diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 95db035b48..154ae048e3 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -116,6 +116,7 @@ if (rtc_include_tests) {
       "../system_wrappers",
       "../system_wrappers:metrics_default",
       "../test:direct_transport",
+      "../test:fake_audio_device",
       "../test:test_support",
       "../test:video_test_common",
       "../video",
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index d772753fa5..63378905d1 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -58,7 +58,7 @@ class TestRtpFeedback : public NullRtpFeedback {
   explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
-  void OnIncomingSSRCChanged(const uint32_t ssrc) override {
+  void OnIncomingSSRCChanged(uint32_t ssrc) override {
     rtp_rtcp_->SetRemoteSSRC(ssrc);
   }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 0aa6361e26..40b8e061b9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -211,7 +211,7 @@ class TestTransport : public Transport,
     return true;
   }
   int OnReceivedPayloadData(const uint8_t* payload_data,
-                            const size_t payload_size,
+                            size_t payload_size,
                             const WebRtcRTPHeader* rtp_header) override {
     return 0;
   }
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 2801c65b90..5e5d8eea49 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -70,11 +70,11 @@ class VerifyingAudioReceiver : public NullRtpData {
 
 class RTPCallback : public NullRtpFeedback {
  public:
-  int32_t OnInitializeDecoder(const int8_t payloadType,
+  int32_t OnInitializeDecoder(int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                              const int frequency,
-                              const size_t channels,
-                              const uint32_t rate) override {
+                              int frequency,
+                              size_t channels,
+                              uint32_t rate) override {
     EXPECT_EQ(0u, rate) << "The rate should be zero";
     return 0;
   }
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index 25ff98d7f2..96379283b1 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -56,7 +56,7 @@ class TestRtpFeedback : public NullRtpFeedback {
   explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
-  void OnIncomingSSRCChanged(const uint32_t ssrc) override {
+  void OnIncomingSSRCChanged(uint32_t ssrc) override {
     rtp_rtcp_->SetRemoteSSRC(ssrc);
   }
 
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 07ffbee412..12eb3cacc3 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -333,6 +333,22 @@ rtc_source_set("direct_transport") {
   ]
 }
 
+rtc_source_set("fake_audio_device") {
+  testonly = true
+  sources = [
+    "fake_audio_device.cc",
+    "fake_audio_device.h",
+  ]
+  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" ]
+  }
+  deps = [
+    "../base:rtc_base_approved",
+    "../modules/audio_device:audio_device",
+  ]
+}
+
 rtc_source_set("test_common") {
   testonly = true
   sources = [
@@ -346,8 +362,6 @@ rtc_source_set("test_common") {
     "drifting_clock.h",
     "encoder_settings.cc",
     "encoder_settings.h",
-    "fake_audio_device.cc",
-    "fake_audio_device.h",
     "fake_decoder.cc",
     "fake_decoder.h",
     "fake_encoder.cc",
@@ -379,13 +393,13 @@ rtc_source_set("test_common") {
 
   deps = [
     ":direct_transport",
+    ":fake_audio_device",
     ":rtp_test_utils",
     ":test_support",
     "..:webrtc_common",
     "../audio",
     "../base:rtc_base_approved",
     "../call",
-    "../modules/audio_device:mock_audio_device",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/audio_processing",
     "../video",