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Add webrtc_perf_tests to BUILD.gn

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Updated the sources in audio_processing:audioproc_test_utils to match the configuration on
"webrtc/modules/audio_processing/audio_processing_tests.gypi"

Removed audio_buffer_tools from modules_unittests to match the gyp file.

BUG=webrtc:6041

Review-Url: https://codereview.webrtc.org/2178963002
Cr-Commit-Position: refs/heads/master@{#13541}
This commit is contained in:
ehmaldonado 2016-07-27 08:14:32 -07:00 committed by Commit bot
parent 81031d6dca
commit 529f83c521
3 changed files with 68 additions and 2 deletions
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66
webrtc/BUILD.gn
View File

@ -619,6 +619,72 @@ if (rtc_include_tests) {
}
}
source_set("video_quality_test") {
testonly = true
configs += [ ":common_config" ]
public_configs = [ ":common_inherited_config" ]
sources = [
"video/video_quality_test.cc",
"video/video_quality_test.h",
]
deps = [
":webrtc",
"system_wrappers",
"//testing/gtest",
]
if (!is_android) {
deps += [ "modules/video_capture:video_capture_internal_impl" ]
}
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
configs -= [ "//build/config/clang:find_bad_constructs" ]
}
}
test("webrtc_perf_tests") {
testonly = true
configs += [
":common_config",
":rtc_unittests_config",
]
public_configs = [ ":common_inherited_config" ]
sources = [
"call/call_perf_tests.cc",
"call/rampup_tests.cc",
"call/rampup_tests.h",
"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/remote_bitrate_estimator/remote_bitrate_estimators_test.cc",
"video/full_stack.cc",
]
deps = [
":video_quality_test",
":webrtc",
"modules/audio_coding:neteq_test_support",
"modules/audio_processing",
"modules/audio_processing:audioproc_test_utils",
"modules/remote_bitrate_estimator:bwe_simulator",
"modules/rtp_rtcp",
"test:channel_transport",
"test:test_common",
"test:test_main",
"test:test_renderer",
"voice_engine",
"//testing/gmock",
"//testing/gtest",
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
}
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
configs -= [ "//build/config/clang:find_bad_constructs" ]
}
}
test("webrtc_nonparallel_tests") {
testonly = true
configs += [

2
webrtc/modules/BUILD.gn
View File

@ -322,8 +322,6 @@ if (rtc_include_tests) {
"audio_processing/high_pass_filter_unittest.cc",
"audio_processing/level_estimator_unittest.cc",
"audio_processing/noise_suppression_unittest.cc",
"audio_processing/test/audio_buffer_tools.cc",
"audio_processing/test/audio_buffer_tools.h",
"audio_processing/test/bitexactness_tools.cc",
"audio_processing/test/bitexactness_tools.h",
"audio_processing/test/debug_dump_replayer.cc",

2
webrtc/modules/audio_processing/BUILD.gn
View File

@ -319,6 +319,8 @@ if (rtc_include_tests) {
source_set("audioproc_test_utils") {
testonly = true
sources = [
"test/audio_buffer_tools.cc",
"test/audio_buffer_tools.h",
"test/test_utils.cc",
"test/test_utils.h",
]