Move ChannelBuffer class to channel_buffer file

No change in functionallity.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 0 insertions, 76 deletions

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_