Commit Graph

22570 Commits

Author SHA1 Message Date
41c11e4cad AEC3: Rounding of estimated call skew
This CL fixes the rounding of the estimated average call skew. Before it
was rounded down (toward INT_MIN). Now it is rounded to the nearest integer.
This avoids unnecessary fluctuations of the estimated call skew (and
unnecessary resets).

Bug: webrtc:9283,chromium:888042
Change-Id: Id5b3c593f812f5f9fd3dcdafb7e388a6ef1ac153
Reviewed-on: https://webrtc-review.googlesource.com/77684
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23338}
2018-05-22 08:15:58 +00:00
5991ac9d22 Remove outdated DEPS rules.
From those I removed, transport.h doesn't exist. For the others
I tried checking that the presubmit doesn't fire if I modify
all lines that include the previously +'d entry (for instance
call/rtp_config.h). I take this to mean that all callers of
for instance rtp_config.h now obtain checkdeps permission
elsewhere, closer to where they're located. This change should
not change checkdeps behaviour, therefore.

Bug: webrtc:4243
Change-Id: Ia909d13c5d79cb244f45b737142d2f47568ba77e
Reviewed-on: https://webrtc-review.googlesource.com/77801
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23337}
2018-05-22 06:36:08 +00:00
0e36a7260f Delete unused class CurrentSpeakerMonitor.
Bug: webrtc:8760
Change-Id: Ib2f84c7d74f1f3187f02dcf697e9c16a4d5f10e3
Reviewed-on: https://webrtc-review.googlesource.com/34652
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23336}
2018-05-22 06:31:08 +00:00
c2ee8e8a46 Removing references to webrtc::VideoSendStream::DegradationPreference.
It was replaced be webrtc::DegradationPreference in this CL:
https://webrtc-review.googlesource.com/c/src/+/77024

But some downstream code was still referencing it.

Bug: webrtc:8830
Change-Id: Ibd0a3d15df7f13473c0f37a2493dd70cec6c0482
Reviewed-on: https://webrtc-review.googlesource.com/78082
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23335}
2018-05-21 20:20:57 +00:00
0327c2ddc1 Move VideoStreamEncoderInterface to api/.
Bug: webrtc:8830
Change-Id: I17908b4ef6a043acf22e2110b9672012d5fa7fc0
Reviewed-on: https://webrtc-review.googlesource.com/74481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23334}
2018-05-21 19:50:37 +00:00
65ec0fc81e Delete unneeded includes of basictypes.h.
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.

Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.

Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
2018-05-21 19:35:08 +00:00
0ce868c60e Recognize additional adapter types on Windows
Specifically, Ethernet, Wi-Fi and cellular interfaces.

Note that this only affects native applications, as chromium already
has its own code for this:
https://cs.chromium.org/chromium/src/net/base/network_interfaces_win.cc?l=29&rcl=568ba7132833eea41fc863dd41c377928f49fa51

Which WebRTC accesses through "IpcNetworkManager".

Bug: webrtc:3149, webrtc:6588
Change-Id: I347f2734d95ea24cea3f89e6ed5bf2d135a9fc77
Reviewed-on: https://webrtc-review.googlesource.com/76622
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23332}
2018-05-21 18:03:26 +00:00
c941b7e28d Roll chromium_revision 911054f7d0..039110971b (559863:560284)
Change log: 911054f7d0..039110971b
Full diff: 911054f7d0..039110971b

Roll chromium third_party 480fd0409d..cc1af82934
Change log: 480fd0409d..cc1af82934

Changed dependencies:
* src/base: b802985ef4..8e89780685
* src/build: fc8308f6b6..66897e4d72
* src/ios: 1562248170..02a22b3900
* src/testing: a5fce03148..671c6a4522
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ce9b3742a1..7ca7a59f02
* src/third_party/depot_tools: 8fe4d8cbef..083eb25f9a
* src/tools: 6c88721b30..ff5c71196b
DEPS diff: 911054f7d0..039110971b/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I015f9dab00cef22f0d30d6f05d3fab6bc27ee7d4
Reviewed-on: https://webrtc-review.googlesource.com/78081
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23331}
2018-05-21 17:17:06 +00:00
b9fc6508c0 Add min and max allowed bitrate in Opus bitrate tests
Instead of checking for an exact bitrate check that the bitrate is between
the min and max values.
Also relax a threshold in a bandwith adaptation test.

Bug: webrtc:9280
Change-Id: I465d785a53759f73242198ee1ccd7da1a26c48b7
Reviewed-on: https://webrtc-review.googlesource.com/78041
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23330}
2018-05-21 16:41:35 +00:00
666becad58 AEC3: ERLE improvements
The ERLE computation was improved by two means:
- The update function was always called and just parts of the internal code reacts to the converged filter flag
- When computing the ERLE, the ratio of energies is now computed using more points and, therefore, a more robust estimation is achieved.

Bug: webrtc:9284
Change-Id: Ie4f871f19cfad1a13741352ddd7b0a27ad6c3fb6
Reviewed-on: https://webrtc-review.googlesource.com/77767
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23329}
2018-05-21 15:11:06 +00:00
8436a699a9 Revert "Reland "Adding absl includes and defines to rtc_* templates.""
This reverts commit bdb0fe42bc46d190ca45fc5a6658eddbfa5eead5.

Reason for revert: https://ci.chromium.org/buildbot/chromium.fyi/Jumbo%20Win%20x64/11502

Original change's description:
> Reland "Adding absl includes and defines to rtc_* templates."
> 
> This reverts commit 85cb19fec7caf558dee7a09aafabe01c5ac78f3f.
> 
> Reason for revert: The new version of Abseil should fix the previous
> issue.
> 
> Original change's description:
> > Revert "Reland "Adding absl includes and defines to rtc_* templates.""
> > 
> > This reverts commit 9632112a16d70a146e917db4de761e6253dfc364.
> > 
> > Reason for revert: It breaks the WebRTC roll into Chromium.
> > https://chromium-review.googlesource.com/c/chromium/src/+/1061476
> > 
> > Original change's description:
> > > Reland "Adding absl includes and defines to rtc_* templates."
> > > 
> > > This reverts commit d161eda477491b2b97fb3f26d229c625a2a0e9b8.
> > > 
> > > Reason for revert: The problem with iOS trybots should be fixed.
> > > 
> > > Original change's description:
> > > > Revert "Adding absl includes and defines to rtc_* templates."
> > > >
> > > > This reverts commit 9d8f3850f4c4faad5dc5ab32ab6f2c9c43df7b6c.
> > > >
> > > > Reason for revert: Breaks some trybots: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Release/builds/12793.
> > > >
> > > > Original change's description:
> > > > > Adding absl includes and defines to rtc_* templates.
> > > > >
> > > > > This CL implicitly adds the -I compiler flag and absl macros to WebRTC
> > > > > templates. In order to include absl headers using relative paths, WebRTC
> > > > > needs to ensure that all its build targets are able to see absl headers.
> > > > >
> > > > > This can also be done with public_deps, but WebRTC is trying to avoid
> > > > > it because it creates problems with other build systems. Given this
> > > > > constraint, using rtc_* templates is the most reliable solution.
> > > > >
> > > > > Please note that rtc_* templates are adding absl includes and defines
> > > > > as public_configs, this means that build targets with WebRTC targets
> > > > > in their public_deps will propagate these configs following the GN
> > > > > guideline.
> > > > >
> > > > > Bug: webrtc:8821
> > > > > Change-Id: I4aa594a524f4bd045bcb3e80d76cc27f06fe01d7
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/70367
> > > > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#22927}
> > > >
> > > > TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> > > >
> > > > Change-Id: Id8e1f881c57553386566eb1970f6b9f8632cab37
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: webrtc:8821
> > > > Reviewed-on: https://webrtc-review.googlesource.com/71000
> > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#22928}
> > > 
> > > TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> > > 
> > > Bug: webrtc:8821
> > > Change-Id: I6ee2eda97bbcd4c9be25c9c4073272192b0373f8
> > > Reviewed-on: https://webrtc-review.googlesource.com/71700
> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#23251}
> > 
> > TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> > 
> > Change-Id: I61fb749797314ca514691b341c66f7f39ef45491
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:8821
> > Reviewed-on: https://webrtc-review.googlesource.com/77220
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23264}
> 
> TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:8821
> Change-Id: I71dea953a002a0d526949c627653bcad0c6518fc
> Reviewed-on: https://webrtc-review.googlesource.com/77781
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23317}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org

Change-Id: I6010f9264dba7bcc4e82c4f4bbfb2eca561e500e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8821, chromium:845158
Reviewed-on: https://webrtc-review.googlesource.com/78061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23328}
2018-05-21 14:38:36 +00:00
9024da84c9 NetEq: Fixing an overflow bug in expand.cc
The overflow currently does not cause any problems, but it has been
found that it can cause crashes after a refactoring that is coming in
the near future.

Bug: webrtc:9180
Change-Id: Ia2c4e545c062c4f8ad13cbc47b8796c6e8a4e906
Reviewed-on: https://webrtc-review.googlesource.com/77667
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23327}
2018-05-21 13:39:25 +00:00
3ca48a69fd Ports base::win:OSInfo from Chrome to rtc_win in WebRTC.
Enables us to do stuff like:

TEST(WindowsVersion, GetVersionGlobalScopeAccessor) {
  if (GetVersion() < VERSION_WIN10) {
    MethodNotSupportedOnWin10AndLater();
  } else {
    MethodSupportedOnWin10AndLater();
  }
}

which is useful when working with Windows.

Note that, I also port a limited part of base::win::RegKey but only
those parts that are needed to implement OSInfo. Hence, I don't expose
any RegKey APIs.

NOTRY=TRUE

No-Presubmit: True
Bug: webrtc:9265
Change-Id: Ia2fc0963f24044ffaad954aa21d28df9c32b3ee7
Reviewed-on: https://webrtc-review.googlesource.com/77723
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23326}
2018-05-21 13:27:46 +00:00
cc02cb595f Add getSupportedCodecs to VideoDecoderFactory interface.
The default implementation of the method is to return an empty list.
Clients should update their implementations before WebRTC starts calling
this method.

Also updates internal WebRTC implentations of this interface to
implement the method.

Bug: webrtc:7925
Change-Id: I258de2f09f6d4cc5dd9f4657e5d54e8411f8f5d8
Reviewed-on: https://webrtc-review.googlesource.com/77641
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23325}
2018-05-21 13:21:45 +00:00
80d02ad93f Suppress warning about exit in destructor, because it intended.
BUG=None

Change-Id: I35323234382aad4f952b2c39a4eecd93ad81e017
Reviewed-on: https://webrtc-review.googlesource.com/77666
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23324}
2018-05-21 12:36:55 +00:00
9ab6eb738a Minor namespace change for CoreAudioUtility
NOTRY=TRUE

TBR: kwiberg@webrtc.org
Bug: webrtc:9265
Change-Id: Ic40634eb5258739ef06becd5db7a70a1e31d29e3
Reviewed-on: https://webrtc-review.googlesource.com/78020
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23323}
2018-05-21 12:28:25 +00:00
79478ad675 Adjust base bitrate for experimental temporal layer count
Bug: webrtc:9260
Change-Id: I15eb24ddf94122d3b70cbf1ee25125a0adbf9f2d
Reviewed-on: https://webrtc-review.googlesource.com/77363
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23322}
2018-05-21 12:18:06 +00:00
195d1d77ea Remove ScreenshareLayerConfig.
Bug: None
Change-Id: I7fe020f9985fa5ca1d9873a126a8518a991ded8e
Reviewed-on: https://webrtc-review.googlesource.com/75509
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23321}
2018-05-21 12:02:36 +00:00
0f405825c7 New class FakePeriodicVideoTrackSource, simplifying shutdown logic.
Previous code had a FakePeriodicVideoSource and a
VideoTrackSource, where the latter is reference counted and
outlives the former. That results in potential races when
RemoveSink is called on the VideoTrackSource after the
FakePeriodicVideoSource is destroyed, with a complicated sequence
to do correct shutdown.

The new class, FakePeriodicVideoTrackSource, owns a
FakePeriodicVideoSource, and they get the same lifetime.

Bug: webrtc:6353
Change-Id: Ic33b393e00a31fa28893dce2018948d3f90e0a9e
Reviewed-on: https://webrtc-review.googlesource.com/76961
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23320}
2018-05-21 10:27:55 +00:00
34b1bc7299 Disable flaky test: FullStackTest.VP9SVC_3SL_High
Following a change in libvpx, FullStackTest.VP9SVC_3SL_High has
become flaky. It will be disabled until the libvpx issue is fixed.

Bug: webrtc:9293
NOTRY: true
Change-Id: Ib375363bdefdbb4104130a1f0f02ea34dc26e7f9
Reviewed-on: https://webrtc-review.googlesource.com/77663
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23319}
2018-05-21 10:02:25 +00:00
6633d41bb0 Reland "Update expected bitrate in Opus tests"
This is a reland of 79ded653fee7183d5c0d94c5addf570bcfb29c9e

Original change's description:
> Update expected bitrate in Opus tests
>
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
>
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}

TBR=henrik.lundin@webrtc.org

Bug: webrtc:9280
Change-Id: I6bfcd1c5e1d5298543024a0faa6a695026434df3
Reviewed-on: https://webrtc-review.googlesource.com/77980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23318}
2018-05-21 08:13:05 +00:00
bdb0fe42bc Reland "Adding absl includes and defines to rtc_* templates."
This reverts commit 85cb19fec7caf558dee7a09aafabe01c5ac78f3f.

Reason for revert: The new version of Abseil should fix the previous
issue.

Original change's description:
> Revert "Reland "Adding absl includes and defines to rtc_* templates.""
> 
> This reverts commit 9632112a16d70a146e917db4de761e6253dfc364.
> 
> Reason for revert: It breaks the WebRTC roll into Chromium.
> https://chromium-review.googlesource.com/c/chromium/src/+/1061476
> 
> Original change's description:
> > Reland "Adding absl includes and defines to rtc_* templates."
> > 
> > This reverts commit d161eda477491b2b97fb3f26d229c625a2a0e9b8.
> > 
> > Reason for revert: The problem with iOS trybots should be fixed.
> > 
> > Original change's description:
> > > Revert "Adding absl includes and defines to rtc_* templates."
> > >
> > > This reverts commit 9d8f3850f4c4faad5dc5ab32ab6f2c9c43df7b6c.
> > >
> > > Reason for revert: Breaks some trybots: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Release/builds/12793.
> > >
> > > Original change's description:
> > > > Adding absl includes and defines to rtc_* templates.
> > > >
> > > > This CL implicitly adds the -I compiler flag and absl macros to WebRTC
> > > > templates. In order to include absl headers using relative paths, WebRTC
> > > > needs to ensure that all its build targets are able to see absl headers.
> > > >
> > > > This can also be done with public_deps, but WebRTC is trying to avoid
> > > > it because it creates problems with other build systems. Given this
> > > > constraint, using rtc_* templates is the most reliable solution.
> > > >
> > > > Please note that rtc_* templates are adding absl includes and defines
> > > > as public_configs, this means that build targets with WebRTC targets
> > > > in their public_deps will propagate these configs following the GN
> > > > guideline.
> > > >
> > > > Bug: webrtc:8821
> > > > Change-Id: I4aa594a524f4bd045bcb3e80d76cc27f06fe01d7
> > > > Reviewed-on: https://webrtc-review.googlesource.com/70367
> > > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#22927}
> > >
> > > TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> > >
> > > Change-Id: Id8e1f881c57553386566eb1970f6b9f8632cab37
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:8821
> > > Reviewed-on: https://webrtc-review.googlesource.com/71000
> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22928}
> > 
> > TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> > 
> > Bug: webrtc:8821
> > Change-Id: I6ee2eda97bbcd4c9be25c9c4073272192b0373f8
> > Reviewed-on: https://webrtc-review.googlesource.com/71700
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23251}
> 
> TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> 
> Change-Id: I61fb749797314ca514691b341c66f7f39ef45491
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8821
> Reviewed-on: https://webrtc-review.googlesource.com/77220
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23264}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8821
Change-Id: I71dea953a002a0d526949c627653bcad0c6518fc
Reviewed-on: https://webrtc-review.googlesource.com/77781
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23317}
2018-05-21 07:51:55 +00:00
d902d58b0a Framerate controller for VP9 screen sharing.
- Limit framerate by dropping frames before encoding.
- The max framerate at screen sharing is set to 5fps.

Bug: webrtc:9261
Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760
Reviewed-on: https://webrtc-review.googlesource.com/76921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23316}
2018-05-19 07:14:48 +00:00
a832019f4e Add qingsi@ as owner of p2p.
Bug: None
Change-Id: Iffcb6eb665b5f4a909f2dcf52471cb57919823c5
Reviewed-on: https://webrtc-review.googlesource.com/77843
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23315}
2018-05-18 21:31:56 +00:00
2d2c888293 Returns RTCError for setting unimplemented RtpParameters.
We have a number of RtpParameters that aren't implemented. If a client
is setting these values it creates unexpected results when the value
doesn't do anything for them. This change incorporates returning the
correct error if the parameter is unimplemented.

It also changes the scale_resolution_down_by and scale_framerate_down_by
RtpEncodingParameters to rtc::Optionals because they aren't implemented.

This change is part of the effort to ship get/setParameters in Chrome.

Bug: webrtc:8772
Change-Id: I9797695e5116e6aeb3c02afddbf460b2a0d7d5ab
Reviewed-on: https://webrtc-review.googlesource.com/75421
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23314}
2018-05-18 17:40:16 +00:00
dfce03af6e Allows injection of network controller factory into peer connection factory.
Bug: webrtc:9155
Change-Id: I0a17024042f154297aba20f5d2dc766feb27f3f7
Reviewed-on: https://webrtc-review.googlesource.com/73123
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23313}
2018-05-18 17:07:16 +00:00
78b0a60223 Add phoglund as root owner.
It seems convenient since the EngProd team make repo-wide changes
every now and then.

Bug: None
Change-Id: If429fc8ed503a3c24c912ac2e8d120f93edc4823
Reviewed-on: https://webrtc-review.googlesource.com/77760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23312}
2018-05-18 15:57:56 +00:00
be71a1ee08 Replace VP9 screen sharing.
- Remove referencing control from encoder wrapper. Use fixed temporal
prediction structure.
- Remove flexible mode from encoder wrapper. It only worked with
referencing control which this CL removes.
- Remove external framerate/bitrate controller. Keep codec's internal
frame dropping enabled at screen sharing.
- Use GetSvcConfig() to configure layering.

Bug: webrtc:9261
Change-Id: I355baa6aab7b98ac5028b3851d1f8ccc82a308e0
Reviewed-on: https://webrtc-review.googlesource.com/76801
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23311}
2018-05-18 15:11:46 +00:00
21219a0e43 Reland "Injectable logging"
Any injected loggable or NativeLogger would be deleted if PCFactory
was reinitialized without calling setInjectableLogger. Now native
logging is not implemented as a Loggable, so it will remain active
unless a Loggable is injected.

This is a reland of 59216ec4a4151b1ba5478c8f2b5c9f01f4683d7f

Original change's description:
> Injectable logging
>
> Allows passing a Loggable to PCFactory.initializationOptions, which
> is then injected to Logging.java and logging.h. Future log messages
> in both Java and native will then be passed to this Loggable.
>
> Bug: webrtc:9225
> Change-Id: I2ff693380639448301a78a93dc11d3a0106f0967
> Reviewed-on: https://webrtc-review.googlesource.com/73243
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23241}

Bug: webrtc:9225
Change-Id: I2fe3fbc8c323814284bb62e43fe1870bdab581ee
TBR: kwiberg
Reviewed-on: https://webrtc-review.googlesource.com/77140
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23310}
2018-05-18 15:04:16 +00:00
812ceafb5a Ensure render time is zero when playout delay is zero so that minimal latency in the render pipeline is ensured.
Bug: webrtc:9135
Change-Id: Id9ae8ec59536808ba8923c73dd46abfe3fa6fe79
Reviewed-on: https://webrtc-review.googlesource.com/75600
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23309}
2018-05-18 14:47:26 +00:00
6bf5a0d5b6 AEC3: High-pass filter delay estimator signals
This CL applies a high pass filter to the delay estimator signals which
improves the adaptation of the matched filters in noisy environments.
This results in faster delay estimation.

Bug: webrtc:9288
Change-Id: I8ffe5442eab7ac2f10a7ba236b08a0f07ec90645
Reviewed-on: https://webrtc-review.googlesource.com/77725
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23308}
2018-05-18 14:33:26 +00:00
77995e744b Revert "Update expected bitrate in Opus tests"
This reverts commit 79ded653fee7183d5c0d94c5addf570bcfb29c9e.

Reason for revert: Different repos have different Opus

Original change's description:
> Update expected bitrate in Opus tests
> 
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
> 
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}

TBR=henrik.lundin@webrtc.org,gustaf@webrtc.org

Change-Id: I3c18db2d6052c4049d836c3e595b00189aebcbc8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9280
Reviewed-on: https://webrtc-review.googlesource.com/77800
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23307}
2018-05-18 14:27:36 +00:00
79ded653fe Update expected bitrate in Opus tests
Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
CL re-enables recently disabled unittests and updates the expected bitrates.

Bug: webrtc:9280
Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
Reviewed-on: https://webrtc-review.googlesource.com/77766
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23306}
2018-05-18 13:45:06 +00:00
2d9a3b1aba Increasing the API call skew hysteresis limit in AEC3
This CL increases the allowed variations in the API call skew limit in
AEC3.

Bug: webrtc:9283,chromium:888042
Change-Id: Ib5e784c6f3dcf1bf3a2cbfe2b1559953db9227a8
Reviewed-on: https://webrtc-review.googlesource.com/77430
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23305}
2018-05-18 13:39:26 +00:00
9a6133e1a0 Roll chromium_revision bcf2616e8e..911054f7d0 (559838:559863)
Change log: bcf2616e8e..911054f7d0
Full diff: bcf2616e8e..911054f7d0

Roll chromium third_party 9d65a3cdda..480fd0409d
Change log: 9d65a3cdda..480fd0409d

Changed dependencies:
* src/ios: 289c450460..1562248170
* src/tools: 6e6e398687..6c88721b30
DEPS diff: bcf2616e8e..911054f7d0/DEPS

No update to Clang.

BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

TBR=phoglund@webrtc.org

Change-Id: I44c7a93cc90729704092b555a6a7ca0fa6846f2a
Reviewed-on: https://webrtc-review.googlesource.com/77765
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23304}
2018-05-18 13:37:06 +00:00
0a8f43580f Move VideoEncoderConfig from call/ to api/.
Bug: webrtc:8830
Change-Id: I42abd45bff9a70fe00733424b34874925c523dc8
Reviewed-on: https://webrtc-review.googlesource.com/77683
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23303}
2018-05-18 12:58:16 +00:00
b5750eb963 Revert "Add presubmit check for changes in 3pp"
This reverts commit 4103b383505575b23222f77fd04116d2f6c10273.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Add presubmit check for changes in 3pp
> 
> Presubmit check will test will new changes be overriden by autoroll
> or not. In more details presubmit will check:
> 1. Each dependency in third_party have to be specified in one of:
>    a. THIRD_PARTY_CHROMIUM_DEPS.json
>    b. THIRD_PARTY_WEBRTC_DEPS.json
> 2. Each dependency not specified in both files from #1
> 3. Changes won't be overriden by chromium third_party deps autoroll:
>    a. Changes were made in WebRTC owned dependency
>    b. Changes were addition of new Chromium owned dependency
> 
> Bug: webrtc:8366
> Change-Id: Ic5db24289e7fa461e0959f75cfbe81ecc65af4b5
> Reviewed-on: https://webrtc-review.googlesource.com/77421
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23301}

TBR=phoglund@webrtc.org,kwiberg@webrtc.org,titovartem@webrtc.org

Change-Id: Ib016ee4ac58729c2c0d302a964dbac71b4ae64af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8366
Reviewed-on: https://webrtc-review.googlesource.com/77780
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23302}
2018-05-18 12:46:26 +00:00
4103b38350 Add presubmit check for changes in 3pp
Presubmit check will test will new changes be overriden by autoroll
or not. In more details presubmit will check:
1. Each dependency in third_party have to be specified in one of:
   a. THIRD_PARTY_CHROMIUM_DEPS.json
   b. THIRD_PARTY_WEBRTC_DEPS.json
2. Each dependency not specified in both files from #1
3. Changes won't be overriden by chromium third_party deps autoroll:
   a. Changes were made in WebRTC owned dependency
   b. Changes were addition of new Chromium owned dependency

Bug: webrtc:8366
Change-Id: Ic5db24289e7fa461e0959f75cfbe81ecc65af4b5
Reviewed-on: https://webrtc-review.googlesource.com/77421
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23301}
2018-05-18 12:39:26 +00:00
a6ccd25eea Fix checkin chromium dep tool's test
TBR=phoglund@webrtc.org

Bug: webrtc:8366
Change-Id: I0c01d640060d6f604d95fa02faff61917c87c7ab
Reviewed-on: https://webrtc-review.googlesource.com/77680
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23300}
2018-05-18 10:49:56 +00:00
c7da266cb8 Android JNI generation: Set JNI namespace in build files
This CL removes the use of the @JNINamespace annotation and instead
sets the correct JNI namespace in the build file.

Bug: webrtc:8278
Change-Id: Ia4490399e45a97d56b02c260fd80df4edfa092bf
Reviewed-on: https://webrtc-review.googlesource.com/76440
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23299}
2018-05-18 10:44:38 +00:00
7e6fcea7de Ports CoreAudioUtil from Chrome to WebRTC.
See https://cs.chromium.org/chromium/src/media/audio/win/core_audio_util_win.h?q=coreaudio&sq=package:chromium&g=0&l=34
for details.

Bug: webrtc:9265
Change-Id: I0fd26620d94a81ccced68d81021c39723a5be2cb
Reviewed-on: https://webrtc-review.googlesource.com/76900
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23298}
2018-05-18 10:25:26 +00:00
7eb8e9fd7b Add RegisterExternalDecoder in VideoCodingModule.
In preparation for landing https://webrtc-review.googlesource.com/c/src/+/72441
a downstream project that uses the VideoCodingModule needs to be able to
inject a decoder object created from the outside, just like how encoders
are possible to inject.

Bug: webrtc:7925
Change-Id: Ibaeffda55f84410436d79f75730e7352e298b9f0
Reviewed-on: https://webrtc-review.googlesource.com/77160
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23297}
2018-05-18 09:43:26 +00:00
460f53bb86 Roll chromium_revision c92ed25217..bcf2616e8e (559015:559838)
Change log: c92ed25217..bcf2616e8e
Full diff: c92ed25217..bcf2616e8e

Roll chromium third_party 51c08cf9af..9d65a3cdda
Change log: 51c08cf9af..9d65a3cdda

Changed dependencies:
* src/base: a7a2409f9b..b802985ef4
* src/build: 03f39fd800..fc8308f6b6
* src/buildtools: a9e946f166..94288c26d2
* src/ios: e070a93062..289c450460
* src/testing: f5b31b58c6..a5fce03148
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d8600ccc2d..ce9b3742a1
* src/third_party/depot_tools: 8de3800ce5..8fe4d8cbef
* src/third_party/googletest/src: 045e7f9ee4..08d5b1f33a
* src/tools: e024720629..6e6e398687
* src/tools/swarming_client: 88229872dd..833f5ebf89
DEPS diff: c92ed25217..bcf2616e8e/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I22bf301fcec0103a1987a92f95ebf86e324dade7
Reviewed-on: https://webrtc-review.googlesource.com/77625
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23296}
2018-05-18 09:37:26 +00:00
c948fe62fd Delete unneeded includes of call/video_config.h.
Bug: webrtc:8830
Change-Id: I6114b47e5524a6d2450108388236478b1ceafb67
Reviewed-on: https://webrtc-review.googlesource.com/77425
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23295}
2018-05-18 09:00:56 +00:00
4639d46eaa Add vpython dependencies needed to run presubmit tests on LUCI
E.g. tools_webrtc/libs/generate_licenses_test.py needs 'mock', which needs the other libs added here.

This is taken from 641bce4223/.vpython (29)

TBR: kwiberg@webrtc.org
No-Try: True
Bug: chromium:749664
Change-Id: I180fc1190d2664c5a82f63857b740334145e7daa
Reviewed-on: https://webrtc-review.googlesource.com/77520
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23294}
2018-05-18 08:10:25 +00:00
638edfc88c Skipping some Opus tests to let the new roll flow.
In order to roll the new version of Opus in WebRTC, this CL disables
some tests that will fail because of [1].

They will be re-enabled and fixed as soon as the new Opus revision is
rolled.

[1] - https://chromium-review.googlesource.com/1061499

TBR=henrik.lundin@webrtc.org

Bug: webrtc:9280
Change-Id: I84870ced66d554f75c2d093dac8103ad7860cae5
Reviewed-on: https://webrtc-review.googlesource.com/77640
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23293}
2018-05-18 07:58:46 +00:00
4c8811b255 Delete some obsolete forward declarations
Bug: None
Change-Id: I3a9b59bf3dd63c206854ab949cf2d606046182c9
Reviewed-on: https://webrtc-review.googlesource.com/77427
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23292}
2018-05-18 07:29:25 +00:00
dd3eae5f94 Revert "Configure and use max bitrate to limit the AIMD controller estimates."
This reverts commit 18d7c7ea7e56444d6d7e6c8fb95b5f426fd7b953.

Reason for revert: 
This seems to cause the auto roller to Chrome to fail on Linux and Mac on the browsertest
WebRtcSimulcastBrowserTest.TestVgaReturnsTwoSimulcastStreams

https://chromium-review.googlesource.com/c/chromium/src/+/1064736


Original change's description:
> Configure and use max bitrate to limit the AIMD controller estimates.
> 
> Bug: webrtc:9275
> Change-Id: I9625cd473e1cb198abe08020f5462f1bd64bf2a5
> Reviewed-on: https://webrtc-review.googlesource.com/77081
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23287}

TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org

Change-Id: I8ed827ab6b2f7d2b70b9889e5a88701bfb974d35
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9275
Reviewed-on: https://webrtc-review.googlesource.com/77660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23291}
2018-05-18 07:12:26 +00:00
0ab95b97c8 Remove all baremetal bots, baremetal tests will be part of *_rel
See https://chromium-review.googlesource.com/c/chromium/tools/build/+/1058799
After the baremetal machines are moved to swarming, that CL will add the tasks that were previously in *_baremetal bots to *_rel bots. *_rel bots will run all the same tasks as before on a pool of generic machines, but also run a few tests on dedicated baremetal machines (pool:WebRTC-baremetal-try).

No-Try: True
Bug: chromium:755660
Change-Id: I99d62a84aac631b1c127bf661546baecb2a3ae9a
Reviewed-on: https://webrtc-review.googlesource.com/76721
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23290}
2018-05-18 07:03:32 +00:00
90394a4db3 Reland "[desktopCapture] Unify the position info in DIP coordinates on Mac."
This is a reland of 89653d5db46419d2a80898635cb27fed64898db2

Original change's description:
> [desktopCapture] Unify the position info in DIP coordinates on Mac.
> 
> On OSX, the logical(DIP) and physical coordinates are used mixingly.
> For example, the captured image has its size in physical pixels(2x) and
> location in logical(DIP) pixels. Same to the cursor position. This
> causes trouble when we check the relative position of image and cursor
> when there are multiple monitors with different DIP setting connected.
> 
> This cl proposed a solution to use DIP pixel for any location info,
> i.e. top-left of a frame and cursor position. Also propose a method to
> get the current scale factor of a window across multiple monitors. And
> save the current scale factor in DPI of the capture frame.
> Then we can check relative position of cursor and frame correctly
> in DIP pixel and compose them in physical pixel.
> 
> Bug: webrtc:9178
> Change-Id: I3c076aeac2d6f2c1f63d000d7fff03500aa375ac
> Reviewed-on: https://webrtc-review.googlesource.com/71621
> Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
> Reviewed-by: Zijie He <zijiehe@chromium.org>
> Commit-Queue: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23263}

Bug: webrtc:9178
Change-Id: I97d9150f7b9a4ed6671733b75613ea9c315d5c1d
Reviewed-on: https://webrtc-review.googlesource.com/77481
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23289}
2018-05-17 18:45:42 +00:00