Roll WebRTC 13830:13857 (26 commits)

Roll WebRTC 13830:13857 (26 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/50a5c3e..79fe0f4

$ git log 50a5c3e..79fe0f4 --date=short --no-merges --format=%ad %ae %s
2016-08-22 deadbeef@webrtc.org Fixing TSan data race warning in video end-to-end tests.
2016-08-22 deadbeef@webrtc.org Some cleanup in BaseChannel RTCP mux code.
2016-08-22 henrik.lundin@webrtc.org Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine
2016-08-22 deadbeef@webrtc.org Adding deadbeef and honghaiz as owners of webrtc/pc.
2016-08-22 peah@webrtc.org Removed the deactivation of the level controller when there is a built-in AGC available
2016-08-22 terelius@webrtc.org Method to parse event log directly from a string.
2016-08-22 ehmaldonado@webrtc.org Add gtest as a dependency for neteq_quality_test_support.
2016-08-22 stefan@webrtc.org Fix issue where the number of packets reported in tests/simulations sometimes are negative.
2016-08-22 kwiberg@webrtc.org Fix trivial lint errors in FileRecorder and FilePlayer
2016-08-22 danilchap@webrtc.org Style cleanup in UpdateTmmbr: function names style updated, unused return type removed. Comment style fixed, redundant comments removed. pass-by-pointer parameter changed to pass-by-value because can't be nullptr any more.
2016-08-22 kwiberg@webrtc.org WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left shift of 1 by 31 overflows)
2016-08-22 danilchap@webrtc.org [rtcp] TransportFeedback adjusted to match other rtcp packets: Derived from rtcp::Rtpfb instead of directly from RtcpPacket Does not depend on RTCPUtility. Parse function takes CommonHeader. TransportFeedback::BlockLength fixed to match size used by Create
2016-08-22 henrika@webrtc.org [Reland] Cleanup of the AudioDeviceBuffer class.
2016-08-22 kjellander@webrtc.org Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ )
2016-08-22 nisse@webrtc.org WebRtcVideoFrame constructor without transport_frame_id.
2016-08-22 danilchap@webrtc.org Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData
2016-08-22 vopatop.skam@gmail.com iOS: add PlistBuddy location to path to avoid build errors
2016-08-22 peah@webrtc.org Disable the software noise suppressor on iOS devices as that functionality is always present in the hardware.
2016-08-22 stefan@webrtc.org Add pps id and sps id parsing to the h.264 depacketizer.
2016-08-22 sakal@webrtc.org Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ )
2016-08-21 arlolra@gmail.com Add field_trial_default dependency to libjingle_peerconnection
2016-08-20 magjed@webrtc.org iOS H264VideoToolBoxEncoder: Stop scaling native CVPixelBuffers
2016-08-19 henrika@webrtc.org Revert of Cleanup of the AudioDeviceBuffer class (patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
2016-08-19 henrika@webrtc.org Cleanup of the AudioDeviceBuffer class.
2016-08-19 danilchap@webrtc.org Reformat rtcp_receiver git cl format --full
2016-08-19 ehmaldonado@webrtc.org Refactor neteq_test_support.

TBR=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=

Committed: https://crrev.com/2693f13c7b19b47bd81583990c540cf607801f36
Cr-Commit-Position: refs/heads/master@{#413694}

[sakal-chromium, 2016-08-23 06:54:07]

The CQ bit was checked by sakal@chromium.org

[commit-bot: I haz the power, 2016-08-23 06:54:21]

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, 2016-08-23 06:54:22]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-08-23 06:54:22]

No L-G-T-M from a valid reviewer yet. 
CQ run can only be started by full committers or once the patch has
received an L-G-T-M from a full committer.
Even if an L-G-T-M may have been provided, it was from a non-committer,
_not_ a full super star committer.
See http://www.chromium.org/getting-involved/become-a-committer
Note that this has nothing to do with OWNERS files.

[sakal-chromium, 2016-08-23 06:59:46]

Description was changed from

==========
Roll WebRTC 13830:13857 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/50a5c...

$ git log 50a5c3e..79fe0f4 --date=short --no-merges --format=%ad %ae %s
2016-08-22 deadbeef@webrtc.org Fixing TSan data race warning in video end-to-end
tests.
2016-08-22 deadbeef@webrtc.org Some cleanup in BaseChannel RTCP mux code.
2016-08-22 henrik.lundin@webrtc.org Add NetEq::FilteredCurrentDelayMs() and use
it in VoiceEngine
2016-08-22 deadbeef@webrtc.org Adding deadbeef and honghaiz as owners of
webrtc/pc.
2016-08-22 peah@webrtc.org Removed the deactivation of the level controller when
there is a built-in AGC available
2016-08-22 terelius@webrtc.org Method to parse event log directly from a string.
2016-08-22 ehmaldonado@webrtc.org Add gtest as a dependency for
neteq_quality_test_support.
2016-08-22 stefan@webrtc.org Fix issue where the number of packets reported in
tests/simulations sometimes are negative.
2016-08-22 kwiberg@webrtc.org Fix trivial lint errors in FileRecorder and
FilePlayer
2016-08-22 danilchap@webrtc.org Style cleanup in UpdateTmmbr: function names
style updated, unused return type removed. Comment style fixed, redundant
comments removed. pass-by-pointer parameter changed to pass-by-value because
can't be nullptr any more.
2016-08-22 kwiberg@webrtc.org WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left
shift of 1 by 31 overflows)
2016-08-22 danilchap@webrtc.org [rtcp] TransportFeedback adjusted to match other
rtcp packets: Derived from rtcp::Rtpfb instead of directly from RtcpPacket Does
not depend on RTCPUtility. Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create
2016-08-22 henrika@webrtc.org [Reland] Cleanup of the AudioDeviceBuffer class.
2016-08-22 kjellander@webrtc.org Revert of Add pps id and sps id parsing to the
h.264 depacketizer. (patchset #5 id:80001 of
https://codereview.webrtc.org/2238253002/ )
2016-08-22 nisse@webrtc.org WebRtcVideoFrame constructor without
transport_frame_id.
2016-08-22 danilchap@webrtc.org Move RTP timestamp calculation from
BuildRTPheader to SendOutgoingData
2016-08-22 vopatop.skam@gmail.com iOS: add PlistBuddy location to path to avoid
build errors
2016-08-22 peah@webrtc.org Disable the software noise suppressor on iOS devices
as that functionality is always present in the hardware.
2016-08-22 stefan@webrtc.org Add pps id and sps id parsing to the h.264
depacketizer.
2016-08-22 sakal@webrtc.org Revert of Add field_trial_default dependency to
libjingle_peerconnection (patchset #3 id:40001 of
https://codereview.webrtc.org/2120673004/ )
2016-08-21 arlolra@gmail.com Add field_trial_default dependency to
libjingle_peerconnection
2016-08-20 magjed@webrtc.org iOS H264VideoToolBoxEncoder: Stop scaling native
CVPixelBuffers
2016-08-19 henrika@webrtc.org Revert of Cleanup of the AudioDeviceBuffer class
(patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
2016-08-19 henrika@webrtc.org Cleanup of the AudioDeviceBuffer class.
2016-08-19 danilchap@webrtc.org Reformat rtcp_receiver git cl format --full
2016-08-19 ehmaldonado@webrtc.org Refactor neteq_test_support.

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 13830:13857 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/50a5c...

$ git log 50a5c3e..79fe0f4 --date=short --no-merges --format=%ad %ae %s
2016-08-22 deadbeef@webrtc.org Fixing TSan data race warning in video end-to-end
tests.
2016-08-22 deadbeef@webrtc.org Some cleanup in BaseChannel RTCP mux code.
2016-08-22 henrik.lundin@webrtc.org Add NetEq::FilteredCurrentDelayMs() and use
it in VoiceEngine
2016-08-22 deadbeef@webrtc.org Adding deadbeef and honghaiz as owners of
webrtc/pc.
2016-08-22 peah@webrtc.org Removed the deactivation of the level controller when
there is a built-in AGC available
2016-08-22 terelius@webrtc.org Method to parse event log directly from a string.
2016-08-22 ehmaldonado@webrtc.org Add gtest as a dependency for
neteq_quality_test_support.
2016-08-22 stefan@webrtc.org Fix issue where the number of packets reported in
tests/simulations sometimes are negative.
2016-08-22 kwiberg@webrtc.org Fix trivial lint errors in FileRecorder and
FilePlayer
2016-08-22 danilchap@webrtc.org Style cleanup in UpdateTmmbr: function names
style updated, unused return type removed. Comment style fixed, redundant
comments removed. pass-by-pointer parameter changed to pass-by-value because
can't be nullptr any more.
2016-08-22 kwiberg@webrtc.org WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left
shift of 1 by 31 overflows)
2016-08-22 danilchap@webrtc.org [rtcp] TransportFeedback adjusted to match other
rtcp packets: Derived from rtcp::Rtpfb instead of directly from RtcpPacket Does
not depend on RTCPUtility. Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create
2016-08-22 henrika@webrtc.org [Reland] Cleanup of the AudioDeviceBuffer class.
2016-08-22 kjellander@webrtc.org Revert of Add pps id and sps id parsing to the
h.264 depacketizer. (patchset #5 id:80001 of
https://codereview.webrtc.org/2238253002/ )
2016-08-22 nisse@webrtc.org WebRtcVideoFrame constructor without
transport_frame_id.
2016-08-22 danilchap@webrtc.org Move RTP timestamp calculation from
BuildRTPheader to SendOutgoingData
2016-08-22 vopatop.skam@gmail.com iOS: add PlistBuddy location to path to avoid
build errors
2016-08-22 peah@webrtc.org Disable the software noise suppressor on iOS devices
as that functionality is always present in the hardware.
2016-08-22 stefan@webrtc.org Add pps id and sps id parsing to the h.264
depacketizer.
2016-08-22 sakal@webrtc.org Revert of Add field_trial_default dependency to
libjingle_peerconnection (patchset #3 id:40001 of
https://codereview.webrtc.org/2120673004/ )
2016-08-21 arlolra@gmail.com Add field_trial_default dependency to
libjingle_peerconnection
2016-08-20 magjed@webrtc.org iOS H264VideoToolBoxEncoder: Stop scaling native
CVPixelBuffers
2016-08-19 henrika@webrtc.org Revert of Cleanup of the AudioDeviceBuffer class
(patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
2016-08-19 henrika@webrtc.org Cleanup of the AudioDeviceBuffer class.
2016-08-19 danilchap@webrtc.org Reformat rtcp_receiver git cl format --full
2016-08-19 ehmaldonado@webrtc.org Refactor neteq_test_support.

TBR=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=
==========

[sakal-chromium, 2016-08-23 06:59:47]

sakal@chromium.org changed reviewers:
+ magjed@chromium.org

[sakal-chromium, 2016-08-23 06:59:51]

The CQ bit was checked by sakal@chromium.org

[commit-bot: I haz the power, 2016-08-23 07:00:09]

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, 2016-08-23 07:00:10]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2016-08-23 07:00:12]

No L-G-T-M from a valid reviewer yet. 
CQ run can only be started by full committers or once the patch has
received an L-G-T-M from a full committer.
Even if an L-G-T-M may have been provided, it was from a non-committer,
_not_ a full super star committer.
See http://www.chromium.org/getting-involved/become-a-committer
Note that this has nothing to do with OWNERS files.

[magjed_chromium, 2016-08-23 07:01:08]

The CQ bit was checked by magjed@chromium.org

[magjed_chromium, 2016-08-23 07:01:08]

lgtm

[sakal-chromium, 2016-08-23 07:01:31]

The CQ bit was checked by sakal@chromium.org

[commit-bot: I haz the power, 2016-08-23 07:01:57]

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, 2016-08-23 08:29:44]

Description was changed from

==========
Roll WebRTC 13830:13857 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/50a5c...

$ git log 50a5c3e..79fe0f4 --date=short --no-merges --format=%ad %ae %s
2016-08-22 deadbeef@webrtc.org Fixing TSan data race warning in video end-to-end
tests.
2016-08-22 deadbeef@webrtc.org Some cleanup in BaseChannel RTCP mux code.
2016-08-22 henrik.lundin@webrtc.org Add NetEq::FilteredCurrentDelayMs() and use
it in VoiceEngine
2016-08-22 deadbeef@webrtc.org Adding deadbeef and honghaiz as owners of
webrtc/pc.
2016-08-22 peah@webrtc.org Removed the deactivation of the level controller when
there is a built-in AGC available
2016-08-22 terelius@webrtc.org Method to parse event log directly from a string.
2016-08-22 ehmaldonado@webrtc.org Add gtest as a dependency for
neteq_quality_test_support.
2016-08-22 stefan@webrtc.org Fix issue where the number of packets reported in
tests/simulations sometimes are negative.
2016-08-22 kwiberg@webrtc.org Fix trivial lint errors in FileRecorder and
FilePlayer
2016-08-22 danilchap@webrtc.org Style cleanup in UpdateTmmbr: function names
style updated, unused return type removed. Comment style fixed, redundant
comments removed. pass-by-pointer parameter changed to pass-by-value because
can't be nullptr any more.
2016-08-22 kwiberg@webrtc.org WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left
shift of 1 by 31 overflows)
2016-08-22 danilchap@webrtc.org [rtcp] TransportFeedback adjusted to match other
rtcp packets: Derived from rtcp::Rtpfb instead of directly from RtcpPacket Does
not depend on RTCPUtility. Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create
2016-08-22 henrika@webrtc.org [Reland] Cleanup of the AudioDeviceBuffer class.
2016-08-22 kjellander@webrtc.org Revert of Add pps id and sps id parsing to the
h.264 depacketizer. (patchset #5 id:80001 of
https://codereview.webrtc.org/2238253002/ )
2016-08-22 nisse@webrtc.org WebRtcVideoFrame constructor without
transport_frame_id.
2016-08-22 danilchap@webrtc.org Move RTP timestamp calculation from
BuildRTPheader to SendOutgoingData
2016-08-22 vopatop.skam@gmail.com iOS: add PlistBuddy location to path to avoid
build errors
2016-08-22 peah@webrtc.org Disable the software noise suppressor on iOS devices
as that functionality is always present in the hardware.
2016-08-22 stefan@webrtc.org Add pps id and sps id parsing to the h.264
depacketizer.
2016-08-22 sakal@webrtc.org Revert of Add field_trial_default dependency to
libjingle_peerconnection (patchset #3 id:40001 of
https://codereview.webrtc.org/2120673004/ )
2016-08-21 arlolra@gmail.com Add field_trial_default dependency to
libjingle_peerconnection
2016-08-20 magjed@webrtc.org iOS H264VideoToolBoxEncoder: Stop scaling native
CVPixelBuffers
2016-08-19 henrika@webrtc.org Revert of Cleanup of the AudioDeviceBuffer class
(patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
2016-08-19 henrika@webrtc.org Cleanup of the AudioDeviceBuffer class.
2016-08-19 danilchap@webrtc.org Reformat rtcp_receiver git cl format --full
2016-08-19 ehmaldonado@webrtc.org Refactor neteq_test_support.

TBR=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=
==========

to

==========
Roll WebRTC 13830:13857 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/50a5c...

$ git log 50a5c3e..79fe0f4 --date=short --no-merges --format=%ad %ae %s
2016-08-22 deadbeef@webrtc.org Fixing TSan data race warning in video end-to-end
tests.
2016-08-22 deadbeef@webrtc.org Some cleanup in BaseChannel RTCP mux code.
2016-08-22 henrik.lundin@webrtc.org Add NetEq::FilteredCurrentDelayMs() and use
it in VoiceEngine
2016-08-22 deadbeef@webrtc.org Adding deadbeef and honghaiz as owners of
webrtc/pc.
2016-08-22 peah@webrtc.org Removed the deactivation of the level controller when
there is a built-in AGC available
2016-08-22 terelius@webrtc.org Method to parse event log directly from a string.
2016-08-22 ehmaldonado@webrtc.org Add gtest as a dependency for
neteq_quality_test_support.
2016-08-22 stefan@webrtc.org Fix issue where the number of packets reported in
tests/simulations sometimes are negative.
2016-08-22 kwiberg@webrtc.org Fix trivial lint errors in FileRecorder and
FilePlayer
2016-08-22 danilchap@webrtc.org Style cleanup in UpdateTmmbr: function names
style updated, unused return type removed. Comment style fixed, redundant
comments removed. pass-by-pointer parameter changed to pass-by-value because
can't be nullptr any more.
2016-08-22 kwiberg@webrtc.org WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left
shift of 1 by 31 overflows)
2016-08-22 danilchap@webrtc.org [rtcp] TransportFeedback adjusted to match other
rtcp packets: Derived from rtcp::Rtpfb instead of directly from RtcpPacket Does
not depend on RTCPUtility. Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create
2016-08-22 henrika@webrtc.org [Reland] Cleanup of the AudioDeviceBuffer class.
2016-08-22 kjellander@webrtc.org Revert of Add pps id and sps id parsing to the
h.264 depacketizer. (patchset #5 id:80001 of
https://codereview.webrtc.org/2238253002/ )
2016-08-22 nisse@webrtc.org WebRtcVideoFrame constructor without
transport_frame_id.
2016-08-22 danilchap@webrtc.org Move RTP timestamp calculation from
BuildRTPheader to SendOutgoingData
2016-08-22 vopatop.skam@gmail.com iOS: add PlistBuddy location to path to avoid
build errors
2016-08-22 peah@webrtc.org Disable the software noise suppressor on iOS devices
as that functionality is always present in the hardware.
2016-08-22 stefan@webrtc.org Add pps id and sps id parsing to the h.264
depacketizer.
2016-08-22 sakal@webrtc.org Revert of Add field_trial_default dependency to
libjingle_peerconnection (patchset #3 id:40001 of
https://codereview.webrtc.org/2120673004/ )
2016-08-21 arlolra@gmail.com Add field_trial_default dependency to
libjingle_peerconnection
2016-08-20 magjed@webrtc.org iOS H264VideoToolBoxEncoder: Stop scaling native
CVPixelBuffers
2016-08-19 henrika@webrtc.org Revert of Cleanup of the AudioDeviceBuffer class
(patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
2016-08-19 henrika@webrtc.org Cleanup of the AudioDeviceBuffer class.
2016-08-19 danilchap@webrtc.org Reformat rtcp_receiver git cl format --full
2016-08-19 ehmaldonado@webrtc.org Refactor neteq_test_support.

TBR=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=
==========

[commit-bot: I haz the power, 2016-08-23 08:29:48]

Committed patchset #1 (id:1)

[commit-bot: I haz the power, 2016-08-23 08:31:33]

Description was changed from

==========
Roll WebRTC 13830:13857 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/50a5c...

$ git log 50a5c3e..79fe0f4 --date=short --no-merges --format=%ad %ae %s
2016-08-22 deadbeef@webrtc.org Fixing TSan data race warning in video end-to-end
tests.
2016-08-22 deadbeef@webrtc.org Some cleanup in BaseChannel RTCP mux code.
2016-08-22 henrik.lundin@webrtc.org Add NetEq::FilteredCurrentDelayMs() and use
it in VoiceEngine
2016-08-22 deadbeef@webrtc.org Adding deadbeef and honghaiz as owners of
webrtc/pc.
2016-08-22 peah@webrtc.org Removed the deactivation of the level controller when
there is a built-in AGC available
2016-08-22 terelius@webrtc.org Method to parse event log directly from a string.
2016-08-22 ehmaldonado@webrtc.org Add gtest as a dependency for
neteq_quality_test_support.
2016-08-22 stefan@webrtc.org Fix issue where the number of packets reported in
tests/simulations sometimes are negative.
2016-08-22 kwiberg@webrtc.org Fix trivial lint errors in FileRecorder and
FilePlayer
2016-08-22 danilchap@webrtc.org Style cleanup in UpdateTmmbr: function names
style updated, unused return type removed. Comment style fixed, redundant
comments removed. pass-by-pointer parameter changed to pass-by-value because
can't be nullptr any more.
2016-08-22 kwiberg@webrtc.org WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left
shift of 1 by 31 overflows)
2016-08-22 danilchap@webrtc.org [rtcp] TransportFeedback adjusted to match other
rtcp packets: Derived from rtcp::Rtpfb instead of directly from RtcpPacket Does
not depend on RTCPUtility. Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create
2016-08-22 henrika@webrtc.org [Reland] Cleanup of the AudioDeviceBuffer class.
2016-08-22 kjellander@webrtc.org Revert of Add pps id and sps id parsing to the
h.264 depacketizer. (patchset #5 id:80001 of
https://codereview.webrtc.org/2238253002/ )
2016-08-22 nisse@webrtc.org WebRtcVideoFrame constructor without
transport_frame_id.
2016-08-22 danilchap@webrtc.org Move RTP timestamp calculation from
BuildRTPheader to SendOutgoingData
2016-08-22 vopatop.skam@gmail.com iOS: add PlistBuddy location to path to avoid
build errors
2016-08-22 peah@webrtc.org Disable the software noise suppressor on iOS devices
as that functionality is always present in the hardware.
2016-08-22 stefan@webrtc.org Add pps id and sps id parsing to the h.264
depacketizer.
2016-08-22 sakal@webrtc.org Revert of Add field_trial_default dependency to
libjingle_peerconnection (patchset #3 id:40001 of
https://codereview.webrtc.org/2120673004/ )
2016-08-21 arlolra@gmail.com Add field_trial_default dependency to
libjingle_peerconnection
2016-08-20 magjed@webrtc.org iOS H264VideoToolBoxEncoder: Stop scaling native
CVPixelBuffers
2016-08-19 henrika@webrtc.org Revert of Cleanup of the AudioDeviceBuffer class
(patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
2016-08-19 henrika@webrtc.org Cleanup of the AudioDeviceBuffer class.
2016-08-19 danilchap@webrtc.org Reformat rtcp_receiver git cl format --full
2016-08-19 ehmaldonado@webrtc.org Refactor neteq_test_support.

TBR=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=
==========

to

==========
Roll WebRTC 13830:13857 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/50a5c...

$ git log 50a5c3e..79fe0f4 --date=short --no-merges --format=%ad %ae %s
2016-08-22 deadbeef@webrtc.org Fixing TSan data race warning in video end-to-end
tests.
2016-08-22 deadbeef@webrtc.org Some cleanup in BaseChannel RTCP mux code.
2016-08-22 henrik.lundin@webrtc.org Add NetEq::FilteredCurrentDelayMs() and use
it in VoiceEngine
2016-08-22 deadbeef@webrtc.org Adding deadbeef and honghaiz as owners of
webrtc/pc.
2016-08-22 peah@webrtc.org Removed the deactivation of the level controller when
there is a built-in AGC available
2016-08-22 terelius@webrtc.org Method to parse event log directly from a string.
2016-08-22 ehmaldonado@webrtc.org Add gtest as a dependency for
neteq_quality_test_support.
2016-08-22 stefan@webrtc.org Fix issue where the number of packets reported in
tests/simulations sometimes are negative.
2016-08-22 kwiberg@webrtc.org Fix trivial lint errors in FileRecorder and
FilePlayer
2016-08-22 danilchap@webrtc.org Style cleanup in UpdateTmmbr: function names
style updated, unused return type removed. Comment style fixed, redundant
comments removed. pass-by-pointer parameter changed to pass-by-value because
can't be nullptr any more.
2016-08-22 kwiberg@webrtc.org WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left
shift of 1 by 31 overflows)
2016-08-22 danilchap@webrtc.org [rtcp] TransportFeedback adjusted to match other
rtcp packets: Derived from rtcp::Rtpfb instead of directly from RtcpPacket Does
not depend on RTCPUtility. Parse function takes CommonHeader.
TransportFeedback::BlockLength fixed to match size used by Create
2016-08-22 henrika@webrtc.org [Reland] Cleanup of the AudioDeviceBuffer class.
2016-08-22 kjellander@webrtc.org Revert of Add pps id and sps id parsing to the
h.264 depacketizer. (patchset #5 id:80001 of
https://codereview.webrtc.org/2238253002/ )
2016-08-22 nisse@webrtc.org WebRtcVideoFrame constructor without
transport_frame_id.
2016-08-22 danilchap@webrtc.org Move RTP timestamp calculation from
BuildRTPheader to SendOutgoingData
2016-08-22 vopatop.skam@gmail.com iOS: add PlistBuddy location to path to avoid
build errors
2016-08-22 peah@webrtc.org Disable the software noise suppressor on iOS devices
as that functionality is always present in the hardware.
2016-08-22 stefan@webrtc.org Add pps id and sps id parsing to the h.264
depacketizer.
2016-08-22 sakal@webrtc.org Revert of Add field_trial_default dependency to
libjingle_peerconnection (patchset #3 id:40001 of
https://codereview.webrtc.org/2120673004/ )
2016-08-21 arlolra@gmail.com Add field_trial_default dependency to
libjingle_peerconnection
2016-08-20 magjed@webrtc.org iOS H264VideoToolBoxEncoder: Stop scaling native
CVPixelBuffers
2016-08-19 henrika@webrtc.org Revert of Cleanup of the AudioDeviceBuffer class
(patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ )
2016-08-19 henrika@webrtc.org Cleanup of the AudioDeviceBuffer class.
2016-08-19 danilchap@webrtc.org Reformat rtcp_receiver git cl format --full
2016-08-19 ehmaldonado@webrtc.org Refactor neteq_test_support.

TBR=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=

Committed: https://crrev.com/2693f13c7b19b47bd81583990c540cf607801f36
Cr-Commit-Position: refs/heads/master@{#413694}
==========

[commit-bot: I haz the power, 2016-08-23 08:31:34]

Patchset 1 (id:??) landed as
https://crrev.com/2693f13c7b19b47bd81583990c540cf607801f36
Cr-Commit-Position: refs/heads/master@{#413694}