DescriptionRoll WebRTC 18617:18665 (33 commits)
Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/f91805c..e6fddec
$ git log f91805c..e6fddec --date=short --no-merges --format=%ad %ae %s
2017-06-19 magjed@webrtc.org Android: Modular WebRTC follow-up
2017-06-19 alexnarest@webrtc.org Fix uploading of available send bitrate statistics.
2017-06-19 minyue@webrtc.org Revert "Adding ANA config event to debug dump."
2017-06-19 andersc@webrtc.org Use uint8 pointer instead of std::vector in NV12Scale.
2017-06-19 minyue@webrtc.org Adding ANA config event to debug dump.
2017-06-19 magjed@webrtc.org Android JNI: Clean up AndroidVideoTrackSource and NativeHandleImpl
2017-06-19 ilnik@webrtc.org Implement timing frames.
2017-06-19 denicija@google.com Remove explicit draw call on MTKView.
2017-06-19 terelius@webrtc.org Remove redundant std::min from ProbeBitrateEstimator.
2017-06-19 oprypin@webrtc.org Use information about blacklisted devices in video_quality_loopback_test
2017-06-18 charujain@webrtc.org Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ )
2017-06-18 charujain@webrtc.org Revert of Opus implementation of the AudioDecoderFactoryTemplate API (patchset #1 id:1 of https://codereview.webrtc.org/2942733003/ )
2017-06-17 zhihuang@webrtc.org Support building WebRTC without audio and video for Android
2017-06-17 kwiberg@webrtc.org Opus implementation of the AudioDecoderFactoryTemplate API
2017-06-17 kwiberg@webrtc.org Opus implementation of the AudioEncoderFactoryTemplate API
2017-06-17 kwiberg@webrtc.org G722 implementation of the AudioEncoderFactoryTemplate API
2017-06-17 kwiberg@webrtc.org G722 implementation of the AudioDecoderFactoryTemplate API
2017-06-17 kwiberg@webrtc.org Templated AudioDecoderFactory
2017-06-16 deadbeef@webrtc.org Fixing incorrect use of erase/remove idiom.
2017-06-16 emadomara@google.com Enable SNI in ssl adapter.
2017-06-16 kwiberg@webrtc.org Templated AudioEncoderFactory
2017-06-16 mellem@webrtc.org Create the VideoEncoderFactory and implement it.
2017-06-16 stefan@webrtc.org Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams.
2017-06-16 eladalon@webrtc.org Style fixes in rtcp_packet/
2017-06-16 ilnik@webrtc.org Add cropping to VIEEncoder to match simulcast streams resolution
2017-06-16 terelius@webrtc.org Add has_value() and value() methods to rtc::Optional.
2017-06-16 henrika@webrtc.org Reduces sensitivity in audio-glitch detector for iOS
2017-06-16 erikvarga@webrtc.org Use RaceChecker instead of ThreadChecker in a few places.
2017-06-16 eladalon@webrtc.org Remove unused #include "libyuv/compare.h"
2017-06-15 andersc@webrtc.org Move setting switches in AppRTCMobile to Settings screen
2017-06-16 sakal@webrtc.org Create AndroidVideoBuffer and allow renderers to consume it.
2017-06-16 nisse@webrtc.org Delete SignalSrtpError.
2017-06-15 glaznev@webrtc.org Support H.264 high profile encoding on Exynos devices.
R=magjed@chromium.org
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng
BUG=
Review-Url: https://codereview.chromium.org/2948753002
Cr-Commit-Position: refs/heads/master@{#480808}
Committed: https://chromium.googlesource.com/chromium/src/+/08b3043932bc085e3cfbe9ccd2b04f52b93e8a1a
Patch Set 1 #Messages
Total messages: 7 (3 generated)
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