Roll 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}

DEPS

@@ -235,7 +235,7 @@ deps = {
     Var('chromium_git') + '/native_client/src/third_party/scons-2.0.1.git' + '@' + '1c1550e17fc26355d08627fbdec13d8291227067',
 
   'src/third_party/webrtc':
-    Var('chromium_git') + '/external/webrtc/trunk/webrtc.git' + '@' + 'f91805c46d7a5067e3b4ce2bd066ca504dc167f3', # commit position 18617
+    Var('chromium_git') + '/external/webrtc/trunk/webrtc.git' + '@' + 'e6fddec2d5f2b9883f3d678c9e5623a8e1c6e28b', # commit position 18665
 
   'src/third_party/openmax_dl':
     Var('chromium_git') + '/external/webrtc/deps/third_party/openmax.git' + '@' +  Var('openmax_dl_revision'),