Roll WebRTC 15623:15652 (26 commits)

Roll WebRTC 15623:15652 (26 commits)

Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/8dca466..5ea8737

$ git log 8dca466..5ea8737 --date=short --no-merges --format=%ad %ae %s
2016-12-16 kthelgason@webrtc.org Fix wrong log message.
2016-12-16 danilchap@webrtc.org Use NtpTime in RTCPSender::RtcpContext instead of pair of uint32_t
2016-12-16 terelius@webrtc.org Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss.
2016-12-16 kthelgason@webrtc.org Move ios_helpers to sdk folder
2016-12-16 mflodman@webrtc.org Adding Åsa and Erik as video owners.
2016-12-16 stefan@webrtc.org Don't report packets with id -1 to the transport feedback adapter as they provide no value.
2016-12-16 kthelgason@webrtc.org Initialize packetization mode in VideoToolbox
2016-12-15 tkchin@webrtc.org iOS: Add trendline filter to field trials.
2016-12-15 sergeyu@chromium.org Fix integer overflow in ProbeController.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double.
2016-12-15 ossu@webrtc.org Add ossu@ to OWNERS of audio/ and modules/audio_coding/
2016-12-15 mflodman@webrtc.org Fixing loopback video test by reconfiguring the encoder to correct size.
2016-12-15 philipel@webrtc.org Create VideoReceiver with external VCMTiming object.
2016-12-15 henrika@webrtc.org Improves release of allocated audio resources on Android.
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of https://codereview.webrtc.org/2577463002/ )
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of https://codereview.webrtc.org/2578543002/ )
2016-12-15 terelius@webrtc.org Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss.
2016-12-15 nisse@webrtc.org Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 michaelt@webrtc.org Wire-up audio BWE with overhead.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double.
2016-12-15 hbos@webrtc.org Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 nisse@webrtc.org Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ )
2016-12-15 hbos@webrtc.org RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs.
2016-12-15 nisse@webrtc.org Reland of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2576673002/ )
2016-12-15 kjellander@webrtc.org MB: Enable memcheck for the linux_memcheck trybot.
2016-12-15 hbos@webrtc.org RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values.

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/c9b1073f434687a1f5ce878372cf5b1541730ad0
Cr-Commit-Position: refs/heads/master@{#439114}

[o1ka, 2016-12-16 13:30:36]

The CQ bit was checked by olka@chromium.org

[commit-bot: I haz the power, 2016-12-16 13:30:52]

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, 2016-12-16 15:44:02]

CQ is committing da patch.

Bot data: {"patchset_id": 1, "attempt_start_ts": 1481895036388330, "parent_rev":
"f415e7368437653ae8399cdd5eb5b1dd7fc27f34", "commit_rev":
"944c0aa43a91e3b6cf52776de428239af05e624a"}

[commit-bot: I haz the power, 2016-12-16 15:44:27]

Description was changed from

==========
Roll WebRTC 15623:15652 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/8dca4...

$ git log 8dca466..5ea8737 --date=short --no-merges --format=%ad %ae %s
2016-12-16 kthelgason@webrtc.org Fix wrong log message.
2016-12-16 danilchap@webrtc.org Use NtpTime in RTCPSender::RtcpContext instead
of pair of uint32_t
2016-12-16 terelius@webrtc.org Pass arrival time as an int64_t rather than a
double to the MedianSlopeEstimator to avoid precision loss.
2016-12-16 kthelgason@webrtc.org Move ios_helpers to sdk folder
2016-12-16 mflodman@webrtc.org Adding Åsa and Erik as video owners.
2016-12-16 stefan@webrtc.org Don't report packets with id -1 to the transport
feedback adapter as they provide no value.
2016-12-16 kthelgason@webrtc.org Initialize packetization mode in VideoToolbox
2016-12-15 tkchin@webrtc.org iOS: Add trendline filter to field trials.
2016-12-15 sergeyu@chromium.org Fix integer overflow in ProbeController.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by
passing the arrival time as an int64_t instead of a double.
2016-12-15 ossu@webrtc.org Add ossu@ to OWNERS of audio/ and
modules/audio_coding/
2016-12-15 mflodman@webrtc.org Fixing loopback video test by reconfiguring the
encoder to correct size.
2016-12-15 philipel@webrtc.org Create VideoReceiver with external VCMTiming
object.
2016-12-15 henrika@webrtc.org Improves release of allocated audio resources on
Android.
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in
TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of
https://codereview.webrtc.org/2577463002/ )
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in
MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of
https://codereview.webrtc.org/2578543002/ )
2016-12-15 terelius@webrtc.org Pass arrival time as an int64_t rather than a
double to the MedianSlopeEstimator to avoid precision loss.
2016-12-15 nisse@webrtc.org Reland of Delete VideoFrame default constructor, and
IsZeroSize method. (patchset #1 id:1 of
https://codereview.webrtc.org/2574123002/ )
2016-12-15 michaelt@webrtc.org Wire-up audio BWE with overhead.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by
passing the arrival time as an int64_t instead of a double.
2016-12-15 hbos@webrtc.org Rename RTCOutboundRTPStreamStats *_rtt members to
*_round_trip_time.
2016-12-15 nisse@webrtc.org Revert of Delete rtc::linked_ptr. Only use, in
statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of
https://codereview.webrtc.org/2581663002/ )
2016-12-15 hbos@webrtc.org RTCStatsIntegrationTest: TestMemberIsIDReference on
all defined IDs.
2016-12-15 nisse@webrtc.org Reland of Delete rtc::linked_ptr. Only use, in
statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of
https://codereview.webrtc.org/2576673002/ )
2016-12-15 kjellander@webrtc.org MB: Enable memcheck for the linux_memcheck
trybot.
2016-12-15 hbos@webrtc.org RTCOutboundRTPStreamStats.roundTripTime: Only report
non-negative values.

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 15623:15652 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/8dca4...

$ git log 8dca466..5ea8737 --date=short --no-merges --format=%ad %ae %s
2016-12-16 kthelgason@webrtc.org Fix wrong log message.
2016-12-16 danilchap@webrtc.org Use NtpTime in RTCPSender::RtcpContext instead
of pair of uint32_t
2016-12-16 terelius@webrtc.org Pass arrival time as an int64_t rather than a
double to the MedianSlopeEstimator to avoid precision loss.
2016-12-16 kthelgason@webrtc.org Move ios_helpers to sdk folder
2016-12-16 mflodman@webrtc.org Adding Åsa and Erik as video owners.
2016-12-16 stefan@webrtc.org Don't report packets with id -1 to the transport
feedback adapter as they provide no value.
2016-12-16 kthelgason@webrtc.org Initialize packetization mode in VideoToolbox
2016-12-15 tkchin@webrtc.org iOS: Add trendline filter to field trials.
2016-12-15 sergeyu@chromium.org Fix integer overflow in ProbeController.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by
passing the arrival time as an int64_t instead of a double.
2016-12-15 ossu@webrtc.org Add ossu@ to OWNERS of audio/ and
modules/audio_coding/
2016-12-15 mflodman@webrtc.org Fixing loopback video test by reconfiguring the
encoder to correct size.
2016-12-15 philipel@webrtc.org Create VideoReceiver with external VCMTiming
object.
2016-12-15 henrika@webrtc.org Improves release of allocated audio resources on
Android.
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in
TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of
https://codereview.webrtc.org/2577463002/ )
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in
MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of
https://codereview.webrtc.org/2578543002/ )
2016-12-15 terelius@webrtc.org Pass arrival time as an int64_t rather than a
double to the MedianSlopeEstimator to avoid precision loss.
2016-12-15 nisse@webrtc.org Reland of Delete VideoFrame default constructor, and
IsZeroSize method. (patchset #1 id:1 of
https://codereview.webrtc.org/2574123002/ )
2016-12-15 michaelt@webrtc.org Wire-up audio BWE with overhead.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by
passing the arrival time as an int64_t instead of a double.
2016-12-15 hbos@webrtc.org Rename RTCOutboundRTPStreamStats *_rtt members to
*_round_trip_time.
2016-12-15 nisse@webrtc.org Revert of Delete rtc::linked_ptr. Only use, in
statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of
https://codereview.webrtc.org/2581663002/ )
2016-12-15 hbos@webrtc.org RTCStatsIntegrationTest: TestMemberIsIDReference on
all defined IDs.
2016-12-15 nisse@webrtc.org Reland of Delete rtc::linked_ptr. Only use, in
statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of
https://codereview.webrtc.org/2576673002/ )
2016-12-15 kjellander@webrtc.org MB: Enable memcheck for the linux_memcheck
trybot.
2016-12-15 hbos@webrtc.org RTCOutboundRTPStreamStats.roundTripTime: Only report
non-negative values.

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/2585763002
==========

[commit-bot: I haz the power, 2016-12-16 15:44:28]

Committed patchset #1 (id:1)

[commit-bot: I haz the power, 2016-12-16 15:46:52]

Description was changed from

==========
Roll WebRTC 15623:15652 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/8dca4...

$ git log 8dca466..5ea8737 --date=short --no-merges --format=%ad %ae %s
2016-12-16 kthelgason@webrtc.org Fix wrong log message.
2016-12-16 danilchap@webrtc.org Use NtpTime in RTCPSender::RtcpContext instead
of pair of uint32_t
2016-12-16 terelius@webrtc.org Pass arrival time as an int64_t rather than a
double to the MedianSlopeEstimator to avoid precision loss.
2016-12-16 kthelgason@webrtc.org Move ios_helpers to sdk folder
2016-12-16 mflodman@webrtc.org Adding Åsa and Erik as video owners.
2016-12-16 stefan@webrtc.org Don't report packets with id -1 to the transport
feedback adapter as they provide no value.
2016-12-16 kthelgason@webrtc.org Initialize packetization mode in VideoToolbox
2016-12-15 tkchin@webrtc.org iOS: Add trendline filter to field trials.
2016-12-15 sergeyu@chromium.org Fix integer overflow in ProbeController.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by
passing the arrival time as an int64_t instead of a double.
2016-12-15 ossu@webrtc.org Add ossu@ to OWNERS of audio/ and
modules/audio_coding/
2016-12-15 mflodman@webrtc.org Fixing loopback video test by reconfiguring the
encoder to correct size.
2016-12-15 philipel@webrtc.org Create VideoReceiver with external VCMTiming
object.
2016-12-15 henrika@webrtc.org Improves release of allocated audio resources on
Android.
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in
TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of
https://codereview.webrtc.org/2577463002/ )
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in
MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of
https://codereview.webrtc.org/2578543002/ )
2016-12-15 terelius@webrtc.org Pass arrival time as an int64_t rather than a
double to the MedianSlopeEstimator to avoid precision loss.
2016-12-15 nisse@webrtc.org Reland of Delete VideoFrame default constructor, and
IsZeroSize method. (patchset #1 id:1 of
https://codereview.webrtc.org/2574123002/ )
2016-12-15 michaelt@webrtc.org Wire-up audio BWE with overhead.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by
passing the arrival time as an int64_t instead of a double.
2016-12-15 hbos@webrtc.org Rename RTCOutboundRTPStreamStats *_rtt members to
*_round_trip_time.
2016-12-15 nisse@webrtc.org Revert of Delete rtc::linked_ptr. Only use, in
statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of
https://codereview.webrtc.org/2581663002/ )
2016-12-15 hbos@webrtc.org RTCStatsIntegrationTest: TestMemberIsIDReference on
all defined IDs.
2016-12-15 nisse@webrtc.org Reland of Delete rtc::linked_ptr. Only use, in
statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of
https://codereview.webrtc.org/2576673002/ )
2016-12-15 kjellander@webrtc.org MB: Enable memcheck for the linux_memcheck
trybot.
2016-12-15 hbos@webrtc.org RTCOutboundRTPStreamStats.roundTripTime: Only report
non-negative values.

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/2585763002
==========

to

==========
Roll WebRTC 15623:15652 (26 commits)

Changes:
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/8dca4...

$ git log 8dca466..5ea8737 --date=short --no-merges --format=%ad %ae %s
2016-12-16 kthelgason@webrtc.org Fix wrong log message.
2016-12-16 danilchap@webrtc.org Use NtpTime in RTCPSender::RtcpContext instead
of pair of uint32_t
2016-12-16 terelius@webrtc.org Pass arrival time as an int64_t rather than a
double to the MedianSlopeEstimator to avoid precision loss.
2016-12-16 kthelgason@webrtc.org Move ios_helpers to sdk folder
2016-12-16 mflodman@webrtc.org Adding Åsa and Erik as video owners.
2016-12-16 stefan@webrtc.org Don't report packets with id -1 to the transport
feedback adapter as they provide no value.
2016-12-16 kthelgason@webrtc.org Initialize packetization mode in VideoToolbox
2016-12-15 tkchin@webrtc.org iOS: Add trendline filter to field trials.
2016-12-15 sergeyu@chromium.org Fix integer overflow in ProbeController.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by
passing the arrival time as an int64_t instead of a double.
2016-12-15 ossu@webrtc.org Add ossu@ to OWNERS of audio/ and
modules/audio_coding/
2016-12-15 mflodman@webrtc.org Fixing loopback video test by reconfiguring the
encoder to correct size.
2016-12-15 philipel@webrtc.org Create VideoReceiver with external VCMTiming
object.
2016-12-15 henrika@webrtc.org Improves release of allocated audio resources on
Android.
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in
TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of
https://codereview.webrtc.org/2577463002/ )
2016-12-15 terelius@webrtc.org Revert of Avoid precision loss in
MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of
https://codereview.webrtc.org/2578543002/ )
2016-12-15 terelius@webrtc.org Pass arrival time as an int64_t rather than a
double to the MedianSlopeEstimator to avoid precision loss.
2016-12-15 nisse@webrtc.org Reland of Delete VideoFrame default constructor, and
IsZeroSize method. (patchset #1 id:1 of
https://codereview.webrtc.org/2574123002/ )
2016-12-15 michaelt@webrtc.org Wire-up audio BWE with overhead.
2016-12-15 terelius@webrtc.org Avoid precision loss in TrendlineEstimator by
passing the arrival time as an int64_t instead of a double.
2016-12-15 hbos@webrtc.org Rename RTCOutboundRTPStreamStats *_rtt members to
*_round_trip_time.
2016-12-15 nisse@webrtc.org Revert of Delete rtc::linked_ptr. Only use, in
statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of
https://codereview.webrtc.org/2581663002/ )
2016-12-15 hbos@webrtc.org RTCStatsIntegrationTest: TestMemberIsIDReference on
all defined IDs.
2016-12-15 nisse@webrtc.org Reland of Delete rtc::linked_ptr. Only use, in
statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of
https://codereview.webrtc.org/2576673002/ )
2016-12-15 kjellander@webrtc.org MB: Enable memcheck for the linux_memcheck
trybot.
2016-12-15 hbos@webrtc.org RTCOutboundRTPStreamStats.roundTripTime: Only report
non-negative values.

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/c9b1073f434687a1f5ce878372cf5b1541730ad0
Cr-Commit-Position: refs/heads/master@{#439114}
==========

[commit-bot: I haz the power, 2016-12-16 15:46:53]

Patchset 1 (id:??) landed as
https://crrev.com/c9b1073f434687a1f5ce878372cf5b1541730ad0
Cr-Commit-Position: refs/heads/master@{#439114}