This makes this target not include the precompiled header, which
is different from GYP.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2402493002
Cr-Commit-Position: refs/heads/master@{#14559}
The clang static analyzer seems unable to resolve cpp locks in ObjC code.
As of current time, the clang analyzer has known limitations documented
http://clang.llvm.org/docs/ThreadSafetyAnalysis.html#known-limitations.
From the documentation: "The analysis currently does not do any checking
inside constructors or destructors.
In other words, every constructor and destructor is treated
as if it was annotated with NO_THREAD_SAFETY_ANALYSIS."
This is 'probably' why the analyzer is unable to resolve the lock when
used in ObjC land (the cpp works fine).
The lock can be removed by using atomic property instead.
It's not on performance critical path and we expect updates on just one queue and reads from others. That's why the thread assurance atomic properties bring is enough.
The CL removes rtc_sdk_peerconnection_objc_warnings_config as well as it's no longer needed.
BUG=webrtc:6308
Review-Url: https://codereview.webrtc.org/2372513004
Cr-Commit-Position: refs/heads/master@{#14450}
This CL adds headers that are present in the /Headers directory but not included in the framework target.
BUG=none
NOTRY=true
Review-Url: https://codereview.webrtc.org/2342293002
Cr-Commit-Position: refs/heads/master@{#14413}
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}
In the migration to GN templates, some targets got the whole
rtc_common_config removed, which can have unpredicted consequences
in terms of different code behavior due to defines not being set
as expected etc.
It's better to enable this config and only disable the warnings
that fails the build.
BUG=webrtc:6306,webrtc:6307,webrtc:6308
NOTRY=True
Review-Url: https://codereview.webrtc.org/2347263002
Cr-Commit-Position: refs/heads/master@{#14280}
The build artifacts don't look completely identical to the ones generated
by the GYP targets, but manual review shows the same symbols are exported.
On iOS, the version generated by the GN follows convention, including
a "Headers" directory, and the .modulemap file. I think this is preferred
over the gyp version.
BUG=webrtc:6320
NOTRY=True
TESTED=Run AppRTCDemo on iOS + Mac and verified with nm that they export the same symbols.
Review-Url: https://codereview.webrtc.org/2340633003
Cr-Commit-Position: refs/heads/master@{#14228}
This is to avoid a naming conflict with webrtc::RTCStatsReport that is
surfaced if you try to include it in peerconnectioninterface.h.
Background: The current stats is very much non-spec-compliant. A new
stats collection API is underway that is meant to be spec-compliant.
Some classes in Chromium and webrtc/sdk/objc have spec-compliant names
but non-spec-compliant behavior. These are being renamed to "Legacy" so
that new spec-compliant classes can be added with the correct names.
BUG=chromium:627816
TBR=tkchin@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2313943002
Cr-Commit-Position: refs/heads/master@{#14150}
There is no clear reason to have them in build_overrides, and
webrtc/build seems to be a better place.
Also, delete build_overrides/webrtc.gni
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2309253004
Cr-Commit-Position: refs/heads/master@{#14108}
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}
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 rotation is currently always applied by AVFoundation by
reconfiguring the capture connection video orientation. This CL sets the
rotation field in the frame instead. This avoids the current flash in
the video when the device is rotated, and also avoids reconfiguring the
local encoder and remote decoder when the device is rotated.
BUG=b/30651939
Review-Url: https://codereview.webrtc.org/2271583003
Cr-Commit-Position: refs/heads/master@{#13916}
This CL adds support in RTCEAGLVideoView for rendering CVPixelBuffers as
OpenGL ES textures directly, compared to the current code that first
converts the CVPixelBuffers to I420, and then reuploads them as
textures. This is only supported on iOS with the use of a
CVOpenGLESTextureCache.
The I420 rendering and native rendering are separated in two different
implementations of a simple shader interface:
@protocol Shader
- (BOOL)drawFrame:(RTCVideoFrame*)frame;
@end
GL resources are allocated when the shader is instantiated and released
when the shader is destroyed. RTCEAGLVideoView will lazily instantiate
the necessary shader when it receives the first frame of that kind. This
is primarily done to avoid allocating GL resources for both I420 and
native rendering.
Some other changes are:
- Print GL shader compilation errors.
- Remove updateTextureSizesForFrame() function. The textures will
resize automatically anyway when the texture data is uploaded with
glTexImage2D().
patch from issue 2154243002 at patchset 140001 (http://crrev.com/2154243002#ps140001)
Continuing magjed@'s work since he is OOO this week.
BUG=
Review-Url: https://codereview.webrtc.org/2202823004
Cr-Commit-Position: refs/heads/master@{#13668}
The AppRTCDemo app on Mac OSX does not show or send local video streams,
as ACFoundation capture session is not compiled in or implemented in
the appropriate places. This is the first part of a two-part patch
that implements local capture on the Mac for AppRTCDemo
P.S. This is my first patch to WebRTC. I didn't see any relevant tests, but I could write some if you can point me at a location. Also, I don't have access to the automated tests (I don't think)
BUG=webrtc:3417
Review-Url: https://codereview.webrtc.org/2046863004
Cr-Commit-Position: refs/heads/master@{#13080}
In https://codereview.webrtc.org/2034923003 it was discovered
that a test binary rtc_sdk_peerconnection_objc_tests was
a dependency to rtc_unittests. Unfortunately gtest doesn't
include dependent executables into the same test executable;
only libraries (so theses tests weren't run).
This CL incorporates those tests into rtc_unittests and
does the same changes to the GN build.
BUG=webrtc:5949
TESTED=Built and ran rtc_unittests locally on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2041743003
Cr-Commit-Position: refs/heads/master@{#13060}
Compile fixes for GN on iOS that finally gets our bots green.
Changes to system_wrappers:
* Updated to only use inclusive sources for maintainability
* Add a few missing GN headers.
* Cleanup GYP hack for atomic32_mac.cc
* Renamed changes sources to avoid problems with GYP/GN file
suffix rules:
- atomic32_mac.cc -> atomic32_darwin.cc
- atomic32_posix.cc -> atomic32_non_darwin_unix.cc
See https://code.google.com/p/chromium/codesearch#chromium/src/build/config/BUILDCONFIG.gn&l=325
for details on which extensions can/cannot be used.
BUG=webrtc:5586
NOTRY=True
Review-Url: https://codereview.webrtc.org/1999723002
Cr-Commit-Position: refs/heads/master@{#12897}
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}
- 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}
