Revert of Refactor webrtc_perf_tests into several source_sets.

webrtc/BUILD.gn

Index: webrtc/BUILD.gn
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 65db50da9605896d159cb885b51b24f13afcf099..3fb9cfea72ebf94abda5ea30f50f0f079ca32b94 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -622,15 +622,38 @@
     testonly = true
     configs += [ ":rtc_unittests_config" ]
 
+    sources = [
+      "call/call_perf_tests.cc",
+      "call/rampup_tests.cc",
+      "call/rampup_tests.h",
+      "modules/audio_coding/codecs/opus/opus_complexity_unittest.cc",
+      "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/audio_processing/residual_echo_detector_complexity_unittest.cc",
+      "modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc",
+      "video/full_stack_tests.cc",
+    ]
     deps = [
-      "call:call_perf_tests",
-      "modules/audio_coding:audio_coding_perf_tests",
-      "modules/audio_processing:audio_processing_perf_tests",
-      "modules/remote_bitrate_estimator:remote_bitrate_estimator_perf_tests",
+      "modules/audio_coding:neteq_test_support",
+      "modules/audio_processing",
+      "modules/audio_processing:audioproc_test_utils",
+      "modules/remote_bitrate_estimator:bwe_simulator_lib",
+      "modules/rtp_rtcp",
+      "test:test_common",
       "test:test_main",
-      "video:video_full_stack_tests",
+      "test:test_renderer",
       "video:video_quality_test",
-    ]
+      "voice_engine",
+      "//testing/gmock",
+      "//testing/gtest",
+    ]
+
+    if (rtc_enable_intelligibility_enhancer) {
+      defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
+    } else {
+      defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
+    }
 
     data = webrtc_perf_tests_resources
     if (is_android) {
@@ -639,6 +662,13 @@
     }
     if (is_ios) {
       deps += [ ":webrtc_perf_tests_bundle_data" ]
+    }
+    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" ]
+    }
+    if (rtc_use_h264) {
+      defines += [ "WEBRTC_USE_H264" ]
     }
   }