Revert of Refactor webrtc_perf_tests into several source_sets.

webrtc/BUILD.gn

 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 604 matching lines @@
       outputs = [
         "{{bundle_resources_dir}}/{{source_file_part}}",
       ]
     }
   }
 
   rtc_test("webrtc_perf_tests") {
     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) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 2700
     }
     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" ]
+    }
   }
 
   rtc_test("webrtc_nonparallel_tests") {
     testonly = true
     configs += [ ":rtc_unittests_config" ]
     sources = [
       "base/nullsocketserver_unittest.cc",
       "base/physicalsocketserver_unittest.cc",
       "base/socket_unittest.cc",
       "base/socket_unittest.h",
@@ skipping 29 matching lines @@
       ]
 
       deps = [
         "//base:base_java_test_support",
         "//webrtc/examples:AppRTCMobile_javalib",
         "//webrtc/sdk/android:libjingle_peerconnection_java",
       ]
     }
   }
 }