The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
The implementation is borrowed from Chromium.
Also change use of raw pointers in VideoSendStreamImpl to use WeakPtr<>
BUG= webrtc:6289
Review-Url: https://codereview.webrtc.org/2367853002
Cr-Commit-Position: refs/heads/master@{#14468}
It's a very general type, and we're about to start needing it in other
places besides AudioCodingModule.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2380463003
Cr-Commit-Position: refs/heads/master@{#14423}
This is done to ensure GN targets are placed in the same directory as of the source files.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2365383004
Cr-Commit-Position: refs/heads/master@{#14411}
Most other targets were converted to static_library to match the
GYP build in https://codereview.webrtc.org/2361623004.
The monolithic 'webrtc' wasn't converted due to a limitation in
libtool on Mac: it will fail if there are no object files to link
(see http://crbug.com/625874).
Because of that, a no_op_function.cc source file was added to
the sources here, even if it's not used. This will be further
refactored in the future as part of the Slim and Modular
effort (bugs.webrtc.org/5716).
BUG=webrtc:6418, webrtc:6410
NOTRY=True
Review-Url: https://codereview.webrtc.org/2372983002
Cr-Commit-Position: refs/heads/master@{#14403}
This CL removes the use_objc_h264 flag. This means that the VideoToolbox
H264 encoder and decoder will always be built.
BUG=webrtc:4081
NOTRY=TRUE
Review-Url: https://codereview.webrtc.org/2366443003
Cr-Commit-Position: refs/heads/master@{#14372}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
Reason for revert:
Breaks internal project.
Original issue's description:
> Rename AppRTCDemo on Android and iOS to AppRTCMobile
>
> The purpose is to make it clearer it is a mobile application.
>
> BUG=webrtc:6359
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/d3af58bdab5b25acd62cd816363becc7003d3e5a
> Cr-Commit-Position: refs/heads/master@{#14356}
TBR=sakal@webrtc.org,kthelgason@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:6359
Review-Url: https://codereview.webrtc.org/2358133003
Cr-Commit-Position: refs/heads/master@{#14358}
The purpose is to make it clearer it is a mobile application.
BUG=webrtc:6359
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2343403002
Cr-Commit-Position: refs/heads/master@{#14356}
For some reason the RtcEventLog unit tests were not building and
running. This CL adds these tests to the rtc_unittests target.
They are only built if protobuf support is enabled.
BUG=webrtc:6379
R=stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2344383002 .
Cr-Commit-Position: refs/heads/master@{#14295}
With the small number of tests these targets contain, it
makes more sense to gather them into android_junit_tests
instead, which was created to be a high-level target containing
all the junit tests.
BUG=chromium:647390
NOTRY=True
Review-Url: https://codereview.webrtc.org/2347403002
Cr-Commit-Position: refs/heads/master@{#14292}
Declare resources for GN targets so that they can be isolated
NOTRY=True
BUG=chromium:497757
Review-Url: https://codereview.webrtc.org/2340753002
Cr-Commit-Position: refs/heads/master@{#14210}
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
BUG=
NOPRESUBMIT=true
Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Original-Commit-Position: refs/heads/master@{#14191}
Cr-Commit-Position: refs/heads/master@{#14198}
Moved it inside the rtc_include_tests if clause
so that it build only when tests flag is set to true.
NOTRY=True
BUG=31425205
Review-Url: https://codereview.webrtc.org/2337463002
Cr-Commit-Position: refs/heads/master@{#14197}
Reason for revert:
Breaks downstream code
Original issue's description:
> Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/
>
> Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
>
> Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
>
> BUG=
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
> Cr-Commit-Position: refs/heads/master@{#14191}
TBR=kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2336123002
Cr-Commit-Position: refs/heads/master@{#14193}
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
BUG=
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Commit-Position: refs/heads/master@{#14191}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Add "//build/config/compiler:optimize_max" to rtc_add_configs and
"//build/config/compiler:default_optimization" to rtc_remove_configs.
This is the default optimization in GYP, and might help explain a 82.5%
regression in webrtc_perf_tests at 13946:13946
BUG=chromium:641966
NOTRY=True
Review-Url: https://codereview.webrtc.org/2307283002
Cr-Commit-Position: refs/heads/master@{#14067}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Remove //build/config/sanitizers:deps as a dependency for
all rtc_executable targets and add it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2308553002
Cr-Commit-Position: refs/heads/master@{#14048}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
The compiler optimization for Windows is O1 by default in GN, but O2 in GYP.
This might help explain the regression observed on neteq_performance_unittest.
NOTRY=True
BUG=641966
Review-Url: https://codereview.webrtc.org/2291253003
Cr-Commit-Position: refs/heads/master@{#13999}
In order to get resource files to be properly packaged into
the .app for a unit test on iOS, the resource files needs
to be listed as sources in a bundle_data target.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2292853002
Cr-Commit-Position: refs/heads/master@{#13968}
iOS tests packaged into an .app uses the same way of
defining resources (the data attribute). Some iOS
simulator tests are failing due to missing resources, so
let's sync them all.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2277753003
Cr-Commit-Position: refs/heads/master@{#13898}
The old and new getStats are very different. This CL proposes rewriting
the new getStats from scratch with a bottom-up approach, starting with
the fundamental stats classes. This will allow cleaner and more
efficient code that is more aligned with the spec.
RTCStats and subclasses are the equivalent to RTCStats and RTCStats-
-derived dictionaries from the specs[1][2]. The dictionary members are
public member variables of type RTCStatsMember<T>, where T is one of the
supported types. All members derive from RTCStatsMemberInterface and
iteration of members is possible with RTCStats::Members().
The members are not stored in a map for performance and readability.
Type checking is supported with static class variables, kType.
Only the supported member types T are specialized and may be
instantiated, and sequences are supported with std::vector<...>. Type
checking is again supported with static class variables, kType.
RTCStatsReport is the equivalent from the spec[3], and maps RTCStats::id
to RTCStats-objects. RTCStatsReport is reference counted. It and its
contained stats may be destroyed on any thread. When the
RTCStatsCollector is added in a follow-up CL, it will return const
references to the RTCStatsReports. This means copies don't have to be
made for multiple stats observers or when jumping threads. In fact, no
copies of any stats will have to be made in surfacing stats to Blink.
[1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstats-dictionary
[2] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html
[3] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object
This adds the new folder webrtc/stats/, with target rtc_stats and binary
rtc_stats_unittests. Public api headers are placed in webrtc/api/ and
.cc files are placed in webrtc/stats/.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2241093002
Cr-Commit-Position: refs/heads/master@{#13879}
When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled.
BUG=webrtc:6215
NOTRY=True
Review-Url: https://codereview.webrtc.org/2250893003
Cr-Commit-Position: refs/heads/master@{#13829}
