In order to switch Chromium to use WebRTC targets instead of
duplicated code listings in src/third_party/libjingle it must
be possible for Chromium to process webrtc/api/api.gyp. This is
currently not possible since it includes build/java.gypi, of which
the path is different in a Chromium checkout. It's not possible
to resolve this in another way since 'includes' processing takes
place early in the GYP cycle, before it's possible to use variables.
They're also processed ignoring conditional statements, resulting
in an error when api.gyp is processed.
BUG=webrtc:4256
TBR=perkj@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2080563002
Cr-Commit-Position: refs/heads/master@{#13208}
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}
For simplicity, this CL removes the need for Design Support Library.
UI is slightly changed for this to be possible. Floating Action Button
for adding favorite is removed and instead there's a button next to the
TextView.
Review-Url: https://codereview.webrtc.org/2003983002
Cr-Commit-Position: refs/heads/master@{#12861}
This allows creating tests for AppRTC Android demo that will be run on
the host machine instead of a device. These tests can mock Android APIs
through Robolectric. Because the tests are run on the host machine,
they run much faster.
BUG=webrtc:5896
NOTRY=True
Review-Url: https://codereview.webrtc.org/1985663002
Cr-Commit-Position: refs/heads/master@{#12769}
- 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}
In addition:
- Introduces RTCAudioSessionTest
- iOS/Mac gtests now have an autoreleasepool
- Moves ScopedAutoreleasePool to rtc_base_approved
- Introduces route change button in AppRTCDemo
BUG=webrtc:5649
Review URL: https://codereview.webrtc.org/1782363002
Cr-Commit-Position: refs/heads/master@{#11971}
Updated peerconnection_server to not need stuff from rtc_base.
BUG=
Review URL: https://codereview.webrtc.org/1789463002
Cr-Commit-Position: refs/heads/master@{#11966}
The roll in https://codereview.webrtc.org/1713493002/
made us start using the Chromium sysroot images for libraries instead
of system libraries. This caused Linux 32-bit builds to break with
an error like this:
../../webrtc/examples/peerconnection/client/linux/main_wnd.cc:82:46: error: missing sentinel in function call [-Werror,-Wsentinel]
"List Items", renderer, "text", 0, NULL);
^
, nullptr
/usr/include/gtk-2.0/gtk/gtktreeviewcolumn.h:128:25: note: function has been explicitly marked sentinel here
GtkTreeViewColumn *gtk_tree_view_column_new_with_attributes (const gchar *title,
^
1 error generated.
This CL suppresses this warning to green up the bots.
TBR=niklase@webrtc.org
Review URL: https://codereview.webrtc.org/1710083003 .
Cr-Commit-Position: refs/heads/master@{#11674}
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
The JNI code for VoiceEngine is not maintained and VoiceEngine is being
refactored. This is not a supported Java interface, use AppRTCDemo as a
starting point instead.
Also renames webrtc/libjingle_examples.gyp webrtc/webrtc_examples.gyp to
replace the previous file (that only contained media_demo).
BUG=
R=henrika@webrtc.org, kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1439593002 .
Cr-Commit-Position: refs/heads/master@{#10599}
JNI already has jstring<->UTF8 string conversion, so using that should
save ~1mb off android binaries (ICU is *large*), probably around
300-400k after compression.
BUG=
Review URL: https://codereview.webrtc.org/1430023005
Cr-Commit-Position: refs/heads/master@{#10545}
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
This makes the build more flexible when linking against
prebuilt external libraries.
Use existing build_* variables for libyuv and json in talk/
(already in use in webrtc/).
Also make it possible to avoid building the GTK parts of the Linux build.
BUG=4242
R=andrew@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44179005
Cr-Commit-Position: refs/heads/master@{#9087}
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
Mainly to pick up recent yasm changes needed for
turning on libvpx in GN.
Summary of changes (git diff 6455c69..deaf2f7 DEPS):
* third_party/boringssl 7bdec13..a70c75c
* third_party/libjpeg_turbo 3963fbc..034e9a9
* third_party/libvpx d95585f..4947d55
* tools/gyp 1972:1977
I had to add src/third_party/junit/src to be ignored
due to http://crbug.com/417292.
BUG=3855, chromium:417292
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7313 4adac7df-926f-26a2-2b94-8c16560cd09d
Targets must now link with implementation of their choice instead of at "gyp"-time.
Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.
Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common
Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests
GN changes:
- Not many since there is almost no test definitions.
Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.
Re-enable android tests by reverting 7026 (some tests left disabled).
TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
into its own targets. Dependencies must link directly with the desired one.
Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.
Targets linking with default/external capture implementation:
- anything dependent on webrtc_test_common
- anything dependent on video_engine_core
Targets linking with internal capture implementation:
- vie_auto_test
- anything dependent on webrtc_test_renderer
GN changes:
- Not many since there is almost no test definitions.
TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3768
R=glaznev@webrtc.orgTBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
The new recipes framework for configuring build explicitly sets the
GYP_DEFINES for Android builds instead of relying on the envsetup.sh script
which probably will be removed at some point in the future.
This causes our build to break since our Android examples relies on the
Android SDK being found using the ANDROID_SDK_ROOT environment variable.
A GYP variable 'android_sdk_root' exists and is set correctly by
common.gypi, which is what I'm using to pass this path correctly to these
tests.
The libjingle example is handled separately in
https://webrtc-codereview.appspot.com/11809004/
BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5907 4adac7df-926f-26a2-2b94-8c16560cd09d
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
