Commit Graph

9190 Commits

Author SHA1 Message Date
f81d8efb24 Promote iOS Simulator 9.0 bot to main waterfall.
Rename existing iOS Simulator bots to match the new one.
Buildbot changes: https://chromium-review.googlesource.com/423055

BUG=chromium:677385
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2601353003 .
Cr-Commit-Position: refs/heads/master@{#15842}
2016-12-30 07:26:24 +00:00
81fa52ff44 Change iOS Simulator bot to 32-bit and iPhone 5.
Before promoting it to the main waterfall I thought it made
sense to ensure it works for a phone that actually is 32-bit
(iPhone 6s isn't). It also makes more sense to run the older
iOS version (9) on the older phone.

BUG=chromium:677385
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2604203002 .
Cr-Commit-Position: refs/heads/master@{#15837}
2016-12-29 20:01:38 +00:00
12166898d9 Rename iOS JSON file.
The currently checked in file from
https://codereview.webrtc.org/2604153002/ contained a
whitespace in the filename.

BUG=chromium:677385
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2602053002 .
Cr-Commit-Position: refs/heads/master@{#15833}
2016-12-29 13:00:23 +00:00
f423593f55 Reland of Refactor webrtc/modules/desktop_capture for GN check
Reason for revert:
Trying to reland this CL.

Original issue's description:
> Revert of Refactor webrtc/modules/desktop_capture for GN check (patchset #1 id:1 of https://codereview.webrtc.org/2593713002/ )
>
> Reason for revert:
> Apparently breaks Chromium compile for unknown reason:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/12314
>
> Original issue's description:
> > Refactor webrtc/modules/desktop_capture for GN check
> >
> > This moves some GN check configurations out of .gn to individual
> > targets.
> >
> > The now checked target is:
> > "//webrtc/modules/desktop_capture/*"
> >
> > BUG=webrtc:6828
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2593713002
> > Cr-Commit-Position: refs/heads/master@{#15725}
> > Committed: 70870b9211
>
> TBR=sergeyu@chromium.org,mbonadei@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6828
>
> Review-Url: https://codereview.webrtc.org/2597923002
> Cr-Commit-Position: refs/heads/master@{#15750}
> Committed: d943c48454

TBR=sergeyu@chromium.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6828

Review-Url: https://codereview.webrtc.org/2609523002
Cr-Commit-Position: refs/heads/master@{#15832}
2016-12-29 11:35:56 +00:00
1f971427e8 Add iOS64 simulator bot running on iOS 9.0 simulator.
Add a new trybot configuration as well.

BUG=chromium:677385
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2604153002 .
Cr-Commit-Position: refs/heads/master@{#15829}
2016-12-29 08:31:05 +00:00
655f7cf355 Prepare to introduce the IceTransportInternal.
The P2PTransportChannel will eventually inherit from IceTransportInternal instead of TransportChannelImpl.
However, the Chromium/remoting depends on TransportChannel and TransportChannelImpl.

The solution to work around this:
Step1:
  Make a WebRTC CL to introduce IceTransportInternal and IceTransportInternal2 by type-defining
  TransportChannel and TransportChannelImpl.
Step2:
  Make a Chromium CL to replace the TransportChannel and TransportChannelImpl with
  IceTransportInternal and IceTransportInternal2.
Step3:
  Make a WebRTC to redefine IceTransportInternal2 to be IceTransportInternal and switch the base
  class of P2PTransportChannel with IceTransportInternal.
Step4"
  Make a Chromium CL to remove the IceTransportInternal2.

This CL is the Step1. The real IceTransportInternal implementation
is commented out temporarily.

BUG=none

Review-Url: https://codereview.webrtc.org/2598103003
Cr-Commit-Position: refs/heads/master@{#15824}
2016-12-28 21:55:02 +00:00
94b880178a Revert of Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9 (patchset #1 id:1 of https://codereview.webrtc.org/2568743007/ )
Reason for revert:
Trying to re-enable this test as we're now using a newer Clang version (289944-2).

Original issue's description:
> Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9
>
> This test is flaky on all platforms, not just Android. Disabling it entirely until webrtc:6057 is fixed.
>
> BUG=webrtc:6057
>
> Committed: https://crrev.com/bb66ec35739830847bfb0146cd029ca41421b2d8
> Cr-Commit-Position: refs/heads/master@{#15594}

TBR=marpan@webrtc.org,sprang@webrtc.org,skvlad@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6057

Review-Url: https://codereview.webrtc.org/2603993002
Cr-Commit-Position: refs/heads/master@{#15821}
2016-12-28 15:22:54 +00:00
ac4a90dfdc Add frame rate throttling to vp8 screenshare_layers.
Drop frames if incoming frame rate is higher than the configured max
framerate.

BUG=webrtc:6897

Review-Url: https://codereview.webrtc.org/2578993002
Cr-Commit-Position: refs/heads/master@{#15819}
2016-12-28 13:58:07 +00:00
49f465f4b2 Refactor bitrate_controller, remote_bitrate_estimator and congestion_contoller for gn check.
These targets are now checked:
- "//webrtc/modules/bitrate_controller/*"
- "//webrtc/modules/congestion_controller/*"
- "//webrtc/modules/remote_bitrate_estimator/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2602563003
Cr-Commit-Position: refs/heads/master@{#15818}
2016-12-28 12:43:46 +00:00
7667db4a74 Fixing init time error in smoothing filter.
BUG=webrtc:6909, webrtc:6303

TBR=tina.legrand@webrtc.org

Review-Url: https://codereview.webrtc.org/2582043002
Cr-Commit-Position: refs/heads/master@{#15817}
2016-12-28 10:57:50 +00:00
4716048ccd Fixing name of a unittest for AudioEncoderOpus.
BUG=webrtc:6936, webrtc:6303

Review-Url: https://codereview.webrtc.org/2606743002
Cr-Commit-Position: refs/heads/master@{#15811}
2016-12-27 21:08:49 +00:00
84f83f8c0c Remove OverUseDetectorOptions from OveruseDetector since it isn't used.
BUG=None

Review-Url: https://codereview.webrtc.org/2580733004
Cr-Commit-Position: refs/heads/master@{#15809}
2016-12-27 18:43:01 +00:00
584c35a334 Make a the decisions of ANA optional for the opus encoder.
BUG=webrtc:6936, webrtc:6303

Review-Url: https://codereview.webrtc.org/2592253004
Cr-Commit-Position: refs/heads/master@{#15807}
2016-12-27 16:21:29 +00:00
e1c2d9b0a8 Reducing calling to SmoothingFilter in Fec Controller.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2585293002
Cr-Commit-Position: refs/heads/master@{#15806}
2016-12-27 14:42:49 +00:00
6d3c57300b Reland "Refactor webrtc/modules/video_processing for GN check"
This reverts commit d39e16ac300d7947d22b953898aaef073e553ad3.

This will fix the missing dependency which was causing the failure of many
buildbots.

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2597643002
Cr-Commit-Position: refs/heads/master@{#15805}
2016-12-27 14:32:08 +00:00
05afa8bce6 Refactor webrtc/modules/utility for gn check
The now checked target is:
"//webrtc/modules/utility/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2606483002
Cr-Commit-Position: refs/heads/master@{#15803}
2016-12-27 13:40:15 +00:00
7d9305f473 Remove WebRTC-specific test runner script for Android
After moving away from the Chromium checkout and the symlinks, we
no longer need to have our own test_runner.py script for Android tests

BUG=webrtc:5006
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2600193002 .
Cr-Commit-Position: refs/heads/master@{#15801}
2016-12-27 08:21:19 +00:00
1e23461d5e Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
Reason for revert:
Broke chromium FYI bot because the chromium mock PC overrides the method whose signature is changing.

Also broke a downstream internal test, which I need to investigate further.

Original issue's description:
> Adding error output param to SetConfiguration, using new RTCError type.
>
> Most notably, will return "INVALID_MODIFICATION" if a field in the
> configuration was modified and modification of that field isn't supported.
>
> Also changing RTCError to a class that wraps an enum type, because it will
> eventually need to hold other information (like SDP line number), to match
> the RTCError that was recently added to the spec:
> https://github.com/w3c/webrtc-pc/pull/850
>
> BUG=webrtc:6916
>
> Review-Url: https://codereview.webrtc.org/2587133004
> Cr-Commit-Position: refs/heads/master@{#15777}
> Committed: 7a5fa6cd61

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6916

Review-Url: https://codereview.webrtc.org/2600813002
Cr-Commit-Position: refs/heads/master@{#15778}
2016-12-24 09:43:32 +00:00
7a5fa6cd61 Adding error output param to SetConfiguration, using new RTCError type.
Most notably, will return "INVALID_MODIFICATION" if a field in the
configuration was modified and modification of that field isn't supported.

Also changing RTCError to a class that wraps an enum type, because it will
eventually need to hold other information (like SDP line number), to match
the RTCError that was recently added to the spec:
https://github.com/w3c/webrtc-pc/pull/850

BUG=webrtc:6916

Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#15777}
2016-12-24 08:47:59 +00:00
d01ed1fe8f Fix an error in Audio Network Adaptor: time constant passed wrong.
BUG=wbrtc:6303

Review-Url: https://codereview.webrtc.org/2595283003
Cr-Commit-Position: refs/heads/master@{#15767}
2016-12-23 09:49:37 +00:00
e97389c505 If network enumeration fails, try binding to the "ANY" address.
This isn't as good as being able to enumerate all networks, but it's better
than doing nothing; it still will provide STUN/TURN candidates for the default
route if one exists.

BUG=webrtc:6932

Review-Url: https://codereview.webrtc.org/2599673003
Cr-Commit-Position: refs/heads/master@{#15766}
2016-12-23 09:43:45 +00:00
40610e24ce Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.
This allows building without SCTP support (and even building/running
tests). The "HAVE_SCTP" define has been functional for a while, but there
wasn't any easy way to turn it on/off.

NOTRY=True
BUG=webrtc:6933

Review-Url: https://codereview.webrtc.org/2593313002
Cr-Commit-Position: refs/heads/master@{#15763}
2016-12-22 18:53:38 +00:00
fe50b4d750 Make class of static functions in rtp_to_ntp.h:
- UpdateRtcpList
- RtpToNtp

class RtpToNtpEstimator
- UpdateMeasurements
- Estimate

List with rtcp measurements is now private.

BUG=none

Review-Url: https://codereview.webrtc.org/2574133003
Cr-Commit-Position: refs/heads/master@{#15762}
2016-12-22 15:53:51 +00:00
bf5f5297c5 Disable flaky VideoSendStreamTest.RemoveOverheadeFromBandwidth
BUG=webrtc:6886
NOTRY=True
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2596223002
Cr-Commit-Position: refs/heads/master@{#15761}
2016-12-22 15:51:54 +00:00
ebafdc8484 Refactor webrtc/modules/rtp_rtcp for GN check
This moves some GN check configurations out of .gn to individual
targets.

This commit also removes the source file 'mocks/mock_rtp_rtcp.h' from
the static_library 'rtp_rtcp' because it depends on a 'testonly = true'
target. After a check this seems only included in the unitest code:

$ grep -Rn "mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/
webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc:18:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc:17:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"

This commit also removes the dependency on
'//webrt/modules/video_coding' because it seems that the following
include can be removed:

#include "webrtc/modules/video_coding/include/video_coding_defines.h"

The now checked target is:
"//webrtc/modules/rtp_rtcp/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2598963002
Cr-Commit-Position: refs/heads/master@{#15760}
2016-12-22 15:35:39 +00:00
000d16396e Refactor webrtc/modules/audio_conference_mixer for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/audio_conference_mixer/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2593003002
Cr-Commit-Position: refs/heads/master@{#15759}
2016-12-22 14:56:21 +00:00
0de11aa130 Landmine to clobber failing Android x86/x64 builds
Clobber to fix Android x86/x64 builds after
https://codereview.webrtc.org/1414343008/

They started failing with errors like
../../third_party/android_tools/ndk/platforms/android-21/arch-x86_64/usr/include/stdint.h:32:20: fatal error: stddef.h: No such file or directory
   #include <stddef.h>
             ^
from https://build.chromium.org/p/tryserver.webrtc/builders/android_compile_x64_dbg/builds/10032/steps/compile/logs/stdio
A clobbered build solved the problem.

BUG=webrtc:5006
TBR=mbonadei@webrtc.org

Review-Url: https://codereview.webrtc.org/2601473002 .
Cr-Commit-Position: refs/heads/master@{#15757}
2016-12-22 11:40:56 +00:00
526248779a Disables AudioDeviceTest.StartStopPlayout on iOS
BUG=webrtc:6889
NOTRY=True

Review-Url: https://codereview.webrtc.org/2595303002
Cr-Commit-Position: refs/heads/master@{#15753}
2016-12-22 09:36:49 +00:00
8d5608880f Do not call OnDecoderTiming before timing values are set.
Wait until first frame is decoded to avoid include zeros in stats.

BUG=b/32659204

Review-Url: https://codereview.webrtc.org/2582313002
Cr-Commit-Position: refs/heads/master@{#15752}
2016-12-22 09:26:18 +00:00
c37ad499da Revert of Make P2PTransportChannel inherit from IceTransportInternal. (patchset #3 id:80001 of https://codereview.webrtc.org/2590063002/ )
Reason for revert:
Breaks Chromium WebRTC FYI bots:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/12337
The error was masked by another breaking change that was committer earlier. This is the first build showing the error.

Original issue's description:
> Make P2PTransportChannel inherit from IceTransportInternal.
>
> Make P2PTransportChannel inherit from IceTransportInternal instead of
> TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
> be separated from P2PTransportChannel.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2590063002
> Cr-Commit-Position: refs/heads/master@{#15743}
> Committed: 12749d89d9

TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none

Review-Url: https://codereview.webrtc.org/2594343002
Cr-Commit-Position: refs/heads/master@{#15751}
2016-12-22 07:52:00 +00:00
d943c48454 Revert of Refactor webrtc/modules/desktop_capture for GN check (patchset #1 id:1 of https://codereview.webrtc.org/2593713002/ )
Reason for revert:
Apparently breaks Chromium compile for unknown reason:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/12314

Original issue's description:
> Refactor webrtc/modules/desktop_capture for GN check
>
> This moves some GN check configurations out of .gn to individual
> targets.
>
> The now checked target is:
> "//webrtc/modules/desktop_capture/*"
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2593713002
> Cr-Commit-Position: refs/heads/master@{#15725}
> Committed: 70870b9211

TBR=sergeyu@chromium.org,mbonadei@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828

Review-Url: https://codereview.webrtc.org/2597923002
Cr-Commit-Position: refs/heads/master@{#15750}
2016-12-22 07:19:59 +00:00
dcccda7e7c Created a java wrapper for the callback OnAddTrack to PeerConnection.Observer
Created a java wrapper for the callback OnAddTrack in this CL since it has been added to native C++ API
The callback function is called when a track is signaled by remote side and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.

BUG=webrtc:6112

Review-Url: https://codereview.webrtc.org/2513723002
Cr-Commit-Position: refs/heads/master@{#15745}
2016-12-21 22:08:03 +00:00
12749d89d9 Make P2PTransportChannel inherit from IceTransportInternal.
Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.

BUG=none

Review-Url: https://codereview.webrtc.org/2590063002
Cr-Commit-Position: refs/heads/master@{#15743}
2016-12-21 18:26:18 +00:00
b29e652b10 Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )"
Problem fixed: RTP header extensions were not properly set in tests.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2593963003
Cr-Commit-Position: refs/heads/master@{#15741}
2016-12-21 14:37:18 +00:00
23368e1aef RTCStatsCollectorTest: ExpectReportContainsCertificateInfo /w EXPECT_EQ
Modify ExpectReportContainsCertificateInfo to use EXPECT_EQ checks of
RTCCertificateStats objects.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2594553003
Cr-Commit-Position: refs/heads/master@{#15738}
2016-12-21 12:29:17 +00:00
c42ba32877 RTCStatsCollectorTest: Remove ExpectReportContainsCandidate.
Remove ExpectReportContainsCandidate in favor of EXPECT_EQ checks of
RTC[Local/Remote]IceCandidateStats objects.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2594753002
Cr-Commit-Position: refs/heads/master@{#15737}
2016-12-21 11:31:45 +00:00
e55b16c664 Drop unneeded include of media_file.h.
BUG=None

Review-Url: https://codereview.webrtc.org/2587403002
Cr-Commit-Position: refs/heads/master@{#15736}
2016-12-21 11:05:44 +00:00
504b95eff8 Avoid creating receiver_time outliers in the VideoAnalyzer.
Prior to this change, the receiver_time metric had huge outliers
whenever FlexFEC was enabled. This was due to a measurement problem,
where the time of the incoming packet was incorrectly set to zero.
This happened for packets that were lost in transit, but recovered
through FEC.

This CL fixes this problem by simply not recording samples where the
incoming packet time is undefined. The CL also removes the possibility
of timestamp collisions in the data structures.

TESTED=Ran './webrtc_perf_tests --gtest_filter="*ForemanCifPlr5H264Flexfec*" | grep receiver_time' locally 10 times, without experiencing any outliers.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2596793002
Cr-Commit-Position: refs/heads/master@{#15735}
2016-12-21 10:54:35 +00:00
d39e16ac30 Revert of Refactor webrtc/modules/video_processing for GN check (patchset #3 id:40001 of https://codereview.webrtc.org/2595543002/ )
Reason for revert:
This CL broke some buildbots. I will investigate it later.

Original issue's description:
> Refactor webrtc/modules/video_processing for GN check
>
> This moves some GN check configurations out of .gn to individual
> targets.
>
> The now checked target is:
> "//webrtc/modules/video_processing/*"
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2595543002
> Cr-Commit-Position: refs/heads/master@{#15732}
> Committed: 00a810b844

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828

Review-Url: https://codereview.webrtc.org/2594973002
Cr-Commit-Position: refs/heads/master@{#15733}
2016-12-21 10:18:54 +00:00
00a810b844 Refactor webrtc/modules/video_processing for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/video_processing/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2595543002
Cr-Commit-Position: refs/heads/master@{#15732}
2016-12-21 10:01:26 +00:00
dbb64d8f27 RTCStatsCollectorTest: Remove ExpectReportContainsDataChannel.
Remove ExpectReportContainsDataChannel in favor of EXPECT_EQ checks of
RTCDataChannelStats objects.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2597433002
Cr-Commit-Position: refs/heads/master@{#15731}
2016-12-21 09:57:46 +00:00
02d2a92d92 RTCStatsReport::AddStats DCHECKs that the ID is unique.
Previously it was allowed to call AddStats with stats of the same ID
multiple times.

This revealed a few things:
- Local and remote streams can have the same label.
  RTCMediaStreamStats's ID is updated to include "local"/"remote".
- The same certificate can show up multiple times (e.g. for local and
  remote in a loopback), so we skip creating RTCCertificateStats for the
  same certificate multiple times

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2593503003
Cr-Commit-Position: refs/heads/master@{#15730}
2016-12-21 09:29:05 +00:00
2a495ca297 Refactor webrtc/modules/pacing for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/pacing/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2594523003
Cr-Commit-Position: refs/heads/master@{#15729}
2016-12-21 08:26:58 +00:00
01c715096f Move nat-related code to target rtc_base_tests_utils.
BUG=None

Review-Url: https://codereview.webrtc.org/2591733002
Cr-Commit-Position: refs/heads/master@{#15728}
2016-12-21 08:23:08 +00:00
70e4053844 Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )
Reason for revert:
Unexpected perf regressions.

Original issue's description:
> Parse FlexFEC RTP headers in Call and add integration with BWE.
>
> BUG=webrtc:5654
>
> Review-Url: https://codereview.webrtc.org/2553863003
> Cr-Commit-Position: refs/heads/master@{#15709}
> Committed: ab2ffa3b28

TBR=philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2589393002
Cr-Commit-Position: refs/heads/master@{#15727}
2016-12-21 08:22:03 +00:00
ba96730bd8 Refactor webrtc/modules/media_file for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/media_file/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2593693002
Cr-Commit-Position: refs/heads/master@{#15726}
2016-12-21 08:20:52 +00:00
70870b9211 Refactor webrtc/modules/desktop_capture for GN check
This moves some GN check configurations out of .gn to individual
targets.

The now checked target is:
"//webrtc/modules/desktop_capture/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2593713002
Cr-Commit-Position: refs/heads/master@{#15725}
2016-12-21 07:42:05 +00:00
fe4a8a41ad Implement current/pending session description methods.
BUG=webrtc:6917

Review-Url: https://codereview.webrtc.org/2590753002
Cr-Commit-Position: refs/heads/master@{#15722}
2016-12-21 01:56:17 +00:00
494dff4c07 Fix a screen capture issue on retina macOS devices.
The CGDisplayStream API returns rects in physical pixel coordinates, not
Density-Independent Pixel coordinates. The code was incorrectly re-applying the
dip_to_pixel scaling.

BUG=chromium:675490

Review-Url: https://codereview.webrtc.org/2588973002
Cr-Commit-Position: refs/heads/master@{#15720}
2016-12-21 01:00:22 +00:00
1b08dc33eb To verify the upcoming code changes it is required
that the level of the output in the audio processing
module is monitored. This CL adds that.

BUG=webrtc:6181, webrtc:6183, webrtc:6220

Review-Url: https://codereview.webrtc.org/2549143004
Cr-Commit-Position: refs/heads/master@{#15718}
2016-12-20 21:45:58 +00:00