diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index b80635ff8e..a3c8b3f188 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -410,68 +410,6 @@ struct AudioDecodingCallStats {
   int decoded_muted_output;  // Number of calls returning a muted state output.
 };
 
-// Type of Noise Suppression.
-enum NsModes {
-  kNsUnchanged = 0,   // previously set mode
-  kNsDefault,         // platform default
-  kNsConference,      // conferencing default
-  kNsLowSuppression,  // lowest suppression
-  kNsModerateSuppression,
-  kNsHighSuppression,
-  kNsVeryHighSuppression,  // highest suppression
-};
-
-// Type of Automatic Gain Control.
-enum AgcModes {
-  kAgcUnchanged = 0,  // previously set mode
-  kAgcDefault,        // platform default
-  // adaptive mode for use when analog volume control exists (e.g. for
-  // PC softphone)
-  kAgcAdaptiveAnalog,
-  // scaling takes place in the digital domain (e.g. for conference servers
-  // and embedded devices)
-  kAgcAdaptiveDigital,
-  // can be used on embedded devices where the capture signal level
-  // is predictable
-  kAgcFixedDigital
-};
-
-// Type of Echo Control.
-enum EcModes {
-  kEcUnchanged = 0,  // previously set mode
-  kEcDefault,        // platform default
-  kEcConference,     // conferencing default (aggressive AEC)
-  kEcAec,            // Acoustic Echo Cancellation
-  kEcAecm,           // AEC mobile
-};
-
-// Mode of AECM.
-enum AecmModes {
-  kAecmQuietEarpieceOrHeadset = 0,
-  // Quiet earpiece or headset use
-  kAecmEarpiece,         // most earpiece use
-  kAecmLoudEarpiece,     // Loud earpiece or quiet speakerphone use
-  kAecmSpeakerphone,     // most speakerphone use (default)
-  kAecmLoudSpeakerphone  // Loud speakerphone
-};
-
-// AGC configuration parameters
-struct AgcConfig {
-  unsigned short targetLeveldBOv;
-  unsigned short digitalCompressionGaindB;
-  bool limiterEnable;
-};
-
-enum StereoChannel { kStereoLeft = 0, kStereoRight, kStereoBoth };
-
-// Audio device layers
-enum AudioLayers {
-  kAudioPlatformDefault = 0,
-  kAudioWindowsCore = 2,
-  kAudioLinuxAlsa = 3,
-  kAudioLinuxPulse = 4
-};
-
 // ==================================================================
 // Video specific types
 // ==================================================================
diff --git a/webrtc/media/engine/apm_helpers.cc b/webrtc/media/engine/apm_helpers.cc
index a806523d62..e11cac73ef 100644
--- a/webrtc/media/engine/apm_helpers.cc
+++ b/webrtc/media/engine/apm_helpers.cc
@@ -45,15 +45,12 @@ void SetAgcConfig(AudioProcessing* apm,
 
 void SetAgcStatus(AudioProcessing* apm,
                   AudioDeviceModule* adm,
-                  bool enable,
-                  AgcModes mode) {
+                  bool enable) {
   RTC_DCHECK(apm);
   RTC_DCHECK(adm);
 #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
-  RTC_DCHECK_EQ(kAgcFixedDigital, mode);
   GainControl::Mode agc_mode = GainControl::kFixedDigital;
 #else
-  RTC_DCHECK_EQ(kAgcAdaptiveAnalog, mode);
   GainControl::Mode agc_mode = GainControl::kAdaptiveAnalog;
 #endif
   GainControl* gc = apm->gain_control();
@@ -70,7 +67,7 @@ void SetAgcStatus(AudioProcessing* apm,
     LOG(LS_ERROR) << "Failed to set AGC mode in ADM: " << enable;
     return;
   }
-  LOG(LS_INFO) << "AGC set to " << enable << " with mode " << mode;
+  LOG(LS_INFO) << "AGC set to " << enable << " with mode " << agc_mode;
 }
 
 void SetEcStatus(AudioProcessing* apm,
diff --git a/webrtc/media/engine/apm_helpers.h b/webrtc/media/engine/apm_helpers.h
index 1bcf0511ec..eff7133c44 100644
--- a/webrtc/media/engine/apm_helpers.h
+++ b/webrtc/media/engine/apm_helpers.h
@@ -11,13 +11,22 @@
 #ifndef WEBRTC_MEDIA_ENGINE_APM_HELPERS_H_
 #define WEBRTC_MEDIA_ENGINE_APM_HELPERS_H_
 
-#include "webrtc/common_types.h"
-
 namespace webrtc {
 
 class AudioProcessing;
 class AudioDeviceModule;
 
+enum EcModes {
+  kEcConference,     // Conferencing default (aggressive AEC).
+  kEcAecm,           // AEC mobile.
+};
+
+struct AgcConfig {
+  unsigned short targetLeveldBOv;
+  unsigned short digitalCompressionGaindB;
+  bool limiterEnable;
+};
+
 namespace apm_helpers {
 
 AgcConfig GetAgcConfig(AudioProcessing* apm);
@@ -25,8 +34,7 @@ void SetAgcConfig(AudioProcessing* apm,
                   const AgcConfig& config);
 void SetAgcStatus(AudioProcessing* apm,
                   AudioDeviceModule* adm,
-                  bool enable,
-                  AgcModes mode);
+                  bool enable);
 void SetEcStatus(AudioProcessing* apm,
                  bool enable,
                  EcModes mode);
diff --git a/webrtc/media/engine/apm_helpers_unittest.cc b/webrtc/media/engine/apm_helpers_unittest.cc
index 8c28db703c..1db77fcbb5 100644
--- a/webrtc/media/engine/apm_helpers_unittest.cc
+++ b/webrtc/media/engine/apm_helpers_unittest.cc
@@ -123,25 +123,21 @@ TEST(ApmHelpersTest, AgcStatus_EnableDisable) {
   TestHelper helper;
   GainControl* gc = helper.apm()->gain_control();
 #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
-  apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), false,
-                            kAgcFixedDigital);
+  apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), false);
   EXPECT_FALSE(gc->is_enabled());
   EXPECT_EQ(GainControl::kFixedDigital, gc->mode());
 
-  apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), true,
-                            kAgcFixedDigital);
+  apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), true);
   EXPECT_TRUE(gc->is_enabled());
   EXPECT_EQ(GainControl::kFixedDigital, gc->mode());
 #else
   EXPECT_CALL(*helper.adm(), SetAGC(false)).WillOnce(testing::Return(0));
-  apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), false,
-                            kAgcAdaptiveAnalog);
+  apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), false);
   EXPECT_FALSE(gc->is_enabled());
   EXPECT_EQ(GainControl::kAdaptiveAnalog, gc->mode());
 
   EXPECT_CALL(*helper.adm(), SetAGC(true)).WillOnce(testing::Return(0));
-  apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), true,
-                            kAgcAdaptiveAnalog);
+  apm_helpers::SetAgcStatus(helper.apm(), helper.adm(), true);
   EXPECT_TRUE(gc->is_enabled());
   EXPECT_EQ(GainControl::kAdaptiveAnalog, gc->mode());
 #endif
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index c95d7582a9..7c6d7940f6 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -681,7 +681,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
 
   // kEcConference is AEC with high suppression.
   webrtc::EcModes ec_mode = webrtc::kEcConference;
-  webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
   if (options.aecm_generate_comfort_noise) {
     LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
                     << *options.aecm_generate_comfort_noise
@@ -700,9 +699,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
 #endif
 
 #if defined(WEBRTC_IOS) || defined(ANDROID)
-  // Set the AGC mode for iOS as well despite disabling it above, to avoid
-  // unsupported configuration errors from webrtc.
-  agc_mode = webrtc::kAgcFixedDigital;
   options.typing_detection = rtc::Optional<bool>(false);
   options.experimental_agc = rtc::Optional<bool>(false);
   options.extended_filter_aec = rtc::Optional<bool>(false);
@@ -770,8 +766,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
         LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead";
       }
     }
-    webrtc::apm_helpers::SetAgcStatus(
-        apm(), adm(), *options.auto_gain_control, agc_mode);
+    webrtc::apm_helpers::SetAgcStatus(apm(), adm(), *options.auto_gain_control);
   }
 
   if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index bac336c017..c777c6c03e 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -25,6 +25,7 @@
 #include "webrtc/call/call.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/rtputils.h"
+#include "webrtc/media/engine/apm_helpers.h"
 #include "webrtc/media/engine/webrtccommon.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"