The function GetExecutablePath is a hack with poor portability. Delete
it and its caller GetTestFilePath. The latter was used in
videoframe_unittest.h, where it is replaced by
webrtc::test::ResourcePath.
Delete unused functions declared in base/testutils.h: ReadFile,
GetSiblingDirectory, GetGoogle3Directory, GetTalkDirectory,
CmpHelperFileEq, EXPECT_FILEEQ, ASSERT_FILEEQ.
Delete unused functions declared in media/base/testutils.h:
GetTestFilePath (see above), LoadPlanarYuvTestImage,
DumpPlanarYuvTestImage, ComputePSNR, ComputeSumSquareError.
The functions LoadPlanarYuvTestImage, DumpPlanarYuvTestImage were used
in yuvscaler_unittests.cc and planarfunctions_unittests.cc, under
webrtc/pc. However, these tests are never compiled or run, and appear
not to have been for the last few years, and are therefore deleted
rather than updated. It might make sense to check if libyuv have
comparable tests, and if not, resurrect them as part of libyuv
unittests.
BUG=
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/2058043002 .
Cr-Commit-Position: refs/heads/master@{#13163}
The RtpReceiverObserverInterface is created.
The SignalFirstPacketReceived will be forwarded from BaseChannel to WebRtcSession.
WebRtcSession will forward SignalFirstAudioPacketReceived and SignalFirstVideoPacketReceived to the RtpReceiverInterface.
The application can listen to the Signal by implementing and registering a RtpReceiverObserver.
Review-Url: https://codereview.webrtc.org/1999853002
Cr-Commit-Position: refs/heads/master@{#13139}
Added the plumbing necessary to get two different lists of codecs from
WebRtcVoiceEngine up to MediaSessionDescriptionFactory.
This should be the last step in this set of CLs. Once
https://codereview.webrtc.org/1991233004/ has landed, it's possible to
implement the ReceiveCodecs getter with the info from the
AudioDecoderFactory. The factory needs to be updated to actually
produce the correct list, as well.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2013053002
Cr-Commit-Position: refs/heads/master@{#13131}
Added initial support for MediaSessionDescriptionFactory to pick different codecs based on communications direction (sendrecv, sendonly, recvonly, inactive) specifically for audio.
This adds some more degradation options for the answer: depending on answer options, it's now possible to degrade to INACTIVE from any offer, as well as to either RECVONLY or SENDONLY from a SENDRECV offer.
The set of "codecs" used for testing the answer was compiled using this spreadsheet:
https://docs.google.com/a/google.com/spreadsheets/d/1nVIfZLsFo5YK10_e80BCAADZnnRQ1devwwwAGmqJPow/edit?usp=sharing
I should probably condense it into a smaller table and put in the source.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1956343002
Cr-Commit-Position: refs/heads/master@{#13126}
Every message will now be traced with the location from which it was
posted, including function name, file and line number.
This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).
This logging should help us identify messages that are taking
longer than expected to be dispatched.
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2019423006 .
Cr-Commit-Position: refs/heads/master@{#13104}
The only thing that differs from the previous attempt in
https://codereview.webrtc.org/1979933002/ is that none of
the new targets are not hooked up to the webrtc target in
webrtc/BUILD.gn, which should make it not break the
chromium.webrtc.fyi bots.
Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.
Changes between previous attempt and the one before that
(https://codereview.webrtc.org/1973313002) are:
* Added libstunprober target
* Adjusted warnings for Chromium's clang plugins
* webrtc/pc/externalhmac.{h,cc} added for Chromium builds.
BUG=webrtc:4256
NOTRY=True
NOPRESUBMIT=True
TBR=perkj@webrtc.org, tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2037983002
Cr-Commit-Position: refs/heads/master@{#13030}
This means there's only one thread hop to the worker thread.
At the video engine level, SetOptions and SetSource
are combined into one method (all within the same critical section)
which ensures that no frame will be encoded while SetVideoSend
is only partially finished.
BUG=webrtc:5691
Review-Url: https://codereview.webrtc.org/1838413002
Cr-Commit-Position: refs/heads/master@{#13022}
By not providing the default implementation of the metrics API
it becomes possible for users of rtc_media to choose which
implementation to use. The dependency is moved into each test
target that uses it instead.
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2026223002
Cr-Commit-Position: refs/heads/master@{#12991}
Reason for revert:
Too many errors to address showed up when trying to land this with Chromium changes in https://codereview.chromium.org/2022833002/.
Will address them separately before relanding.
Original issue's description:
> Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
>
> Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
> preparation for removing src/third_party/libjingle in Chromium.
>
> Changes from previous attempt:
> * Added libstunprober target
> * Adjusted warnings for Chromium's clang plugins
> * webrtc/pc/externalhmac.{h,cc} added for Chromium builds.
>
> As soon this has landed a roll including the changes in
> https://codereview.chromium.org/2022833002/ is needed to make
> Chromium build cleanly.
>
> BUG=webrtc:4256
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/164e978f981c7810c4260c4184f41e26bae90230
> Cr-Commit-Position: refs/heads/master@{#12983}
TBR=perkj@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256
Review-Url: https://codereview.webrtc.org/2023233002
Cr-Commit-Position: refs/heads/master@{#12988}
Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.
Changes from previous attempt:
* Added libstunprober target
* Adjusted warnings for Chromium's clang plugins
* webrtc/pc/externalhmac.{h,cc} added for Chromium builds.
As soon this has landed a roll including the changes in
https://codereview.chromium.org/2022833002/ is needed to make
Chromium build cleanly.
BUG=webrtc:4256
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/1979933002
Cr-Commit-Position: refs/heads/master@{#12983}
This fixes a client breakage by adding back the RtpHeaderExtension temporarily
so that it can be fixed in the client before being removed in webrtc.
BUG=
CQ_INCLUDE_TRYBOTS=tryserver.chromium.linux:linux_chromium_rel_ng;tryserver.chromium.win:win_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2024153002
Cr-Commit-Position: refs/heads/master@{#12977}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
With this change, when max-bundle and rtcp-mux are both enabled, we no
longer create and destroy a temporary transport channel when a media
channel gets added. Instead, the media channel uses the correct bundled
transport channel from the start.
This fixes a bug where adding a media type would cause the ICE state to
briefly become Disconnected and then immediately recover. The temporary
channel was created in a non-writable state, which caused the
TransportController to declare the ICE state to be Disconnected (as not
all transport channels were writable). Right after creation, the
temporary channel was then destroyed and the ICE state went back to the
correct one.
BUG=webrtc:5856
Review-Url: https://codereview.webrtc.org/1972493002
Cr-Commit-Position: refs/heads/master@{#12781}
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.
R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1917193008 .
Cr-Commit-Position: refs/heads/master@{#12761}
Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.
BUG=webrtc:4256
NOTRY=True
TBR=perkj@webrtc.org
Review-Url: https://codereview.webrtc.org/1973313002
Cr-Commit-Position: refs/heads/master@{#12731}
The caller can set a negative or zero file size to avoid using a limit.
BUG=
Review-Url: https://codereview.webrtc.org/1974453002
Cr-Commit-Position: refs/heads/master@{#12730}
Reason for revert:
Breaks GN in Chromium.
Original issue's description:
> GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
>
> Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
> preparation for removing src/third_party/libjingle in Chromium.
>
> BUG=webrtc:4256
> NOTRY=True
>
> Committed: https://crrev.com/4d02a358b4205bd0f7b5f794b6fb8c157e075b9e
> Cr-Commit-Position: refs/heads/master@{#12724}
TBR=perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256
Review-Url: https://codereview.webrtc.org/1977853002
Cr-Commit-Position: refs/heads/master@{#12726}
Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.
BUG=webrtc:4256
NOTRY=True
Review-Url: https://codereview.webrtc.org/1929633002
Cr-Commit-Position: refs/heads/master@{#12724}
Some tests were passing in a local description created from hard-coded
SDP strings, which won't work in the future (since some attributes such
as the fingerprint and ICE ufrag/pwd are non-modifiable). These tests
now do the typical approach of calling CreateOffer and modifying the
result if necessary.
Also added some non-const versions of the SessionDescription accessor
helper functions, since that makes it much easier to modify a
SessionDescription. Previous alternatives were re-implementing the
helper methods from scratch, or converting the description to SDP,
modifying it, and converting it back.
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1966333002 .
Cr-Commit-Position: refs/heads/master@{#12704}
BaseChannel do calls to transport_channel on network_thread,
while keep calls to media_engine on worker_thread.
It still works when network_thread == worker_thread.
BUG=webrtc:5645
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1903393004 .
Cr-Commit-Position: refs/heads/master@{#12690}
Reason for revert:
Breaks user code. Said code needs to stop using scoped_ptr!
Original issue's description:
> Remove webrtc/base/scoped_ptr.h
>
> BUG=webrtc:5520
>
> NOTRY=True
>
> Committed: https://crrev.com/65fc62e9dd8a8716db625aaef76ab92f542ecc5a
> Cr-Commit-Position: refs/heads/master@{#12684}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520
Review-Url: https://codereview.webrtc.org/1965063003
Cr-Commit-Position: refs/heads/master@{#12686}
The CNAME is generated in the PeerConnection constructor and is populated through the MediaSessionOptions.
A default cname will be set in the MediaSessionOptions constructor.
BUG=webrtc:3431
Review-Url: https://codereview.webrtc.org/1871993002
Cr-Commit-Position: refs/heads/master@{#12650}
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.
BUG=webrtc:5520
Review-Url: https://codereview.webrtc.org/1937693002
Cr-Commit-Position: refs/heads/master@{#12581}
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
This propagated into various other places. Also had to #include headers that
were implicitly pulled by "scoped_ptr.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1920043002
Cr-Commit-Position: refs/heads/master@{#12501}
Reason for revert:
Breaks downstream for SRTP include paths. Will rework this and reland without that one.
Original issue's description:
> Remove the rtc_relative_path GYP variable and similar defines
>
> The defines that can be used to alter the include paths for Expat, SRTP
> and gtest are no longer needed in WebRTC or Chromium. Let's remove them
> to simplify the GYP a little.
>
> Removed defines:
> EXPAT_RELATIVE_PATH
> GTEST_RELATIVE_PATH
> SRTP_RELATIVE_PATH
>
> They're all set in the Chromium build so this shouldn't affect Chromium:
> https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp
>
> BUG=webrtc:4256
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/e19cf59eb6ee44fd4d7e7fbcfdd1a6ea75063605
> Cr-Commit-Position: refs/heads/master@{#12467}
TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256
Review URL: https://codereview.webrtc.org/1913043003
Cr-Commit-Position: refs/heads/master@{#12468}
The defines that can be used to alter the include paths for Expat, SRTP
and gtest are no longer needed in WebRTC or Chromium. Let's remove them
to simplify the GYP a little.
Removed defines:
EXPAT_RELATIVE_PATH
GTEST_RELATIVE_PATH
SRTP_RELATIVE_PATH
They're all set in the Chromium build so this shouldn't affect Chromium:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp
BUG=webrtc:4256
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1903553003
Cr-Commit-Position: refs/heads/master@{#12467}
Reason for revert:
Broke the Chromium build by introducing static initializers.
Original issue's description:
> Accept all the media profiles required by JSEP.
>
> JSEP section 5.1.3 states that:
> Any profile matching the following patterns MUST be accepted:
> "RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"
>
> NOTRY=True
> BUG=webrtc:5638
>
> Committed: https://crrev.com/b7f425ab68ec58e2a5beaaf5ef79f50f1982c6f9
> Cr-Commit-Position: refs/heads/master@{#12338}
TBR=deadbeef@webrtc.org,pthatcher@webrtc.org,avi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5638
Review URL: https://codereview.webrtc.org/1882923002
Cr-Commit-Position: refs/heads/master@{#12351}
This field only existed as an implementation detail for getting the
codecs sorted, so it doesn't need to be in the public interface.
It cluttered the code and undesirably affected codec comparisons,
causing the video encoder to be reconfigured if a codec's preference
changed but nothing else did.
BUG=webrtc:5690
Review URL: https://codereview.webrtc.org/1845673002
Cr-Commit-Position: refs/heads/master@{#12349}
JSEP section 5.1.3 states that:
Any profile matching the following patterns MUST be accepted:
"RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"
NOTRY=True
BUG=webrtc:5638
Review URL: https://codereview.webrtc.org/1880913002
Cr-Commit-Position: refs/heads/master@{#12338}
This change builds on top of the refactoring in https://codereview.webrtc.org/1841083008/, and enables WebRTC client applications to control the max send bitrate for every audio stream through RtpParameters.
The AudioSendStream now stores the last codec spec, and whenever a global or per-stream bitrate limit changes, the effective limit (smaller of the two) is recomputed and the codec is reconfigured with that bitrate.
TBR=pthatcher
BUG=
Review URL: https://codereview.webrtc.org/1847353004
Cr-Commit-Position: refs/heads/master@{#12290}
so that the call knows which packet ids were sent on the previous candidate pair.
Note that packet_id is actually 16bits, so we can use -1 for values that are not set.
Also moved the tests for candidate pair changes to TestSelectConnectionBeforeNomination.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1842093002 .
Cr-Commit-Position: refs/heads/master@{#12184}
The re-land moves the isolate build targets for media.gyp
and pc.gyp into the include_tests==1 condition.
This has been tested in a Chromium checkout and no longer
causes the error that was seen after landing
https://codereview.webrtc.org/1839763004/
Original issue's description:
> Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ )
>
> Reason for revert:
> Breaks Chromium: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/11313/steps/gclient%20runhooks/logs/stdio:
>
> Updating projects from gyp files...
> Using overrides found in /Users/chrome-bot/.gyp/include.gypi
> Traceback (most recent call last):
> File "src/build/gyp_chromium", line 12, in <module>
> execfile(__file__ + '.py')
> File "src/build/gyp_chromium.py", line 341, in <module>
> sys.exit(main())
> File "src/build/gyp_chromium.py", line 328, in main
> gyp_rc = gyp.main(args)
> File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 538, in main
> return gyp_main(args)
> File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 514, in gyp_main
> options.duplicate_basename_check)
> File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 130, in Load
> params['parallel'], params['root_targets'])
> File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 2800, in Load
> RemoveLinkDependenciesFromNoneTargets(targets)
> File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 1510, in RemoveLinkDependenciesFromNoneTargets
> if targets[t].get('variables', {}).get('link_dependency', 0):
> KeyError: '/b/build/slave/Mac_Builder/build/src/third_party/webrtc/media/media.gyp:rtc_media_unittests#target'
> Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/Mac_Builder/build
> Hook '/usr/bin/python src/build/gyp_chromium' took 20.29 secs
>
> Original issue's description:
> > Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files.
> >
> > These contributes to circular dependency problems in WebRTC
> > since one have to depend on webrtc.gyp in order to depend on
> > a target in them.
> >
> > This reduces the number of cyclic dependencies in WebRTC from 21
> > to 16.
> >
> > BUG=webrtc:4243
> > NOTRY=True
> > NOPRESUBMIT=True
> >
> > Committed: https://crrev.com/231b69f28dd22f4e2d98e5048f8eaae7b20915e6
> > Cr-Commit-Position: refs/heads/master@{#12166}
>
> TBR=pthatcher@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4243
>
> Committed: https://crrev.com/72644d2cf6b14bbc4a107f79158eaa225f3196b5
> Cr-Commit-Position: refs/heads/master@{#12167}
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243
Review URL: https://codereview.webrtc.org/1843193002
Cr-Commit-Position: refs/heads/master@{#12180}
- Remove WVoE::SetAudioDeviceModule() - the ADM is now supplied in ctor.
- Remove WVoE::Init() and WVoE::Terminate().
- Remove MediaEngineInterface::Terminate().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1830213002
Cr-Commit-Position: refs/heads/master@{#12173}
Reason for revert:
Breaks Chromium: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/11313/steps/gclient%20runhooks/logs/stdio:
Updating projects from gyp files...
Using overrides found in /Users/chrome-bot/.gyp/include.gypi
Traceback (most recent call last):
File "src/build/gyp_chromium", line 12, in <module>
execfile(__file__ + '.py')
File "src/build/gyp_chromium.py", line 341, in <module>
sys.exit(main())
File "src/build/gyp_chromium.py", line 328, in main
gyp_rc = gyp.main(args)
File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 538, in main
return gyp_main(args)
File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 514, in gyp_main
options.duplicate_basename_check)
File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/__init__.py", line 130, in Load
params['parallel'], params['root_targets'])
File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 2800, in Load
RemoveLinkDependenciesFromNoneTargets(targets)
File "/b/build/slave/Mac_Builder/build/src/tools/gyp/pylib/gyp/input.py", line 1510, in RemoveLinkDependenciesFromNoneTargets
if targets[t].get('variables', {}).get('link_dependency', 0):
KeyError: '/b/build/slave/Mac_Builder/build/src/third_party/webrtc/media/media.gyp:rtc_media_unittests#target'
Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/Mac_Builder/build
Hook '/usr/bin/python src/build/gyp_chromium' took 20.29 secs
Original issue's description:
> Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files.
>
> These contributes to circular dependency problems in WebRTC
> since one have to depend on webrtc.gyp in order to depend on
> a target in them.
>
> This reduces the number of cyclic dependencies in WebRTC from 21
> to 16.
>
> BUG=webrtc:4243
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/231b69f28dd22f4e2d98e5048f8eaae7b20915e6
> Cr-Commit-Position: refs/heads/master@{#12166}
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243
Review URL: https://codereview.webrtc.org/1846693002
Cr-Commit-Position: refs/heads/master@{#12167}
These contributes to circular dependency problems in WebRTC
since one have to depend on webrtc.gyp in order to depend on
a target in them.
This reduces the number of cyclic dependencies in WebRTC from 21
to 16.
BUG=webrtc:4243
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1839763004
Cr-Commit-Position: refs/heads/master@{#12166}
