1 files changed, 175 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h
new file mode 100644
index 0000000000..bb25c34a9e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h
@@ -0,0 +1,175 @@
+/*
+ *  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 MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
+
+#include <complex>
+#include <memory>
+
+#include "common_audio/real_fourier.h"
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+#include "rtc_base/memory/aligned_malloc.h"
+
+namespace webrtc {
+
+// Wrapper class for aligned arrays. Every row (and the first dimension) are
+// aligned to the given byte alignment.
+template <typename T>
+class AlignedArray {
+ public:
+  AlignedArray(size_t rows, size_t cols, size_t alignment)
+      : rows_(rows), cols_(cols) {
+    RTC_CHECK_GT(alignment, 0);
+    head_row_ =
+        static_cast<T**>(AlignedMalloc(rows_ * sizeof(*head_row_), alignment));
+    for (size_t i = 0; i < rows_; ++i) {
+      head_row_[i] = static_cast<T*>(
+          AlignedMalloc(cols_ * sizeof(**head_row_), alignment));
+    }
+  }
+
+  ~AlignedArray() {
+    for (size_t i = 0; i < rows_; ++i) {
+      AlignedFree(head_row_[i]);
+    }
+    AlignedFree(head_row_);
+  }
+
+  T* const* Array() { return head_row_; }
+
+  const T* const* Array() const { return head_row_; }
+
+  T* Row(size_t row) {
+    RTC_CHECK_LE(row, rows_);
+    return head_row_[row];
+  }
+
+  const T* Row(size_t row) const {
+    RTC_CHECK_LE(row, rows_);
+    return head_row_[row];
+  }
+
+ private:
+  size_t rows_;
+  size_t cols_;
+  T** head_row_;
+};
+
+// 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,
+                                   size_t num_in_channels,
+                                   size_t frames,
+                                   size_t num_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); `window` should have length equal to `block_length`.
+  // `block_length` defines the length of a block, in samples.
+  // `shift_amount` is in samples. `callback` is the caller-owned audio
+  // processing function called for each block of the input chunk.
+  LappedTransform(size_t num_in_channels,
+                  size_t num_out_channels,
+                  size_t chunk_length,
+                  const float* window,
+                  size_t block_length,
+                  size_t 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);
+
+  // Get the chunk length.
+  //
+  // The chunk length is the number of samples per channel that must be passed
+  // to ProcessChunk via the parameter in_chunk.
+  //
+  // Returns the same chunk_length passed to the LappedTransform constructor.
+  size_t chunk_length() const { return chunk_length_; }
+
+  // Get the number of input channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // in_chunk.
+  //
+  // Returns the same num_in_channels passed to the LappedTransform constructor.
+  size_t num_in_channels() const { return num_in_channels_; }
+
+  // Get the number of output channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // out_chunk.
+  //
+  // Returns the same num_out_channels passed to the LappedTransform
+  // constructor.
+  size_t num_out_channels() const { return num_out_channels_; }
+
+  // Returns the initial delay.
+  //
+  // This is the delay introduced by the `blocker_` to be able to get and return
+  // chunks of `chunk_length`, but process blocks of `block_length`.
+  size_t initial_delay() const { return blocker_.initial_delay(); }
+
+ private:
+  // Internal middleware callback, given to the blocker. Transforms each block
+  // and hands it over to the processing method given at construction time.
+  class BlockThunk : public BlockerCallback {
+   public:
+    explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
+
+    void ProcessBlock(const float* const* input,
+                      size_t num_frames,
+                      size_t num_input_channels,
+                      size_t num_output_channels,
+                      float* const* output) override;
+
+   private:
+    LappedTransform* const parent_;
+  } blocker_callback_;
+
+  const size_t num_in_channels_;
+  const size_t num_out_channels_;
+
+  const size_t block_length_;
+  const size_t chunk_length_;
+
+  Callback* const block_processor_;
+  Blocker blocker_;
+
+  // TODO(alessiob): Replace RealFourier with a different FFT library.
+  std::unique_ptr<RealFourier> fft_;
+  const size_t cplx_length_;
+  AlignedArray<float> real_buf_;
+  AlignedArray<std::complex<float> > cplx_pre_;
+  AlignedArray<std::complex<float> > cplx_post_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_