Roll WebRTC 13633:13652 (13 commits)

Roll WebRTC 13633:13652 (13 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/518d267..1462743

$ git log 518d267..1462743 --date=short --no-merges --format=%ad %ae %s
2016-08-05 danilchap@webrtc.org TMMBRHelp::FindBoundingSet function cleaned it become static to clearly state input, output and temporary variables style updated with minor improvments to actual algorithm.
2016-08-04 tkchin@webrtc.org Fix case-sensitive build system errors.
2016-08-04 deadbeef@webrtc.org Revert of Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. (patchset #8 id:280001 of https://codereview.webrtc.org/2166873002/ )
2016-08-04 zhihuang@webrtc.org Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-04 terelius@webrtc.org Add fraction loss in visualization tool.
2016-08-04 sprang@webrtc.org Remove rtt limit on retransmission rate limiter
2016-08-04 ehmaldonado@webrtc.org Revert of Enable socketservers unittests on windows (patchset #7 id:260001 of https://codereview.webrtc.org/2199193004/ )
2016-08-04 sakal@webrtc.org Reland of GN: Create target for video_capture_tests on Linux. (patchset #1 id:1 of https://codereview.webrtc.org/2217463002/ )
2016-08-04 guidou@webrtc.org Revert of GN: Create target for video_capture_tests on Linux. (patchset #2 id:20001 of https://codereview.webrtc.org/2207463004/ )
2016-08-04 stefan@webrtc.org Tune BWE to be more sensitive on low capacity networks.
2016-08-04 sakal@webrtc.org GN: Create target for video_capture_tests on Linux.
2016-08-04 aleloi@webrtc.org Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
2016-08-04 jbauch@webrtc.org Add support for GCM cipher suites from RFC 7714.

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

BUG=

[ehmaldonado_chromium, 2016-08-05 11:57:35]

Description was changed from

==========
Roll WebRTC 13633:13652 (13 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/518d2...

$ git log 518d267..1462743 --date=short --no-merges --format=%ad %ae %s
2016-08-05 danilchap@webrtc.org TMMBRHelp::FindBoundingSet function cleaned it
become static to clearly state input, output and temporary variables style
updated with minor improvments to actual algorithm.
2016-08-04 tkchin@webrtc.org Fix case-sensitive build system errors.
2016-08-04 deadbeef@webrtc.org Revert of Modified PeerConnection and
WebRtcSession for end-to-end QuicDataChannel usage. (patchset #8 id:280001 of
https://codereview.webrtc.org/2166873002/ )
2016-08-04 zhihuang@webrtc.org Modified PeerConnection and WebRtcSession for
end-to-end QuicDataChannel usage.
2016-08-04 terelius@webrtc.org Add fraction loss in visualization tool.
2016-08-04 sprang@webrtc.org Remove rtt limit on retransmission rate limiter
2016-08-04 ehmaldonado@webrtc.org Revert of Enable socketservers unittests on
windows (patchset #7 id:260001 of https://codereview.webrtc.org/2199193004/ )
2016-08-04 sakal@webrtc.org Reland of GN: Create target for video_capture_tests
on Linux. (patchset #1 id:1 of https://codereview.webrtc.org/2217463002/ )
2016-08-04 guidou@webrtc.org Revert of GN: Create target for video_capture_tests
on Linux. (patchset #2 id:20001 of https://codereview.webrtc.org/2207463004/ )
2016-08-04 stefan@webrtc.org Tune BWE to be more sensitive on low capacity
networks.
2016-08-04 sakal@webrtc.org GN: Create target for video_capture_tests on Linux.
2016-08-04 aleloi@webrtc.org Removed calls to VoE::SetPlayout() from
WebRTCVoiceEngine.
2016-08-04 jbauch@webrtc.org Add support for GCM cipher suites from RFC 7714.

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

Sigh


This seems to be it.


Merge branch 'master' of https://chromium.googlesource.com/chromium/src


And another stash.


Stashing again.


Stashing again.


Stashing 2.


Merge branch 'master' of https://chromium.googlesource.com/chromium/src


Stashing.

BUG=
==========

to

==========
Roll WebRTC 13633:13652 (13 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/518d2...

$ git log 518d267..1462743 --date=short --no-merges --format=%ad %ae %s
2016-08-05 danilchap@webrtc.org TMMBRHelp::FindBoundingSet function cleaned it
become static to clearly state input, output and temporary variables style
updated with minor improvments to actual algorithm.
2016-08-04 tkchin@webrtc.org Fix case-sensitive build system errors.
2016-08-04 deadbeef@webrtc.org Revert of Modified PeerConnection and
WebRtcSession for end-to-end QuicDataChannel usage. (patchset #8 id:280001 of
https://codereview.webrtc.org/2166873002/ )
2016-08-04 zhihuang@webrtc.org Modified PeerConnection and WebRtcSession for
end-to-end QuicDataChannel usage.
2016-08-04 terelius@webrtc.org Add fraction loss in visualization tool.
2016-08-04 sprang@webrtc.org Remove rtt limit on retransmission rate limiter
2016-08-04 ehmaldonado@webrtc.org Revert of Enable socketservers unittests on
windows (patchset #7 id:260001 of https://codereview.webrtc.org/2199193004/ )
2016-08-04 sakal@webrtc.org Reland of GN: Create target for video_capture_tests
on Linux. (patchset #1 id:1 of https://codereview.webrtc.org/2217463002/ )
2016-08-04 guidou@webrtc.org Revert of GN: Create target for video_capture_tests
on Linux. (patchset #2 id:20001 of https://codereview.webrtc.org/2207463004/ )
2016-08-04 stefan@webrtc.org Tune BWE to be more sensitive on low capacity
networks.
2016-08-04 sakal@webrtc.org GN: Create target for video_capture_tests on Linux.
2016-08-04 aleloi@webrtc.org Removed calls to VoE::SetPlayout() from
WebRTCVoiceEngine.
2016-08-04 jbauch@webrtc.org Add support for GCM cipher suites from RFC 7714.

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

BUG=
==========

[ehmaldonado_chromium, 2016-08-05 12:00:09]

ehmaldonado@chromium.org changed reviewers:
+ kjellander@chromium.org

[ehmaldonado_chromium, 2016-08-05 12:00:10]

Not meant to be submitted.
An example of what the script generates.

[ehmaldonado_chromium, 2016-08-05 12:01:27]

The modifications in tools/roll_webrtc.py are there because I launched it from
the branch I was modifying it.

[kjellander_chromium, 2016-08-05 12:55:15]

On 2016/08/05 12:01:27, ehmaldonado_chromium wrote:
> The modifications in tools/roll_webrtc.py are there because I launched it from
> the branch I was modifying it.

I see. You can just edit the CL and set it as closed to keep it for the future
(but not do anything with it). That prevents it from showing up in reviewer's
queue.