diff --git a/webrtc/common_audio/include/audio_util.h b/webrtc/common_audio/include/audio_util.h
index 9972a0e067..0ce034be34 100644
--- a/webrtc/common_audio/include/audio_util.h
+++ b/webrtc/common_audio/include/audio_util.h
@@ -62,7 +62,7 @@ void ScaleToFloat(const int16_t* src, size_t size, float* dest);
 // per buffer).
 template <typename T>
 void Deinterleave(const T* interleaved, int samples_per_channel,
-                  int num_channels, T** deinterleaved) {
+                  int num_channels, T* const* deinterleaved) {
   for (int i = 0; i < num_channels; ++i) {
     T* channel = deinterleaved[i];
     int interleaved_idx = i;
diff --git a/webrtc/modules/audio_processing/common.h b/webrtc/modules/audio_processing/common.h
index 98e36cb098..c1bae896ce 100644
--- a/webrtc/modules/audio_processing/common.h
+++ b/webrtc/modules/audio_processing/common.h
@@ -14,6 +14,7 @@
 #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"
 
@@ -42,37 +43,66 @@ class ChannelBuffer {
         channels_(new T*[num_channels]),
         samples_per_channel_(samples_per_channel),
         num_channels_(num_channels) {
-    memset(data_.get(), 0, sizeof(T) * samples_per_channel * num_channels);
-    for (int i = 0; i < num_channels; ++i)
-      channels_[i] = &data_[i * samples_per_channel];
+    SetChannelPtrs();
   }
+
+  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) {
+    SetChannelPtrs();
+    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) {
+    SetChannelPtrs();
+    for (int i = 0; i < num_channels_; ++i)
+      CopyFrom(channels[i], i);
+  }
+
   ~ChannelBuffer() {}
 
   void CopyFrom(const void* channel_ptr, int i) {
-    assert(i < num_channels_);
+    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 {
-    assert(i >= 0 && i < num_channels_);
+    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** channels() { return channels_.get(); }
 
-  int samples_per_channel() { return samples_per_channel_; }
-  int num_channels() { return num_channels_; }
-  int length() { return samples_per_channel_ * num_channels_; }
+  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 SetChannelPtrs() {
+    for (int i = 0; i < num_channels_; ++i)
+      channels_[i] = &data_[i * samples_per_channel_];
+  }
+
   scoped_ptr<T[]> data_;
   scoped_ptr<T*[]> channels_;
-  int samples_per_channel_;
-  int num_channels_;
+  const int samples_per_channel_;
+  const int num_channels_;
 };
 
 }  // namespace webrtc