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2014-06-19Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-06-19Revert 6481 and 6482fgalligan@google.com
2014-06-19Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buff...turaj@webrtc.org
2014-06-19Adding an empty constructor implementation to the AudioSink classhenrik.lundin@webrtc.org
2014-06-19Changes to tests and tools in audio_processing.bjornv@webrtc.org
2014-06-19Ensure that the start bitrate can be set multiple times.stefan@webrtc.org
2014-06-19Adding test::AudioSink interface and derived classeshenrik.lundin@webrtc.org
2014-06-19Fixes and re-enables tests disabled on Androidbjornv@webrtc.org
2014-06-19Update makefiles after merge of Chromium at 278252Android Chromium Automerger
2014-06-18Update webrtc to fix unpack_lib expansion.fgalligan@google.com
2014-06-18Update generated asm offsets scripts.fgalligan@google.com
2014-06-18Neon version of FilterAdaptation()bjornv@webrtc.org
2014-06-18Update PacketSource and RtpFileSourcehenrik.lundin@webrtc.org
2014-06-18Revert "Restore ptypes.txt file"henrik.lundin@webrtc.org
2014-06-17Revert 6473 "Update generated asm offsets scripts."turaj@webrtc.org
2014-06-17Update generated asm offsets scripts.fgalligan@google.com
2014-06-17Add round-robin selection of send stream to pad on.stefan@webrtc.org
2014-06-17Add high perf mode to VP8niklas.enbom@webrtc.org
2014-06-17base: Renaming + conforming: post commit review changes for https://webrtc-co...henrike@webrtc.org
2014-06-17Rebase webrtc/base with r6464 version of talk/base:henrike@webrtc.org
2014-06-17Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."minyue@webrtc.org
2014-06-17Initial GN work for WebRTCkjellander@webrtc.org
2014-06-17Restore ptypes.txt filehenrik.lundin@webrtc.org
2014-06-17Updated W3C getusermedia tests to the latest version of the spec.phoglund@webrtc.org
2014-06-17Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-...minyue@webrtc.org
2014-06-16Makes it possible to prevent some third party libraries (jsoncpp and openssl)...henrike@webrtc.org
2014-06-16Update makefiles after merge of Chromium at 277521Android Chromium Automerger
2014-06-16Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-06-16Update makefiles after merge of Chromium at 277428Android Chromium Automerger
2014-06-16Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-06-16Add max limit of number for overuses. When limit is reached always apply the ...asapersson@webrtc.org
2014-06-16Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.asapersson@webrtc.org
2014-06-16Remove ivinnichenko from webrtc/test/OWNERSkjellander@webrtc.org
2014-06-16Importing ThreadChecker class from Chromiumhenrik.lundin@webrtc.org
2014-06-16Adds aluebs@webrtc.org as owner to audio_processingbjornv@webrtc.org
2014-06-16common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16bjornv@webrtc.org
2014-06-16Implements start bitrate for new video API.mflodman@webrtc.org
2014-06-16Add thread annotations to parts of ACMGenericCodechenrik.lundin@webrtc.org
2014-06-16Add missing sources to webrtc/base/base.gypkjellander@webrtc.org
2014-06-13Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc.glaznev@webrtc.org
2014-06-13Neon version of OverdriveAndSuppress()bjornv@webrtc.org
2014-06-13Pass GYP DEPTH variable to isolate.kjellander@webrtc.org
2014-06-12Revert 6415 "Update generated asm offsets scripts."wu@webrtc.org
2014-06-12json.h include different header files depending on WEBRTC_CHROMIUM_BUILD bein...henrike@webrtc.org
2014-06-12Enable pacing by default and remove the option to disable it from the new API.stefan@webrtc.org
2014-06-12Update generated asm offsets scripts.fgalligan@google.com
2014-06-12Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..."kjellander@webrtc.org
2014-06-12Revert 6407 "Revert 6405 "Update generated asm offsets scripts.""minyue@webrtc.org
2014-06-12Increased kMaxRampUpDelayMs (120 to 240s).asapersson@webrtc.org
2014-06-12Revert 6405 "Update generated asm offsets scripts."henrike@webrtc.org
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