diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 530c5d4e3c..0c36221b7d 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -331,6 +331,8 @@ if (!build_with_chromium) {
 source_set("webrtc_common") {
   sources = [
     "audio_sink.h",
+    "common.cc",
+    "common.h",
     "common_types.cc",
     "common_types.h",
     "config.cc",
diff --git a/webrtc/common.cc b/webrtc/common.cc
new file mode 100644
index 0000000000..bc24818a60
--- /dev/null
+++ b/webrtc/common.cc
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "webrtc/common.h"
+
+namespace webrtc {
+
+Config::Config() {}
+
+Config::~Config() {
+  for (OptionMap::iterator it = options_.begin(); it != options_.end(); ++it) {
+    delete it->second;
+  }
+}
+
+}  // namespace webrtc
diff --git a/webrtc/common.gyp b/webrtc/common.gyp
index 2970877309..c9ac71c898 100644
--- a/webrtc/common.gyp
+++ b/webrtc/common.gyp
@@ -13,6 +13,8 @@
       'type': 'static_library',
       'sources': [
         'audio_sink.h',
+        'common.cc',
+        'common.h',
         'common_types.cc',
         'common_types.h',
         'config.h',
diff --git a/webrtc/common.h b/webrtc/common.h
index 8795064fd0..79db50c5b5 100644
--- a/webrtc/common.h
+++ b/webrtc/common.h
@@ -71,15 +71,8 @@ class Config {
   // This instance gets ownership of the newly set value.
   template<typename T> void Set(T* value);
 
-  Config() {}
-  ~Config() {
-    // Note: this method is inline so webrtc public API depends only
-    // on the headers.
-    for (OptionMap::iterator it = options_.begin();
-         it != options_.end(); ++it) {
-      delete it->second;
-    }
-  }
+  Config();
+  ~Config();
 
  private:
   struct BaseOption {
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 2925e0871f..72a471d9a6 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -917,11 +917,6 @@ if (rtc_include_tests) {
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
 
-    if (is_clang) {
-      # Suppress warnings from the Chromium Clang plugins (bugs.webrtc.org/163).
-      configs -= [ "//build/config/clang:find_bad_constructs" ]
-    }
-
     defines = audio_coding_defines
 
     deps = audio_coding_deps + [
@@ -943,12 +938,6 @@ if (rtc_include_tests) {
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
 
-    if (is_clang) {
-      # Suppress warnings from Chrome's Clang plugins.
-      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-      configs -= [ "//build/config/clang:find_bad_constructs" ]
-    }
-
     deps = [
       ":audio_coding",
       "../../:webrtc_common",
diff --git a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
index 84db4911d7..e16e55bb61 100644
--- a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
@@ -47,6 +47,8 @@ AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
   acm_->RegisterTransportCallback(this);
 }
 
+AcmSendTestOldApi::~AcmSendTestOldApi() = default;
+
 bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
                                       int sampling_freq_hz,
                                       int channels,
diff --git a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
index c752878186..50b51a5559 100644
--- a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
+++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
@@ -32,7 +32,7 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
   AcmSendTestOldApi(InputAudioFile* audio_source,
                     int source_rate_hz,
                     int test_duration_ms);
-  virtual ~AcmSendTestOldApi() {}
+  ~AcmSendTestOldApi() override;
 
   // Registers the send codec. Returns true on success, false otherwise.
   bool RegisterCodec(const char* payload_name,
diff --git a/webrtc/modules/audio_coding/test/Channel.h b/webrtc/modules/audio_coding/test/Channel.h
index 5910fade25..c45864acbb 100644
--- a/webrtc/modules/audio_coding/test/Channel.h
+++ b/webrtc/modules/audio_coding/test/Channel.h
@@ -47,7 +47,7 @@ class Channel : public AudioPacketizationCallback {
  public:
 
   Channel(int16_t chID = -1);
-  ~Channel();
+  ~Channel() override;
 
   int32_t SendData(FrameType frameType,
                    uint8_t payloadType,
diff --git a/webrtc/modules/audio_coding/test/PCMFile.cc b/webrtc/modules/audio_coding/test/PCMFile.cc
index 9289d73baa..b4acf35005 100644
--- a/webrtc/modules/audio_coding/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/test/PCMFile.cc
@@ -46,6 +46,12 @@ PCMFile::PCMFile(uint32_t timestamp)
   timestamp_ = timestamp;
 }
 
+PCMFile::~PCMFile() {
+  if (pcm_file_) {
+    fclose(pcm_file_);
+  }
+}
+
 int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len,
                             uint16_t* frequency_hz) {
   char tmp_name[MAX_FILE_NAME_LENGTH_BYTE];
diff --git a/webrtc/modules/audio_coding/test/PCMFile.h b/webrtc/modules/audio_coding/test/PCMFile.h
index 840933a1bd..b5ced0bac9 100644
--- a/webrtc/modules/audio_coding/test/PCMFile.h
+++ b/webrtc/modules/audio_coding/test/PCMFile.h
@@ -26,11 +26,7 @@ class PCMFile {
  public:
   PCMFile();
   PCMFile(uint32_t timestamp);
-  ~PCMFile() {
-    if (pcm_file_ != NULL) {
-      fclose(pcm_file_);
-    }
-  }
+  ~PCMFile();
 
   void Open(const std::string& filename, uint16_t frequency, const char* mode,
             bool auto_rewind = false);
diff --git a/webrtc/modules/audio_coding/test/utility.h b/webrtc/modules/audio_coding/test/utility.h
index 23869be7ed..dbd398ee7d 100644
--- a/webrtc/modules/audio_coding/test/utility.h
+++ b/webrtc/modules/audio_coding/test/utility.h
@@ -118,10 +118,8 @@ bool FixedPayloadTypeCodec(const char* payloadName);
 class VADCallback : public ACMVADCallback {
  public:
   VADCallback();
-  ~VADCallback() {
-  }
 
-  int32_t InFrameType(FrameType frame_type);
+  int32_t InFrameType(FrameType frame_type) override;
 
   void PrintFrameTypes();
   void Reset();