Refactor webrtc_perf_tests into several source_sets.

webrtc/modules/audio_processing/BUILD.gn

Index: webrtc/modules/audio_processing/BUILD.gn
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 049a4e1d048ddc966c304915da60a1dc9d0150a6..bacad3108acc5c4a56bf668f707cdf30e86fe6ae 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -411,6 +411,25 @@ if (rtc_include_tests) {
     }
   }
 
+  rtc_source_set("audio_processing_perf_tests") {
+    testonly = true
+    sources = [
+      "audio_processing_performance_unittest.cc",
+      "level_controller/level_controller_complexity_unittest.cc",
+      "residual_echo_detector_complexity_unittest.cc",
+    ]
+    deps = [
+      ":audio_processing",
+      ":audioproc_test_utils",
+      "//testing/gtest",
+    ]
+    if (rtc_enable_intelligibility_enhancer) {
+      defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
+    } else {
+      defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
+    }
+  }
+
   rtc_executable("unpack_aecdump") {
     testonly = true
     sources = [