Chromium Code Reviews| Index: webrtc/modules/audio_processing/BUILD.gn |
| diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn |
| index 5686dc66908dced9a188cb2929b8cf2ee035667d..c1aee93b9b4daa7045f697c7172b58258e576108 100644 |
| --- a/webrtc/modules/audio_processing/BUILD.gn |
| +++ b/webrtc/modules/audio_processing/BUILD.gn |
| @@ -500,6 +500,7 @@ if (rtc_include_tests) { |
| "config_unittest.cc", |
| "echo_cancellation_impl_unittest.cc", |
| "splitting_filter_unittest.cc", |
| + "test/fake_recording_device_unittest.cc", |
| "transient/dyadic_decimator_unittest.cc", |
| "transient/file_utils.cc", |
| "transient/file_utils.h", |
| @@ -524,6 +525,7 @@ if (rtc_include_tests) { |
| deps = [ |
| ":audio_processing", |
| ":audioproc_test_utils", |
| + ":fake_recording_device", |
| "../..:webrtc_common", |
| "../../base:gtest_prod", |
| "../../base:protobuf_utils", |
| @@ -670,6 +672,16 @@ if (rtc_include_tests) { |
| } |
| } |
| + rtc_source_set("fake_recording_device") { |
| + sources = [ |
| + "test/fake_recording_device.cc", |
| + "test/fake_recording_device.h", |
| + ] |
| + deps = [ |
| + "../../base:rtc_base_approved", |
| + ] |
| + } |
| + |
|
AleBzk
2017/05/17 11:52:22
I spoke to Alex. Maybe it's too early to consider
peah-webrtc
2017/05/17 14:52:12
Good point! Then we need to do this. But then I'd
AleBzk
2017/05/23 13:56:41
Right, thanks.
fake_recording_device -> analog_mic
|
| if (rtc_enable_protobuf) { |
| rtc_executable("unpack_aecdump") { |
| testonly = true |
| @@ -708,6 +720,7 @@ if (rtc_include_tests) { |
| ":audioproc_debug_proto", |
| ":audioproc_protobuf_utils", |
| ":audioproc_test_utils", |
| + ":fake_recording_device", |
| "../../base:protobuf_utils", |
| "../../base:rtc_base_approved", |
| "../../common_audio:common_audio", |