| Index: webrtc/test/BUILD.gn
|
| diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
|
| index 07ffbee41238236d6923e967897ba1541e59e9a9..12eb3cacc32090d17a5a494a86b4a70989f1b986 100644
|
| --- a/webrtc/test/BUILD.gn
|
| +++ b/webrtc/test/BUILD.gn
|
| @@ -333,6 +333,22 @@ rtc_source_set("direct_transport") {
|
| ]
|
| }
|
|
|
| +rtc_source_set("fake_audio_device") {
|
| + testonly = true
|
| + sources = [
|
| + "fake_audio_device.cc",
|
| + "fake_audio_device.h",
|
| + ]
|
| + 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_base_approved",
|
| + "../modules/audio_device:audio_device",
|
| + ]
|
| +}
|
| +
|
| rtc_source_set("test_common") {
|
| testonly = true
|
| sources = [
|
| @@ -346,8 +362,6 @@ rtc_source_set("test_common") {
|
| "drifting_clock.h",
|
| "encoder_settings.cc",
|
| "encoder_settings.h",
|
| - "fake_audio_device.cc",
|
| - "fake_audio_device.h",
|
| "fake_decoder.cc",
|
| "fake_decoder.h",
|
| "fake_encoder.cc",
|
| @@ -379,13 +393,13 @@ rtc_source_set("test_common") {
|
|
|
| deps = [
|
| ":direct_transport",
|
| + ":fake_audio_device",
|
| ":rtp_test_utils",
|
| ":test_support",
|
| "..:webrtc_common",
|
| "../audio",
|
| "../base:rtc_base_approved",
|
| "../call",
|
| - "../modules/audio_device:mock_audio_device",
|
| "../modules/audio_mixer:audio_mixer_impl",
|
| "../modules/audio_processing",
|
| "../video",
|
|
|