diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index e1d583f8e5..e89350e89a 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -48,7 +48,9 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":audio",
+      "../base:rtc_base_approved",
       "../modules/audio_device:mock_audio_device",
+      "../test:test_common",
       "//testing/gmock",
       "//testing/gtest",
     ]
diff --git a/webrtc/audio/DEPS b/webrtc/audio/DEPS
index e53d28578e..6e6259b9e5 100644
--- a/webrtc/audio/DEPS
+++ b/webrtc/audio/DEPS
@@ -4,6 +4,7 @@ include_rules = [
   "+webrtc/logging/rtc_event_log",
   "+webrtc/modules/audio_coding/codecs/mock",
   "+webrtc/modules/audio_device",
+  "+webrtc/modules/audio_mixer",
   "+webrtc/modules/audio_processing/include",
   "+webrtc/modules/bitrate_controller",
   "+webrtc/modules/congestion_controller",
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 0fc93b8b34..f263948fc4 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -15,6 +15,7 @@
 #include "webrtc/audio/conversion.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
@@ -84,6 +85,7 @@ struct ConfigHelper {
 
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
+    config.audio_mixer = AudioMixerImpl::Create();
     audio_state_ = AudioState::Create(config);
 
     EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 7e0ea45bb2..dbc49662e6 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -16,6 +16,7 @@
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
@@ -81,6 +82,7 @@ struct ConfigHelper {
 
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
+    config.audio_mixer = AudioMixerImpl::Create();
     audio_state_ = AudioState::Create(config);
 
     SetupDefaultChannelProxy();
diff --git a/webrtc/audio/audio_state.cc b/webrtc/audio/audio_state.cc
index 95a90a5bf1..c15ddd7795 100644
--- a/webrtc/audio/audio_state.cc
+++ b/webrtc/audio/audio_state.cc
@@ -26,6 +26,8 @@ AudioState::AudioState(const AudioState::Config& config)
                              voe_base_->audio_processing(),
                              config_.audio_mixer) {
   process_thread_checker_.DetachFromThread();
+  RTC_DCHECK(config_.audio_mixer);
+
   // Only one AudioState should be created per VoiceEngine.
   RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
 
@@ -48,6 +50,7 @@ VoiceEngine* AudioState::voice_engine() {
 }
 
 rtc::scoped_refptr<AudioMixer> AudioState::mixer() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_.audio_mixer;
 }
 
diff --git a/webrtc/audio/audio_state_unittest.cc b/webrtc/audio/audio_state_unittest.cc
index bd39baac1b..38485a88bf 100644
--- a/webrtc/audio/audio_state_unittest.cc
+++ b/webrtc/audio/audio_state_unittest.cc
@@ -11,6 +11,7 @@
 #include <memory>
 
 #include "webrtc/audio/audio_state.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
 
@@ -29,6 +30,7 @@ struct ConfigHelper {
     EXPECT_CALL(voice_engine_, audio_transport());
 
     config_.voice_engine = &voice_engine_;
+    config_.audio_mixer = AudioMixerImpl::Create();
   }
   AudioState::Config& config() { return config_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index ad20f340fc..d77dfc6b60 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -49,7 +49,10 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":call",
+      "../base:rtc_base_approved",
       "../modules/audio_device:mock_audio_device",
+      "../modules/audio_mixer",
+      "../test:test_common",
       "//testing/gmock",
       "//testing/gtest",
     ]
diff --git a/webrtc/call/DEPS b/webrtc/call/DEPS
index 01ad1d69cb..2ef22fca8f 100644
--- a/webrtc/call/DEPS
+++ b/webrtc/call/DEPS
@@ -4,6 +4,7 @@ include_rules = [
   "+webrtc/logging/rtc_event_log",
   "+webrtc/modules/audio_coding",
   "+webrtc/modules/audio_device",
+  "+webrtc/modules/audio_mixer",
   "+webrtc/modules/audio_processing",
   "+webrtc/modules/bitrate_controller",
   "+webrtc/modules/congestion_controller",
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index b09d73c424..354e092a4f 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -20,6 +20,7 @@
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics_default.h"
@@ -159,6 +160,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
 
   AudioState::Config send_audio_state_config;
   send_audio_state_config.voice_engine = voice_engine;
+  send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
   Call::Config sender_config(&event_log_);
   sender_config.audio_state = AudioState::Create(send_audio_state_config);
   Call::Config receiver_config(&event_log_);
@@ -264,7 +266,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   Start();
 
   fake_audio_device.Start();
-  EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
+  audio_receive_stream->Start();
   EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
 
   EXPECT_TRUE(observer.Wait())
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index d981b0b013..2d75be5eda 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -14,6 +14,7 @@
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
@@ -26,6 +27,7 @@ struct CallHelper {
       : voice_engine_(decoder_factory) {
     webrtc::AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &voice_engine_;
+    audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
     EXPECT_CALL(voice_engine_, audio_device_module());
     EXPECT_CALL(voice_engine_, audio_processing());
     EXPECT_CALL(voice_engine_, audio_transport());
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index c3c456c297..c3d173aac8 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -168,6 +168,7 @@ rtc_static_library("rtc_media") {
     "../api:call_api",
     "../base:rtc_base_approved",
     "../call",
+    "../modules/audio_mixer:audio_mixer_impl",
     "../modules/video_coding",
     "../p2p",
     "../system_wrappers",
diff --git a/webrtc/media/DEPS b/webrtc/media/DEPS
index 580ac5c452..e6d642daa1 100644
--- a/webrtc/media/DEPS
+++ b/webrtc/media/DEPS
@@ -6,6 +6,7 @@ include_rules = [
   "+webrtc/logging/rtc_event_log",
   "+webrtc/modules/audio_coding",
   "+webrtc/modules/audio_device",
+  "+webrtc/modules/audio_mixer",
   "+webrtc/modules/audio_processing",
   "+webrtc/modules/video_capture",
   "+webrtc/modules/video_coding",
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 2c67c56406..937078c3e9 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -37,6 +37,7 @@
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -279,6 +280,7 @@ void GetOpusConfig(const AudioCodec& codec,
 webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) {
   webrtc::AudioState::Config config;
   config.voice_engine = voe_wrapper->engine();
+  config.audio_mixer = webrtc::AudioMixerImpl::Create();
   return config;
 }
 
diff --git a/webrtc/modules/audio_mixer/BUILD.gn b/webrtc/modules/audio_mixer/BUILD.gn
index 9b34d54300..addffdd9f4 100644
--- a/webrtc/modules/audio_mixer/BUILD.gn
+++ b/webrtc/modules/audio_mixer/BUILD.gn
@@ -16,10 +16,6 @@ group("audio_mixer") {
 }
 
 rtc_static_library("audio_mixer_impl") {
-  visibility = [
-    "../../audio:audio",
-    "../../modules/*",
-  ]
   sources = [
     "audio_mixer_impl.cc",
     "audio_mixer_impl.h",
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 28251e6cfb..126857b50c 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -334,6 +334,7 @@ rtc_source_set("test_common") {
     "../base:rtc_base_approved",
     "../call",
     "../modules/audio_device:mock_audio_device",
+    "../modules/audio_mixer:audio_mixer_impl",
     "../modules/audio_processing",
     "../modules/media_file",
     "../modules/video_capture:video_capture_module",
diff --git a/webrtc/test/DEPS b/webrtc/test/DEPS
index 9c41bd4ce8..59167aaed1 100644
--- a/webrtc/test/DEPS
+++ b/webrtc/test/DEPS
@@ -6,6 +6,7 @@ include_rules = [
   "+webrtc/media/base",
   "+webrtc/modules/audio_coding",
   "+webrtc/modules/audio_device",
+  "+webrtc/modules/audio_mixer",
   "+webrtc/modules/audio_processing",
   "+webrtc/modules/media_file",
   "+webrtc/modules/rtp_rtcp",
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index ddff552565..28971af986 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -15,6 +15,8 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc/test/call_test.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
@@ -52,6 +54,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
     CreateVoiceEngines();
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe_send_.voice_engine;
+    audio_state_config.audio_mixer = AudioMixerImpl::Create();
     send_config.audio_state = AudioState::Create(audio_state_config);
   }
   CreateSenderCall(send_config);
@@ -60,6 +63,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
     if (num_audio_streams_ > 0) {
       AudioState::Config audio_state_config;
       audio_state_config.voice_engine = voe_recv_.voice_engine;
+      audio_state_config.audio_mixer = AudioMixerImpl::Create();
       recv_config.audio_state = AudioState::Create(audio_state_config);
     }
     CreateReceiverCall(recv_config);
diff --git a/webrtc/video/DEPS b/webrtc/video/DEPS
index fafad7b9ed..1cdad193e3 100644
--- a/webrtc/video/DEPS
+++ b/webrtc/video/DEPS
@@ -4,6 +4,7 @@ include_rules = [
   "+webrtc/common_video",
   "+webrtc/logging/rtc_event_log",
   "+webrtc/media/base",
+  "+webrtc/modules/audio_mixer",
   "+webrtc/modules/bitrate_controller",
   "+webrtc/modules/congestion_controller",
   "+webrtc/modules/pacing",
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index b48c692376..d0624c0a52 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -26,6 +26,7 @@
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
@@ -1341,6 +1342,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
     CreateVoiceEngine(&voe, decoder_factory_);
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe.voice_engine;
+    audio_state_config.audio_mixer = AudioMixerImpl::Create();
     call_config.audio_state = AudioState::Create(audio_state_config);
   }