Roll WebRTC 14746:14770 (23 commits)

Roll WebRTC 14746:14770 (23 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/7938471..697aba7

$ git log 7938471..697aba7 --date=short --no-merges --format=%ad %ae %s
2016-10-25 peah@webrtc.org Moved the AGC render sample queue into the audio processing module
2016-10-25 peah@webrtc.org Several subcomponents inside APM copy render audio from the render side to the capture side using the same pattern. Currently this is done independently for the submodules.
2016-10-25 hbos@webrtc.org RTCStats equality operator added.
2016-10-25 stefan@webrtc.org Reland of Deflake ChangingNetworkRoute test.
2016-10-25 ehmaldonado@webrtc.org Revert of Deflake ChangingNetworkRoute test. (patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ )
2016-10-25 stefan@webrtc.org Deflake ChangingNetworkRoute test.
2016-10-25 danilchap@webrtc.org Remove now unused code in RtpHeaderExtensionMap Remove functions to enumerate all extensions, Remove concept of the inactive extension. Decision if extension should be included into rtp header is done by rtp_sender GetTotalLengthInBytes now calculates all extension, included or not. That is used only for calculating how much space to reserve for fec. Since extension might suddenly be included in the next packet (which still might belong to same fec group), it is safer to calculate all registered extension.
2016-10-25 sprang@webrtc.org Make WebRTC compatible with OpenH264 v1.6.
2016-10-25 kjellander@webrtc.org Remove dead dependencies on xmllite and xmpp.
2016-10-25 magjed@webrtc.org codec_unittest.cc: Fix TEST vs TEST_F mismatch
2016-10-25 magjed@webrtc.org Revert of H264 codec: Check profile-level-id when matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
2016-10-25 nisse@webrtc.org Delete left-over file profiler_unittest.cc.
2016-10-25 nisse@webrtc.org Delete unused file screencastid.h.
2016-10-24 honghaiz@webrtc.org Fix more swarming test failures by using the fake clock or longer timeout. In the swarming test, the machines sometimes were blocked for 1-2 seconds without processing anything. This CL makes sure that 1 second timeout is only used with fake clock.
2016-10-24 kwiberg@webrtc.org Use relative names in GN to make Chromium happy
2016-10-24 sergeyu@chromium.org Update XServerPixelBuffer to handle errors returned from XGetImage().
2016-10-24 kwiberg@webrtc.org Stop using old AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 solenberg@webrtc.org Remove unused function implementations from FakeWebRtcVoiceEngine.
2016-10-24 deadbeef@webrtc.org Preventing TURN redirects to loopback addresses.
2016-10-24 terelius@webrtc.org Revert of Fix chromium-style warnings. (patchset #1 id:1 of https://codereview.webrtc.org/2400993002/ )
2016-10-24 hbos@webrtc.org rtcstats_objects.h updated with TODOs about stats not being collected or not being collected correctly.
2016-10-24 minyue@webrtc.org Simplifying audio network adaptor by moving receiver frame length range to ctor.
2016-10-24 ossu@webrtc.org NetEq now works with packets as values, rather than pointers.

TBR=
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/5a4427d288f832af7ffe39d4b16efa5b7b3901a5
Cr-Commit-Position: refs/heads/master@{#427625}

[sakal-chromium, 2016-10-25 13:03:36]

The CQ bit was checked by sakal@chromium.org

[commit-bot: I haz the power, 2016-10-25 13:03:52]

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, 2016-10-25 13:03:54]

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

[commit-bot: I haz the power, 2016-10-25 13:03:54]

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, 2016-10-25 14:03:15]

Description was changed from

==========
Roll WebRTC 14746:14770 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/79384...

$ git log 7938471..697aba7 --date=short --no-merges --format=%ad %ae %s
2016-10-25 peah@webrtc.org Moved the AGC render sample queue into the audio
processing module
2016-10-25 peah@webrtc.org Several subcomponents inside APM copy render audio
from the render side to the capture side using the same pattern. Currently this
is done independently for the submodules.
2016-10-25 hbos@webrtc.org RTCStats equality operator added.
2016-10-25 stefan@webrtc.org Reland of Deflake ChangingNetworkRoute test.
2016-10-25 ehmaldonado@webrtc.org Revert of Deflake ChangingNetworkRoute test.
(patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ )
2016-10-25 stefan@webrtc.org Deflake ChangingNetworkRoute test.
2016-10-25 danilchap@webrtc.org Remove now unused code in RtpHeaderExtensionMap
Remove functions to enumerate all extensions, Remove concept of the inactive
extension. Decision if extension should be included into rtp header is done by
rtp_sender GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec. Since
extension might suddenly be included in the next packet (which still might
belong to same fec group), it is safer to calculate all registered extension.
2016-10-25 sprang@webrtc.org Make WebRTC compatible with OpenH264 v1.6.
2016-10-25 kjellander@webrtc.org Remove dead dependencies on xmllite and xmpp.
2016-10-25 magjed@webrtc.org codec_unittest.cc: Fix TEST vs TEST_F mismatch
2016-10-25 magjed@webrtc.org Revert of H264 codec: Check profile-level-id when
matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
2016-10-25 nisse@webrtc.org Delete left-over file profiler_unittest.cc.
2016-10-25 nisse@webrtc.org Delete unused file screencastid.h.
2016-10-24 honghaiz@webrtc.org Fix more swarming test failures by using the fake
clock or longer timeout. In the swarming test, the machines sometimes were
blocked for 1-2 seconds without processing anything. This CL makes sure that 1
second timeout is only used with fake clock.
2016-10-24 kwiberg@webrtc.org Use relative names in GN to make Chromium happy
2016-10-24 sergeyu@chromium.org Update XServerPixelBuffer to handle errors
returned from XGetImage().
2016-10-24 kwiberg@webrtc.org Stop using old
AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 solenberg@webrtc.org Remove unused function implementations from
FakeWebRtcVoiceEngine.
2016-10-24 deadbeef@webrtc.org Preventing TURN redirects to loopback addresses.
2016-10-24 terelius@webrtc.org Revert of Fix chromium-style warnings. (patchset
#1 id:1 of https://codereview.webrtc.org/2400993002/ )
2016-10-24 hbos@webrtc.org rtcstats_objects.h updated with TODOs about stats not
being collected or not being collected correctly.
2016-10-24 minyue@webrtc.org Simplifying audio network adaptor by moving
receiver frame length range to ctor.
2016-10-24 ossu@webrtc.org NetEq now works with packets as values, rather than
pointers.

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 14746:14770 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/79384...

$ git log 7938471..697aba7 --date=short --no-merges --format=%ad %ae %s
2016-10-25 peah@webrtc.org Moved the AGC render sample queue into the audio
processing module
2016-10-25 peah@webrtc.org Several subcomponents inside APM copy render audio
from the render side to the capture side using the same pattern. Currently this
is done independently for the submodules.
2016-10-25 hbos@webrtc.org RTCStats equality operator added.
2016-10-25 stefan@webrtc.org Reland of Deflake ChangingNetworkRoute test.
2016-10-25 ehmaldonado@webrtc.org Revert of Deflake ChangingNetworkRoute test.
(patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ )
2016-10-25 stefan@webrtc.org Deflake ChangingNetworkRoute test.
2016-10-25 danilchap@webrtc.org Remove now unused code in RtpHeaderExtensionMap
Remove functions to enumerate all extensions, Remove concept of the inactive
extension. Decision if extension should be included into rtp header is done by
rtp_sender GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec. Since
extension might suddenly be included in the next packet (which still might
belong to same fec group), it is safer to calculate all registered extension.
2016-10-25 sprang@webrtc.org Make WebRTC compatible with OpenH264 v1.6.
2016-10-25 kjellander@webrtc.org Remove dead dependencies on xmllite and xmpp.
2016-10-25 magjed@webrtc.org codec_unittest.cc: Fix TEST vs TEST_F mismatch
2016-10-25 magjed@webrtc.org Revert of H264 codec: Check profile-level-id when
matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
2016-10-25 nisse@webrtc.org Delete left-over file profiler_unittest.cc.
2016-10-25 nisse@webrtc.org Delete unused file screencastid.h.
2016-10-24 honghaiz@webrtc.org Fix more swarming test failures by using the fake
clock or longer timeout. In the swarming test, the machines sometimes were
blocked for 1-2 seconds without processing anything. This CL makes sure that 1
second timeout is only used with fake clock.
2016-10-24 kwiberg@webrtc.org Use relative names in GN to make Chromium happy
2016-10-24 sergeyu@chromium.org Update XServerPixelBuffer to handle errors
returned from XGetImage().
2016-10-24 kwiberg@webrtc.org Stop using old
AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 solenberg@webrtc.org Remove unused function implementations from
FakeWebRtcVoiceEngine.
2016-10-24 deadbeef@webrtc.org Preventing TURN redirects to loopback addresses.
2016-10-24 terelius@webrtc.org Revert of Fix chromium-style warnings. (patchset
#1 id:1 of https://codereview.webrtc.org/2400993002/ )
2016-10-24 hbos@webrtc.org rtcstats_objects.h updated with TODOs about stats not
being collected or not being collected correctly.
2016-10-24 minyue@webrtc.org Simplifying audio network adaptor by moving
receiver frame length range to ctor.
2016-10-24 ossu@webrtc.org NetEq now works with packets as values, rather than
pointers.

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

[sakal-chromium, 2016-10-25 14:03:15]

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

[sakal-chromium, 2016-10-25 14:03:20]

The CQ bit was checked by sakal@chromium.org to run a CQ dry run

[sakal-chromium, 2016-10-25 14:03:31]

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

[sakal-chromium, 2016-10-25 14:03:32]

PTAL

[commit-bot: I haz the power, 2016-10-25 14:03:38]

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

[magjed_chromium, 2016-10-25 15:29:16]

lgtm

[commit-bot: I haz the power, 2016-10-25 16:39:58]

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

[commit-bot: I haz the power, 2016-10-25 16:39:58]

Dry run: This issue passed the CQ dry run.

[sakal-chromium, 2016-10-26 07:14:14]

The CQ bit was checked by sakal@chromium.org

[commit-bot: I haz the power, 2016-10-26 07:14:32]

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, 2016-10-26 07:18:11]

Description was changed from

==========
Roll WebRTC 14746:14770 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/79384...

$ git log 7938471..697aba7 --date=short --no-merges --format=%ad %ae %s
2016-10-25 peah@webrtc.org Moved the AGC render sample queue into the audio
processing module
2016-10-25 peah@webrtc.org Several subcomponents inside APM copy render audio
from the render side to the capture side using the same pattern. Currently this
is done independently for the submodules.
2016-10-25 hbos@webrtc.org RTCStats equality operator added.
2016-10-25 stefan@webrtc.org Reland of Deflake ChangingNetworkRoute test.
2016-10-25 ehmaldonado@webrtc.org Revert of Deflake ChangingNetworkRoute test.
(patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ )
2016-10-25 stefan@webrtc.org Deflake ChangingNetworkRoute test.
2016-10-25 danilchap@webrtc.org Remove now unused code in RtpHeaderExtensionMap
Remove functions to enumerate all extensions, Remove concept of the inactive
extension. Decision if extension should be included into rtp header is done by
rtp_sender GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec. Since
extension might suddenly be included in the next packet (which still might
belong to same fec group), it is safer to calculate all registered extension.
2016-10-25 sprang@webrtc.org Make WebRTC compatible with OpenH264 v1.6.
2016-10-25 kjellander@webrtc.org Remove dead dependencies on xmllite and xmpp.
2016-10-25 magjed@webrtc.org codec_unittest.cc: Fix TEST vs TEST_F mismatch
2016-10-25 magjed@webrtc.org Revert of H264 codec: Check profile-level-id when
matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
2016-10-25 nisse@webrtc.org Delete left-over file profiler_unittest.cc.
2016-10-25 nisse@webrtc.org Delete unused file screencastid.h.
2016-10-24 honghaiz@webrtc.org Fix more swarming test failures by using the fake
clock or longer timeout. In the swarming test, the machines sometimes were
blocked for 1-2 seconds without processing anything. This CL makes sure that 1
second timeout is only used with fake clock.
2016-10-24 kwiberg@webrtc.org Use relative names in GN to make Chromium happy
2016-10-24 sergeyu@chromium.org Update XServerPixelBuffer to handle errors
returned from XGetImage().
2016-10-24 kwiberg@webrtc.org Stop using old
AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 solenberg@webrtc.org Remove unused function implementations from
FakeWebRtcVoiceEngine.
2016-10-24 deadbeef@webrtc.org Preventing TURN redirects to loopback addresses.
2016-10-24 terelius@webrtc.org Revert of Fix chromium-style warnings. (patchset
#1 id:1 of https://codereview.webrtc.org/2400993002/ )
2016-10-24 hbos@webrtc.org rtcstats_objects.h updated with TODOs about stats not
being collected or not being collected correctly.
2016-10-24 minyue@webrtc.org Simplifying audio network adaptor by moving
receiver frame length range to ctor.
2016-10-24 ossu@webrtc.org NetEq now works with packets as values, rather than
pointers.

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 14746:14770 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/79384...

$ git log 7938471..697aba7 --date=short --no-merges --format=%ad %ae %s
2016-10-25 peah@webrtc.org Moved the AGC render sample queue into the audio
processing module
2016-10-25 peah@webrtc.org Several subcomponents inside APM copy render audio
from the render side to the capture side using the same pattern. Currently this
is done independently for the submodules.
2016-10-25 hbos@webrtc.org RTCStats equality operator added.
2016-10-25 stefan@webrtc.org Reland of Deflake ChangingNetworkRoute test.
2016-10-25 ehmaldonado@webrtc.org Revert of Deflake ChangingNetworkRoute test.
(patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ )
2016-10-25 stefan@webrtc.org Deflake ChangingNetworkRoute test.
2016-10-25 danilchap@webrtc.org Remove now unused code in RtpHeaderExtensionMap
Remove functions to enumerate all extensions, Remove concept of the inactive
extension. Decision if extension should be included into rtp header is done by
rtp_sender GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec. Since
extension might suddenly be included in the next packet (which still might
belong to same fec group), it is safer to calculate all registered extension.
2016-10-25 sprang@webrtc.org Make WebRTC compatible with OpenH264 v1.6.
2016-10-25 kjellander@webrtc.org Remove dead dependencies on xmllite and xmpp.
2016-10-25 magjed@webrtc.org codec_unittest.cc: Fix TEST vs TEST_F mismatch
2016-10-25 magjed@webrtc.org Revert of H264 codec: Check profile-level-id when
matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
2016-10-25 nisse@webrtc.org Delete left-over file profiler_unittest.cc.
2016-10-25 nisse@webrtc.org Delete unused file screencastid.h.
2016-10-24 honghaiz@webrtc.org Fix more swarming test failures by using the fake
clock or longer timeout. In the swarming test, the machines sometimes were
blocked for 1-2 seconds without processing anything. This CL makes sure that 1
second timeout is only used with fake clock.
2016-10-24 kwiberg@webrtc.org Use relative names in GN to make Chromium happy
2016-10-24 sergeyu@chromium.org Update XServerPixelBuffer to handle errors
returned from XGetImage().
2016-10-24 kwiberg@webrtc.org Stop using old
AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 solenberg@webrtc.org Remove unused function implementations from
FakeWebRtcVoiceEngine.
2016-10-24 deadbeef@webrtc.org Preventing TURN redirects to loopback addresses.
2016-10-24 terelius@webrtc.org Revert of Fix chromium-style warnings. (patchset
#1 id:1 of https://codereview.webrtc.org/2400993002/ )
2016-10-24 hbos@webrtc.org rtcstats_objects.h updated with TODOs about stats not
being collected or not being collected correctly.
2016-10-24 minyue@webrtc.org Simplifying audio network adaptor by moving
receiver frame length range to ctor.
2016-10-24 ossu@webrtc.org NetEq now works with packets as values, rather than
pointers.

TBR=
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, 2016-10-26 07:18:12]

Committed patchset #1 (id:1)

[commit-bot: I haz the power, 2016-10-26 07:20:15]

Description was changed from

==========
Roll WebRTC 14746:14770 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/79384...

$ git log 7938471..697aba7 --date=short --no-merges --format=%ad %ae %s
2016-10-25 peah@webrtc.org Moved the AGC render sample queue into the audio
processing module
2016-10-25 peah@webrtc.org Several subcomponents inside APM copy render audio
from the render side to the capture side using the same pattern. Currently this
is done independently for the submodules.
2016-10-25 hbos@webrtc.org RTCStats equality operator added.
2016-10-25 stefan@webrtc.org Reland of Deflake ChangingNetworkRoute test.
2016-10-25 ehmaldonado@webrtc.org Revert of Deflake ChangingNetworkRoute test.
(patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ )
2016-10-25 stefan@webrtc.org Deflake ChangingNetworkRoute test.
2016-10-25 danilchap@webrtc.org Remove now unused code in RtpHeaderExtensionMap
Remove functions to enumerate all extensions, Remove concept of the inactive
extension. Decision if extension should be included into rtp header is done by
rtp_sender GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec. Since
extension might suddenly be included in the next packet (which still might
belong to same fec group), it is safer to calculate all registered extension.
2016-10-25 sprang@webrtc.org Make WebRTC compatible with OpenH264 v1.6.
2016-10-25 kjellander@webrtc.org Remove dead dependencies on xmllite and xmpp.
2016-10-25 magjed@webrtc.org codec_unittest.cc: Fix TEST vs TEST_F mismatch
2016-10-25 magjed@webrtc.org Revert of H264 codec: Check profile-level-id when
matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
2016-10-25 nisse@webrtc.org Delete left-over file profiler_unittest.cc.
2016-10-25 nisse@webrtc.org Delete unused file screencastid.h.
2016-10-24 honghaiz@webrtc.org Fix more swarming test failures by using the fake
clock or longer timeout. In the swarming test, the machines sometimes were
blocked for 1-2 seconds without processing anything. This CL makes sure that 1
second timeout is only used with fake clock.
2016-10-24 kwiberg@webrtc.org Use relative names in GN to make Chromium happy
2016-10-24 sergeyu@chromium.org Update XServerPixelBuffer to handle errors
returned from XGetImage().
2016-10-24 kwiberg@webrtc.org Stop using old
AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 solenberg@webrtc.org Remove unused function implementations from
FakeWebRtcVoiceEngine.
2016-10-24 deadbeef@webrtc.org Preventing TURN redirects to loopback addresses.
2016-10-24 terelius@webrtc.org Revert of Fix chromium-style warnings. (patchset
#1 id:1 of https://codereview.webrtc.org/2400993002/ )
2016-10-24 hbos@webrtc.org rtcstats_objects.h updated with TODOs about stats not
being collected or not being collected correctly.
2016-10-24 minyue@webrtc.org Simplifying audio network adaptor by moving
receiver frame length range to ctor.
2016-10-24 ossu@webrtc.org NetEq now works with packets as values, rather than
pointers.

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 14746:14770 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/79384...

$ git log 7938471..697aba7 --date=short --no-merges --format=%ad %ae %s
2016-10-25 peah@webrtc.org Moved the AGC render sample queue into the audio
processing module
2016-10-25 peah@webrtc.org Several subcomponents inside APM copy render audio
from the render side to the capture side using the same pattern. Currently this
is done independently for the submodules.
2016-10-25 hbos@webrtc.org RTCStats equality operator added.
2016-10-25 stefan@webrtc.org Reland of Deflake ChangingNetworkRoute test.
2016-10-25 ehmaldonado@webrtc.org Revert of Deflake ChangingNetworkRoute test.
(patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ )
2016-10-25 stefan@webrtc.org Deflake ChangingNetworkRoute test.
2016-10-25 danilchap@webrtc.org Remove now unused code in RtpHeaderExtensionMap
Remove functions to enumerate all extensions, Remove concept of the inactive
extension. Decision if extension should be included into rtp header is done by
rtp_sender GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec. Since
extension might suddenly be included in the next packet (which still might
belong to same fec group), it is safer to calculate all registered extension.
2016-10-25 sprang@webrtc.org Make WebRTC compatible with OpenH264 v1.6.
2016-10-25 kjellander@webrtc.org Remove dead dependencies on xmllite and xmpp.
2016-10-25 magjed@webrtc.org codec_unittest.cc: Fix TEST vs TEST_F mismatch
2016-10-25 magjed@webrtc.org Revert of H264 codec: Check profile-level-id when
matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
2016-10-25 nisse@webrtc.org Delete left-over file profiler_unittest.cc.
2016-10-25 nisse@webrtc.org Delete unused file screencastid.h.
2016-10-24 honghaiz@webrtc.org Fix more swarming test failures by using the fake
clock or longer timeout. In the swarming test, the machines sometimes were
blocked for 1-2 seconds without processing anything. This CL makes sure that 1
second timeout is only used with fake clock.
2016-10-24 kwiberg@webrtc.org Use relative names in GN to make Chromium happy
2016-10-24 sergeyu@chromium.org Update XServerPixelBuffer to handle errors
returned from XGetImage().
2016-10-24 kwiberg@webrtc.org Stop using old
AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 solenberg@webrtc.org Remove unused function implementations from
FakeWebRtcVoiceEngine.
2016-10-24 deadbeef@webrtc.org Preventing TURN redirects to loopback addresses.
2016-10-24 terelius@webrtc.org Revert of Fix chromium-style warnings. (patchset
#1 id:1 of https://codereview.webrtc.org/2400993002/ )
2016-10-24 hbos@webrtc.org rtcstats_objects.h updated with TODOs about stats not
being collected or not being collected correctly.
2016-10-24 minyue@webrtc.org Simplifying audio network adaptor by moving
receiver frame length range to ctor.
2016-10-24 ossu@webrtc.org NetEq now works with packets as values, rather than
pointers.

TBR=
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/5a4427d288f832af7ffe39d4b16efa5b7b3901a5
Cr-Commit-Position: refs/heads/master@{#427625}
==========

[commit-bot: I haz the power, 2016-10-26 07:20:16]

Patchset 1 (id:??) landed as
https://crrev.com/5a4427d288f832af7ffe39d4b16efa5b7b3901a5
Cr-Commit-Position: refs/heads/master@{#427625}