview.webrtc.org/1687303002/ )
by deadbeef
· 9 years ago
5e7834e
Android: Make VideoCapturer an interface for all VideoCapturers to implement
by Magnus Jedvert
· 9 years ago
e78765b
Removes Nexus 5 from AEC and NS blacklists
by henrika
· 9 years ago
579e832
Fix race on VCM protection callback.
by Peter Boström
· 9 years ago
b72dada
Remove Reset from conditionally-compiled decoders.
by Peter Boström
· 9 years ago
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