Index: webrtc/modules/audio_processing/BUILD.gn |
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn |
index a569bd100625bfc8e6c8fc44554a5597d08c84d2..2455db41d2ea177e5ee394b6bed4be1251ecfa97 100644 |
--- a/webrtc/modules/audio_processing/BUILD.gn |
+++ b/webrtc/modules/audio_processing/BUILD.gn |
@@ -494,6 +494,7 @@ if (rtc_include_tests) { |
":audioproc_f", |
":audioproc_unittest_proto", |
":unpack_aecdump", |
+ "aec_dump:aec_dump_unittests", |
"test/conversational_speech", |
"test/py_quality_assessment", |
] |
@@ -524,6 +525,7 @@ 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", |
@@ -546,6 +548,7 @@ if (rtc_include_tests) { |
] |
deps = [ |
+ ":analog_mic_simulation", |
":audio_processing", |
":audioproc_test_utils", |
"..:module_api", |
@@ -707,6 +710,16 @@ if (rtc_include_tests) { |
} |
} |
+ rtc_source_set("analog_mic_simulation") { |
+ sources = [ |
+ "test/fake_recording_device.cc", |
+ "test/fake_recording_device.h", |
+ ] |
+ deps = [ |
+ "../../rtc_base:rtc_base_approved", |
+ ] |
+ } |
+ |
if (rtc_enable_protobuf) { |
rtc_executable("unpack_aecdump") { |
testonly = true |
@@ -741,6 +754,7 @@ if (rtc_include_tests) { |
] |
deps = [ |
+ ":analog_mic_simulation", |
":audio_processing", |
":audioproc_debug_proto", |
":audioproc_protobuf_utils", |