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2014-12-08Merge from Chromium at DEPS revision 39.0.2171.95android-5.1.1_r5android-5.1.1_r28android-5.1.1_r22android-5.1.1_r17android-5.1.1_r12lollipop-mr1-wfc-releaselollipop-mr1-devBen Murdoch
2014-12-08Update makefiles after merge of Chromium at 39.0.2171.95Ben Murdoch
2014-12-08Merge third_party/webrtc from https://chromium.googlesource.com/a/external/we...Ben Murdoch
2014-12-02Merge 7729 "Build fix for MIPS Android Webview build."andrew@webrtc.org
2014-12-02Merge 7343 "Changed mips_arch_variant variable value correspondi..."andrew@webrtc.org
2014-12-02Merge 7580 "Build fix for MIPS32R6."andrew@webrtc.org
2014-10-29Merge from Chromium at DEPS revision 39.0.2171.44Ben Murdoch
2014-10-29Update makefiles after merge of Chromium at 39.0.2171.44Ben Murdoch
2014-10-29Merge third_party/webrtc from https://chromium.googlesource.com/a/external/we...Ben Murdoch
2014-10-27Merge r7418 to 39 branchtnakamura@webrtc.org
2014-10-14Merge from Chromium at DEPS revision 39.0.2171.26Torne (Richard Coles)
2014-10-14Update makefiles after merge of Chromium at 39.0.2171.26Torne (Richard Coles)
2014-10-14Merge third_party/webrtc from https://chromium.googlesource.com/a/external/we...Torne (Richard Coles)
2014-10-06Merge webrtc r7310 to M39.jiayl@webrtc.org
2014-10-06Merge webrtc r7301 to M39 branch. jiayl@webrtc.org
2014-09-30Create WebRTC branch 39 from trunk@7296tnakamura@webrtc.org
2014-09-30Merge from Chromium at DEPS revision 267aeeb8d85cPrimiano Tucci
2014-09-25Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-09-25Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.henrik.lundin@webrtc.org
2014-09-25Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.henrik.lundin@webrtc.org
2014-09-25Fix typo from RtpPacketizerH264.pbos@webrtc.org
2014-09-25Fix typo from RtpPacketizerH264.pbos@webrtc.org
2014-09-25Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293).andresp@webrtc.org
2014-09-25Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293).andresp@webrtc.org
2014-09-24Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.jiayl@webrtc.org
2014-09-24Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.jiayl@webrtc.org
2014-09-24Enable render downmixing to mono in AudioProcessing.andrew@webrtc.org
2014-09-24Enable render downmixing to mono in AudioProcessing.andrew@webrtc.org
2014-09-24Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMacjiayl@webrtc.org
2014-09-24Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMacjiayl@webrtc.org
2014-09-24Fix a problem in Thread::Send.jiayl@webrtc.org
2014-09-24Fix a problem in Thread::Send.jiayl@webrtc.org
2014-09-24Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-09-24Call NS AnalyzeCaptureAudio before AECaluebs@webrtc.org
2014-09-24Call NS AnalyzeCaptureAudio before AECaluebs@webrtc.org
2014-09-24Reduce jitter delay for low fps streams.sprang@webrtc.org
2014-09-24Reduce jitter delay for low fps streams.sprang@webrtc.org
2014-09-24Moved the filter calculation from analyze to process in ns_corealuebs@webrtc.org
2014-09-24Moved the filter calculation from analyze to process in ns_corealuebs@webrtc.org
2014-09-24audioproc: Now also writes to output file in simulation modebjornv@webrtc.org
2014-09-24audioproc: Now also writes to output file in simulation modebjornv@webrtc.org
2014-09-24WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_tkwiberg@webrtc.org
2014-09-24WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_tkwiberg@webrtc.org
2014-09-24Thread annotation of rtc::CriticalSection.pbos@webrtc.org
2014-09-24Thread annotation of rtc::CriticalSection.pbos@webrtc.org
2014-09-24Move thread_annotations.h to webrtc/base/.pbos@webrtc.org
2014-09-24Move thread_annotations.h to webrtc/base/.pbos@webrtc.org
2014-09-23Use VPX_IMG_FMT_*/VPX_PLANE_* definesjohannkoenig@google.com
2014-09-23Use VPX_IMG_FMT_*/VPX_PLANE_* definesjohannkoenig@google.com
2014-09-23Revert "Remove DTMF status methods from Voice Engine" r7276henrik.lundin@webrtc.org
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