Roll WebRTC 18023:18079 (28 commits)

Roll WebRTC 18023:18079 (28 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/d6b3a36..b2a8855

$ git log d6b3a36..b2a8855 --date=short --no-merges --format=%ad %ae %s
2017-05-10 minyue@webrtc.org Fixing video loopback test with encoder factory.
2017-05-10 nisse@webrtc.org Deleted unused method EstimateMTU, and the WinPing class.
2017-05-09 jtteh@webrtc.org iOS audio session isInterrupted flag does not get reset correctly:
2017-05-09 nisse@webrtc.org Delete helper class MediaTypePacketReceiver.
2017-05-09 oprypin@webrtc.org Drop deprecated AudioFrameOperations::Scale method signatures
2017-05-09 sakal@webrtc.org Test for Gradle project generation.
2017-05-09 mbonadei@webrtc.org Moving scripts to download and build apprtc/collider.
2017-05-09 kjellander@webrtc.org Rename tools-webrtc -> tools_webrtc
2017-05-08 zstein@webrtc.org This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008).
2017-05-08 deadbeef@webrtc.org Fixing invalid IPv6 address parsing stack underflow on Windows.
2017-05-08 mellem@webrtc.org Make WebRtcAudioEffects and its create method public.
2017-05-08 nisse@webrtc.org Delete unused class SharedExclusiveLock.
2017-05-08 kwiberg@webrtc.org Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 magjed@webrtc.org Actually move CoreVideoFrameBuffer from webrtc/common_video/ to webrtc/sdk/objc/
2017-05-08 nisse@webrtc.org Delete method MessageQueue::set_socketserver
2017-05-08 nisse@webrtc.org Update comments for removal of MediaController.
2017-05-08 ilnik@webrtc.org Remove deprecated API
2017-05-08 nisse@webrtc.org Refactor TestClient to use std::unique_ptr, and fix VirtualSocketServerTest leaks.
2017-05-07 deadbeef@webrtc.org Unflaking PeerConnectionIntegrationTest.DtmfSenderObserver.
2017-05-05 kwiberg@webrtc.org Add myself as OWNER of webrtc/api/ and webrtc/base/
2017-05-05 kwiberg@webrtc.org Create an OWNERS file in webrtc/api/audio_codecs/
2017-05-05 ehmaldonado@webrtc.org Fix webrtcsdp_unittest.
2017-05-05 deadbeef@webrtc.org When a data channel fails to be created, return nil instead of crashing.
2017-05-05 ivoc@webrtc.org Prevent residual echo likelihood values greater than 1.0.
2017-05-05 henrik.lundin@webrtc.org NetEq: Fix a bug in expand_rate and speech_expand_rate calculation
2017-05-05 nisse@webrtc.org Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 henrika@webrtc.org Blacklisting of HW-AEC/NS and OpenSL must now be done by the WebRTC client.
2017-05-05 sakal@webrtc.org Add support for media recorders in Camera1Capturer.

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/2872293002
Cr-Commit-Position: refs/heads/master@{#470550}
Committed: https://chromium.googlesource.com/chromium/src/+/714a743dbf866eb7bd2084c58217d36ce6f4881b

[magjed_chromium, 2017-05-10 12:04:28]

The CQ bit was checked by magjed@chromium.org

[commit-bot: I haz the power, 2017-05-10 12:05:00]

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, 2017-05-10 13:18:50]

CQ is committing da patch.

Bot data: {"patchset_id": 1, "attempt_start_ts": 1494417868038460, "parent_rev":
"2c54e2be976f7c4e8d8ac2438f5f1e97f6566a6c", "commit_rev":
"714a743dbf866eb7bd2084c58217d36ce6f4881b"}

[commit-bot: I haz the power, 2017-05-10 13:19:08]

Description was changed from

==========
Roll WebRTC 18023:18079 (28 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/d6b3a...

$ git log d6b3a36..b2a8855 --date=short --no-merges --format=%ad %ae %s
2017-05-10 minyue@webrtc.org Fixing video loopback test with encoder factory.
2017-05-10 nisse@webrtc.org Deleted unused method EstimateMTU, and the WinPing
class.
2017-05-09 jtteh@webrtc.org iOS audio session isInterrupted flag does not get
reset correctly:
2017-05-09 nisse@webrtc.org Delete helper class MediaTypePacketReceiver.
2017-05-09 oprypin@webrtc.org Drop deprecated AudioFrameOperations::Scale method
signatures
2017-05-09 sakal@webrtc.org Test for Gradle project generation.
2017-05-09 mbonadei@webrtc.org Moving scripts to download and build
apprtc/collider.
2017-05-09 kjellander@webrtc.org Rename tools-webrtc -> tools_webrtc
2017-05-08 zstein@webrtc.org This will allow me to test that Call invokes
SendSideCongestionController::SetBweBitrates as expected (for
https://codereview.chromium.org/2793913008).
2017-05-08 deadbeef@webrtc.org Fixing invalid IPv6 address parsing stack
underflow on Windows.
2017-05-08 mellem@webrtc.org Make WebRtcAudioEffects and its create method
public.
2017-05-08 nisse@webrtc.org Delete unused class SharedExclusiveLock.
2017-05-08 kwiberg@webrtc.org Don't add or rename files in webrtc/ and
webrtc/api/ without a proper review
2017-05-08 magjed@webrtc.org Actually move CoreVideoFrameBuffer from
webrtc/common_video/ to webrtc/sdk/objc/
2017-05-08 nisse@webrtc.org Delete method MessageQueue::set_socketserver
2017-05-08 nisse@webrtc.org Update comments for removal of MediaController.
2017-05-08 ilnik@webrtc.org Remove deprecated API
2017-05-08 nisse@webrtc.org Refactor TestClient to use std::unique_ptr, and fix
VirtualSocketServerTest leaks.
2017-05-07 deadbeef@webrtc.org Unflaking
PeerConnectionIntegrationTest.DtmfSenderObserver.
2017-05-05 kwiberg@webrtc.org Add myself as OWNER of webrtc/api/ and
webrtc/base/
2017-05-05 kwiberg@webrtc.org Create an OWNERS file in webrtc/api/audio_codecs/
2017-05-05 ehmaldonado@webrtc.org Fix webrtcsdp_unittest.
2017-05-05 deadbeef@webrtc.org When a data channel fails to be created, return
nil instead of crashing.
2017-05-05 ivoc@webrtc.org Prevent residual echo likelihood values greater than
1.0.
2017-05-05 henrik.lundin@webrtc.org NetEq: Fix a bug in expand_rate and
speech_expand_rate calculation
2017-05-05 nisse@webrtc.org Delete MediaController class, move Call ownership to
PeerConnection.
2017-05-05 henrika@webrtc.org Blacklisting of HW-AEC/NS and OpenSL must now be
done by the WebRTC client.
2017-05-05 sakal@webrtc.org Add support for media recorders in Camera1Capturer.

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 18023:18079 (28 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/d6b3a...

$ git log d6b3a36..b2a8855 --date=short --no-merges --format=%ad %ae %s
2017-05-10 minyue@webrtc.org Fixing video loopback test with encoder factory.
2017-05-10 nisse@webrtc.org Deleted unused method EstimateMTU, and the WinPing
class.
2017-05-09 jtteh@webrtc.org iOS audio session isInterrupted flag does not get
reset correctly:
2017-05-09 nisse@webrtc.org Delete helper class MediaTypePacketReceiver.
2017-05-09 oprypin@webrtc.org Drop deprecated AudioFrameOperations::Scale method
signatures
2017-05-09 sakal@webrtc.org Test for Gradle project generation.
2017-05-09 mbonadei@webrtc.org Moving scripts to download and build
apprtc/collider.
2017-05-09 kjellander@webrtc.org Rename tools-webrtc -> tools_webrtc
2017-05-08 zstein@webrtc.org This will allow me to test that Call invokes
SendSideCongestionController::SetBweBitrates as expected (for
https://codereview.chromium.org/2793913008).
2017-05-08 deadbeef@webrtc.org Fixing invalid IPv6 address parsing stack
underflow on Windows.
2017-05-08 mellem@webrtc.org Make WebRtcAudioEffects and its create method
public.
2017-05-08 nisse@webrtc.org Delete unused class SharedExclusiveLock.
2017-05-08 kwiberg@webrtc.org Don't add or rename files in webrtc/ and
webrtc/api/ without a proper review
2017-05-08 magjed@webrtc.org Actually move CoreVideoFrameBuffer from
webrtc/common_video/ to webrtc/sdk/objc/
2017-05-08 nisse@webrtc.org Delete method MessageQueue::set_socketserver
2017-05-08 nisse@webrtc.org Update comments for removal of MediaController.
2017-05-08 ilnik@webrtc.org Remove deprecated API
2017-05-08 nisse@webrtc.org Refactor TestClient to use std::unique_ptr, and fix
VirtualSocketServerTest leaks.
2017-05-07 deadbeef@webrtc.org Unflaking
PeerConnectionIntegrationTest.DtmfSenderObserver.
2017-05-05 kwiberg@webrtc.org Add myself as OWNER of webrtc/api/ and
webrtc/base/
2017-05-05 kwiberg@webrtc.org Create an OWNERS file in webrtc/api/audio_codecs/
2017-05-05 ehmaldonado@webrtc.org Fix webrtcsdp_unittest.
2017-05-05 deadbeef@webrtc.org When a data channel fails to be created, return
nil instead of crashing.
2017-05-05 ivoc@webrtc.org Prevent residual echo likelihood values greater than
1.0.
2017-05-05 henrik.lundin@webrtc.org NetEq: Fix a bug in expand_rate and
speech_expand_rate calculation
2017-05-05 nisse@webrtc.org Delete MediaController class, move Call ownership to
PeerConnection.
2017-05-05 henrika@webrtc.org Blacklisting of HW-AEC/NS and OpenSL must now be
done by the WebRTC client.
2017-05-05 sakal@webrtc.org Add support for media recorders in Camera1Capturer.

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/2872293002
Cr-Commit-Position: refs/heads/master@{#470550}
Committed:
https://chromium.googlesource.com/chromium/src/+/714a743dbf866eb7bd2084c58217...
==========

[commit-bot: I haz the power, 2017-05-10 13:19:09]

Committed patchset #1 (id:1) as
https://chromium.googlesource.com/chromium/src/+/714a743dbf866eb7bd2084c58217...