diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index e568fc7c7e..cf43954c4c 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -462,15 +462,38 @@ if (rtc_include_tests) {
     testonly = true
     configs += [ ":rtc_unittests_config" ]
 
-    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",
-      "test:test_main",
-      "video:video_full_stack_tests",
-      "video:video_quality_test",
+    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 = [
+      "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",
+      "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) {
@@ -480,6 +503,13 @@ if (rtc_include_tests) {
     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") {
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 195c37f614..4a32385348 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -75,21 +75,4 @@ if (rtc_include_tests) {
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
-
-  rtc_source_set("call_perf_tests") {
-    testonly = true
-    sources = [
-      "call_perf_tests.cc",
-      "rampup_tests.cc",
-      "rampup_tests.h",
-    ]
-    deps = [
-      "//testing/gtest",
-      "//webrtc/test:test_common",
-    ]
-    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" ]
-    }
-  }
 }
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index c44d022994..70ecc55dc2 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -1136,27 +1136,6 @@ if (rtc_include_tests) {
     }
   }
 
-  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 = [
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 5cbd97d90a..049a4e1d04 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -411,35 +411,6 @@ if (rtc_include_tests) {
     }
   }
 
-  rtc_source_set("audio_processing_perf_tests") {
-    # Has problems with autogenerated targets on Android and iOS
-    # Dependency chain (there may also be others):
-    #  //webrtc/modules/audio_processing:audio_processing_perf_tests -->
-    #  //webrtc/modules:modules_unittests --[private]-->
-    #  //webrtc/modules:modules_unittests_apk -->
-    #  //webrtc/modules:modules_unittests_apk__create -->
-    #  //webrtc/modules:modules_unittests_apk__create__finalize -->
-    #  //webrtc/modules:modules_unittests_apk__create__package --[private]-->
-    #  //webrtc/modules:_modules_unittests__library
-    check_includes = false
-    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 = [
diff --git a/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
index 7ddb0bb1e8..f73fa22067 100644
--- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn
+++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
@@ -124,21 +124,4 @@ if (rtc_include_tests) {
       "//testing/gtest",
     ]
   }
-
-  rtc_source_set("remote_bitrate_estimator_perf_tests") {
-    testonly = true
-    sources = [
-      "remote_bitrate_estimators_test.cc",
-    ]
-    deps = [
-      ":bwe_simulator_lib",
-      ":remote_bitrate_estimator",
-      "../../base:rtc_base_approved",
-      "../../test:fileutils",
-    ]
-    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" ]
-    }
-  }
 }
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 54464cf8d9..87a9800b85 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -89,7 +89,6 @@ if (rtc_include_tests) {
       "../media:rtc_media_base",
       "../system_wrappers",
       "//testing/gtest",
-      "//webrtc/test:test_renderer",
     ]
     if (!is_android) {
       deps += [ "../modules/video_capture:video_capture_internal_impl" ]
@@ -100,25 +99,6 @@ if (rtc_include_tests) {
     }
   }
 
-  rtc_source_set("video_full_stack_tests") {
-    testonly = true
-    sources = [
-      "full_stack_tests.cc",
-    ]
-    deps = [
-      ":video_quality_test",
-      "//testing/gtest",
-      "//webrtc/test:test_common",
-    ]
-    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_executable("video_loopback") {
     testonly = true
     sources = [