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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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277e06d314608f334b662aac914c4da4a7394db9
277e06d
Remove unused libudev on Linux.
by solenberg
· 9 years ago
102362b
Truly disable tests.
by Stefan Holmer
· 9 years ago
1d50ee4
Stop using some scoped_ptr features that unique_ptr doesn't have
by kwiberg
· 9 years ago
15622c0
WebRtcIsacfix_PitchFilter: Don't read uninitialized array entries
by kwiberg
· 9 years ago
076c7b5
Roll chromium_revision 6822f00..ba603a0 (381689:381748)
by kjellander
· 9 years ago
b031955
Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API
by aluebs
· 9 years ago
da116c4
Use ProcessReverseStream in VoiceEngines OutputMixer
by aluebs
· 9 years ago
56d4d05
Detect and report camera close timeout.
by Alex Glaznev
· 9 years ago
246b527
Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
by deadbeef
· 9 years ago
4cd331b
Remove code interfacing legacy openssl.
by torbjorng
· 9 years ago
e0897c0
Remove webrtc/sound/ subdir.
by solenberg
· 9 years ago
c9022f5
Delete empty API files and cleaned up includes.
by perkj
· 9 years ago
8811b35
Enable Continual gathering in apprtcdemo.
by honghaiz
· 9 years ago
22feaa3
Replace scoped_ptr with unique_ptr in webrtc/modules/
by kwiberg
· 9 years ago
b213795
Fix confusing naming of static class variables
by kwiberg
· 9 years ago
55d6e7c
Disable tests due to issue 5659.
by Stefan Holmer
· 9 years ago
ed05137
Roll chromium_revision a310422..6822f00 (381642:381689)
by kjellander
· 9 years ago
b4c8247
Added function for parsing single rtcp packet in tests.
by Danil Chapovalov
· 9 years ago
505945a
Delete unused VideoCapturer statistics.
by Niels Möller
· 9 years ago
94a23f0
Reland "Add check_deps rules in DEPS files."
by kjellander@webrtc.org
· 9 years ago
d8ddb79
SurfaceTextureHelper: Fix startListening()/stopListening() race
by magjed
· 9 years ago
0de1c13
Adding DebugDumpReplayer.
by minyue
· 9 years ago
d6c3954
Refactor VideoTracks to forward all sinks to its source
by perkj
· 9 years ago
292d658
Fix for intermittent tsan2 errors from SendRtpToRtpOnThread and SendSrtpToSrtpOnThread.
by ossu
· 9 years ago
8300641
Roll chromium_revision f1e8fe4..a310422 (381494:381642)
by kjellander
· 9 years ago
dc1c62c
Enable setting the maximum bitrate limit in RtpSender.
by skvlad
· 9 years ago
df6416a
Dont always downsample to 16kHz in the reverse stream in APM
by aluebs
· 9 years ago
2bb3afa
Replace scoped_ptr with unique_ptr in webrtc/modules/desktop_capture/
by kwiberg
· 9 years ago
672b78f
Roll chromium_revision 3001f17..f1e8fe4 (381427:381494)
by kjellander
· 9 years ago
2bbff99
Helpers in peer connection unit tests: Use scoped_ptr instead of raw pointers
by kwiberg
· 9 years ago
8ad582d
Remove DeviceManager and DeviceInfo.
by solenberg
· 9 years ago
34b11eb
Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p.
by honghaiz
· 9 years ago
c4a74e9
Revert of VCMCodecTimer: Change filter from max to 95th percentile (patchset #5 id:180001 of https://codereview.webrtc.org/1742323002/ )
by magjed
· 9 years ago
0e2e5d9
Revert of Revert opus memcheck suppression (patchset #1 id:1 of https://codereview.webrtc.org/1801233002/ )
by stefan
· 9 years ago
b69395b
Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (patchset #2 id:20001 of https://codereview.webrtc.org/1802993002/ )
by solenberg
· 9 years ago
69a8199
Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
by solenberg
· 9 years ago
bf31d3c
Roll chromium_revision 334c2c6..3001f17 (381380:381427)
by kjellander
· 9 years ago
b4855cc
Delete file acm_neteq_unittest.cc
by henrik.lundin
· 9 years ago
f62d107
Revert opus memcheck suppression
by flim
· 9 years ago
0510331
Drop VideoOptions from VideoSendParameters.
by nisse
· 9 years ago
5a83380
Roll chromium_revision b22628f..334c2c6 (381270:381380)
by kjellander
· 9 years ago
56cf60e
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
by kjellander
· 9 years ago
086f851
Add check_deps rules in DEPS files.
by kjellander@webrtc.org
· 9 years ago
e54467f
Use RTCAudioSessionDelegateAdapter in AudioDeviceIOS.
by tkchin
· 9 years ago
4557d33
Roll chromium_revision e1a2958..b22628f (381201:381270)
by kjellander
· 9 years ago
776593b
Reland: Drop the 16kHz sample rate restriction on AECM and zero out higher bands
by aluebs
· 9 years ago
f5d4786
SSLCertificate::GetChain: Return scoped_ptr
by kwiberg
· 9 years ago
6021fe2
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
by solenberg
· 9 years ago
6baec03
Port::GetStunMessage: Write to scoped_ptr instead of raw pointer
by kwiberg
· 9 years ago
0540242
Add more conditions for CPU detection in denoiser filter.
by jackychen
· 9 years ago
88dec83
Fixing flaky "TestExpireTime" test.
by Taylor Brandstetter
· 9 years ago
7021b92
introduced rtcp::CommonHeader class
by Danil Chapovalov
· 9 years ago
b58a158
Removed the AudioProcessing dependency in EchoCancellerImpl.
by peah
· 9 years ago
8e3949c
Roll chromium_revision 1a84b14..e1a2958 (381152:381201)
by kjellander
· 9 years ago
80c2cd9
Android: Add more info for createPbufferSurface() exceptions
by magjed
· 9 years ago
253534d
Removed the dependency on AudioProcessingImpl in EchoControlMobileImpl
by peah
· 9 years ago
b8fbb54
Removed the dependency on AudioProcessingImpl in GainControlImpl
by peah
· 9 years ago
f8cdd18
Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs"
by asapersson
· 9 years ago
a1cf366
Handle iOS devices with no rear-facing camera
by hjon
· 9 years ago
e5d5e51
Roll chromium_revision 390847b..1a84b14 (381002:381152)
by kjellander
· 9 years ago
e50872b
Remove unused method OutputMixer::PlayDtmfTone() and infrastructure.
by solenberg
· 9 years ago
c4ec4a2
Add breaks in switch statement to fix AppRTCDemo crash
by hjon
· 9 years ago
a9635b8
Use the right mirroring state when switching cameras in AppRTCDemo.
by hjon
· 9 years ago
8bbbf2c
Rename RTCIceConnectionStateMax to RTCIceConnectionStateCount in Objective-C API.
by hjon
· 9 years ago
5ad6bf1
Roll chromium_revision 6a56b54..390847b (380836:381002)
by kjellander
· 9 years ago
7fb69db
Reland the CL to remove candidates when doing continual gathering
by Honghai Zhang
· 9 years ago
1122dc0
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
fb647a6
Initialize/configure video encoders asychronously.
by Peter Boström
· 9 years ago
4c83c05
Implemented more general version of ForwardDiff/RevereseDiff.
by philipel
· 9 years ago
7a4116a
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
by danilchap
· 9 years ago
31642aa
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
d67717f
Make opus memcheck suppression more generic
by Stefan Holmer
· 9 years ago
3b41170
Remove sparse macros (RTC_HISTOGRAM_*_SPARSE_*) that are no longer used.
by asapersson
· 9 years ago
0dc2316
VideoCapturer: Update interface
by magjed
· 9 years ago
b2a24ec
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
479a04c
Suppress invalid read in Opus.
by Stefan Holmer
· 9 years ago
2875077
Roll chromium_revision db8316d..6a56b54 (380688:380836): MSVS 2015 switch
by kjellander@webrtc.org
· 9 years ago
79858f8
Update iOS AppRTCDemo to use the updated Objective-C API.
by hjon
· 9 years ago
9f987d3
Refactor AVAudioSession intialization code.
by tkchin
· 9 years ago
0ce3bf9
Fix lock behavior on RTCAudioSession.
by tkchin
· 9 years ago
b25345e
Replace scoped_ptr with unique_ptr in webrtc/call/
by kwiberg
· 9 years ago
83f831a
Experiment for the nack module.
by philipel
· 9 years ago
84cc918
Replace scoped_ptr with unique_ptr in talk/
by kwiberg
· 9 years ago
2db1dbb
Remove references to build_with_libjingle and libjingle_java GYP variables.
by kjellander@webrtc.org
· 9 years ago
bad7b09
Update examples GYP to avoid rtc_base_approved warning.
by tkchin
· 9 years ago
d44c077
Revert of Safe numeric library: base/numerics (copied from Chromium) (patchset #11 id:250001 of https://codereview.webrtc.org/1753293002/ )
by Tommi
· 9 years ago
35c5336
Revert of Added webrtc/base/safe_conversions.h as a pseudonym (patchset #1 id:20001 of https://codereview.webrtc.org/1774933003/ )
by Tommi
· 9 years ago
e7ba086
Reconfigure video encoders even when not sending.
by Peter Boström
· 9 years ago
0149e75
Remove the (previosly deprecated) Pass methods
by kwiberg
· 9 years ago
3102294
Replace scoped_ptr with unique_ptr in webrtc/pc/
by kwiberg
· 9 years ago
6f59a4f
Revert of Remove candidates when doing continual gathering (patchset #15 id:560001 of https://codereview.webrtc.org/1648813004/ )
by tommi
· 9 years ago
916c76e
Add new files for VideoSourceBase to roll into Chrome in preparation for implementatin.
by perkj
· 9 years ago
84430da
When doing candidate re-gathering in the same generation, Remove the existing local candidate on the same network
by honghaiz
· 9 years ago
44d7392
Roll chromium_revision 5778d35..db8316d (380596:380688)
by kjellander
· 9 years ago
3ad4bd3
Skinmap improvement.
by jackychen
· 9 years ago
86aabb2
Move BitrateAllocator reference from ViEEncoder to VideoSendStream.
by mflodman
· 9 years ago
a590605
Roll chromium_revision 1ff3458..5778d35 (380437:380596)
by kjellander
· 9 years ago
8842c3e
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
4bf0c71
VCMCodecTimer: Change filter from max to 95th percentile
by magjed
· 9 years ago
43166b8
Add IsAcceptableCipher, use instead of GetDefaultCipher.
by torbjorng
· 9 years ago
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