diff --git a/webrtc/common_audio/audio_converter.h b/webrtc/common_audio/audio_converter.h
index 6365f58799..c71a3fdcc5 100644
--- a/webrtc/common_audio/audio_converter.h
+++ b/webrtc/common_audio/audio_converter.h
@@ -13,7 +13,7 @@
 
 // TODO(ajm): Move channel buffer to common_audio.
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_processing/common.h"
+#include "webrtc/modules/audio_processing/channel_buffer.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/scoped_vector.h"
 
diff --git a/webrtc/common_audio/audio_converter_unittest.cc b/webrtc/common_audio/audio_converter_unittest.cc
index 91836f9cef..3e509ceb01 100644
--- a/webrtc/common_audio/audio_converter_unittest.cc
+++ b/webrtc/common_audio/audio_converter_unittest.cc
@@ -15,7 +15,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
-#include "webrtc/modules/audio_processing/common.h"
+#include "webrtc/modules/audio_processing/channel_buffer.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
diff --git a/webrtc/common_audio/blocker.h b/webrtc/common_audio/blocker.h
index f3f99cd31e..1cf95dbc38 100644
--- a/webrtc/common_audio/blocker.h
+++ b/webrtc/common_audio/blocker.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
 #define WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
 
-#include "webrtc/modules/audio_processing/common.h"
+#include "webrtc/modules/audio_processing/channel_buffer.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index 5d542d038f..33af43e6b4 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -13,6 +13,7 @@
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/channel_buffer.h"
+#include "webrtc/modules/audio_processing/common.h"
 
 namespace webrtc {
 namespace {
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index b92e499e4c..fe2cf36c30 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/common_audio/include/audio_util.h"
-#include "webrtc/modules/audio_processing/common.h"
+#include "webrtc/modules/audio_processing/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/splitting_filter.h"
 #include "webrtc/modules/interface/module_common_types.h"
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 1b16c7c609..0c50ecb060 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -16,6 +16,7 @@
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/channel_buffer.h"
 #include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
 #include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
diff --git a/webrtc/modules/audio_processing/channel_buffer.h b/webrtc/modules/audio_processing/channel_buffer.h
index 07990e7326..6ecc07ab53 100644
--- a/webrtc/modules/audio_processing/channel_buffer.h
+++ b/webrtc/modules/audio_processing/channel_buffer.h
@@ -11,11 +11,86 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
 
+#include <string.h>
+
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/include/audio_util.h"
-#include "webrtc/modules/audio_processing/common.h"
 
 namespace webrtc {
 
+// Helper to encapsulate a contiguous data buffer with access to a pointer
+// array of the deinterleaved channels.
+template <typename T>
+class ChannelBuffer {
+ public:
+  ChannelBuffer(int samples_per_channel, int num_channels)
+      : data_(new T[samples_per_channel * num_channels]),
+        channels_(new T*[num_channels]),
+        samples_per_channel_(samples_per_channel),
+        num_channels_(num_channels) {
+    Initialize();
+  }
+
+  ChannelBuffer(const T* data, int samples_per_channel, int num_channels)
+      : data_(new T[samples_per_channel * num_channels]),
+        channels_(new T*[num_channels]),
+        samples_per_channel_(samples_per_channel),
+        num_channels_(num_channels) {
+    Initialize();
+    memcpy(data_.get(), data, length() * sizeof(T));
+  }
+
+  ChannelBuffer(const T* const* channels, int samples_per_channel,
+                int num_channels)
+      : data_(new T[samples_per_channel * num_channels]),
+        channels_(new T*[num_channels]),
+        samples_per_channel_(samples_per_channel),
+        num_channels_(num_channels) {
+    Initialize();
+    for (int i = 0; i < num_channels_; ++i)
+      CopyFrom(channels[i], i);
+  }
+
+  ~ChannelBuffer() {}
+
+  void CopyFrom(const void* channel_ptr, int i) {
+    DCHECK_LT(i, num_channels_);
+    memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
+  }
+
+  T* data() { return data_.get(); }
+  const T* data() const { return data_.get(); }
+
+  const T* channel(int i) const {
+    DCHECK_GE(i, 0);
+    DCHECK_LT(i, num_channels_);
+    return channels_[i];
+  }
+  T* channel(int i) {
+    const ChannelBuffer<T>* t = this;
+    return const_cast<T*>(t->channel(i));
+  }
+
+  T* const* channels() { return channels_.get(); }
+  const T* const* channels() const { return channels_.get(); }
+
+  int samples_per_channel() const { return samples_per_channel_; }
+  int num_channels() const { return num_channels_; }
+  int length() const { return samples_per_channel_ * num_channels_; }
+
+ private:
+  void Initialize() {
+    memset(data_.get(), 0, sizeof(T) * length());
+    for (int i = 0; i < num_channels_; ++i)
+      channels_[i] = &data_[i * samples_per_channel_];
+  }
+
+  scoped_ptr<T[]> data_;
+  scoped_ptr<T*[]> channels_;
+  const int samples_per_channel_;
+  const int num_channels_;
+};
+
 // One int16_t and one float ChannelBuffer that are kept in sync. The sync is
 // broken when someone requests write access to either ChannelBuffer, and
 // reestablished when someone requests the outdated ChannelBuffer. It is
diff --git a/webrtc/modules/audio_processing/common.h b/webrtc/modules/audio_processing/common.h
index 98c624127c..ed8a0544c3 100644
--- a/webrtc/modules/audio_processing/common.h
+++ b/webrtc/modules/audio_processing/common.h
@@ -12,11 +12,8 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
 
 #include <assert.h>
-#include <string.h>
 
-#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -33,79 +30,6 @@ static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
   return -1;
 }
 
-// Helper to encapsulate a contiguous data buffer with access to a pointer
-// array of the deinterleaved channels.
-template <typename T>
-class ChannelBuffer {
- public:
-  ChannelBuffer(int samples_per_channel, int num_channels)
-      : data_(new T[samples_per_channel * num_channels]),
-        channels_(new T*[num_channels]),
-        samples_per_channel_(samples_per_channel),
-        num_channels_(num_channels) {
-    Initialize();
-  }
-
-  ChannelBuffer(const T* data, int samples_per_channel, int num_channels)
-      : data_(new T[samples_per_channel * num_channels]),
-        channels_(new T*[num_channels]),
-        samples_per_channel_(samples_per_channel),
-        num_channels_(num_channels) {
-    Initialize();
-    memcpy(data_.get(), data, length() * sizeof(T));
-  }
-
-  ChannelBuffer(const T* const* channels, int samples_per_channel,
-                int num_channels)
-      : data_(new T[samples_per_channel * num_channels]),
-        channels_(new T*[num_channels]),
-        samples_per_channel_(samples_per_channel),
-        num_channels_(num_channels) {
-    Initialize();
-    for (int i = 0; i < num_channels_; ++i)
-      CopyFrom(channels[i], i);
-  }
-
-  ~ChannelBuffer() {}
-
-  void CopyFrom(const void* channel_ptr, int i) {
-    DCHECK_LT(i, num_channels_);
-    memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
-  }
-
-  T* data() { return data_.get(); }
-  const T* data() const { return data_.get(); }
-
-  const T* channel(int i) const {
-    DCHECK_GE(i, 0);
-    DCHECK_LT(i, num_channels_);
-    return channels_[i];
-  }
-  T* channel(int i) {
-    const ChannelBuffer<T>* t = this;
-    return const_cast<T*>(t->channel(i));
-  }
-
-  T* const* channels() { return channels_.get(); }
-  const T* const* channels() const { return channels_.get(); }
-
-  int samples_per_channel() const { return samples_per_channel_; }
-  int num_channels() const { return num_channels_; }
-  int length() const { return samples_per_channel_ * num_channels_; }
-
- private:
-  void Initialize() {
-    memset(data_.get(), 0, sizeof(T) * length());
-    for (int i = 0; i < num_channels_; ++i)
-      channels_[i] = &data_[i * samples_per_channel_];
-  }
-
-  scoped_ptr<T[]> data_;
-  scoped_ptr<T*[]> channels_;
-  const int samples_per_channel_;
-  const int num_channels_;
-};
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index 401391aa2b..78e79a5eb9 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -18,6 +18,7 @@
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/modules/interface/module_common_types.h"
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index d0d08cb8cf..2243590bb1 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -14,7 +14,7 @@
 #include "webrtc/audio_processing/debug.pb.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/wav_file.h"
-#include "webrtc/modules/audio_processing/common.h"
+#include "webrtc/modules/audio_processing/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"