Roll WebRTC 15358:15399 (25 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/42ad2d4..6926a7d

$ git log 42ad2d4..6926a7d --date=short --no-merges --format=%ad %ae %s
2016-12-02 deadbeef@webrtc.org Revert of Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan (patchset #1 id:1 of https://codereview.webrtc.org/2546913003/ )
2016-12-02 deadbeef@webrtc.org Allow locally rendered video to be downscaled in end-to-end tests.
2016-12-02 zhihuang@webrtc.org Modify the parameter type of PeerConnectionObserver callback OnAddTrack.
2016-12-02 deadbeef@webrtc.org Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )
2016-12-02 sergeyu@chromium.org Fix exponential probing in ProbeController.
2016-12-02 asapersson@webrtc.org Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
2016-12-02 stefan@webrtc.org Use different restrictions of acked bitrate lag depending on operating point.
2016-12-02 sprang@webrtc.org Wire up rtcp xr target bitrate on receive side.
2016-12-02 charujain@webrtc.org Added tool for reference less video analysis (go/refless-video-analysis)
2016-12-02 nisse@webrtc.org New gn target video_frame_api.
2016-12-02 kjellander@webrtc.org Remove xdisplaycheck
2016-12-02 howtofly@gmail.com fix coding and documentary ambiguity in AimdRateControl::TimeToReduceFurther.
2016-12-02 henrik.lundin@webrtc.org Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan
2016-12-02 magjed@webrtc.org VP8DecoderImpl: Fix uninitialized memory crash
2016-12-02 ossu@webrtc.org Deprecated SetAudioPacketSize from RTPSender and removed calls to it.
2016-12-02 henrik.lundin@webrtc.org Remove API-related #defines from voice_engine_configurations.h
2016-12-02 kthelgason@webrtc.org Sanity check parsed QP values from H264 bitstream
2016-12-01 deadbeef@webrtc.org In end-to-end PeerConnection tests, allow video to be downscaled.
2016-12-01 deadbeef@webrtc.org Revert of Disabled flaky P2PTestConductor tests on ASAN and MSAN. (patchset #1 id:1 of https://codereview.webrtc.org/2539103002/ )
2016-12-01 deadbeef@webrtc.org Relaxing timeouts for TestMediaMonitor.
2016-12-01 deadbeef@webrtc.org Relaxing DCHECK for packets sent before SRTP is enabled.
2016-12-01 henrik.lundin@webrtc.org Add linearly spaced counting histograms
2016-12-01 danilchap@webrtc.org Cleanup RtpHeaderExtensionMap removing use of two legacy functions
2016-12-01 terelius@webrtc.org Remove bitrate cap for AdaptiveVideoSource and increase other caps to 25 Mbps.
2016-12-01 sprang@webrtc.org Wire up BitrateAllocation to be sent as RTCP TargetBitrate

TBR=
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/2549043003
Cr-Commit-Position: refs/heads/master@{#436254}

DEPS

@@ -224,7 +224,7 @@ 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' + '@' + '42ad2d474da82fbcb90a432d463a765dee22fad1', # commit position 15358
+    Var('chromium_git') + '/external/webrtc/trunk/webrtc.git' + '@' + '6926a7dafe23d323971945fdf298c26a7aa411a5', # commit position 15399
 
   'src/third_party/openmax_dl':
     Var('chromium_git') + '/external/webrtc/deps/third_party/openmax.git' + '@' +  Var('openmax_dl_revision'),