Roll WebRTC 14722:14746 (23 commits)

Roll WebRTC 14722:14746 (23 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/5a1b3fe..7938471

$ git log 5a1b3fe..7938471 --date=short --no-merges --format=%ad %ae %s
2016-10-24 ehmaldonado@google.com Roll chromium_revision 1362287708..9b5bb47fa0 (426760:426837) + roll Android SDK to N
2016-10-24 philipel@webrtc.org Drop VP8 frames older than the last sync frame in the RtpFrameReferenceFinder.
2016-10-24 aleloi@webrtc.org Elimiteted race condition in the AudioMixer.
2016-10-24 terelius@webrtc.org Revert of Fix some chromium style warnings in remote_bitrate_estimator.h (patchset #1 id:1 of https://codereview.webrtc.org/2387113008/ )
2016-10-24 danilchap@webrtc.org Add tests and fix thread annotations
2016-10-24 kwiberg@webrtc.org Eliminate left shift of negative value by using multiplication instead
2016-10-24 hbos@webrtc.org RTCTransportStats[1] added, supporting all members.
2016-10-24 terelius@webrtc.org Remove unused variable from delay based BWE.
2016-10-24 terelius@webrtc.org Fix chromium-style warnings.
2016-10-24 terelius@webrtc.org Fix some chromium style warnings in remote_bitrate_estimator.h
2016-10-24 kjellander@webrtc.org GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 sakal@webrtc.org Implement framesEncoded stat in video send ssrc stats.
2016-10-24 perkj@webrtc.org Remove cricket::VideoCodec with, height and framerate properties Since WebRtcVideoSendStream have reconfigures the send codec to match the incoming captured frames widht and height they have not been used. Framerate has just been set when parsing sdp to 60fps and not changed elsewhere.
2016-10-24 nisse@webrtc.org Delete always-zero ByteBufferWriter::start_.
2016-10-24 nisse@webrtc.org Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 sakal@webrtc.org Allow using Java classes that don't require JNI in Chromium.
2016-10-24 nisse@webrtc.org Revert of Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. (patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 nisse@webrtc.org Delete unused class AudioSourceWithMixStatus.
2016-10-23 brandtr@webrtc.org Integrate FlexfecReceiveStream with Call.
2016-10-22 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.
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the AECM bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the gain controller bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the AEC bitexactness test

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/7bda5b2db9ce07f1b2d62f576f7517fe1885fb21
Cr-Commit-Position: refs/heads/master@{#427301}

[sakal-chromium, 2016-10-24 14:55:57]

Description was changed from

==========
Roll WebRTC 14722:14746 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/5a1b3...

$ git log 5a1b3fe..7938471 --date=short --no-merges --format=%ad %ae %s
2016-10-24 ehmaldonado@google.com Roll chromium_revision 1362287708..9b5bb47fa0
(426760:426837) + roll Android SDK to N
2016-10-24 philipel@webrtc.org Drop VP8 frames older than the last sync frame in
the RtpFrameReferenceFinder.
2016-10-24 aleloi@webrtc.org Elimiteted race condition in the AudioMixer.
2016-10-24 terelius@webrtc.org Revert of Fix some chromium style warnings in
remote_bitrate_estimator.h (patchset #1 id:1 of
https://codereview.webrtc.org/2387113008/ )
2016-10-24 danilchap@webrtc.org Add tests and fix thread annotations
2016-10-24 kwiberg@webrtc.org Eliminate left shift of negative value by using
multiplication instead
2016-10-24 hbos@webrtc.org RTCTransportStats[1] added, supporting all members.
2016-10-24 terelius@webrtc.org Remove unused variable from delay based BWE.
2016-10-24 terelius@webrtc.org Fix chromium-style warnings.
2016-10-24 terelius@webrtc.org Fix some chromium style warnings in
remote_bitrate_estimator.h
2016-10-24 kjellander@webrtc.org GN: move webrtc/video/ targets from
webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 sakal@webrtc.org Implement framesEncoded stat in video send ssrc
stats.
2016-10-24 perkj@webrtc.org Remove cricket::VideoCodec with, height and
framerate properties Since WebRtcVideoSendStream have reconfigures the send
codec to match the incoming captured frames widht and height they have not been
used. Framerate has just been set when parsing sdp to 60fps and not changed
elsewhere.
2016-10-24 nisse@webrtc.org Delete always-zero ByteBufferWriter::start_.
2016-10-24 nisse@webrtc.org Reland of Delete webrtc::VideoFrame::CopyFrame.
(patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 sakal@webrtc.org Allow using Java classes that don't require JNI in
Chromium.
2016-10-24 nisse@webrtc.org Revert of Add method
cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
(patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 nisse@webrtc.org Delete unused class AudioSourceWithMixStatus.
2016-10-23 brandtr@webrtc.org Integrate FlexfecReceiveStream with Call.
2016-10-22 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.
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AECM bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
gain controller bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AEC bitexactness test

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 14722:14746 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/5a1b3...

$ git log 5a1b3fe..7938471 --date=short --no-merges --format=%ad %ae %s
2016-10-24 ehmaldonado@google.com Roll chromium_revision 1362287708..9b5bb47fa0
(426760:426837) + roll Android SDK to N
2016-10-24 philipel@webrtc.org Drop VP8 frames older than the last sync frame in
the RtpFrameReferenceFinder.
2016-10-24 aleloi@webrtc.org Elimiteted race condition in the AudioMixer.
2016-10-24 terelius@webrtc.org Revert of Fix some chromium style warnings in
remote_bitrate_estimator.h (patchset #1 id:1 of
https://codereview.webrtc.org/2387113008/ )
2016-10-24 danilchap@webrtc.org Add tests and fix thread annotations
2016-10-24 kwiberg@webrtc.org Eliminate left shift of negative value by using
multiplication instead
2016-10-24 hbos@webrtc.org RTCTransportStats[1] added, supporting all members.
2016-10-24 terelius@webrtc.org Remove unused variable from delay based BWE.
2016-10-24 terelius@webrtc.org Fix chromium-style warnings.
2016-10-24 terelius@webrtc.org Fix some chromium style warnings in
remote_bitrate_estimator.h
2016-10-24 kjellander@webrtc.org GN: move webrtc/video/ targets from
webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 sakal@webrtc.org Implement framesEncoded stat in video send ssrc
stats.
2016-10-24 perkj@webrtc.org Remove cricket::VideoCodec with, height and
framerate properties Since WebRtcVideoSendStream have reconfigures the send
codec to match the incoming captured frames widht and height they have not been
used. Framerate has just been set when parsing sdp to 60fps and not changed
elsewhere.
2016-10-24 nisse@webrtc.org Delete always-zero ByteBufferWriter::start_.
2016-10-24 nisse@webrtc.org Reland of Delete webrtc::VideoFrame::CopyFrame.
(patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 sakal@webrtc.org Allow using Java classes that don't require JNI in
Chromium.
2016-10-24 nisse@webrtc.org Revert of Add method
cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
(patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 nisse@webrtc.org Delete unused class AudioSourceWithMixStatus.
2016-10-23 brandtr@webrtc.org Integrate FlexfecReceiveStream with Call.
2016-10-22 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.
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AECM bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
gain controller bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AEC bitexactness test

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, 2016-10-24 14:55:57]

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

[sakal-chromium, 2016-10-24 14:56:04]

PTAL

[sakal-chromium, 2016-10-24 14:57:56]

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

[commit-bot: I haz the power, 2016-10-24 14:58:09]

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

[kjellander_chromium, 2016-10-24 17:03:22]

lgtm, although notice that revision hasn't yet been fully built and tested in
https://build.chromium.org/p/chromium.webrtc.fyi/console

[commit-bot: I haz the power, 2016-10-24 17:23:40]

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

[commit-bot: I haz the power, 2016-10-24 17:23:41]

Dry run: Try jobs failed on following builders:
  android_arm64_dbg_recipe on master.tryserver.chromium.android (JOB_FAILED,
https://build.chromium.org/p/tryserver.chromium.android/builders/android_arm6...)

[kjellander_chromium, 2016-10-24 19:23:20]

On 2016/10/24 17:03:22, kjellander_chromium wrote:
> lgtm, although notice that revision hasn't yet been fully built and tested in
> https://build.chromium.org/p/chromium.webrtc.fyi/console

Oh, it failed Chromium trybots. Hmm, I guess we should look into if that's
missing coverage in WebRTC standalone.

[sakal-chromium, 2016-10-25 07:40:24]

The CQ bit was checked by sakal@chromium.org

[commit-bot: I haz the power, 2016-10-25 07:40:41]

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 08:38:00]

Description was changed from

==========
Roll WebRTC 14722:14746 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/5a1b3...

$ git log 5a1b3fe..7938471 --date=short --no-merges --format=%ad %ae %s
2016-10-24 ehmaldonado@google.com Roll chromium_revision 1362287708..9b5bb47fa0
(426760:426837) + roll Android SDK to N
2016-10-24 philipel@webrtc.org Drop VP8 frames older than the last sync frame in
the RtpFrameReferenceFinder.
2016-10-24 aleloi@webrtc.org Elimiteted race condition in the AudioMixer.
2016-10-24 terelius@webrtc.org Revert of Fix some chromium style warnings in
remote_bitrate_estimator.h (patchset #1 id:1 of
https://codereview.webrtc.org/2387113008/ )
2016-10-24 danilchap@webrtc.org Add tests and fix thread annotations
2016-10-24 kwiberg@webrtc.org Eliminate left shift of negative value by using
multiplication instead
2016-10-24 hbos@webrtc.org RTCTransportStats[1] added, supporting all members.
2016-10-24 terelius@webrtc.org Remove unused variable from delay based BWE.
2016-10-24 terelius@webrtc.org Fix chromium-style warnings.
2016-10-24 terelius@webrtc.org Fix some chromium style warnings in
remote_bitrate_estimator.h
2016-10-24 kjellander@webrtc.org GN: move webrtc/video/ targets from
webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 sakal@webrtc.org Implement framesEncoded stat in video send ssrc
stats.
2016-10-24 perkj@webrtc.org Remove cricket::VideoCodec with, height and
framerate properties Since WebRtcVideoSendStream have reconfigures the send
codec to match the incoming captured frames widht and height they have not been
used. Framerate has just been set when parsing sdp to 60fps and not changed
elsewhere.
2016-10-24 nisse@webrtc.org Delete always-zero ByteBufferWriter::start_.
2016-10-24 nisse@webrtc.org Reland of Delete webrtc::VideoFrame::CopyFrame.
(patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 sakal@webrtc.org Allow using Java classes that don't require JNI in
Chromium.
2016-10-24 nisse@webrtc.org Revert of Add method
cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
(patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 nisse@webrtc.org Delete unused class AudioSourceWithMixStatus.
2016-10-23 brandtr@webrtc.org Integrate FlexfecReceiveStream with Call.
2016-10-22 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.
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AECM bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
gain controller bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AEC bitexactness test

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 14722:14746 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/5a1b3...

$ git log 5a1b3fe..7938471 --date=short --no-merges --format=%ad %ae %s
2016-10-24 ehmaldonado@google.com Roll chromium_revision 1362287708..9b5bb47fa0
(426760:426837) + roll Android SDK to N
2016-10-24 philipel@webrtc.org Drop VP8 frames older than the last sync frame in
the RtpFrameReferenceFinder.
2016-10-24 aleloi@webrtc.org Elimiteted race condition in the AudioMixer.
2016-10-24 terelius@webrtc.org Revert of Fix some chromium style warnings in
remote_bitrate_estimator.h (patchset #1 id:1 of
https://codereview.webrtc.org/2387113008/ )
2016-10-24 danilchap@webrtc.org Add tests and fix thread annotations
2016-10-24 kwiberg@webrtc.org Eliminate left shift of negative value by using
multiplication instead
2016-10-24 hbos@webrtc.org RTCTransportStats[1] added, supporting all members.
2016-10-24 terelius@webrtc.org Remove unused variable from delay based BWE.
2016-10-24 terelius@webrtc.org Fix chromium-style warnings.
2016-10-24 terelius@webrtc.org Fix some chromium style warnings in
remote_bitrate_estimator.h
2016-10-24 kjellander@webrtc.org GN: move webrtc/video/ targets from
webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 sakal@webrtc.org Implement framesEncoded stat in video send ssrc
stats.
2016-10-24 perkj@webrtc.org Remove cricket::VideoCodec with, height and
framerate properties Since WebRtcVideoSendStream have reconfigures the send
codec to match the incoming captured frames widht and height they have not been
used. Framerate has just been set when parsing sdp to 60fps and not changed
elsewhere.
2016-10-24 nisse@webrtc.org Delete always-zero ByteBufferWriter::start_.
2016-10-24 nisse@webrtc.org Reland of Delete webrtc::VideoFrame::CopyFrame.
(patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 sakal@webrtc.org Allow using Java classes that don't require JNI in
Chromium.
2016-10-24 nisse@webrtc.org Revert of Add method
cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
(patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 nisse@webrtc.org Delete unused class AudioSourceWithMixStatus.
2016-10-23 brandtr@webrtc.org Integrate FlexfecReceiveStream with Call.
2016-10-22 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.
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AECM bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
gain controller bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AEC bitexactness test

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, 2016-10-25 08:38:01]

Committed patchset #1 (id:1)

[commit-bot: I haz the power, 2016-10-25 08:40:29]

Description was changed from

==========
Roll WebRTC 14722:14746 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/5a1b3...

$ git log 5a1b3fe..7938471 --date=short --no-merges --format=%ad %ae %s
2016-10-24 ehmaldonado@google.com Roll chromium_revision 1362287708..9b5bb47fa0
(426760:426837) + roll Android SDK to N
2016-10-24 philipel@webrtc.org Drop VP8 frames older than the last sync frame in
the RtpFrameReferenceFinder.
2016-10-24 aleloi@webrtc.org Elimiteted race condition in the AudioMixer.
2016-10-24 terelius@webrtc.org Revert of Fix some chromium style warnings in
remote_bitrate_estimator.h (patchset #1 id:1 of
https://codereview.webrtc.org/2387113008/ )
2016-10-24 danilchap@webrtc.org Add tests and fix thread annotations
2016-10-24 kwiberg@webrtc.org Eliminate left shift of negative value by using
multiplication instead
2016-10-24 hbos@webrtc.org RTCTransportStats[1] added, supporting all members.
2016-10-24 terelius@webrtc.org Remove unused variable from delay based BWE.
2016-10-24 terelius@webrtc.org Fix chromium-style warnings.
2016-10-24 terelius@webrtc.org Fix some chromium style warnings in
remote_bitrate_estimator.h
2016-10-24 kjellander@webrtc.org GN: move webrtc/video/ targets from
webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 sakal@webrtc.org Implement framesEncoded stat in video send ssrc
stats.
2016-10-24 perkj@webrtc.org Remove cricket::VideoCodec with, height and
framerate properties Since WebRtcVideoSendStream have reconfigures the send
codec to match the incoming captured frames widht and height they have not been
used. Framerate has just been set when parsing sdp to 60fps and not changed
elsewhere.
2016-10-24 nisse@webrtc.org Delete always-zero ByteBufferWriter::start_.
2016-10-24 nisse@webrtc.org Reland of Delete webrtc::VideoFrame::CopyFrame.
(patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 sakal@webrtc.org Allow using Java classes that don't require JNI in
Chromium.
2016-10-24 nisse@webrtc.org Revert of Add method
cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
(patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 nisse@webrtc.org Delete unused class AudioSourceWithMixStatus.
2016-10-23 brandtr@webrtc.org Integrate FlexfecReceiveStream with Call.
2016-10-22 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.
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AECM bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
gain controller bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AEC bitexactness test

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 14722:14746 (23 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/5a1b3...

$ git log 5a1b3fe..7938471 --date=short --no-merges --format=%ad %ae %s
2016-10-24 ehmaldonado@google.com Roll chromium_revision 1362287708..9b5bb47fa0
(426760:426837) + roll Android SDK to N
2016-10-24 philipel@webrtc.org Drop VP8 frames older than the last sync frame in
the RtpFrameReferenceFinder.
2016-10-24 aleloi@webrtc.org Elimiteted race condition in the AudioMixer.
2016-10-24 terelius@webrtc.org Revert of Fix some chromium style warnings in
remote_bitrate_estimator.h (patchset #1 id:1 of
https://codereview.webrtc.org/2387113008/ )
2016-10-24 danilchap@webrtc.org Add tests and fix thread annotations
2016-10-24 kwiberg@webrtc.org Eliminate left shift of negative value by using
multiplication instead
2016-10-24 hbos@webrtc.org RTCTransportStats[1] added, supporting all members.
2016-10-24 terelius@webrtc.org Remove unused variable from delay based BWE.
2016-10-24 terelius@webrtc.org Fix chromium-style warnings.
2016-10-24 terelius@webrtc.org Fix some chromium style warnings in
remote_bitrate_estimator.h
2016-10-24 kjellander@webrtc.org GN: move webrtc/video/ targets from
webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 sakal@webrtc.org Implement framesEncoded stat in video send ssrc
stats.
2016-10-24 perkj@webrtc.org Remove cricket::VideoCodec with, height and
framerate properties Since WebRtcVideoSendStream have reconfigures the send
codec to match the incoming captured frames widht and height they have not been
used. Framerate has just been set when parsing sdp to 60fps and not changed
elsewhere.
2016-10-24 nisse@webrtc.org Delete always-zero ByteBufferWriter::start_.
2016-10-24 nisse@webrtc.org Reland of Delete webrtc::VideoFrame::CopyFrame.
(patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 sakal@webrtc.org Allow using Java classes that don't require JNI in
Chromium.
2016-10-24 nisse@webrtc.org Revert of Add method
cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
(patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 nisse@webrtc.org Delete unused class AudioSourceWithMixStatus.
2016-10-23 brandtr@webrtc.org Integrate FlexfecReceiveStream with Call.
2016-10-22 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.
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AECM bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
gain controller bitexactness test
2016-10-22 peah@webrtc.org Added the missing ReadQueuedRenderData() call to the
AEC bitexactness test

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/7bda5b2db9ce07f1b2d62f576f7517fe1885fb21
Cr-Commit-Position: refs/heads/master@{#427301}
==========

[commit-bot: I haz the power, 2016-10-25 08:40:31]

Patchset 1 (id:??) landed as
https://crrev.com/7bda5b2db9ce07f1b2d62f576f7517fe1885fb21
Cr-Commit-Position: refs/heads/master@{#427301}