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AgeCommit message (Expand)Author
2014-09-05common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processingbjornv@webrtc.org
2014-09-04Revert 7070 "TurnPort should retry allocation with a new address on errorhenrike@webrtc.org
2014-09-04TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOC...jiayl@webrtc.org
2014-09-04Add support for WAV output in audioprocaluebs@webrtc.org
2014-09-04Add direct_dependent_config to desktop_capture in GN build.brettw@chromium.org
2014-09-04Fix strange owners files with comments that crashs "git cl presubmit"andresp@webrtc.org
2014-09-04[MIPS] Fix gn gen failure for MIPS in webrtckjellander@webrtc.org
2014-09-04Moving the api.js and bot.js to /rtcbot/bot/ to be shared betweenhoussainy@google.com
2014-09-04Reland rev 7041 with BUILD.gn files.andresp@webrtc.org
2014-09-04Fix MSVC warnings about value truncations, webrtc/common_audio/ edition.bjornv@webrtc.org
2014-09-04Rename Audio[Multi]Vector.CopyFrom to .CopyTohenrik.lundin@webrtc.org
2014-09-04Change gflags and gmock includes to be full paths.kjellander@webrtc.org
2014-09-04ACMOpus: Remove useless member variable fec_enabled_kwiberg@webrtc.org
2014-09-04Add support for multi-channel DTMF tone generationhenrik.lundin@webrtc.org
2014-09-04Change return value for number of discarded packets to be int.asapersson@webrtc.org
2014-09-04Fix audio/video sync when FEC is enabled.stefan@webrtc.org
2014-09-04Fix compile errors on webrtc/base.andresp@webrtc.org
2014-09-04Remove ambiguous call to MakeCheckOpString.andresp@webrtc.org
2014-09-03cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile e...fbarchard@google.com
2014-09-03Fix leak of NSAutoreleasePool.tkchin@webrtc.org
2014-09-03Revert 7041 " Audio codecs to include webrtc/typedefs.h"henrike@webrtc.org
2014-09-03Network up/down signaling in Call.pbos@webrtc.org
2014-09-03Audio codecs to include webrtc/typedefs.handresp@webrtc.org
2014-09-03Partial revert of r7014 (Android APK refactor)kjellander@webrtc.org
2014-09-03Use the sample rate as a temporary solution to unpack aecdumps with wrong sizesaluebs@webrtc.org
2014-09-03Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRateminyue@webrtc.org
2014-09-03Setting marker bit on DTMF correctlystefan@webrtc.org
2014-09-03Fix issues in audioproc for float aecdumpsaluebs@webrtc.org
2014-09-03audio_processing/nsx: Bug fix that could cause divide by zerobjornv@webrtc.org
2014-09-02Remove the checks.h dependence on logging.h in a standalone build.andrew@webrtc.org
2014-09-02Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator f...stefan@webrtc.org
2014-09-02Create a copy of talk/xmllite under webrtc/xmllite.henrike@webrtc.org
2014-09-02Disable video_engine_tests and webrtc_perf_tests on Android.kjellander@webrtc.org
2014-09-02Divide-by-zero problem in NetEq's Normal::Process fixedhenrik.lundin@webrtc.org
2014-09-02Disable video_capture_tests for Android.kjellander@webrtc.org
2014-09-02GN: Update webrtc/base to recent GYP changes.kjellander@webrtc.org
2014-09-02Update makefiles after merge of Chromium at a804d98340beAndroid Chromium Automerger
2014-09-02Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-09-02RTCBot is a framework that allows to write tests where logic runs on a singleandresp@webrtc.org
2014-09-02Remove build_with_chromium==1 conditions for Androidkjellander@webrtc.org
2014-09-02Unpacking aecdumps generates wav filesaluebs@webrtc.org
2014-09-01Fix audio_decoder_unittests.isolatekjellander@webrtc.org
2014-09-01Adding more codecs to the AcmSenderBitExactnesshenrik.lundin@webrtc.org
2014-09-01Android APK tests built from a normal WebRTC checkout.kjellander@webrtc.org
2014-09-01GN: Audio device modulekjellander@webrtc.org
2014-08-31GN: Implement voice engine, common audio, audio coding and audio processingkjellander@webrtc.org
2014-08-29GN: Fix webrtc/video/BUILD.gn for Chromium build.kjellander@webrtc.org
2014-08-29MIPS optimizations for AEC audio processing moduleandrew@webrtc.org
2014-08-29Add LTO support for Android Chromium.andrew@webrtc.org
2014-08-29Allow same src and dst in InputAudioFile::DuplicateInterleavedhenrik.lundin@webrtc.org
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