diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 8a7ca03608..cf3073d500 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -36,6 +36,7 @@ rtc_source_set("call_api") {
 
   deps = [
     # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
+    ":audio_mixer_api",
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../modules/audio_coding:audio_encoder_interface",
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index b50dfd5683..70efb5a27e 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -99,6 +99,7 @@
       'type': 'static_library',
       'dependencies': [
         # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
+        ':audio_mixer_api',
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface',
diff --git a/webrtc/api/call/audio_state.h b/webrtc/api/call/audio_state.h
index ac912773aa..1941cbf7bc 100644
--- a/webrtc/api/call/audio_state.h
+++ b/webrtc/api/call/audio_state.h
@@ -10,6 +10,7 @@
 #ifndef WEBRTC_API_CALL_AUDIO_STATE_H_
 #define WEBRTC_API_CALL_AUDIO_STATE_H_
 
+#include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
@@ -33,8 +34,9 @@ class AudioState : public rtc::RefCountInterface {
     // the AudioState itself.
     VoiceEngine* voice_engine = nullptr;
 
-    // The AudioDeviceModule associated with the Calls.
-    AudioDeviceModule* audio_device_module = nullptr;
+    // The audio mixer connected to active receive streams. One per
+    // AudioState.
+    rtc::scoped_refptr<AudioMixer> audio_mixer;
   };
 
   // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.