diff --git a/BUILD.gn b/BUILD.gn
index de5e773036..25a2fb4148 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -425,6 +425,7 @@ if (rtc_include_tests) {
     deps = [
       ":webrtc_common",
       "api:rtc_api_unittests",
+      "api/audio/test:audio_api_unittests",
       "api/audio_codecs/test:audio_codecs_api_unittests",
       "p2p:libstunprober_unittests",
       "p2p:rtc_p2p_unittests",
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index 864188dee4..6fb0ffaa0c 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -8,8 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <string.h>
-
 #include "api/audio/audio_frame.h"
 
 #include "rtc_base/checks.h"
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index d69f607ce6..39840e5e6a 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -11,6 +11,8 @@
 #ifndef API_AUDIO_AUDIO_FRAME_H_
 #define API_AUDIO_AUDIO_FRAME_H_
 
+#include <stddef.h>
+
 #include "rtc_base/constructormagic.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
diff --git a/api/audio/test/BUILD.gn b/api/audio/test/BUILD.gn
new file mode 100644
index 0000000000..9a797788f6
--- /dev/null
+++ b/api/audio/test/BUILD.gn
@@ -0,0 +1,27 @@
+# Copyright (c) 2018 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+if (rtc_include_tests) {
+  rtc_source_set("audio_api_unittests") {
+    testonly = true
+    sources = [
+      "audio_frame_unittest.cc",
+    ]
+    deps = [
+      "..:audio_frame_api",
+      "../../../rtc_base:rtc_base_approved",
+      "../../../test:test_support",
+    ]
+  }
+}
diff --git a/api/audio/test/audio_frame_unittest.cc b/api/audio/test/audio_frame_unittest.cc
new file mode 100644
index 0000000000..2a41e76f2c
--- /dev/null
+++ b/api/audio/test/audio_frame_unittest.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright 2018 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 <string.h>  // memcmp
+
+#include "api/audio/audio_frame.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
+  const int16_t* frame_data = frame.data();
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
+    if (frame_data[i] != sample) {
+      return false;
+    }
+  }
+  return true;
+}
+
+constexpr uint32_t kTimestamp = 27;
+constexpr int kSampleRateHz = 16000;
+constexpr size_t kNumChannels = 1;
+constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
+
+}  // namespace
+
+TEST(AudioFrameTest, FrameStartsMuted) {
+  AudioFrame frame;
+  EXPECT_TRUE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
+  AudioFrame frame;
+  frame.mutable_data();
+  EXPECT_FALSE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
+  AudioFrame frame;
+  int16_t* frame_data = frame.mutable_data();
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
+    frame_data[i] = 17;
+  }
+  ASSERT_TRUE(AllSamplesAre(17, frame));
+  frame.Mute();
+  EXPECT_TRUE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, UpdateFrame) {
+  AudioFrame frame;
+  int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
+  frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
+                    AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
+
+  EXPECT_EQ(kTimestamp, frame.timestamp_);
+  EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel_);
+  EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz_);
+  EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_);
+  EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);
+  EXPECT_EQ(kNumChannels, frame.num_channels_);
+
+  EXPECT_FALSE(frame.muted());
+  EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
+
+  frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
+                    kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+                    kNumChannels);
+  EXPECT_TRUE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, CopyFrom) {
+  AudioFrame frame1;
+  AudioFrame frame2;
+
+  int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
+  frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel,
+                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+                     kNumChannels);
+  frame1.CopyFrom(frame2);
+
+  EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
+  EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
+  EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
+  EXPECT_EQ(frame2.speech_type_, frame1.speech_type_);
+  EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
+  EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
+
+  EXPECT_EQ(frame2.muted(), frame1.muted());
+  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
+
+  frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
+                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+                     kNumChannels);
+  frame1.CopyFrom(frame2);
+
+  EXPECT_EQ(frame2.muted(), frame1.muted());
+  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
+}
+
+}  // namespace webrtc
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 3880eab61c..8981dd3353 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -52,6 +52,7 @@ rtc_static_library("audio") {
     "../api:optional",
     "../api:transport_api",
     "../api/audio:aec3_factory",
+    "../api/audio:audio_frame_api",
     "../api/audio:audio_mixer_api",
     "../api/audio_codecs:audio_codecs_api",
     "../api/audio_codecs:builtin_audio_encoder_factory",
@@ -62,7 +63,6 @@ rtc_static_library("audio") {
     "../common_audio:common_audio_c",
     "../logging:rtc_event_audio",
     "../logging:rtc_event_log_api",
-    "../modules:module_api",
     "../modules/audio_coding",
     "../modules/audio_coding:audio_format_conversion",
     "../modules/audio_coding:audio_network_adaptor_config",
@@ -127,13 +127,13 @@ if (rtc_include_tests) {
       ":audio",
       ":audio_end_to_end_test",
       "../api:mock_audio_mixer",
+      "../api/audio:audio_frame_api",
       "../call:mock_call_interfaces",
       "../call:mock_rtp_interfaces",
       "../call:rtp_interfaces",
       "../call:rtp_receiver",
       "../common_audio",
       "../logging:mocks",
-      "../modules:module_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/audio_processing:audio_processing_statistics",
diff --git a/audio/audio_level.cc b/audio/audio_level.cc
index ca5252200a..f1c5b68dc1 100644
--- a/audio/audio_level.cc
+++ b/audio/audio_level.cc
@@ -10,8 +10,8 @@
 
 #include "audio/audio_level.h"
 
+#include "api/audio/audio_frame.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace voe {
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index a47b59c3e9..09007f0497 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "api/audio/audio_mixer.h"
+#include "api/rtp_headers.h"
 #include "audio/audio_state.h"
 #include "call/audio_receive_stream.h"
 #include "call/rtp_packet_sink_interface.h"
diff --git a/audio/channel.cc b/audio/channel.cc
index a7325c32a9..df34ca22ce 100644
--- a/audio/channel.cc
+++ b/audio/channel.cc
@@ -26,7 +26,6 @@
 #include "modules/audio_coding/codecs/audio_format_conversion.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "modules/include/module_common_types.h"
 #include "modules/pacing/packet_router.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/include/rtp_payload_registry.h"
diff --git a/audio/remix_resample.cc b/audio/remix_resample.cc
index 52a491fdd9..69038cd0c9 100644
--- a/audio/remix_resample.cc
+++ b/audio/remix_resample.cc
@@ -10,11 +10,11 @@
 
 #include "audio/remix_resample.h"
 
+#include "api/audio/audio_frame.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "common_audio/resampler/include/push_resampler.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "common_types.h"  // NOLINT(build/include)
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 
diff --git a/audio/remix_resample.h b/audio/remix_resample.h
index ddd8086957..a45270b39a 100644
--- a/audio/remix_resample.h
+++ b/audio/remix_resample.h
@@ -11,12 +11,10 @@
 #ifndef AUDIO_REMIX_RESAMPLE_H_
 #define AUDIO_REMIX_RESAMPLE_H_
 
+#include "api/audio/audio_frame.h"
 #include "common_audio/resampler/include/push_resampler.h"
 
 namespace webrtc {
-
-class AudioFrame;
-
 namespace voe {
 
 // Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
diff --git a/audio/remix_resample_unittest.cc b/audio/remix_resample_unittest.cc
index 753584b46d..1d8cce7cc4 100644
--- a/audio/remix_resample_unittest.cc
+++ b/audio/remix_resample_unittest.cc
@@ -12,8 +12,8 @@
 
 #include "audio/remix_resample.h"
 #include "common_audio/resampler/include/push_resampler.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/format_macros.h"
 #include "test/gtest.h"
 
diff --git a/audio/transport_feedback_packet_loss_tracker.h b/audio/transport_feedback_packet_loss_tracker.h
index 7e73210327..4ad49024a8 100644
--- a/audio/transport_feedback_packet_loss_tracker.h
+++ b/audio/transport_feedback_packet_loss_tracker.h
@@ -15,7 +15,6 @@
 #include <vector>
 
 #include "api/optional.h"
-#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/audio/utility/BUILD.gn b/audio/utility/BUILD.gn
index aa8445c90b..1572469629 100644
--- a/audio/utility/BUILD.gn
+++ b/audio/utility/BUILD.gn
@@ -23,7 +23,7 @@ rtc_static_library("audio_frame_operations") {
   deps = [
     "../..:webrtc_common",
     "../../:typedefs",
-    "../../modules:module_api",
+    "../../api/audio:audio_frame_api",
     "../../modules/audio_coding:audio_format_conversion",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
@@ -38,7 +38,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":audio_frame_operations",
-      "../../modules:module_api",
       "../../rtc_base:checks",
       "../../rtc_base:rtc_base_approved",
       "../../test:test_support",
diff --git a/audio/utility/audio_frame_operations.cc b/audio/utility/audio_frame_operations.cc
index a7c77821f6..a53629ae6f 100644
--- a/audio/utility/audio_frame_operations.cc
+++ b/audio/utility/audio_frame_operations.cc
@@ -12,7 +12,6 @@
 
 #include <algorithm>
 
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
diff --git a/audio/utility/audio_frame_operations.h b/audio/utility/audio_frame_operations.h
index cd55f19fc1..65a2bad9c7 100644
--- a/audio/utility/audio_frame_operations.h
+++ b/audio/utility/audio_frame_operations.h
@@ -13,12 +13,11 @@
 
 #include <stddef.h>
 
+#include "api/audio/audio_frame.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 
-class AudioFrame;
-
 // TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
 // Change reference parameters to pointers. Consider using a namespace rather
 // than a class.
diff --git a/audio/utility/audio_frame_operations_unittest.cc b/audio/utility/audio_frame_operations_unittest.cc
index 6d23731a05..1d08d7ee41 100644
--- a/audio/utility/audio_frame_operations_unittest.cc
+++ b/audio/utility/audio_frame_operations_unittest.cc
@@ -9,7 +9,6 @@
  */
 
 #include "audio/utility/audio_frame_operations.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "test/gtest.h"
 
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index f6c6920f90..c350d76ba7 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -97,7 +97,6 @@ rtc_source_set("audio_coding_module_typedefs") {
     "include/audio_coding_module_typedefs.h",
   ]
   deps = [
-    "..:module_api",
     "../..:typedefs",
     "../..:webrtc_common",
   ]
@@ -136,12 +135,13 @@ rtc_static_library("audio_coding") {
   }
 
   deps = audio_coding_deps + [
+           "../../api/audio:audio_frame_api",
+           "..:module_api",
            "../../common_audio:common_audio_c",
            "../..:typedefs",
            "../../rtc_base:deprecation",
            "../../rtc_base:checks",
            "../../system_wrappers:metrics_api",
-           "..:module_api",
            "../../api:array_view",
            "../../api/audio_codecs:audio_codecs_api",
            ":audio_coding_module_typedefs",
@@ -1086,6 +1086,7 @@ rtc_static_library("neteq") {
     "../..:webrtc_common",
     "../../api:libjingle_peerconnection_api",
     "../../api:optional",
+    "../../api/audio:audio_frame_api",
     "../../api/audio_codecs:audio_codecs_api",
     "../../common_audio",
     "../../common_audio:common_audio_c",
@@ -1129,11 +1130,11 @@ rtc_source_set("neteq_tools_minimal") {
 
   deps = [
     ":neteq",
-    "..:module_api",
     "../..:typedefs",
     "../..:webrtc_common",
     "../../api:libjingle_peerconnection_api",
     "../../api:optional",
+    "../../api/audio:audio_frame_api",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs:builtin_audio_decoder_factory",
     "../../rtc_base:checks",
@@ -1168,7 +1169,6 @@ rtc_source_set("neteq_test_tools") {
 
   deps = [
     ":pcm16b",
-    "..:module_api",
     "../..:typedefs",
     "../..:webrtc_common",
     "../../api:array_view",
@@ -1220,6 +1220,7 @@ rtc_source_set("neteq_tools") {
   }
 
   deps = [
+    "..:module_api",
     "../..:typedefs",
     "../..:webrtc_common",
     "../../api:array_view",
@@ -1379,6 +1380,7 @@ if (rtc_include_tests) {
       "../..:typedefs",
       "../..:webrtc_common",
       "../../api:optional",
+      "../../api/audio:audio_frame_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../rtc_base:rtc_base_approved",
       "../../rtc_base/synchronization:rw_lock_wrapper",
@@ -1438,6 +1440,7 @@ if (rtc_include_tests) {
     defines = audio_coding_defines
 
     deps = audio_coding_deps + [
+             "..:module_api",
              ":audio_coding",
              ":audio_format_conversion",
              "../../api/audio_codecs:audio_codecs_api",
@@ -1459,6 +1462,7 @@ if (rtc_include_tests) {
     defines = audio_coding_defines
 
     deps = audio_coding_deps + [
+             "../../api/audio:audio_frame_api",
              "../../rtc_base:checks",
              ":audio_coding",
              ":neteq_tools",
@@ -1490,6 +1494,7 @@ if (rtc_include_tests) {
       "../..:typedefs",
       "../../:webrtc_common",
       "../../api:optional",
+      "../../api/audio:audio_frame_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
@@ -1523,6 +1528,7 @@ if (rtc_include_tests) {
       "../..:typedefs",
       "../../:webrtc_common",
       "../../api:optional",
+      "../../api/audio:audio_frame_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
@@ -1602,7 +1608,6 @@ if (rtc_include_tests) {
       testonly = true
       defines = []
       deps = [
-        "..:module_api",
         "../..:typedefs",
         "../../rtc_base:checks",
         "../../test:fileutils",
@@ -1713,9 +1718,9 @@ if (rtc_include_tests) {
       ":neteq",
       ":neteq_test_tools",
       ":pcm16b",
-      "..:module_api",
       "../..:typedefs",
       "../..:webrtc_common",
+      "../../api/audio:audio_frame_api",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../rtc_base:checks",
@@ -1742,7 +1747,6 @@ if (rtc_include_tests) {
     deps = [
       ":neteq",
       ":neteq_test_tools",
-      "..:module_api",
       "../..:typedefs",
       "../..:webrtc_common",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
@@ -1761,6 +1765,7 @@ if (rtc_include_tests) {
              "../..:typedefs",
              ":audio_coding",
              ":neteq_input_audio_tools",
+             "../../api/audio:audio_frame_api",
              "../../api/audio_codecs/g711:audio_encoder_g711",
              "../../api/audio_codecs/L16:audio_encoder_L16",
              "../../api/audio_codecs/g722:audio_encoder_g722",
@@ -2222,6 +2227,7 @@ if (rtc_include_tests) {
       "..:module_api",
       "../..:typedefs",
       "../..:webrtc_common",
+      "../../api/audio:audio_frame_api",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../api/audio_codecs:builtin_audio_encoder_factory",
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 082506ab56..473b6519ab 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -21,6 +21,7 @@
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
 #include "modules/audio_coding/neteq/tools/packet.h"
 #include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/include/module_common_types.h"
 #include "test/gtest.h"
 
 namespace webrtc {
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 4d0209bc92..41a23a7b1d 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -22,6 +22,7 @@
 #include "modules/audio_coding/acm2/call_statistics.h"
 #include "modules/audio_coding/acm2/rent_a_codec.h"
 #include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/format_macros.h"
 #include "rtc_base/logging.h"
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 5c6b36fdb0..ce1e1f2ff1 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -16,6 +16,7 @@
 #include <string>
 #include <vector>
 
+#include "api/audio/audio_frame.h"
 #include "api/array_view.h"
 #include "api/optional.h"
 #include "common_audio/vad/include/webrtc_vad.h"
@@ -23,7 +24,6 @@
 #include "modules/audio_coding/acm2/call_statistics.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/neteq/include/neteq.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/thread_annotations.h"
 #include "typedefs.h"  // NOLINT(build/include)
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 8d0b2f100e..78778214a0 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -17,6 +17,7 @@
 #include "modules/audio_coding/acm2/rent_a_codec.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "system_wrappers/include/clock.h"
diff --git a/modules/audio_coding/acm2/acm_send_test.h b/modules/audio_coding/acm2/acm_send_test.h
index 6aea0f1b52..68ba9e1fb6 100644
--- a/modules/audio_coding/acm2/acm_send_test.h
+++ b/modules/audio_coding/acm2/acm_send_test.h
@@ -14,6 +14,7 @@
 #include <memory>
 #include <vector>
 
+#include "api/audio/audio_frame.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/neteq/tools/packet_source.h"
 #include "rtc_base/constructormagic.h"
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 0c296636f5..4545810b56 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -16,6 +16,7 @@
 #include "modules/audio_coding/acm2/acm_resampler.h"
 #include "modules/audio_coding/acm2/codec_manager.h"
 #include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 24f0c3f416..ff5431dba9 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -32,7 +32,6 @@
 #include "modules/audio_coding/neteq/tools/output_wav_file.h"
 #include "modules/audio_coding/neteq/tools/packet.h"
 #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/messagedigest.h"
 #include "rtc_base/numerics/safe_conversions.h"
diff --git a/modules/audio_coding/acm2/call_statistics.h b/modules/audio_coding/acm2/call_statistics.h
index 9dd052f003..9dced6475a 100644
--- a/modules/audio_coding/acm2/call_statistics.h
+++ b/modules/audio_coding/acm2/call_statistics.h
@@ -11,8 +11,8 @@
 #ifndef MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
 #define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
 
+#include "api/audio/audio_frame.h"
 #include "common_types.h"  // NOLINT(build/include)
-#include "modules/include/module_common_types.h"
 
 //
 // This class is for book keeping of calls to ACM. It is not useful to log API
diff --git a/modules/audio_coding/acm2/rent_a_codec_unittest.cc b/modules/audio_coding/acm2/rent_a_codec_unittest.cc
index c949c1cae6..ca469e7cdc 100644
--- a/modules/audio_coding/acm2/rent_a_codec_unittest.cc
+++ b/modules/audio_coding/acm2/rent_a_codec_unittest.cc
@@ -10,6 +10,7 @@
 
 #include <memory>
 
+#include "common_types.h"
 #include "modules/audio_coding/acm2/rent_a_codec.h"
 #include "rtc_base/arraysize.h"
 #include "test/gtest.h"
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 85225c27c6..3c193a41fe 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -21,7 +21,6 @@
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "modules/audio_coding/neteq/include/neteq.h"
-#include "modules/include/module.h"
 #include "rtc_base/deprecation.h"
 #include "rtc_base/function_view.h"
 #include "system_wrappers/include/clock.h"
@@ -278,9 +277,7 @@ class AudioCodingModule {
   //
   // Input:
   //   -audio_frame        : the input audio frame, containing raw audio
-  //                         sampling frequency etc.,
-  //                         c.f. module_common_types.h for definition of
-  //                         AudioFrame.
+  //                         sampling frequency etc.
   //
   // Return value:
   //   >= 0   number of bytes encoded.
@@ -663,9 +660,7 @@ class AudioCodingModule {
   //
   // Output:
   //   -audio_frame        : output audio frame which contains raw audio data
-  //                         and other relevant parameters, c.f.
-  //                         module_common_types.h for the definition of
-  //                         AudioFrame.
+  //                         and other relevant parameters.
   //   -muted              : if true, the sample data in audio_frame is not
   //                         populated, and must be interpreted as all zero.
   //
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index ad71ef1f97..85a6bf932f 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -13,7 +13,6 @@
 
 #include <map>
 
-#include "modules/include/module_common_types.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc
index 966d5c376d..6ab27168bd 100644
--- a/modules/audio_coding/neteq/decision_logic.cc
+++ b/modules/audio_coding/neteq/decision_logic.cc
@@ -19,6 +19,7 @@
 #include "modules/audio_coding/neteq/expand.h"
 #include "modules/audio_coding/neteq/packet_buffer.h"
 #include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/modules/audio_coding/neteq/decision_logic_normal.cc b/modules/audio_coding/neteq/decision_logic_normal.cc
index 10f501ac9a..1429bb7d13 100644
--- a/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -20,7 +20,6 @@
 #include "modules/audio_coding/neteq/expand.h"
 #include "modules/audio_coding/neteq/packet_buffer.h"
 #include "modules/audio_coding/neteq/sync_buffer.h"
-#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/modules/audio_coding/neteq/nack_tracker.cc b/modules/audio_coding/neteq/nack_tracker.cc
index d187883916..c62cdf88e4 100644
--- a/modules/audio_coding/neteq/nack_tracker.cc
+++ b/modules/audio_coding/neteq/nack_tracker.cc
@@ -14,7 +14,6 @@
 
 #include <algorithm>  // For std::max.
 
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
diff --git a/modules/audio_coding/neteq/nack_tracker.h b/modules/audio_coding/neteq/nack_tracker.h
index 4f88d917c2..66383ce088 100644
--- a/modules/audio_coding/neteq/nack_tracker.h
+++ b/modules/audio_coding/neteq/nack_tracker.h
@@ -15,6 +15,7 @@
 #include <map>
 
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/include/module_common_types.h"
 #include "rtc_base/gtest_prod_util.h"
 
 //
diff --git a/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index ec166277f8..03f5aa3cc0 100644
--- a/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -12,13 +12,13 @@
 
 #include <memory>
 
+#include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "modules/include/module_common_types.h"
 #include "test/gmock.h"
 #include "test/testsupport/fileutils.h"
 
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 6ce6a12637..742683c73c 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -41,7 +41,6 @@
 #include "modules/audio_coding/neteq/sync_buffer.h"
 #include "modules/audio_coding/neteq/tick_timer.h"
 #include "modules/audio_coding/neteq/timestamp_scaler.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 3b7070f9f7..5e58453ad6 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "api/optional.h"
+#include "api/audio/audio_frame.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/defines.h"
 #include "modules/audio_coding/neteq/expand_uma_logger.h"
@@ -24,7 +25,6 @@
 #include "modules/audio_coding/neteq/rtcp.h"
 #include "modules/audio_coding/neteq/statistics_calculator.h"
 #include "modules/audio_coding/neteq/tick_timer.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/thread_annotations.h"
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index e163992a60..de24cda0f2 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -27,7 +27,6 @@
 #include "modules/audio_coding/neteq/preemptive_expand.h"
 #include "modules/audio_coding/neteq/sync_buffer.h"
 #include "modules/audio_coding/neteq/timestamp_scaler.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index 334715f8d1..b3170995c7 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -10,10 +10,10 @@
 
 #include <memory>
 
+#include "api/audio/audio_frame.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "modules/include/module_common_types.h"
 #include "test/gmock.h"
 
 namespace webrtc {
diff --git a/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index 1bef9c83a2..49facdd1ec 100644
--- a/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -15,13 +15,13 @@
 #include <string>
 #include <list>
 
+#include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "modules/audio_coding/neteq/include/neteq.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "modules/include/module_common_types.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index ca93cf5533..430ebdb9bf 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -20,12 +20,12 @@
 #include <string>
 #include <vector>
 
+#include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "modules/audio_coding/neteq/tools/audio_loop.h"
 #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/ignore_wundef.h"
 #include "rtc_base/messagedigest.h"
 #include "rtc_base/numerics/safe_conversions.h"
diff --git a/modules/audio_coding/neteq/rtcp.cc b/modules/audio_coding/neteq/rtcp.cc
index 2885398584..551eb5f75b 100644
--- a/modules/audio_coding/neteq/rtcp.cc
+++ b/modules/audio_coding/neteq/rtcp.cc
@@ -15,8 +15,6 @@
 
 #include <algorithm>
 
-#include "modules/include/module_common_types.h"
-
 namespace webrtc {
 
 void Rtcp::Init(uint16_t start_sequence_number) {
diff --git a/modules/audio_coding/neteq/sync_buffer.h b/modules/audio_coding/neteq/sync_buffer.h
index ab9ff525ef..d880356163 100644
--- a/modules/audio_coding/neteq/sync_buffer.h
+++ b/modules/audio_coding/neteq/sync_buffer.h
@@ -11,8 +11,8 @@
 #ifndef MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
 #define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
 
+#include "api/audio/audio_frame.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/constructormagic.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
diff --git a/modules/audio_coding/neteq/tools/audio_sink.h b/modules/audio_coding/neteq/tools/audio_sink.h
index ecec51b895..18ac6fcd9f 100644
--- a/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/modules/audio_coding/neteq/tools/audio_sink.h
@@ -11,7 +11,7 @@
 #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
 
-#include "modules/include/module_common_types.h"
+#include "api/audio/audio_frame.h"
 #include "rtc_base/constructormagic.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.h b/modules/audio_coding/neteq/tools/encode_neteq_input.h
index b44d4ac08c..13b39b3f50 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.h
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.h
@@ -15,7 +15,6 @@
 
 #include "api/audio_codecs/audio_encoder.h"
 #include "modules/audio_coding/neteq/tools/neteq_input.h"
-#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
index 882f82321c..ba0b2174f3 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
@@ -17,6 +17,7 @@
 #include <limits>
 #include <utility>
 
+#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
diff --git a/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 68dde5296f..2c23e5c90b 100644
--- a/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -11,6 +11,7 @@
 
 #include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 
+#include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "rtc_base/format_macros.h"
 #include "test/gtest.h"
diff --git a/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index aefa62e4d1..b8670a3f68 100644
--- a/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -17,7 +17,6 @@
 #include "api/audio_codecs/audio_decoder.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/neteq/include/neteq.h"
-#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 27ecdf4679..80aa809132 100644
--- a/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -10,13 +10,13 @@
 
 #include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
 
+#include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "modules/audio_coding/neteq/include/neteq.h"
 #include "modules/audio_coding/neteq/tools/audio_loop.h"
 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "system_wrappers/include/clock.h"
 #include "test/testsupport/fileutils.h"
@@ -103,7 +103,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
         return -1;
       payload_len = WebRtcPcm16b_Encode(input_samples.data(),
                                         input_samples.size(), input_payload);
-      assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+      RTC_DCHECK_EQ(payload_len, kInputBlockSizeSamples * sizeof(int16_t));
     }
 
     // Get output audio, but don't do anything with it.
@@ -113,8 +113,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
     if (error != NetEq::kOK)
       return -1;
 
-    assert(out_frame.samples_per_channel_ ==
-           static_cast<size_t>(kSampRateHz * 10 / 1000));
+    RTC_DCHECK_EQ(out_frame.samples_per_channel_, (kSampRateHz * 10) / 1000);
 
     static const int kOutputBlockSizeMs = 10;
     time_now_ms += kOutputBlockSizeMs;
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.h b/modules/audio_coding/neteq/tools/neteq_quality_test.h
index 531a0805ab..2b82b0aec8 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -19,7 +19,6 @@
 #include "modules/audio_coding/neteq/tools/audio_sink.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/flags.h"
 #include "test/gtest.h"
 #include "typedefs.h"  // NOLINT(build/include)
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 1620ebde6c..fb937cd707 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -32,7 +32,6 @@
 #include "modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "modules/audio_coding/neteq/tools/output_wav_file.h"
 #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/flags.h"
 #include "test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/neteq/tools/packet.cc b/modules/audio_coding/neteq/tools/packet.cc
index 71337b6e68..9505a2953c 100644
--- a/modules/audio_coding/neteq/tools/packet.cc
+++ b/modules/audio_coding/neteq/tools/packet.cc
@@ -14,7 +14,6 @@
 
 #include <memory>
 
-#include "modules/include/module_common_types.h"
 #include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "rtc_base/checks.h"
 
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index ce0719980f..66e7a285ad 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -21,6 +21,7 @@
 #include <map>
 #include <string>
 
+#include "api/audio/audio_frame.h"
 #include "api/audio_codecs/L16/audio_encoder_L16.h"
 #include "api/audio_codecs/g711/audio_encoder_g711.h"
 #include "api/audio_codecs/g722/audio_encoder_g722.h"
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index 5351a2a4bf..3855bc7d38 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -18,6 +18,7 @@
 #include "modules/audio_coding/test/ACMTest.h"
 #include "modules/audio_coding/test/PCMFile.h"
 #include "modules/audio_coding/test/RTPFile.h"
+#include "modules/include/module_common_types.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
diff --git a/modules/audio_coding/test/PCMFile.cc b/modules/audio_coding/test/PCMFile.cc
index e9e9430a33..bdb46eb5a8 100644
--- a/modules/audio_coding/test/PCMFile.cc
+++ b/modules/audio_coding/test/PCMFile.cc
@@ -14,7 +14,6 @@
 #include <stdio.h>
 #include <string.h>
 
-#include "modules/include/module_common_types.h"
 #include "test/gtest.h"
 
 namespace webrtc {
diff --git a/modules/audio_coding/test/PCMFile.h b/modules/audio_coding/test/PCMFile.h
index bbf2571980..84386dc945 100644
--- a/modules/audio_coding/test/PCMFile.h
+++ b/modules/audio_coding/test/PCMFile.h
@@ -16,8 +16,8 @@
 
 #include <string>
 
+#include "api/audio/audio_frame.h"
 #include "api/optional.h"
-#include "modules/include/module_common_types.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
diff --git a/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
index 6ea5354154..8cc5bd9f99 100644
--- a/modules/audio_coding/test/RTPFile.cc
+++ b/modules/audio_coding/test/RTPFile.cc
@@ -19,8 +19,7 @@
 #   include <arpa/inet.h>
 #endif
 
-#include "audio_coding_module.h"
-#include "rtc_base/synchronization/rw_lock_wrapper.h"
+#include "modules/include/module_common_types.h"
 // TODO(tlegrand): Consider removing usage of gtest.
 #include "test/gtest.h"
 #include "typedefs.h"  // NOLINT(build/include)
diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h
index d7e8c26606..b9afe2f1c4 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -15,7 +15,6 @@
 #include <queue>
 
 #include "modules/audio_coding/include/audio_coding_module.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/synchronization/rw_lock_wrapper.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
diff --git a/modules/audio_coding/test/insert_packet_with_timing.cc b/modules/audio_coding/test/insert_packet_with_timing.cc
index d34fa20d77..2fe52a1ccf 100644
--- a/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -19,7 +19,6 @@
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_coding/test/Channel.h"
 #include "modules/audio_coding/test/PCMFile.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/flags.h"
 #include "system_wrappers/include/clock.h"
 #include "test/gtest.h"
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 575dd706b5..89bf34f43b 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -10,6 +10,7 @@
 
 #include <memory>
 
+#include "api/audio/audio_frame.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index bfa095b788..d1d3159dac 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -177,7 +177,6 @@ rtc_source_set("audio_device_impl") {
     ":audio_device_api",
     ":audio_device_buffer",
     ":audio_device_generic",
-    "..:module_api",
     "../..:webrtc_common",
     "../../:typedefs",
     "../../api:array_view",
diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn
index fb45c4a3bb..b8294073e0 100644
--- a/modules/audio_mixer/BUILD.gn
+++ b/modules/audio_mixer/BUILD.gn
@@ -36,7 +36,6 @@ rtc_static_library("audio_mixer_impl") {
 
   deps = [
     ":audio_frame_manipulator",
-    "..:module_api",
     "../..:webrtc_common",
     "../../:typedefs",
     "../../api:array_view",
@@ -67,7 +66,7 @@ rtc_static_library("audio_frame_manipulator") {
   ]
 
   deps = [
-    "..:module_api",
+    "../../api/audio:audio_frame_api",
     "../../audio/utility:audio_frame_operations",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
@@ -91,8 +90,8 @@ if (rtc_include_tests) {
     deps = [
       ":audio_frame_manipulator",
       ":audio_mixer_impl",
-      "..:module_api",
       "../../api:array_view",
+      "../../api/audio:audio_frame_api",
       "../../api/audio:audio_mixer_api",
       "../../audio/utility:audio_frame_operations",
       "../../rtc_base:checks",
diff --git a/modules/audio_mixer/audio_frame_manipulator.cc b/modules/audio_mixer/audio_frame_manipulator.cc
index 0f5a83fed7..92526a7d09 100644
--- a/modules/audio_mixer/audio_frame_manipulator.cc
+++ b/modules/audio_mixer/audio_frame_manipulator.cc
@@ -10,7 +10,6 @@
 
 #include "modules/audio_mixer/audio_frame_manipulator.h"
 #include "audio/utility/audio_frame_operations.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
diff --git a/modules/audio_mixer/audio_frame_manipulator.h b/modules/audio_mixer/audio_frame_manipulator.h
index fe87169676..dc24caea5a 100644
--- a/modules/audio_mixer/audio_frame_manipulator.h
+++ b/modules/audio_mixer/audio_frame_manipulator.h
@@ -11,7 +11,7 @@
 #ifndef MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
 #define MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
 
-#include "modules/include/module_common_types.h"
+#include "api/audio/audio_frame.h"
 
 namespace webrtc {
 
diff --git a/modules/audio_mixer/audio_frame_manipulator_unittest.cc b/modules/audio_mixer/audio_frame_manipulator_unittest.cc
index b3be883ce0..36148212ea 100644
--- a/modules/audio_mixer/audio_frame_manipulator_unittest.cc
+++ b/modules/audio_mixer/audio_frame_manipulator_unittest.cc
@@ -11,7 +11,6 @@
 #include <algorithm>
 
 #include "modules/audio_mixer/audio_frame_manipulator.h"
-#include "modules/include/module_common_types.h"
 #include "test/gtest.h"
 
 namespace webrtc {
diff --git a/modules/audio_mixer/audio_mixer_impl.h b/modules/audio_mixer/audio_mixer_impl.h
index cb74ad7b27..86fcc1bf1e 100644
--- a/modules/audio_mixer/audio_mixer_impl.h
+++ b/modules/audio_mixer/audio_mixer_impl.h
@@ -18,7 +18,6 @@
 #include "modules/audio_mixer/frame_combiner.h"
 #include "modules/audio_mixer/output_rate_calculator.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/scoped_ref_ptr.h"
 #include "rtc_base/thread_annotations.h"
diff --git a/modules/audio_mixer/frame_combiner.h b/modules/audio_mixer/frame_combiner.h
index 14257d27dd..2b77d6eb44 100644
--- a/modules/audio_mixer/frame_combiner.h
+++ b/modules/audio_mixer/frame_combiner.h
@@ -16,7 +16,6 @@
 
 #include "modules/audio_processing/agc2/fixed_gain_controller.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 class ApmDataDumper;
diff --git a/modules/audio_mixer/sine_wave_generator.h b/modules/audio_mixer/sine_wave_generator.h
index b5c4674e20..4a50c0e88e 100644
--- a/modules/audio_mixer/sine_wave_generator.h
+++ b/modules/audio_mixer/sine_wave_generator.h
@@ -11,7 +11,7 @@
 #ifndef MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
 #define MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
 
-#include "modules/include/module_common_types.h"
+#include "api/audio/audio_frame.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 95761c3d33..a3b83e34f3 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -122,12 +122,12 @@ rtc_static_library("audio_processing") {
     ":audio_generator_interface",
     ":audio_processing_c",
     ":audio_processing_statistics",
-    "..:module_api",
     "../..:typedefs",
     "../..:webrtc_common",
     "../../api:array_view",
     "../../api:optional",
     "../../api/audio:aec3_config",
+    "../../api/audio:audio_frame_api",
     "../../api/audio:echo_control",
     "../../audio/utility:audio_frame_operations",
     "../../common_audio:common_audio_c",
@@ -214,6 +214,7 @@ rtc_source_set("aec_dump_interface") {
   deps = [
     ":audio_frame_view",
     "../../api:array_view",
+    "../../api/audio:audio_frame_api",
     "../../rtc_base:rtc_base_approved",
   ]
 }
@@ -526,7 +527,6 @@ if (rtc_include_tests) {
       ":audioproc_test_utils",
       ":file_audio_generator_unittests",
       ":mocks",
-      "..:module_api",
       "../..:typedefs",
       "../..:webrtc_common",
       "../../api:array_view",
@@ -581,6 +581,7 @@ if (rtc_include_tests) {
         ":audioproc_protobuf_utils",
         ":audioproc_test_utils",
         ":audioproc_unittest_proto",
+        "../../api/audio:audio_frame_api",
         "../../rtc_base:rtc_task_queue",
         "aec_dump",
         "aec_dump:aec_dump_unittests",
@@ -629,7 +630,6 @@ if (rtc_include_tests) {
       ":audio_processing",
       ":audioproc_test_utils",
       "../../api:array_view",
-      "../../modules:module_api",
       "../../rtc_base:protobuf_utils",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
@@ -668,8 +668,9 @@ if (rtc_include_tests) {
     ]
     deps = [
       "../../api:array_view",
+      "../../api:optional",
+      "../../api/audio:audio_frame_api",
       "../../common_audio:common_audio",
-      "../../modules:module_api",
       "../../rtc_base:checks",
       "../../rtc_base:rtc_base_approved",
     ]
@@ -743,9 +744,9 @@ if (rtc_include_tests) {
 
     deps = [
       ":audio_processing",
-      "..:module_api",
       "../../api:array_view",
       "../../api:optional",
+      "../../api/audio:audio_frame_api",
       "../../common_audio",
       "../../rtc_base:checks",
       "../../rtc_base:rtc_base_approved",
@@ -766,7 +767,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":audio_processing",
-      "..:module_api",
       "../..:typedefs",
       "../..:webrtc_common",
       "../../common_audio:common_audio",
diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn
index 4e9af38c73..152c290cbe 100644
--- a/modules/audio_processing/aec_dump/BUILD.gn
+++ b/modules/audio_processing/aec_dump/BUILD.gn
@@ -29,7 +29,6 @@ rtc_source_set("mock_aec_dump") {
 
   deps = [
     "..:aec_dump_interface",
-    "../..:module_api",
     "../../../test:test_support",
   ]
 }
@@ -63,7 +62,7 @@ if (rtc_enable_protobuf) {
     deps = [
       ":aec_dump",
       "..:aec_dump_interface",
-      "../../../modules:module_api",
+      "../../../api/audio:audio_frame_api",
       "../../../rtc_base:checks",
       "../../../rtc_base:protobuf_utils",
       "../../../rtc_base:rtc_base_approved",
@@ -83,7 +82,6 @@ if (rtc_enable_protobuf) {
       ":aec_dump_impl",
       "..:aec_dump_interface",
       "..:audioproc_debug_proto",
-      "../../../modules:module_api",
       "../../../rtc_base:rtc_task_queue",
       "../../../test:fileutils",
       "../../../test:test_support",
diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.h b/modules/audio_processing/aec_dump/aec_dump_impl.h
index 8ff3398b2f..0d88a7ed96 100644
--- a/modules/audio_processing/aec_dump/aec_dump_impl.h
+++ b/modules/audio_processing/aec_dump/aec_dump_impl.h
@@ -15,10 +15,10 @@
 #include <string>
 #include <vector>
 
+#include "api/audio/audio_frame.h"
 #include "modules/audio_processing/aec_dump/capture_stream_info.h"
 #include "modules/audio_processing/aec_dump/write_to_file_task.h"
 #include "modules/audio_processing/include/aec_dump.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/ignore_wundef.h"
 #include "rtc_base/platform_file.h"
 #include "rtc_base/race_checker.h"
diff --git a/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/modules/audio_processing/aec_dump/aec_dump_unittest.cc
index 965ac03bd7..98640b9824 100644
--- a/modules/audio_processing/aec_dump/aec_dump_unittest.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_unittest.cc
@@ -12,7 +12,6 @@
 
 #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
 
-#include "modules/include/module_common_types.h"
 #include "rtc_base/task_queue.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
diff --git a/modules/audio_processing/aec_dump/capture_stream_info.h b/modules/audio_processing/aec_dump/capture_stream_info.h
index 91bb1faf5b..da8fb58895 100644
--- a/modules/audio_processing/aec_dump/capture_stream_info.h
+++ b/modules/audio_processing/aec_dump/capture_stream_info.h
@@ -15,9 +15,9 @@
 #include <utility>
 #include <vector>
 
+#include "api/audio/audio_frame.h"
 #include "modules/audio_processing/aec_dump/write_to_file_task.h"
 #include "modules/audio_processing/include/aec_dump.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/ignore_wundef.h"
 #include "rtc_base/logging.h"
diff --git a/modules/audio_processing/aec_dump/mock_aec_dump.h b/modules/audio_processing/aec_dump/mock_aec_dump.h
index 8cfabdd648..c2088c0f4c 100644
--- a/modules/audio_processing/aec_dump/mock_aec_dump.h
+++ b/modules/audio_processing/aec_dump/mock_aec_dump.h
@@ -14,7 +14,6 @@
 #include <memory>
 
 #include "modules/audio_processing/include/aec_dump.h"
-#include "modules/include/module_common_types.h"
 #include "test/gmock.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/agc/agc.cc b/modules/audio_processing/agc/agc.cc
index e1616765b1..12c8cfb7fb 100644
--- a/modules/audio_processing/agc/agc.cc
+++ b/modules/audio_processing/agc/agc.cc
@@ -18,7 +18,6 @@
 
 #include "modules/audio_processing/agc/loudness_histogram.h"
 #include "modules/audio_processing/agc/utility.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index 5ba5f4ff48..2d6ee81521 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -18,7 +18,6 @@
 
 #include "modules/audio_processing/agc/gain_map_internal.h"
 #include "modules/audio_processing/gain_control_impl.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_minmax.h"
diff --git a/modules/audio_processing/agc/loudness_histogram.cc b/modules/audio_processing/agc/loudness_histogram.cc
index 63d5f7cad0..0ed5850e72 100644
--- a/modules/audio_processing/agc/loudness_histogram.cc
+++ b/modules/audio_processing/agc/loudness_histogram.cc
@@ -13,7 +13,6 @@
 #include <cmath>
 #include <cstring>
 
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/agc/mock_agc.h b/modules/audio_processing/agc/mock_agc.h
index b27d28cd89..cf2a859d61 100644
--- a/modules/audio_processing/agc/mock_agc.h
+++ b/modules/audio_processing/agc/mock_agc.h
@@ -13,7 +13,6 @@
 
 #include "modules/audio_processing/agc/agc.h"
 
-#include "modules/include/module_common_types.h"
 #include "test/gmock.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index 8451bdeeaa..508f96f651 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -14,10 +14,10 @@
 #include <memory>
 #include <vector>
 
+#include "api/audio/audio_frame.h"
 #include "common_audio/channel_buffer.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/splitting_filter.h"
-#include "modules/include/module_common_types.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 27388ead27..2e3689561a 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -43,7 +43,6 @@
 #include "modules/audio_processing/residual_echo_detector.h"
 #include "modules/audio_processing/transient/transient_suppressor.h"
 #include "modules/audio_processing/voice_detection_impl.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/atomicops.h"
 #include "rtc_base/system/file_wrapper.h"
 #include "system_wrappers/include/metrics.h"
@@ -1375,8 +1374,8 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
   processing_config.reverse_output_stream() = output_config;
 
   RETURN_ON_ERR(MaybeInitializeRender(processing_config));
-  assert(input_config.num_frames() ==
-         formats_.api_format.reverse_input_stream().num_frames());
+  RTC_DCHECK_EQ(input_config.num_frames(),
+                formats_.api_format.reverse_input_stream().num_frames());
 
   if (aec_dump_) {
     const size_t channel_size =
diff --git a/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index d4cff4582b..39f8b8b4c6 100644
--- a/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -16,7 +16,6 @@
 
 #include "api/array_view.h"
 #include "modules/audio_processing/test/test_utils.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/event.h"
 #include "rtc_base/platform_thread.h"
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index e152befc5c..08ac6233d5 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -11,7 +11,6 @@
 #include "modules/audio_processing/audio_processing_impl.h"
 
 #include "modules/audio_processing/test/test_utils.h"
-#include "modules/include/module_common_types.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
diff --git a/modules/audio_processing/audio_processing_performance_unittest.cc b/modules/audio_processing/audio_processing_performance_unittest.cc
index 8dd81b2a58..d3121370e7 100644
--- a/modules/audio_processing/audio_processing_performance_unittest.cc
+++ b/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -17,7 +17,6 @@
 
 #include "api/array_view.h"
 #include "modules/audio_processing/test/test_utils.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/atomicops.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/platform_thread.h"
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 89d6cb9ee3..6993da59b1 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -27,7 +27,6 @@
 #include "modules/audio_processing/include/mock_audio_processing.h"
 #include "modules/audio_processing/test/protobuf_utils.h"
 #include "modules/audio_processing/test/test_utils.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/gtest_prod_util.h"
diff --git a/modules/audio_processing/include/aec_dump.h b/modules/audio_processing/include/aec_dump.h
index 2035bf4742..d4d45692b8 100644
--- a/modules/audio_processing/include/aec_dump.h
+++ b/modules/audio_processing/include/aec_dump.h
@@ -16,12 +16,11 @@
 #include <vector>
 
 #include "api/array_view.h"
+#include "api/audio/audio_frame.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
 
 namespace webrtc {
 
-class AudioFrame;
-
 // Struct for passing current config from APM without having to
 // include protobuf headers.
 struct InternalAPMConfig {
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index 65669aba4c..e1c67633e6 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -14,6 +14,7 @@
 
 #include "modules/audio_processing/test/protobuf_utils.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
 
 namespace webrtc {
 namespace test {
diff --git a/modules/audio_processing/test/fake_recording_device.cc b/modules/audio_processing/test/fake_recording_device.cc
index aee3dcef61..3260ec1e61 100644
--- a/modules/audio_processing/test/fake_recording_device.cc
+++ b/modules/audio_processing/test/fake_recording_device.cc
@@ -12,6 +12,7 @@
 
 #include <algorithm>
 
+#include "api/optional.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/ptr_util.h"
 
diff --git a/modules/audio_processing/test/fake_recording_device.h b/modules/audio_processing/test/fake_recording_device.h
index b1e37a331d..0d93b7a249 100644
--- a/modules/audio_processing/test/fake_recording_device.h
+++ b/modules/audio_processing/test/fake_recording_device.h
@@ -15,9 +15,9 @@
 #include <memory>
 #include <vector>
 
+#include "api/audio/audio_frame.h"
 #include "api/array_view.h"
 #include "common_audio/channel_buffer.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/test/test_utils.h b/modules/audio_processing/test/test_utils.h
index 57dc7b3797..81684f210b 100644
--- a/modules/audio_processing/test/test_utils.h
+++ b/modules/audio_processing/test/test_utils.h
@@ -18,10 +18,10 @@
 #include <string>
 #include <vector>
 
+#include "api/audio/audio_frame.h"
 #include "common_audio/channel_buffer.h"
 #include "common_audio/wav_file.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/constructormagic.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/transient/transient_suppression_test.cc b/modules/audio_processing/transient/transient_suppression_test.cc
index 14fe4f8267..8512e0128a 100644
--- a/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/modules/audio_processing/transient/transient_suppression_test.cc
@@ -19,7 +19,6 @@
 
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/agc/agc.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/flags.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
diff --git a/modules/audio_processing/typing_detection.h b/modules/audio_processing/typing_detection.h
index fe74a5956e..14dfe1d0ea 100644
--- a/modules/audio_processing/typing_detection.h
+++ b/modules/audio_processing/typing_detection.h
@@ -11,7 +11,6 @@
 #ifndef MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 #define MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 
-#include "modules/include/module_common_types.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
diff --git a/modules/audio_processing/vad/BUILD.gn b/modules/audio_processing/vad/BUILD.gn
index 43329cb5bb..f9f59745ea 100644
--- a/modules/audio_processing/vad/BUILD.gn
+++ b/modules/audio_processing/vad/BUILD.gn
@@ -35,7 +35,6 @@ rtc_static_library("vad") {
     "voice_gmm_tables.h",
   ]
   deps = [
-    "../..:module_api",
     "../../..:typedefs",
     "../../../audio/utility:audio_frame_operations",
     "../../../common_audio",
@@ -70,7 +69,6 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":vad",
-      "../..:module_api",
       "../../../common_audio",
       "../../../test:fileutils",
       "../../../test:test_support",
diff --git a/modules/audio_processing/vad/pitch_based_vad.cc b/modules/audio_processing/vad/pitch_based_vad.cc
index bca2552c35..240ec63f29 100644
--- a/modules/audio_processing/vad/pitch_based_vad.cc
+++ b/modules/audio_processing/vad/pitch_based_vad.cc
@@ -17,7 +17,6 @@
 #include "modules/audio_processing/vad/common.h"
 #include "modules/audio_processing/vad/noise_gmm_tables.h"
 #include "modules/audio_processing/vad/voice_gmm_tables.h"
-#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/modules/audio_processing/vad/standalone_vad.cc b/modules/audio_processing/vad/standalone_vad.cc
index 004cefebb8..f7ae449709 100644
--- a/modules/audio_processing/vad/standalone_vad.cc
+++ b/modules/audio_processing/vad/standalone_vad.cc
@@ -11,7 +11,6 @@
 #include "modules/audio_processing/vad/standalone_vad.h"
 
 #include "audio/utility/audio_frame_operations.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 #include "typedefs.h"  // NOLINT(build/include)
 
diff --git a/modules/audio_processing/vad/standalone_vad_unittest.cc b/modules/audio_processing/vad/standalone_vad_unittest.cc
index 28d1349396..512276ecb5 100644
--- a/modules/audio_processing/vad/standalone_vad_unittest.cc
+++ b/modules/audio_processing/vad/standalone_vad_unittest.cc
@@ -14,7 +14,6 @@
 
 #include <memory>
 
-#include "modules/include/module_common_types.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 
diff --git a/modules/audio_processing/vad/vad_audio_proc.cc b/modules/audio_processing/vad/vad_audio_proc.cc
index b1841d0b21..98fcf1912d 100644
--- a/modules/audio_processing/vad/vad_audio_proc.cc
+++ b/modules/audio_processing/vad/vad_audio_proc.cc
@@ -24,7 +24,6 @@ extern "C" {
 #include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
 #include "modules/audio_coding/codecs/isac/main/source/structs.h"
 }
-#include "modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/modules/audio_processing/vad/vad_audio_proc_unittest.cc
index c520257f8e..5b96be624d 100644
--- a/modules/audio_processing/vad/vad_audio_proc_unittest.cc
+++ b/modules/audio_processing/vad/vad_audio_proc_unittest.cc
@@ -20,7 +20,6 @@
 #include <string>
 
 #include "modules/audio_processing/vad/common.h"
-#include "modules/include/module_common_types.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 
diff --git a/modules/module_common_types_unittest.cc b/modules/module_common_types_unittest.cc
index c8bb5f9850..3045c0dc6a 100644
--- a/modules/module_common_types_unittest.cc
+++ b/modules/module_common_types_unittest.cc
@@ -10,108 +10,10 @@
 
 #include "modules/include/module_common_types.h"
 
-#include <string.h>  // memcmp
-
 #include "test/gtest.h"
 
 namespace webrtc {
 
-namespace {
-
-bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
-  const int16_t* frame_data = frame.data();
-  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
-    if (frame_data[i] != sample) {
-      return false;
-    }
-  }
-  return true;
-}
-
-constexpr uint32_t kTimestamp = 27;
-constexpr int kSampleRateHz = 16000;
-constexpr size_t kNumChannels = 1;
-constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
-
-}  // namespace
-
-TEST(AudioFrameTest, FrameStartsMuted) {
-  AudioFrame frame;
-  EXPECT_TRUE(frame.muted());
-  EXPECT_TRUE(AllSamplesAre(0, frame));
-}
-
-TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
-  AudioFrame frame;
-  frame.mutable_data();
-  EXPECT_FALSE(frame.muted());
-  EXPECT_TRUE(AllSamplesAre(0, frame));
-}
-
-TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
-  AudioFrame frame;
-  int16_t* frame_data = frame.mutable_data();
-  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
-    frame_data[i] = 17;
-  }
-  ASSERT_TRUE(AllSamplesAre(17, frame));
-  frame.Mute();
-  EXPECT_TRUE(frame.muted());
-  EXPECT_TRUE(AllSamplesAre(0, frame));
-}
-
-TEST(AudioFrameTest, UpdateFrame) {
-  AudioFrame frame;
-  int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
-  frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
-                    AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
-
-  EXPECT_EQ(kTimestamp, frame.timestamp_);
-  EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel_);
-  EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz_);
-  EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_);
-  EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);
-  EXPECT_EQ(kNumChannels, frame.num_channels_);
-
-  EXPECT_FALSE(frame.muted());
-  EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
-
-  frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
-                    kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
-                    kNumChannels);
-  EXPECT_TRUE(frame.muted());
-  EXPECT_TRUE(AllSamplesAre(0, frame));
-}
-
-TEST(AudioFrameTest, CopyFrom) {
-  AudioFrame frame1;
-  AudioFrame frame2;
-
-  int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
-  frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel,
-                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
-                     kNumChannels);
-  frame1.CopyFrom(frame2);
-
-  EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
-  EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
-  EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
-  EXPECT_EQ(frame2.speech_type_, frame1.speech_type_);
-  EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
-  EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
-
-  EXPECT_EQ(frame2.muted(), frame1.muted());
-  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
-
-  frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
-                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
-                     kNumChannels);
-  frame1.CopyFrom(frame2);
-
-  EXPECT_EQ(frame2.muted(), frame1.muted());
-  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
-}
-
 TEST(IsNewerSequenceNumber, Equal) {
   EXPECT_FALSE(IsNewerSequenceNumber(0x0001, 0x0001));
 }
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index f57ef40fdb..bc0648d334 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -290,7 +290,7 @@ if (rtc_include_tests) {
     }
 
     deps = [
-      "../modules:module_api",
+      "../api/audio:audio_frame_api",
       "../modules/audio_processing",
       "../modules/audio_processing/vad",
       "../rtc_base:rtc_base_approved",
diff --git a/rtc_tools/agc/activity_metric.cc b/rtc_tools/agc/activity_metric.cc
index 3c65d02b4a..ed48543b4a 100644
--- a/rtc_tools/agc/activity_metric.cc
+++ b/rtc_tools/agc/activity_metric.cc
@@ -16,6 +16,7 @@
 #include <algorithm>
 #include <memory>
 
+#include "api/audio/audio_frame.h"
 #include "modules/audio_processing/agc/agc.h"
 #include "modules/audio_processing/agc/loudness_histogram.h"
 #include "modules/audio_processing/agc/utility.h"
@@ -23,7 +24,6 @@
 #include "modules/audio_processing/vad/pitch_based_vad.h"
 #include "modules/audio_processing/vad/standalone_vad.h"
 #include "modules/audio_processing/vad/vad_audio_proc.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/flags.h"
 #include "rtc_base/numerics/safe_minmax.h"
 #include "test/gtest.h"
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 639c54b121..6160dfcb65 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -430,7 +430,7 @@ rtc_static_library("audio_processing_fuzzer_helper") {
   deps = [
     ":fuzz_data_helper",
     "../../api:optional",
-    "../../modules:module_api",
+    "../../api/audio:audio_frame_api",
     "../../modules/audio_processing",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
diff --git a/test/fuzzers/audio_processing_fuzzer_helper.cc b/test/fuzzers/audio_processing_fuzzer_helper.cc
index 868532632f..b07e177539 100644
--- a/test/fuzzers/audio_processing_fuzzer_helper.cc
+++ b/test/fuzzers/audio_processing_fuzzer_helper.cc
@@ -15,8 +15,8 @@
 #include <cmath>
 #include <limits>
 
+#include "api/audio/audio_frame.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "modules/include/module_common_types.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {