diff --git a/common_audio/resampler/sinc_resampler.cc b/common_audio/resampler/sinc_resampler.cc
index acf7d0f..2f17ded 100644
--- a/common_audio/resampler/sinc_resampler.cc
+++ b/common_audio/resampler/sinc_resampler.cc
@@ -160,12 +160,12 @@
           AlignedMalloc(sizeof(float) * kKernelStorageSize, 16))),
       input_buffer_(static_cast<float*>(
           AlignedMalloc(sizeof(float) * input_buffer_size_, 16))),
-#if defined(WEBRTC_CPU_DETECTION)
+#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__))
       convolve_proc_(nullptr),
 #endif
       r1_(input_buffer_.get()),
       r2_(input_buffer_.get() + kKernelSize / 2) {
-#if defined(WEBRTC_CPU_DETECTION)
+#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__))
   InitializeCPUSpecificFeatures();
   RTC_DCHECK(convolve_proc_);
 #endif
diff --git a/common_audio/resampler/sinc_resampler.h b/common_audio/resampler/sinc_resampler.h
index fa92163..50059e5 100644
--- a/common_audio/resampler/sinc_resampler.h
+++ b/common_audio/resampler/sinc_resampler.h
@@ -149,7 +149,7 @@
   // TODO(ajm): Move to using a global static which must only be initialized
   // once by the user. We're not doing this initially, because we don't have
   // e.g. a LazyInstance helper in webrtc.
-#if defined(WEBRTC_CPU_DETECTION)
+#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__))
   typedef float (*ConvolveProc)(const float*, const float*, const float*,
                                 double);
   ConvolveProc convolve_proc_;
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 7858e84..f4ace1b 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -935,7 +935,7 @@
           static_cast<uint32_t>(output_size_samples_));
       // Skipping break on purpose. Execution should move on into the
       // next case.
-      FALLTHROUGH();
+      RTC_FALLTHROUGH();
     }
     case kAudioRepetition: {
       // TODO(hlundin): Write test for this.
diff --git a/modules/audio_device/ios/voice_processing_audio_unit.mm b/modules/audio_device/ios/voice_processing_audio_unit.mm
index 3701f3f..591a176 100644
--- a/modules/audio_device/ios/voice_processing_audio_unit.mm
+++ b/modules/audio_device/ios/voice_processing_audio_unit.mm
@@ -434,12 +434,12 @@
       case kStarted:
         Stop();
         // Fall through.
-        FALLTHROUGH();
+        RTC_FALLTHROUGH();
       case kInitialized:
         Uninitialize();
         break;
       case kUninitialized:
-        FALLTHROUGH();
+        RTC_FALLTHROUGH();
       case kInitRequired:
         break;
     }
diff --git a/modules/remote_bitrate_estimator/test/bwe.cc b/modules/remote_bitrate_estimator/test/bwe.cc
index abc301c..566f93b 100644
--- a/modules/remote_bitrate_estimator/test/bwe.cc
+++ b/modules/remote_bitrate_estimator/test/bwe.cc
@@ -97,7 +97,7 @@
     case kBbrEstimator:
       return new BbrBweSender(observer, clock);
     case kTcpEstimator:
-      FALLTHROUGH();
+      RTC_FALLTHROUGH();
     case kNullEstimator:
       return new NullBweSender();
   }
diff --git a/modules/rtp_rtcp/source/rtp_format_h264.cc b/modules/rtp_rtcp/source/rtp_format_h264.cc
index 5aa7e27..eb33bcc 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -574,7 +574,7 @@
       }
       case H264::NaluType::kIdr:
         parsed_payload->frame_type = kVideoFrameKey;
-        FALLTHROUGH();
+        RTC_FALLTHROUGH();
       case H264::NaluType::kSlice: {
         rtc::Optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
             &payload_data[start_offset], end_offset - start_offset);
diff --git a/modules/video_coding/jitter_buffer.cc b/modules/video_coding/jitter_buffer.cc
index a556080..eaf09be 100644
--- a/modules/video_coding/jitter_buffer.cc
+++ b/modules/video_coding/jitter_buffer.cc
@@ -749,7 +749,7 @@
           frame_event_->Set();
         }
       }
-      FALLTHROUGH();
+      RTC_FALLTHROUGH();
     }
     // Note: There is no break here - continuing to kDecodableSession.
     case kDecodableSession: {
diff --git a/modules/video_coding/rtp_frame_reference_finder.cc b/modules/video_coding/rtp_frame_reference_finder.cc
index e167e15..82af339 100644
--- a/modules/video_coding/rtp_frame_reference_finder.cc
+++ b/modules/video_coding/rtp_frame_reference_finder.cc
@@ -71,7 +71,7 @@
         case kHandOff:
           complete_frame = true;
           frame_callback_->OnCompleteFrame(std::move(*frame_it));
-          FALLTHROUGH();
+          RTC_FALLTHROUGH();
         case kDrop:
           frame_it = stashed_frames_.erase(frame_it);
       }
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 902024f..76de9c0 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -98,6 +98,8 @@
   all_dependent_configs = [ ":rtc_base_approved_all_dependent_config" ]
 
   sources = [
+    "annotations.h",
+    "arch.h",
     "array_view.h",
     "arraysize.h",
     "atomicops.h",
diff --git a/rtc_base/annotations.h b/rtc_base/annotations.h
new file mode 100644
index 0000000..25dac33
--- /dev/null
+++ b/rtc_base/annotations.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#ifndef WEBRTC_RTC_BASE_ANNOTATIONS_H_
+#define WEBRTC_RTC_BASE_ANNOTATIONS_H_
+
+// Annotate a function indicating the caller must examine the return value.
+// Use like:
+//   int foo() RTC_WARN_UNUSED_RESULT;
+// To explicitly ignore a result, cast to void.
+// TODO(kwiberg): Remove when we can use [[nodiscard]] from C++17.
+#if defined(__clang__)
+#define RTC_WARN_UNUSED_RESULT __attribute__((__warn_unused_result__))
+#elif defined(__GNUC__)
+// gcc has a __warn_unused_result__ attribute, but you can't quiet it by
+// casting to void, so we don't use it.
+#define RTC_WARN_UNUSED_RESULT
+#else
+#define RTC_WARN_UNUSED_RESULT
+#endif
+
+// Macro to be used for switch-case fallthrough (required for enabling
+// -Wimplicit-fallthrough warning on Clang).
+#if defined(__clang__)
+#define RTC_FALLTHROUGH() [[clang::fallthrough]]
+#else
+#define RTC_FALLTHROUGH() do { } while (0)
+#endif
+
+// Prevent the compiler from warning about an unused variable. For example:
+//   int result = DoSomething();
+//   assert(result == 17);
+//   RTC_UNUSED(result);
+// Note: In most cases it is better to remove the unused variable rather than
+// suppressing the compiler warning.
+#define RTC_UNUSED(x) static_cast<void>(x)
+
+#endif  // WEBRTC_RTC_BASE_ANNOTATIONS_H_
diff --git a/rtc_base/arch.h b/rtc_base/arch.h
new file mode 100644
index 0000000..0137503
--- /dev/null
+++ b/rtc_base/arch.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+// This file contains platform-specific defines.
+// Much of it is derived from Chromium's build/build_config.h.
+
+#ifndef WEBRTC_RTC_BASE_ARCH_H_
+#define WEBRTC_RTC_BASE_ARCH_H_
+
+// Processor architecture detection.  For more info on what's defined, see:
+//   http://msdn.microsoft.com/en-us/library/b0084kay.aspx
+//   http://www.agner.org/optimize/calling_conventions.pdf
+//   or with gcc, run: "echo | gcc -E -dM -"
+#if defined(_M_X64) || defined(__x86_64__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86_64
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__aarch64__)
+#define WEBRTC_ARCH_ARM_FAMILY
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(_M_IX86) || defined(__i386__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__ARMEL__)
+#define WEBRTC_ARCH_ARM_FAMILY
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__MIPSEL__)
+#define WEBRTC_ARCH_MIPS_FAMILY
+#if defined(__LP64__)
+#define WEBRTC_ARCH_64_BITS
+#else
+#define WEBRTC_ARCH_32_BITS
+#endif
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__pnacl__)
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#else
+#error Please add support for your architecture in rtc_base/arch.h
+#endif
+
+#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
+#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
+#endif
+
+#endif  // WEBRTC_RTC_BASE_ARCH_H_
diff --git a/rtc_base/checks.cc b/rtc_base/checks.cc
index 533240f..ee0bbb8 100644
--- a/rtc_base/checks.cc
+++ b/rtc_base/checks.cc
@@ -100,7 +100,7 @@
   delete result;
 }
 
-NO_RETURN FatalMessage::~FatalMessage() {
+RTC_NO_RETURN FatalMessage::~FatalMessage() {
   fflush(stdout);
   fflush(stderr);
   stream_ << std::endl << "#" << std::endl;
@@ -136,6 +136,8 @@
 }  // namespace rtc
 
 // Function to call from the C version of the RTC_CHECK and RTC_DCHECK macros.
-NO_RETURN void rtc_FatalMessage(const char* file, int line, const char* msg) {
+RTC_NO_RETURN void rtc_FatalMessage(const char* file,
+                                    int line,
+                                    const char* msg) {
   rtc::FatalMessage(file, line).stream() << msg;
 }
diff --git a/rtc_base/checks.h b/rtc_base/checks.h
index f9507e2..d1a3ccb 100644
--- a/rtc_base/checks.h
+++ b/rtc_base/checks.h
@@ -11,8 +11,6 @@
 #ifndef WEBRTC_RTC_BASE_CHECKS_H_
 #define WEBRTC_RTC_BASE_CHECKS_H_
 
-#include "webrtc/typedefs.h"
-
 // If you for some reson need to know if DCHECKs are on, test the value of
 // RTC_DCHECK_IS_ON. (Test its value, not if it's defined; it'll always be
 // defined, to either a true or a false value.)
@@ -22,10 +20,20 @@
 #define RTC_DCHECK_IS_ON 0
 #endif
 
+// Annotate a function that will not return control flow to the caller.
+#if defined(_MSC_VER)
+#define RTC_NO_RETURN __declspec(noreturn)
+#elif defined(__GNUC__)
+#define RTC_NO_RETURN __attribute__ ((__noreturn__))
+#else
+#define RTC_NO_RETURN
+#endif
+
 #ifdef __cplusplus
 extern "C" {
 #endif
-NO_RETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
+RTC_NO_RETURN
+    void rtc_FatalMessage(const char* file, int line, const char* msg);
 #ifdef __cplusplus
 }  // extern "C"
 #endif
@@ -232,7 +240,7 @@
   FatalMessage(const char* file, int line);
   // Used for RTC_CHECK_EQ(), etc. Takes ownership of the given string.
   FatalMessage(const char* file, int line, std::string* result);
-  NO_RETURN ~FatalMessage();
+  RTC_NO_RETURN ~FatalMessage();
 
   std::ostream& stream() { return stream_; }
 
diff --git a/rtc_base/httpbase.cc b/rtc_base/httpbase.cc
index 64cb5d9..cb7bac1 100644
--- a/rtc_base/httpbase.cc
+++ b/rtc_base/httpbase.cc
@@ -537,7 +537,7 @@
       case SR_EOS:
         // Clean close, with no error.
         read_error = 0;
-        FALLTHROUGH();  // Fall through to HandleStreamClose.
+        RTC_FALLTHROUGH();  // Fall through to HandleStreamClose.
       case SR_ERROR:
         *error = HandleStreamClose(read_error);
         return true;
diff --git a/typedefs.h b/typedefs.h
index bd0bee7..84a7c08 100644
--- a/typedefs.h
+++ b/typedefs.h
@@ -8,118 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// This file contains platform-specific typedefs and defines.
-// Much of it is derived from Chromium's build/build_config.h.
-
 #ifndef WEBRTC_TYPEDEFS_H_
 #define WEBRTC_TYPEDEFS_H_
 
-// Processor architecture detection.  For more info on what's defined, see:
-//   http://msdn.microsoft.com/en-us/library/b0084kay.aspx
-//   http://www.agner.org/optimize/calling_conventions.pdf
-//   or with gcc, run: "echo | gcc -E -dM -"
-#if defined(_M_X64) || defined(__x86_64__)
-#define WEBRTC_ARCH_X86_FAMILY
-#define WEBRTC_ARCH_X86_64
-#define WEBRTC_ARCH_64_BITS
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#elif defined(__aarch64__)
-#define WEBRTC_ARCH_ARM_FAMILY
-#define WEBRTC_ARCH_64_BITS
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#elif defined(_M_IX86) || defined(__i386__)
-#define WEBRTC_ARCH_X86_FAMILY
-#define WEBRTC_ARCH_X86
-#define WEBRTC_ARCH_32_BITS
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#elif defined(__ARMEL__)
-#define WEBRTC_ARCH_ARM_FAMILY
-#define WEBRTC_ARCH_32_BITS
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#elif defined(__MIPSEL__)
-#define WEBRTC_ARCH_MIPS_FAMILY
-#if defined(__LP64__)
-#define WEBRTC_ARCH_64_BITS
-#else
-#define WEBRTC_ARCH_32_BITS
-#endif
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#elif defined(__pnacl__)
-#define WEBRTC_ARCH_32_BITS
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#else
-#error Please add support for your architecture in typedefs.h
-#endif
-
-#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
-#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
-#endif
-
-// TODO(zhongwei.yao): WEBRTC_CPU_DETECTION is only used in one place; we should
-// probably just remove it.
-#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__))
-#define WEBRTC_CPU_DETECTION
-#endif
+// TODO(solenberg): Delete this file once downstream projects have been updated.
 
 #include <stdint.h>
 
-// Annotate a function indicating the caller must examine the return value.
-// Use like:
-//   int foo() RTC_WARN_UNUSED_RESULT;
-// To explicitly ignore a result, cast to void.
-// TODO(kwiberg): Remove when we can use [[nodiscard]] from C++17.
-#if defined(__clang__)
-#define RTC_WARN_UNUSED_RESULT __attribute__((__warn_unused_result__))
-#elif defined(__GNUC__)
-// gcc has a __warn_unused_result__ attribute, but you can't quiet it by
-// casting to void, so we don't use it.
-#define RTC_WARN_UNUSED_RESULT
-#else
-#define RTC_WARN_UNUSED_RESULT
-#endif
-
-// Put after a variable that might not be used, to prevent compiler warnings:
-//   int result ATTRIBUTE_UNUSED = DoSomething();
-//   assert(result == 17);
-// Deprecated since it only works with GCC & clang. See RTC_UNUSED below.
-// TODO(terelius): Remove.
-#ifndef ATTRIBUTE_UNUSED
-#if defined(__GNUC__) || defined(__clang__)
-#define ATTRIBUTE_UNUSED __attribute__ ((__unused__))
-#else
-#define ATTRIBUTE_UNUSED
-#endif
-#endif
-
-// Macro to be used for switch-case fallthrough (required for enabling
-// -Wimplicit-fallthrough warning on Clang).
-#ifndef FALLTHROUGH
-#if defined(__clang__)
-#define FALLTHROUGH() [[clang::fallthrough]]
-#else
-#define FALLTHROUGH() do { } while (0)
-#endif
-#endif
-
-#ifndef NO_RETURN
-// Annotate a function that will not return control flow to the caller.
-#if defined(_MSC_VER)
-#define NO_RETURN __declspec(noreturn)
-#elif defined(__GNUC__)
-#define NO_RETURN __attribute__ ((__noreturn__))
-#else
-#define NO_RETURN
-#endif
-#endif
-
-// Prevent the compiler from warning about an unused variable. For example:
-//   int result = DoSomething();
-//   assert(result == 17);
-//   RTC_UNUSED(result);
-// Note: In most cases it is better to remove the unused variable rather than
-// suppressing the compiler warning.
-#ifndef RTC_UNUSED
-#define RTC_UNUSED(x) static_cast<void>(x)
-#endif  // RTC_UNUSED
+#include "webrtc/rtc_base/arch.h"
+#include "webrtc/rtc_base/annotations.h"
 
 #endif  // WEBRTC_TYPEDEFS_H_
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index a47055a..500c0bc 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -268,7 +268,7 @@
     switch (tracker_.CopyAndFixBitstream(&packet)) {
       case video_coding::H264SpsPpsTracker::kRequestKeyframe:
         keyframe_request_sender_->RequestKeyFrame();
-        FALLTHROUGH();
+        RTC_FALLTHROUGH();
       case video_coding::H264SpsPpsTracker::kDrop:
         return 0;
       case video_coding::H264SpsPpsTracker::kInsert:
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index 52dcbf0..7d810dc 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -978,7 +978,7 @@
         break;
       }
       // Scale down resolution.
-      FALLTHROUGH();
+      RTC_FALLTHROUGH();
     }
     case VideoSendStream::DegradationPreference::kMaintainFramerate:
       // Scale down resolution.
@@ -1056,7 +1056,7 @@
         break;
       }
       // Scale up resolution.
-      FALLTHROUGH();
+      RTC_FALLTHROUGH();
     }
     case VideoSendStream::DegradationPreference::kMaintainFramerate: {
       // Scale up resolution.