Import LappedTransform and friends.

Add code for doing block-based frequency domain processing. Developed
and reviewed in isolation. Corresponding export CL:
https://chromereviews.googleplex.com/95187013/

R=bercic@google.com, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_audio/lapped_transform.h b/webrtc/common_audio/lapped_transform.h
new file mode 100644
index 0000000..330886a
--- /dev/null
+++ b/webrtc/common_audio/lapped_transform.h
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2014 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_COMMON_AUDIO_LAPPED_TRANSFORM_H_
+#define WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
+
+#include <complex>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/blocker.h"
+#include "webrtc/common_audio/real_fourier.h"
+#include "webrtc/system_wrappers/interface/aligned_array.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+// Helper class for audio processing modules which operate on frequency domain
+// input derived from the windowed time domain audio stream.
+//
+// The input audio chunk is sliced into possibly overlapping blocks, multiplied
+// by a window and transformed with an FFT implementation. The transformed data
+// is supplied to the given callback for processing. The processed output is
+// then inverse transformed into the time domain and spliced back into a chunk
+// which constitutes the final output of this processing module.
+class LappedTransform {
+ public:
+  class Callback {
+   public:
+    virtual ~Callback() {}
+
+    virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
+                                   int in_channels, int frames,
+                                   int out_channels,
+                                   std::complex<float>* const* out_block) = 0;
+  };
+
+  // Construct a transform instance. |chunk_length| is the number of samples in
+  // each channel. |window| defines the window, owned by the caller (a copy is
+  // made internally); can be NULL to disable windowing entirely.
+  // |block_length| defines the length of a block, in samples, even when
+  // windowing is disabled. |shift_length| is in samples. |callback| is the
+  // caller-owned audio processing function called for each block of the input
+  // chunk.
+  LappedTransform(int in_channels, int out_channels, int chunk_length,
+                  const float* window, int block_length, int shift_amount,
+                  Callback* callback);
+  ~LappedTransform();
+
+  // Main audio processing helper method. Internally slices |in_chunk| into
+  // blocks, transforms them to frequency domain, calls the callback for each
+  // block and returns a de-blocked time domain chunk of audio through
+  // |out_chunk|. Both buffers are caller-owned.
+  void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
+
+ private:
+  // Internal middleware callback, given to the blocker. Transforms each block
+  // and hands it over to the processing method given at construction time.
+  friend class BlockThunk;
+  class BlockThunk : public BlockerCallback {
+   public:
+    explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
+    virtual ~BlockThunk() {}
+
+    virtual void ProcessBlock(const float* const* input, int num_frames,
+                              int num_input_channels, int num_output_channels,
+                              float* const* output);
+
+   private:
+    LappedTransform* parent_;
+  } blocker_callback_;
+
+  int in_channels_;
+  int out_channels_;
+
+  const float* window_;
+  bool own_window_;
+  int window_shift_amount_;
+
+  int block_length_;
+  int chunk_length_;
+  Callback* block_processor_;
+  scoped_ptr<Blocker> blocker_;
+
+  RealFourier fft_;
+  int cplx_length_;
+  AlignedArray<float> real_buf_;
+  AlignedArray<std::complex<float> > cplx_pre_;
+  AlignedArray<std::complex<float> > cplx_post_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
+