Roll WebRTC 16045:16087 (20 commits)

Roll WebRTC 16045:16087 (20 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/1638114..cda50c3

$ git log 1638114..cda50c3 --date=short --no-merges --format=%ad %ae %s
2017-01-16 nisse@webrtc.org Delete unused file faketaskrunner.h.
2017-01-16 henrikg@webrtc.org Revert of Remove the IceTransportInternal2. (patchset #1 id:1 of https://codereview.webrtc.org/2632563002/ )
2017-01-15 zhihuang@webrtc.org Remove the IceTransportInternal2.
2017-01-15 ivoc@webrtc.org Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-13 deadbeef@webrtc.org Stop using deprecated constraints-based version of CreateAudioSource.
2017-01-13 deadbeef@webrtc.org Provide better message for when RTCP mux "require" policy is triggered.
2017-01-13 terelius@webrtc.org Rename base/analytics/ to base/numerics/
2017-01-13 brandtr@webrtc.org Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.
2017-01-13 sprang@webrtc.org Signal target bitrate only for screenshare streams
2017-01-13 brandtr@webrtc.org Explicitly only add transport-cc RTCP feedback param to default FlexFEC codec.
2017-01-13 magjed@webrtc.org Remove use of PCH files in webrtc/sdk/objc
2017-01-13 nisse@webrtc.org Replace left-over ASSERTs in httpcommon.h and webrtcsession.cc.
2017-01-13 sakal@webrtc.org Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
2017-01-13 michaelt@webrtc.org Log audio network adapter decisions in event log.
2017-01-13 michaelt@webrtc.org Pass event log to ANA.
2017-01-13 kthelgason@webrtc.org Update ice server provider response format in iOS AppRTCMobile
2017-01-13 palmkvist@webrtc.org Add UMA stats to bad call detection.
2017-01-13 perkj@webrtc.org Remove WebRtcVideoSendStream2::VideoSink inheritance. Remove sending black frame on source removal.
2017-01-13 sakal@webrtc.org Drop dependency to Chromium in WebRtcJniBootTest.
2017-01-13 nisse@webrtc.org Delete unused rtpdump code in media/base.

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/2634923002
Cr-Commit-Position: refs/heads/master@{#443865}
Committed: https://chromium.googlesource.com/chromium/src/+/980d3ef4c00833fdccfdfe4475518c1777e9f667

[Henrik Grunell, 2017-01-16 09:04:38]

The CQ bit was checked by grunell@chromium.org

[commit-bot: I haz the power, 2017-01-16 09:04:52]

CQ is trying da patch. Follow status at
 
https://chromium-cq-status.appspot.com/v2/patch-status/codereview.chromium.or...

[commit-bot: I haz the power, 2017-01-16 09:59:30]

CQ is committing da patch.

Bot data: {"patchset_id": 1, "attempt_start_ts": 1484557478454630, "parent_rev":
"67f8202c63cc39ca0a4c3bd444158a7587b9bb4d", "commit_rev":
"980d3ef4c00833fdccfdfe4475518c1777e9f667"}

[commit-bot: I haz the power, 2017-01-16 10:00:00]

Description was changed from

==========
Roll WebRTC 16045:16087 (20 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/16381...

$ git log 1638114..cda50c3 --date=short --no-merges --format=%ad %ae %s
2017-01-16 nisse@webrtc.org Delete unused file faketaskrunner.h.
2017-01-16 henrikg@webrtc.org Revert of Remove the IceTransportInternal2.
(patchset #1 id:1 of https://codereview.webrtc.org/2632563002/ )
2017-01-15 zhihuang@webrtc.org Remove the IceTransportInternal2.
2017-01-15 ivoc@webrtc.org Added a new echo likelihood stat that reports the
maximum value from a previous time period.
2017-01-13 deadbeef@webrtc.org Stop using deprecated constraints-based version
of CreateAudioSource.
2017-01-13 deadbeef@webrtc.org Provide better message for when RTCP mux
"require" policy is triggered.
2017-01-13 terelius@webrtc.org Rename base/analytics/ to base/numerics/
2017-01-13 brandtr@webrtc.org Make |rtcp_send_transport| mandatory in
FlexfecReceiveStream::Config.
2017-01-13 sprang@webrtc.org Signal target bitrate only for screenshare streams
2017-01-13 brandtr@webrtc.org Explicitly only add transport-cc RTCP feedback
param to default FlexFEC codec.
2017-01-13 magjed@webrtc.org Remove use of PCH files in webrtc/sdk/objc
2017-01-13 nisse@webrtc.org Replace left-over ASSERTs in httpcommon.h and
webrtcsession.cc.
2017-01-13 sakal@webrtc.org Revert of Log audio network adapter decisions in
event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/
)
2017-01-13 michaelt@webrtc.org Log audio network adapter decisions in event log.
2017-01-13 michaelt@webrtc.org Pass event log to ANA.
2017-01-13 kthelgason@webrtc.org Update ice server provider response format in
iOS AppRTCMobile
2017-01-13 palmkvist@webrtc.org Add UMA stats to bad call detection.
2017-01-13 perkj@webrtc.org Remove WebRtcVideoSendStream2::VideoSink
inheritance. Remove sending black frame on source removal.
2017-01-13 sakal@webrtc.org Drop dependency to Chromium in WebRtcJniBootTest.
2017-01-13 nisse@webrtc.org Delete unused rtpdump code in media/base.

TBR=
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng
BUG=
==========

to

==========
Roll WebRTC 16045:16087 (20 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/16381...

$ git log 1638114..cda50c3 --date=short --no-merges --format=%ad %ae %s
2017-01-16 nisse@webrtc.org Delete unused file faketaskrunner.h.
2017-01-16 henrikg@webrtc.org Revert of Remove the IceTransportInternal2.
(patchset #1 id:1 of https://codereview.webrtc.org/2632563002/ )
2017-01-15 zhihuang@webrtc.org Remove the IceTransportInternal2.
2017-01-15 ivoc@webrtc.org Added a new echo likelihood stat that reports the
maximum value from a previous time period.
2017-01-13 deadbeef@webrtc.org Stop using deprecated constraints-based version
of CreateAudioSource.
2017-01-13 deadbeef@webrtc.org Provide better message for when RTCP mux
"require" policy is triggered.
2017-01-13 terelius@webrtc.org Rename base/analytics/ to base/numerics/
2017-01-13 brandtr@webrtc.org Make |rtcp_send_transport| mandatory in
FlexfecReceiveStream::Config.
2017-01-13 sprang@webrtc.org Signal target bitrate only for screenshare streams
2017-01-13 brandtr@webrtc.org Explicitly only add transport-cc RTCP feedback
param to default FlexFEC codec.
2017-01-13 magjed@webrtc.org Remove use of PCH files in webrtc/sdk/objc
2017-01-13 nisse@webrtc.org Replace left-over ASSERTs in httpcommon.h and
webrtcsession.cc.
2017-01-13 sakal@webrtc.org Revert of Log audio network adapter decisions in
event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/
)
2017-01-13 michaelt@webrtc.org Log audio network adapter decisions in event log.
2017-01-13 michaelt@webrtc.org Pass event log to ANA.
2017-01-13 kthelgason@webrtc.org Update ice server provider response format in
iOS AppRTCMobile
2017-01-13 palmkvist@webrtc.org Add UMA stats to bad call detection.
2017-01-13 perkj@webrtc.org Remove WebRtcVideoSendStream2::VideoSink
inheritance. Remove sending black frame on source removal.
2017-01-13 sakal@webrtc.org Drop dependency to Chromium in WebRtcJniBootTest.
2017-01-13 nisse@webrtc.org Delete unused rtpdump code in media/base.

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/2634923002
Cr-Commit-Position: refs/heads/master@{#443865}
Committed:
https://chromium.googlesource.com/chromium/src/+/980d3ef4c00833fdccfdfe447551...
==========

[commit-bot: I haz the power, 2017-01-16 10:00:01]

Committed patchset #1 (id:1) as
https://chromium.googlesource.com/chromium/src/+/980d3ef4c00833fdccfdfe447551...