The warnings in question are benign and should be supressed locally so
we can keep the analysis for the rest of the files.
BUG=webrtc:5748
Review-Url: https://codereview.webrtc.org/2790883002
Cr-Commit-Position: refs/heads/master@{#17485}
This makes a few things a lot clearer when looking at perf trace data:
* What module instances (where they were created) are called
* On what thread
* How frequently
* For how long
ProcessThread will be replaced by TaskQueue moving forward and this is a step towards understanding the behavior of the affected code.
BUG=webrtc:7219
Review-Url: https://codereview.webrtc.org/2729053002
Cr-Commit-Position: refs/heads/master@{#16998}
Reason for revert:
I unfortunately have to revert this since trybots are still trying to run these tests and now they fail because the tests aren't there :-/ Before relanding, can you work with engprod to make sure we don't run them?
Original issue's description:
> Removes a test suite that is no longer used or maintained.
>
> BUG=NONE
>
> Review-Url: https://codereview.webrtc.org/2716843002
> Cr-Commit-Position: refs/heads/master@{#16825}
> Committed: bb5136f940TBR=solenberg@webrtc.org,nisse@webrtc.org,henrika@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2721433002
Cr-Commit-Position: refs/heads/master@{#16845}
This implementation uses various legacy classes such as EventTimeWrapper,
CriticalSectionWrapper, EventWrapper etc and hasn't been maintained
(or used?) for a long time.
Instead of spending time on testing and updating the class, I think
we should just remove it. For versions of Windows that we support,
following Win7, we use the CoreAudio implementation.
BUG=webrtc:7183
R=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2700983002 .
Cr-Commit-Position: refs/heads/master@{#16678}
Reason for revert:
It seems that we cannot skip the generation of "//webrtc/base/base_java" in chromium without some refactoring because it is included as a dependency in some places.
Original issue's description:
> Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )
>
> Reason for revert:
> This breaks some chromium.webrtc.fyi buildbots with the following error:
>
> ERROR Unresolved dependencies.
> //third_party/webrtc/base:base(//build/toolchain/android:android_arm)
> needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
>
>
> Original issue's description:
> > Creating libwebrtc bundle jar
> >
> > Creates a JAR which includes:
> > - //webrtc/base:base_java
> > - //webrtc/modules/audio_device:audio_device_java
> > - //webrtc/sdk/android:libjingle_peerconnection_java
> > - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
> >
> > The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
> >
> > BUG=webrtc:6356
> >
> > Review-Url: https://codereview.webrtc.org/2646443002
> > Cr-Commit-Position: refs/heads/master@{#16189}
> > Committed: a62a82b7e7
>
> TBR=kjellander@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2640023010
> Cr-Commit-Position: refs/heads/master@{#16190}
> Committed: 3c9151b953TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646093004
Cr-Commit-Position: refs/heads/master@{#16203}
Reason for revert:
This breaks some chromium.webrtc.fyi buildbots with the following error:
ERROR Unresolved dependencies.
//third_party/webrtc/base:base(//build/toolchain/android:android_arm)
needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
Original issue's description:
> Creating libwebrtc bundle jar
>
> Creates a JAR which includes:
> - //webrtc/base:base_java
> - //webrtc/modules/audio_device:audio_device_java
> - //webrtc/sdk/android:libjingle_peerconnection_java
> - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
>
> The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
>
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2646443002
> Cr-Commit-Position: refs/heads/master@{#16189}
> Committed: a62a82b7e7TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2640023010
Cr-Commit-Position: refs/heads/master@{#16190}
Creates a JAR which includes:
- //webrtc/base:base_java
- //webrtc/modules/audio_device:audio_device_java
- //webrtc/sdk/android:libjingle_peerconnection_java
- //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646443002
Cr-Commit-Position: refs/heads/master@{#16189}
Reason for revert:
Skipping the build of the target "//webrtc/modules/audio_device:audio_device_java" when building webrtc with chromium.
This was causing a failure because in that case it is not possible to refer to the script used by the template "android_shared_srcjar" with an absolute path (which seems to be necessary since the script attribute of action is relative to the invocation target and not to the template declaration).
Original issue's description:
> Revert of Fixing package-boundary violation with srjar_deps (patchset #5 id:80001 of https://codereview.webrtc.org/2610823002/ )
>
> Reason for revert:
> This CL is breaking a chromium.webrtc.fyi: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/226
>
> I am trying to reproduce the issue on my local machine and I will try to re-land the CL later.
>
> Original issue's description:
> > Fixing package-boundary violation with srjar_deps
> >
> > Without the usage of the srcjar_deps attribute we were not able to
> > include .java files from other packages without violating the package
> > boundary contraint.
> >
> > As an example, in this CL the target "libjingle_peerconnection_java" was
> > directly including .java files from another packages in its "java_files"
> > attribute.
> >
> > Using srcjar_deps we are able to declare the dependency of the target
> > avoiding to create hidden dependencies in the codebase.
> >
> > This is not fixing the webrtc:6356 bug directly but it is a first step to
> > include ThreadUtils classes in libjingle_peerconnection_client_java.jar
> > again.
> >
> > It seems also to be related to the chromium:648244 bug. This can be solved
> > if we can find a way to perform srcjar generation in the android_library
> > target without changing the semantic of the target.
> >
> > BUG=webrtc:6356
> >
> > Review-Url: https://codereview.webrtc.org/2610823002
> > Cr-Commit-Position: refs/heads/master@{#15914}
> > Committed: 10a76592a7
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2617533005
> Cr-Commit-Position: refs/heads/master@{#15915}
> Committed: eb731ed09eTBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2612953004
Cr-Commit-Position: refs/heads/master@{#15958}
Reason for revert:
This CL is breaking a chromium.webrtc.fyi: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/226
I am trying to reproduce the issue on my local machine and I will try to re-land the CL later.
Original issue's description:
> Fixing package-boundary violation with srjar_deps
>
> Without the usage of the srcjar_deps attribute we were not able to
> include .java files from other packages without violating the package
> boundary contraint.
>
> As an example, in this CL the target "libjingle_peerconnection_java" was
> directly including .java files from another packages in its "java_files"
> attribute.
>
> Using srcjar_deps we are able to declare the dependency of the target
> avoiding to create hidden dependencies in the codebase.
>
> This is not fixing the webrtc:6356 bug directly but it is a first step to
> include ThreadUtils classes in libjingle_peerconnection_client_java.jar
> again.
>
> It seems also to be related to the chromium:648244 bug. This can be solved
> if we can find a way to perform srcjar generation in the android_library
> target without changing the semantic of the target.
>
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2610823002
> Cr-Commit-Position: refs/heads/master@{#15914}
> Committed: 10a76592a7TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2617533005
Cr-Commit-Position: refs/heads/master@{#15915}
Without the usage of the srcjar_deps attribute we were not able to
include .java files from other packages without violating the package
boundary contraint.
As an example, in this CL the target "libjingle_peerconnection_java" was
directly including .java files from another packages in its "java_files"
attribute.
Using srcjar_deps we are able to declare the dependency of the target
avoiding to create hidden dependencies in the codebase.
This is not fixing the webrtc:6356 bug directly but it is a first step to
include ThreadUtils classes in libjingle_peerconnection_client_java.jar
again.
It seems also to be related to the chromium:648244 bug. This can be solved
if we can find a way to perform srcjar generation in the android_library
target without changing the semantic of the target.
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2610823002
Cr-Commit-Position: refs/heads/master@{#15914}
The Bazel build format doesn't support having separate
lists of compilation flags for C and C++; it just has a single
copts list for cc_library:
https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts
This makes it hard to convert our GN targets to Bazel when there are
compiler warnings that aren't supported for C (like -Woverloaded-virtual
being added in bugs.webrtc.org/6653).
The solution for this is to move all .c files to their own targets
and remove C++-only compiler flags during conversion.
New targets:
//webrtc/common_audio:common_audio_c
//webrtc/common_audio:common_audio_neon_c
//webrtc/modules/audio_coding:g711_c
//webrtc/modules/audio_coding:g722_c
//webrtc/modules/audio_coding:ilbc_c
//webrtc/modules/audio_coding:isac_c
//webrtc/modules/audio_coding:isac_fix_c
//webrtc/modules/audio_coding:isac_test_util
//webrtc/modules/audio_coding:pcm16b_c
//webrtc/modules/audio_coding:webrtc_opusj_c
//webrtc/modules/audio_device:mac_portaudio
//webrtc/modules/audio_procssing:audio_processing_c
//webrtc/modules/audio_procssing:audio_processing_neon_c
This CL also adds a PRESUBMIT.py check that will throw an error
if targets are mixing .c and .cc files, to preven this from regressing.
BUG=webrtc:6653
NOTRY=True
Review-Url: https://codereview.webrtc.org/2550563003
Cr-Commit-Position: refs/heads/master@{#15433}
- Out from modules/utility/ and into modules/audio_device/ios/ - there they are used.
BUG=none
Review-Url: https://codereview.webrtc.org/2526273002
Cr-Commit-Position: refs/heads/master@{#15236}
Due to bugs.webrtc.org/216 code that includes
//webrtc/modules/audio_device:mock_audio_device fails to compile on
Windows unless a warning flag is switched off. It seems that
googlemock changes types of overridden virtual method parameters from
'const int' to 'int', which causes the VS compiler to report an error.
The problem was not solved by suppressing the flag wd4373 in
//webrtc/modules/audio_device:mock_audio_device, because targets that
include the headers are compiled separately.
This CL adds the flag suppression to the GN variable
|all_dependent_configs|. Then GN will apply the configuration to all
reachable dependencies. This is needed to reduce clutter and extra
conditions in dependency build targets.
The reason for |all_dependent_configs| before |public_configs| is
for a situation in which a targets headers include the headers from
this target. Then dependencies of dependencies will have a copy of
this targets' code after preprocessing, and compilation will fail.
This will happen if we e.g. change mock_voice_engine to return a
MockAudioTransport or MockAudioDevice.
This change has been tested by compiling a dependent CL
(https://codereview.webrtc.org/2454373002/) which uses these mocks
on Windows without suppressing the flag.
There is no GYP change, because test code has been removed from GYP.
NOTRY=True
BUG=webrtc:216
Review-Url: https://codereview.webrtc.org/2492713003
Cr-Commit-Position: refs/heads/master@{#15024}
There are currently two nearly identical classes called
MockAudioTransport defined in two unit tests:
android/audio_transport_unittest.cc and
/ios/audio_transport_unittest_ios.cc
This change defines a common mocked AudioTransport. The two current
mocks are rewritten to use the common one. A GN target is created for
this mock and MockAudioDevice.
This change will allow to provide a mocked AudioTransport to
AudioState in a dependent CL https://codereview.webrtc.org/2454373002/
BUG=webrtc:6346
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2493483002
Cr-Commit-Position: refs/heads/master@{#15010}
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}
NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that
OpenSL ES is not accidentally activated in existing clients. There are still some
unresolved issues to sort out before it can be utilized.
Enables possibility to use OpenSL ES based audio in both directions for WebRTC.
All unit tests and demo clients have been tested with the new implementation but
the new support is behind a flag (see above).
More testing is needed before it can be used in the field and additional support for
hardware effects is still missing.
BUG=webrtc:5925
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2119633004 .
Cr-Commit-Position: refs/heads/master@{#14290}
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}
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}
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}
Conceptually, dummy audio file devices are a "platform", like
win/mac/linux, and so the conditional slots under
include_internal_audio_device. When enabled, use_dummy_audio_file_devices
disables whatever platform-specific audio layer would have been used and
turns on dummy file device support.
BUG=
Review-Url: https://codereview.webrtc.org/2250483002
Cr-Commit-Position: refs/heads/master@{#13790}
Changes:
* Enabled protobuf for iOS globally.
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global
scope similar to GYP since tests depend on it.
* Added missing rtc_libvpx_build_vp9 variable.
* Moved out audio_coding defines into .gni file to avoid code duplication
* Renamed files to avoid object naming conflicts that GN disallows:
* webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc}
* webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac. Also ran:
gn gen out/Default --args="rtc_enable_bwe_test_logging=true"
and verified that more objects are being built (1885 vs 1883)
when compiling modules_unittests.
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2041233006
Cr-Commit-Position: refs/heads/master@{#13108}
Changes:
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope
to match GYP.
* Enable sctpdataengine_unittest.cc for iOS, which should have
been done in https://codereview.webrtc.org/1587193006
* Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils
to match GYP.
* Added dependencies on call, modules/video_coding and video for
rtc_media.
* Added dependency on audio for rtc_media_unitttests (couldn't be
added to rtc_media due to circular dependency problem).
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2050313002
Cr-Commit-Position: refs/heads/master@{#13106}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2025343002
Cr-Commit-Position: refs/heads/master@{#13007}
- Places most ObjC code into webrtc/sdk/objc instead.
- New gyp targets to build, strip and export symbols for dylib.
- Removes old script used to generate dylib.
BUG=
Review URL: https://codereview.webrtc.org/1903663002
Cr-Commit-Position: refs/heads/master@{#12524}
Part 3 of refactor. Also:
- better weak pointer delegate storage + tests
- we now ignore route changes when we're interrupted
- fixed bug where preferred sample rate wasn't set if audio session
wasn't active
BUG=
Review URL: https://codereview.webrtc.org/1796983004
Cr-Commit-Position: refs/heads/master@{#12007}
