8c1dd8d32f
This effectively reverts commit c3e1cabc696240e4b5a128653264785292878205 (https://codereview.webrtc.org/2589703002/). The reason the test was failing before was missing resource dependencies in the GN file. This is now fixed. Furthermore, the test did not trigger the complexity adaptation that it was supposed to test, since the hysteresis window of the bitrate was not taken into account. This is also fixed. Finally, a percent label was added to a printout, to match the same printout in the other test. BUG=webrtc:6708 Review-Url: https://codereview.webrtc.org/2580383002 Cr-Commit-Position: refs/heads/master@{#15679}
719 lines
21 KiB
Plaintext
719 lines
21 KiB
Plaintext
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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# TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
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import("//build/config/linux/pkg_config.gni")
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import("//build/config/sanitizers/sanitizers.gni")
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import("build/webrtc.gni")
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import("//third_party/protobuf/proto_library.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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# Contains the defines and includes in common.gypi that are duplicated both as
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# target_defaults and direct_dependent_settings.
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config("common_inherited_config") {
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defines = []
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cflags = []
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ldflags = []
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if (build_with_mozilla) {
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defines += [ "WEBRTC_MOZILLA_BUILD" ]
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}
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if (build_with_chromium) {
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defines = [
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# TODO(kjellander): Cleanup unused ones and move defines closer to
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# the source when webrtc:4256 is completed.
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"FEATURE_ENABLE_SSL",
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"FEATURE_ENABLE_VOICEMAIL",
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"EXPAT_RELATIVE_PATH",
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"GTEST_RELATIVE_PATH",
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"NO_MAIN_THREAD_WRAPPING",
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"NO_SOUND_SYSTEM",
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"WEBRTC_CHROMIUM_BUILD",
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]
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include_dirs = [
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# The overrides must be included first as that is the mechanism for
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# selecting the override headers in Chromium.
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"../webrtc_overrides",
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# Allow includes to be prefixed with webrtc/ in case it is not an
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# immediate subdirectory of the top-level.
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"..",
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]
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}
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if (is_posix) {
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defines += [ "WEBRTC_POSIX" ]
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}
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if (is_ios) {
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defines += [
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"WEBRTC_MAC",
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"WEBRTC_IOS",
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]
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}
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if (is_linux) {
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defines += [ "WEBRTC_LINUX" ]
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}
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if (is_mac) {
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defines += [ "WEBRTC_MAC" ]
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}
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if (is_win) {
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defines += [
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"WEBRTC_WIN",
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"_CRT_SECURE_NO_WARNINGS", # Suppress warnings about _vsnprinf
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]
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}
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if (is_android) {
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defines += [
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"WEBRTC_LINUX",
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"WEBRTC_ANDROID",
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]
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}
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if (is_chromeos) {
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defines += [ "CHROMEOS" ]
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}
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if (rtc_sanitize_coverage != "") {
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assert(is_clang, "sanitizer coverage requires clang")
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cflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
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ldflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
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}
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# TODO(GYP): Support these in GN.
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# if (is_bsd) {
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# defines += [ "BSD" ]
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# }
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# if (is_openbsd) {
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# defines += [ "OPENBSD" ]
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# }
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# if (is_freebsd) {
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# defines += [ "FREEBSD" ]
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# }
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}
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config("common_config") {
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cflags = []
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cflags_cc = []
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defines = []
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if (rtc_restrict_logging) {
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defines += [ "WEBRTC_RESTRICT_LOGGING" ]
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}
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if (rtc_include_internal_audio_device) {
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defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
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}
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if (rtc_relative_path) {
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defines += [ "EXPAT_RELATIVE_PATH" ]
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}
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if (!rtc_libvpx_build_vp9) {
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defines += [ "RTC_DISABLE_VP9" ]
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}
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if (build_with_chromium) {
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defines += [
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# NOTICE: Since common_inherited_config is used in public_configs for our
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# targets, there's no point including the defines in that config here.
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# TODO(kjellander): Cleanup unused ones and move defines closer to the
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# source when webrtc:4256 is completed.
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"ENABLE_EXTERNAL_AUTH",
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"HAVE_OPENSSL_SSL_H",
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"HAVE_SCTP",
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"HAVE_SRTP",
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"HAVE_WEBRTC_VIDEO",
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"HAVE_WEBRTC_VOICE",
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"LOGGING_INSIDE_WEBRTC",
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"SSL_USE_OPENSSL",
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"USE_WEBRTC_DEV_BRANCH",
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]
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} else {
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if (is_posix) {
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# Enable more warnings: -Wextra is currently disabled in Chromium.
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cflags = [
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"-Wextra",
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# Repeat some flags that get overridden by -Wextra.
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"-Wno-unused-parameter",
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"-Wno-missing-field-initializers",
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"-Wno-strict-overflow",
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]
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cflags_cc = [
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"-Wnon-virtual-dtor",
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# This is enabled for clang; enable for gcc as well.
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"-Woverloaded-virtual",
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]
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}
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if (is_clang) {
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cflags += [
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"-Wimplicit-fallthrough",
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"-Wthread-safety",
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"-Winconsistent-missing-override",
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"-Wundef",
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]
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}
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}
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if (current_cpu == "arm64") {
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defines += [ "WEBRTC_ARCH_ARM64" ]
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defines += [ "WEBRTC_HAS_NEON" ]
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}
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if (current_cpu == "arm") {
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defines += [ "WEBRTC_ARCH_ARM" ]
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if (arm_version >= 7) {
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defines += [ "WEBRTC_ARCH_ARM_V7" ]
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if (arm_use_neon) {
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defines += [ "WEBRTC_HAS_NEON" ]
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}
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}
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}
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if (current_cpu == "mipsel") {
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defines += [ "MIPS32_LE" ]
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if (mips_float_abi == "hard") {
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defines += [ "MIPS_FPU_LE" ]
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}
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if (mips_arch_variant == "r2") {
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defines += [ "MIPS32_R2_LE" ]
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}
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if (mips_dsp_rev == 1) {
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defines += [ "MIPS_DSP_R1_LE" ]
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} else if (mips_dsp_rev == 2) {
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defines += [
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"MIPS_DSP_R1_LE",
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"MIPS_DSP_R2_LE",
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]
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}
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}
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if (is_android && !is_clang) {
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# The Android NDK doesn"t provide optimized versions of these
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# functions. Ensure they are disabled for all compilers.
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cflags += [
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"-fno-builtin-cos",
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"-fno-builtin-sin",
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"-fno-builtin-cosf",
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"-fno-builtin-sinf",
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]
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}
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if (use_libfuzzer || use_drfuzz || use_afl) {
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# Used in Chromium's overrides to disable logging
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defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
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}
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}
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config("common_objc") {
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libs = [ "Foundation.framework" ]
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precompiled_header = "sdk/objc/WebRTC-Prefix.pch"
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precompiled_source = "sdk/objc/WebRTC-Prefix.pch"
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}
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if (!build_with_chromium) {
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# Target to build all the WebRTC production code.
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rtc_static_library("webrtc") {
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# Only the root target should depend on this.
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visibility = [ "//:default" ]
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sources = [
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# TODO(kjellander): Remove this whenever possible. GN's static_library
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# target type requires at least one object to avoid errors linking.
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"build/no_op_function.cc",
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# TODO(ossu): Keep this here until donwstream projects have updated.
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# http://bugs.webrtc.org/6716
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"call.h",
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"config.h",
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]
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defines = []
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deps = [
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":webrtc_common",
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"api",
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"api:transport_api",
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"audio",
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"base",
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"call",
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"common_audio",
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"common_video",
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"libjingle/xmllite",
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"libjingle/xmpp",
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"logging",
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"media",
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"modules",
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"modules/video_capture:video_capture_internal_impl",
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"p2p",
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"pc",
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"sdk",
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"stats",
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"system_wrappers",
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"video",
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"voice_engine",
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]
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if (rtc_enable_protobuf) {
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defines += [ "ENABLE_RTC_EVENT_LOG" ]
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deps += [ "logging:rtc_event_log_proto" ]
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}
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}
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if (rtc_include_tests) {
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# Target to build all the WebRTC tests (but not examples or tools).
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# Executable in order to get a target that links all WebRTC code.
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rtc_executable("webrtc_tests") {
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testonly = true
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# Only the root target should depend on this.
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visibility = [ "//:default" ]
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deps = [
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":rtc_unittests",
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":video_engine_tests",
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":webrtc_nonparallel_tests",
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":webrtc_perf_tests",
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"api:peerconnection_unittests",
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"common_audio:common_audio_unittests",
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"common_video:common_video_unittests",
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"libjingle:xmllite_xmpp_unittests",
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"media:rtc_media_unittests",
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"modules:modules_tests",
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"modules:modules_unittests",
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"modules/audio_coding:audio_coding_tests",
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"modules/audio_processing:audio_processing_tests",
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"modules/rtp_rtcp:test_packet_masks_metrics",
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"modules/video_capture:video_capture_internal_impl",
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"pc:rtc_pc_unittests",
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"stats:rtc_stats_unittests",
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"system_wrappers:system_wrappers_unittests",
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"test",
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"video:screenshare_loopback",
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"video:video_loopback",
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"video:video_tests",
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"voice_engine:voe_cmd_test",
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"voice_engine:voice_engine_unittests",
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]
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if (is_android) {
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deps += [
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":android_junit_tests",
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"//webrtc/sdk/android:libjingle_peerconnection_android_unittest",
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]
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} else {
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deps += [ "modules/video_capture:video_capture_tests" ]
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}
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if (!is_ios) {
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deps += [
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"modules/audio_device:audio_device_tests",
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"voice_engine:voe_auto_test",
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]
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}
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}
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}
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}
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rtc_static_library("webrtc_common") {
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sources = [
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"common_types.cc",
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"common_types.h",
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"config.cc",
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"config.h",
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"typedefs.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (use_libfuzzer || use_drfuzz || use_afl) {
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# This target is only here for gn to discover fuzzer build targets under
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# webrtc/test/fuzzers/.
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group("webrtc_fuzzers_dummy") {
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testonly = true
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deps = [
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"test/fuzzers:webrtc_fuzzer_main",
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]
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}
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}
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if (rtc_include_tests) {
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config("rtc_unittests_config") {
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# GN orders flags on a target before flags from configs. The default config
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# adds -Wall, and this flag have to be after -Wall -- so they need to
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# come from a config and can"t be on the target directly.
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if (is_clang) {
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cflags = [
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"-Wno-missing-braces",
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"-Wno-sign-compare",
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"-Wno-unused-const-variable",
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]
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}
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}
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rtc_test("rtc_unittests") {
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testonly = true
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sources = [
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"api/fakemetricsobserver.cc",
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"base/analytics/exp_filter_unittest.cc",
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"base/analytics/percentile_filter_unittest.cc",
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"base/array_view_unittest.cc",
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"base/atomicops_unittest.cc",
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"base/autodetectproxy_unittest.cc",
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"base/base64_unittest.cc",
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"base/basictypes_unittest.cc",
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"base/bind_unittest.cc",
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"base/bitbuffer_unittest.cc",
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"base/buffer_unittest.cc",
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"base/bufferqueue_unittest.cc",
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"base/bytebuffer_unittest.cc",
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"base/byteorder_unittest.cc",
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"base/callback_unittest.cc",
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"base/copyonwritebuffer_unittest.cc",
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"base/crc32_unittest.cc",
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"base/criticalsection_unittest.cc",
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"base/event_tracer_unittest.cc",
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"base/event_unittest.cc",
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"base/file_unittest.cc",
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"base/filerotatingstream_unittest.cc",
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"base/fileutils_unittest.cc",
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"base/function_view_unittest.cc",
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"base/helpers_unittest.cc",
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"base/httpbase_unittest.cc",
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"base/httpcommon_unittest.cc",
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"base/httpserver_unittest.cc",
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"base/ipaddress_unittest.cc",
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"base/logging_unittest.cc",
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"base/md5digest_unittest.cc",
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"base/messagedigest_unittest.cc",
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"base/messagequeue_unittest.cc",
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"base/mod_ops_unittest.cc",
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"base/nat_unittest.cc",
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"base/network_unittest.cc",
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"base/onetimeevent_unittest.cc",
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"base/optional_unittest.cc",
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"base/optionsfile_unittest.cc",
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"base/pathutils_unittest.cc",
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"base/platform_thread_unittest.cc",
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"base/proxy_unittest.cc",
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"base/proxydetect_unittest.cc",
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"base/random_unittest.cc",
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"base/rate_limiter_unittest.cc",
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"base/rate_statistics_unittest.cc",
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"base/ratelimiter_unittest.cc",
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"base/ratetracker_unittest.cc",
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|
"base/refcountedobject_unittest.cc",
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"base/rollingaccumulator_unittest.cc",
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|
"base/rtccertificate_unittest.cc",
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"base/rtccertificategenerator_unittest.cc",
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|
"base/safe_compare_unittest.cc",
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|
"base/scopedptrcollection_unittest.cc",
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|
"base/sequenced_task_checker_unittest.cc",
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|
"base/sha1digest_unittest.cc",
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|
"base/sharedexclusivelock_unittest.cc",
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|
"base/signalthread_unittest.cc",
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|
"base/sigslot_unittest.cc",
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|
"base/sigslottester_unittest.cc",
|
|
"base/stream_unittest.cc",
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|
"base/stringencode_unittest.cc",
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|
"base/stringutils_unittest.cc",
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"base/swap_queue_unittest.cc",
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|
|
# TODO(ronghuawu): Reenable this test.
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# "systeminfo_unittest.cc",
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"base/task_queue_unittest.cc",
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|
"base/task_unittest.cc",
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|
"base/testclient_unittest.cc",
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|
"base/thread_annotations_unittest.cc",
|
|
"base/thread_checker_unittest.cc",
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|
"base/thread_unittest.cc",
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|
"base/timestampaligner_unittest.cc",
|
|
"base/timeutils_unittest.cc",
|
|
"base/weak_ptr_unittest.cc",
|
|
"p2p/base/asyncstuntcpsocket_unittest.cc",
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|
"p2p/base/dtlstransportchannel_unittest.cc",
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|
"p2p/base/fakeportallocator.h",
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|
"p2p/base/faketransportcontroller.h",
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|
"p2p/base/jseptransport_unittest.cc",
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|
"p2p/base/p2ptransportchannel_unittest.cc",
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|
"p2p/base/port_unittest.cc",
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|
"p2p/base/portallocator_unittest.cc",
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|
"p2p/base/pseudotcp_unittest.cc",
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|
"p2p/base/relayport_unittest.cc",
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|
"p2p/base/relayserver_unittest.cc",
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|
"p2p/base/stun_unittest.cc",
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|
"p2p/base/stunport_unittest.cc",
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|
"p2p/base/stunrequest_unittest.cc",
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|
"p2p/base/stunserver_unittest.cc",
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|
"p2p/base/tcpport_unittest.cc",
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|
"p2p/base/testrelayserver.h",
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|
"p2p/base/teststunserver.h",
|
|
"p2p/base/testturnserver.h",
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|
"p2p/base/transportcontroller_unittest.cc",
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|
"p2p/base/transportdescriptionfactory_unittest.cc",
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|
"p2p/base/turnport_unittest.cc",
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|
"p2p/base/turnserver_unittest.cc",
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|
"p2p/base/udptransportchannel_unittest.cc",
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|
"p2p/client/basicportallocator_unittest.cc",
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|
"p2p/stunprober/stunprober_unittest.cc",
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]
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|
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if (is_win) {
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sources += [
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"base/win32_unittest.cc",
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|
"base/win32window_unittest.cc",
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]
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|
}
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|
|
|
if (is_mac) {
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|
sources += [ "base/macutils_unittest.cc" ]
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|
}
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|
|
|
if (is_posix) {
|
|
sources += [
|
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"base/ssladapter_unittest.cc",
|
|
"base/sslidentity_unittest.cc",
|
|
"base/sslstreamadapter_unittest.cc",
|
|
]
|
|
}
|
|
if (rtc_use_quic) {
|
|
sources += [
|
|
"p2p/quic/quicconnectionhelper_unittest.cc",
|
|
"p2p/quic/quicsession_unittest.cc",
|
|
"p2p/quic/quictransport_unittest.cc",
|
|
"p2p/quic/quictransportchannel_unittest.cc",
|
|
"p2p/quic/reliablequicstream_unittest.cc",
|
|
]
|
|
}
|
|
|
|
configs += [ ":rtc_unittests_config" ]
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
|
|
deps = [
|
|
"base:rtc_analytics",
|
|
"base:rtc_base",
|
|
"base:rtc_base_tests_utils",
|
|
"base:rtc_task_queue",
|
|
"p2p:libstunprober",
|
|
"p2p:rtc_p2p",
|
|
"//testing/gmock",
|
|
"//testing/gtest",
|
|
]
|
|
|
|
if (rtc_enable_protobuf) {
|
|
deps += [ "logging:rtc_event_log_tests" ]
|
|
}
|
|
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_support" ]
|
|
shard_timeout = 900
|
|
}
|
|
|
|
if (is_ios || is_mac) {
|
|
deps += [
|
|
"sdk:rtc_sdk_peerconnection_objc",
|
|
"system_wrappers:system_wrappers_default",
|
|
"//third_party/ocmock",
|
|
]
|
|
sources += [
|
|
"sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm",
|
|
"sdk/objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm",
|
|
"sdk/objc/Framework/UnitTests/RTCIceCandidateTest.mm",
|
|
"sdk/objc/Framework/UnitTests/RTCIceServerTest.mm",
|
|
"sdk/objc/Framework/UnitTests/RTCMediaConstraintsTest.mm",
|
|
"sdk/objc/Framework/UnitTests/RTCSessionDescriptionTest.mm",
|
|
"sdk/objc/Framework/UnitTests/avformatmappertests.mm",
|
|
]
|
|
|
|
# TODO(tkchin): Cleanup this warning.
|
|
cflags = [ "-Wno-objc-property-no-attribute" ]
|
|
|
|
# |-ObjC| flag needed to make sure category method implementations
|
|
# are included:
|
|
# https://developer.apple.com/library/mac/qa/qa1490/_index.html
|
|
ldflags = [ "-ObjC" ]
|
|
}
|
|
}
|
|
|
|
# TODO(pbos): Rename test suite, this is no longer "just" for video targets.
|
|
video_engine_tests_resources = [
|
|
"//resources/foreman_cif_short.yuv",
|
|
"//resources/voice_engine/audio_long16.pcm",
|
|
]
|
|
|
|
if (is_ios) {
|
|
bundle_data("video_engine_tests_bundle_data") {
|
|
testonly = true
|
|
sources = video_engine_tests_resources
|
|
outputs = [
|
|
"{{bundle_resources_dir}}/{{source_file_part}}",
|
|
]
|
|
}
|
|
}
|
|
|
|
rtc_test("video_engine_tests") {
|
|
testonly = true
|
|
deps = [
|
|
"audio:audio_tests",
|
|
"call:call_tests",
|
|
"modules/video_capture",
|
|
"test:test_common",
|
|
"test:test_main",
|
|
"video:video_tests",
|
|
]
|
|
data = video_engine_tests_resources
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_native_code" ]
|
|
shard_timeout = 900
|
|
}
|
|
if (is_ios) {
|
|
deps += [ ":video_engine_tests_bundle_data" ]
|
|
}
|
|
}
|
|
|
|
webrtc_perf_tests_resources = [
|
|
"//resources/audio_coding/speech_mono_16kHz.pcm",
|
|
"//resources/audio_coding/speech_mono_32_48kHz.pcm",
|
|
"//resources/audio_coding/testfile32kHz.pcm",
|
|
"//resources/ConferenceMotion_1280_720_50.yuv",
|
|
"//resources/difficult_photo_1850_1110.yuv",
|
|
"//resources/foreman_cif.yuv",
|
|
"//resources/google-wifi-3mbps.rx",
|
|
"//resources/paris_qcif.yuv",
|
|
"//resources/photo_1850_1110.yuv",
|
|
"//resources/presentation_1850_1110.yuv",
|
|
"//resources/verizon4g-downlink.rx",
|
|
"//resources/voice_engine/audio_long16.pcm",
|
|
"//resources/web_screenshot_1850_1110.yuv",
|
|
]
|
|
|
|
if (is_ios) {
|
|
bundle_data("webrtc_perf_tests_bundle_data") {
|
|
testonly = true
|
|
sources = webrtc_perf_tests_resources
|
|
outputs = [
|
|
"{{bundle_resources_dir}}/{{source_file_part}}",
|
|
]
|
|
}
|
|
}
|
|
|
|
rtc_test("webrtc_perf_tests") {
|
|
testonly = true
|
|
configs += [ ":rtc_unittests_config" ]
|
|
|
|
sources = [
|
|
"call/call_perf_tests.cc",
|
|
"call/rampup_tests.cc",
|
|
"call/rampup_tests.h",
|
|
"modules/audio_coding/codecs/opus/opus_complexity_unittest.cc",
|
|
"modules/audio_coding/neteq/test/neteq_performance_unittest.cc",
|
|
"modules/audio_processing/audio_processing_performance_unittest.cc",
|
|
"modules/audio_processing/level_controller/level_controller_complexity_unittest.cc",
|
|
"modules/audio_processing/residual_echo_detector_complexity_unittest.cc",
|
|
"modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc",
|
|
"video/full_stack_tests.cc",
|
|
]
|
|
deps = [
|
|
"modules/audio_coding:neteq_test_support",
|
|
"modules/audio_processing",
|
|
"modules/audio_processing:audioproc_test_utils",
|
|
"modules/remote_bitrate_estimator:bwe_simulator_lib",
|
|
"modules/rtp_rtcp",
|
|
"test:test_common",
|
|
"test:test_main",
|
|
"test:test_renderer",
|
|
"video:video_quality_test",
|
|
"voice_engine",
|
|
"//testing/gmock",
|
|
"//testing/gtest",
|
|
]
|
|
|
|
if (rtc_enable_intelligibility_enhancer) {
|
|
defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
|
|
} else {
|
|
defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
|
|
}
|
|
|
|
data = webrtc_perf_tests_resources
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_native_code" ]
|
|
shard_timeout = 2700
|
|
}
|
|
if (is_ios) {
|
|
deps += [ ":webrtc_perf_tests_bundle_data" ]
|
|
}
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
if (rtc_use_h264) {
|
|
defines += [ "WEBRTC_USE_H264" ]
|
|
}
|
|
}
|
|
|
|
rtc_test("webrtc_nonparallel_tests") {
|
|
testonly = true
|
|
configs += [ ":rtc_unittests_config" ]
|
|
sources = [
|
|
"base/nullsocketserver_unittest.cc",
|
|
"base/physicalsocketserver_unittest.cc",
|
|
"base/socket_unittest.cc",
|
|
"base/socket_unittest.h",
|
|
"base/socketaddress_unittest.cc",
|
|
"base/virtualsocket_unittest.cc",
|
|
]
|
|
deps = [
|
|
"base:rtc_base",
|
|
"base:rtc_base_tests_utils",
|
|
"//testing/gtest",
|
|
]
|
|
if (is_win) {
|
|
sources += [ "base/win32socketserver_unittest.cc" ]
|
|
}
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_support" ]
|
|
shard_timeout = 900
|
|
}
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
}
|
|
|
|
if (is_android) {
|
|
junit_binary("android_junit_tests") {
|
|
java_files = [
|
|
"androidjunit/src/org/webrtc/CameraEnumerationTest.java",
|
|
"examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java",
|
|
"examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java",
|
|
"examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java",
|
|
]
|
|
|
|
deps = [
|
|
"//base:base_java_test_support",
|
|
"//webrtc/examples:AppRTCMobile_javalib",
|
|
"//webrtc/sdk/android:libjingle_peerconnection_java",
|
|
]
|
|
}
|
|
}
|
|
}
|