Roll WebRTC 15456:15513 (40 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/666ab2c..f23e926

$ git log 666ab2c..f23e926 --date=short --no-merges --format=%ad %ae %s
2016-12-09 pbos@webrtc.org Remove extra uses of basictypes.h.
2016-12-09 brandtr@webrtc.org Only store sequence numbers for media stream in FlexFEC end-to-end test.
2016-12-09 kthelgason@webrtc.org Disable failing perf test on Android.
2016-12-09 minyue@webrtc.org Modify JavaToStdString to allow ISO-8859-1 encoded strings.
2016-12-09 terelius@webrtc.org Implement Theil-Sen's method for fitting a line to noisy data (used in bandwidth estimation).
2016-12-09 hbos@webrtc.org RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector.
2016-12-09 hbos@webrtc.org RTCIceCandidatePairStats.requestsReceived defined by RTCStatsCollector.
2016-12-09 kthelgason@webrtc.org Reland of Bump up scaling limit for MediaCodec. (patchset #1 id:1 of https://codereview.webrtc.org/2562963002/ )
2016-12-09 peah@webrtc.org During AEC development, it is handy to be able to simulate different orders of the ProcessStream and ProcessReverseStream API calls.
2016-12-09 peah@webrtc.org When recreating a call based on an aecdump recording the nearend used is the one stored in the aecdump.
2016-12-09 asapersson@webrtc.org Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 kthelgason@webrtc.org Revert of Bump up scaling limit for MediaCodec. (patchset #3 id:40001 of https://codereview.webrtc.org/2566533002/ )
2016-12-09 kthelgason@webrtc.org Bump up scaling limit for MediaCodec.
2016-12-08 stefan@webrtc.org Fix issue with deprecated CongestionController interface not working.
2016-12-08 zijiehe@chromium.org Enable screen capturer tests for Linux / DirectX capturer / magnifier capturer
2016-12-08 ehmaldonado@webrtc.org Refactor webrtc/modules/video_{capture,coding} for GN check
2016-12-08 brandtr@webrtc.org Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled.
2016-12-08 asapersson@webrtc.org Do not update OnReceivedRtcpReceiverReport if report block list is empty (and rtt zero).
2016-12-08 kthelgason@webrtc.org Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ )
2016-12-08 nisse@webrtc.org Delete deprecated CongestionController constructor and packet_router method.
2016-12-08 kjellander@webrtc.org Re-enable disabled VideoProcessorIntegrationTest tests
2016-12-08 brandtr@webrtc.org Add FlexFEC settings toggle in Android AppRTCMobile.
2016-12-08 nisse@webrtc.org Simplify an always true condition.
2016-12-08 sakal@webrtc.org Fix error in VideoFileRenderer_nativeI420Scale.
2016-12-08 brandtr@webrtc.org Generalize FlexfecReceiveStream::Config.
2016-12-08 brandtr@webrtc.org Clean up FlexfecReceiveStream ctor signatures.
2016-12-08 ehmaldonado@webrtc.org Refactor webrtc/modules/audio_processing for GN check
2016-12-08 johan@webrtc.org Decode h264 fmtp sprop-parameter-sets to binary.
2016-12-08 aleloi@webrtc.org Injectable output rate calculater for AudioMixer.
2016-12-08 asapersson@webrtc.org Remove unused arguments and variable in MediaOptimization.
2016-12-08 kthelgason@webrtc.org Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ )
2016-12-08 kthelgason@webrtc.org Add ability to scale to arbitrary factors
2016-12-08 hta@webrtc.org Refactoring: Declare cricket::Codec constructors protected.
2016-12-08 magjed@webrtc.org Android classreferenceholder.h: Reorder function declaration keywords
2016-12-07 zijiehe@chromium.org Log BitBlt failure
2016-12-07 zhihuang@webrtc.org Create the Java Wrapper of RtpReceiverObserverInterface.
2016-12-07 kjellander@webrtc.org Refactor webrtc/{api,audio} and modules/audio_coding for GN check
2016-12-07 ossu@webrtc.org Moved call.h and most of api/call/* into call/
2016-12-07 ehmaldonado@webrtc.org Clean up redundant include of ../webrtc_overrides
2016-12-07 minyue@webrtc.org Adding OnReceivedOverhead to AudioEncoder.

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/2567643002
Cr-Commit-Position: refs/heads/master@{#437564}

DEPS

@@ -228,7 +228,7 @@ 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' + '@' + '666ab2c1173e5411c3f81ebb0277644c978d136a', # commit position 15456
+    Var('chromium_git') + '/external/webrtc/trunk/webrtc.git' + '@' + 'f23e9262fa5dc35022d9af4acabad3690c728135', # commit position 15513
 
   'src/third_party/openmax_dl':
     Var('chromium_git') + '/external/webrtc/deps/third_party/openmax.git' + '@' +  Var('openmax_dl_revision'),