Roll src/third_party/webrtc 44974e143c90..38332cdcb107 (7 commits)

https://webrtc.googlesource.com/src.git/+log/44974e143c90..38332cdcb107


git log 44974e143c90..38332cdcb107 --date=short --no-merges --format='%ad %ae %s'
2018-11-20 orphis@webrtc.org Add RTCP and simulcast support for RTCRtpReceiver::getParameters()
2018-11-20 alessiob@webrtc.org Add output directory option for audioproc_f data dump files.
2018-11-20 jonasolsson@webrtc.org Make RTC_LOG_FILE_LINE use its parameters
2018-11-20 terelius@webrtc.org Remove unused variables in RtcEventAudioXStreamConfig::Copy()
2018-11-20 nisse@webrtc.org Move VideoCodecType from common_types.h to api/video/video_codec_type.h
2018-11-20 mbonadei@webrtc.org Fix threshold in VideoCodecTestLibvpx.ChangeFramerateVP9.
2018-11-20 alessiob@webrtc.org APM audioproc_f: flag for AGC2 adaptive level estimator.


Created with:
  gclient setdep -r src/third_party/webrtc@38332cdcb107

The AutoRoll server is located here: https://autoroll.skia.org/r/webrtc-chromium-autoroll

Documentation for the AutoRoller is here:
https://skia.googlesource.com/buildbot/+/master/autoroll/README.md

If the roll is causing failures, please contact the current sheriff, who should
be CC'd on the roll, and stop the roller if necessary.

CQ_INCLUDE_TRYBOTS=luci.chromium.try:linux_chromium_archive_rel_ng;luci.chromium.try:mac_chromium_archive_rel_ng

BUG=chromium:None
TBR=webrtc-chromium-sheriffs-robots@google.com

Change-Id: I404c6b156fffc919b2d8199cdca67467c0b791bc
Reviewed-on: https://chromium-review.googlesource.com/c/1344272
Reviewed-by: chromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com>
Commit-Queue: chromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#609904}
1 file changed