Refactor webrtc_perf_tests into several source_sets.

webrtc/modules/audio_coding/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.
 
 import("../../build/webrtc.gni")
 import("audio_coding.gni")
@@ skipping 1118 matching lines @@
       ":neteq_speed_test",
       ":rtp_analyze",
       ":rtpcat",
       ":webrtc_opus_fec_test",
     ]
     if (rtc_enable_protobuf) {
       public_deps += [ ":neteq_rtpplay" ]
     }
   }
 
+  rtc_source_set("audio_coding_perf_tests") {
+    testonly = true
+    sources = [
+      "codecs/opus/opus_complexity_unittest.cc",
+      "neteq/test/neteq_performance_unittest.cc",
+    ]
+    deps = [
+      ":neteq_test_support",
+      ":neteq_unittest_tools",
+      ":webrtc_opus",
+      "../..:webrtc_common",
+      "../../base:rtc_base_approved",
+      "../../system_wrappers:system_wrappers",
+      "../../test:test_support",
+    ]
+    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" ]
+    }
+  }
+
   rtc_source_set("acm_receive_test") {
     testonly = true
     sources = [
       "acm2/acm_receive_test.cc",
       "acm2/acm_receive_test.h",
     ]
 
     defines = audio_coding_defines
 
     deps = audio_coding_deps + [
@@ skipping 766 matching lines @@
       "../../test:test_main",
       "//testing/gtest",
     ]
 
     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" ]
     }
   }
 }