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2014-08-04Update makefiles after merge of Chromium at 287308Android Chromium Automerger
2014-07-29Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-07-24Make sure padding is sent on the first sending RTP module.mflodman@webrtc.org
2014-07-23Fix flaky ramp-up test.stefan@webrtc.org
2014-07-22The lastest commit on this file was inminyue@webrtc.org
2014-07-22Remove no longer used SkipEncodingUnusedStreams.andresp@webrtc.org
2014-07-22Remove remains of WEBRTC_NO_STL.andresp@webrtc.org
2014-07-21MIPS optimizations for ISAC (patch #2)andrew@webrtc.org
2014-07-20Check before send/receive rtp header extensions.pbos@webrtc.org
2014-07-18This is to re-open an earlier CLminyue@webrtc.org
2014-07-18Runtime guard for iOS7 property.tkchin@webrtc.org
2014-07-18Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.tkchin@webrtc.org
2014-07-18This is related to an earlier CL of enabling Opus 48 kHz.minyue@webrtc.org
2014-07-18Sleep in ThreadTest thread functions.pbos@webrtc.org
2014-07-18AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->floatkwiberg@webrtc.org
2014-07-17Reduce runtime of RingBufferTest by a factor of 100.andrew@webrtc.org
2014-07-17Use _numMixedParticipants instead of audioFrameList->size() to determine if t...wu@webrtc.org
2014-07-17Fix issue where padding is sent before media with undefined timestamps if not...stefan@webrtc.org
2014-07-17Remove unused ExperimentalNS API in AudioProcessingaluebs@webrtc.org
2014-07-17AudioBuffer: Eliminate the SplitChannelBuffer classkwiberg@webrtc.org
2014-07-17Simplify AudioBuffer::mixed_low_pass_data APIaluebs@webrtc.org
2014-07-17AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameterkwiberg@webrtc.org
2014-07-17Add unit test for MediaFile WAV file writingkwiberg@webrtc.org
2014-07-17Fixes up rtc so that it compiles on iOS 8 SDK.tkchin@webrtc.org
2014-07-16r6709 lacks a change in BUILD.gnminyue@webrtc.org
2014-07-16Raw packet loss rate reported by RTP_RTCP module may vary too drastically ove...minyue@webrtc.org
2014-07-16Compile-time guard for iOS7 specific property.tkchin@webrtc.org
2014-07-16Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-07-16Print an info log instead of return an error if an external encoder is de-reg...stefan@webrtc.org
2014-07-16Remove old padding path in RTPSender.pbos@webrtc.org
2014-07-16int16<->float conversions: Use size_t for array length argument, not intkwiberg@webrtc.org
2014-07-16Define convenient FATAL_ERROR() and FATAL_ERROR_IF() macroskwiberg@webrtc.org
2014-07-16nrsh1 is written before tmp321 is read, so needs to be earlyclobberkwiberg@webrtc.org
2014-07-15Fix an invalid memory access due to typo in win/cursor.cc.jiayl@webrtc.org
2014-07-15After an audio interruption the audio unit no longer invokes its render callb...tkchin@webrtc.org
2014-07-15Remove Thread::RunningForChannelManager().tkchin@webrtc.org
2014-07-15Improvements to the pacer where it lost some budget due to truncation errors.stefan@webrtc.org
2014-07-15Fix breakage introduced by r6691.pbos@webrtc.org
2014-07-15Make RTCP sender report send media bytes.pbos@webrtc.org
2014-07-15Eliminate unnecessary #includekwiberg@webrtc.org
2014-07-15rtc::Fatal output: Print space between # and messagekwiberg@webrtc.org
2014-07-15Remove the VPM denoiser.pbos@webrtc.org
2014-07-14Rebase webrtc/base with r6682 version of talk/base:henrike@webrtc.org
2014-07-14Fix deadlock in Android stopCapture() call.glaznev@webrtc.org
2014-07-13GN: Fix include paths for WebRTC in Chromium build.kjellander@webrtc.org
2014-07-11Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .tommi@webrtc.org
2014-07-11Remove always-true expression.tommi@webrtc.org
2014-07-11Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.tommi@webrtc.org
2014-07-11Thread annotate RTCPSender.pbos@webrtc.org
2014-07-11Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.stefan@webrtc.org
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