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+/*
+ *  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_BLOCKER_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
+
+#include <memory>
+
+#include "common_audio/channel_buffer.h"
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+namespace webrtc {
+
+// The callback function to process audio in the time domain. Input has already
+// been windowed, and output will be windowed. The number of input channels
+// must be >= the number of output channels.
+class BlockerCallback {
+ public:
+  virtual ~BlockerCallback() {}
+
+  virtual void ProcessBlock(const float* const* input,
+                            size_t num_frames,
+                            size_t num_input_channels,
+                            size_t num_output_channels,
+                            float* const* output) = 0;
+};
+
+// The main purpose of Blocker is to abstract away the fact that often we
+// receive a different number of audio frames than our transform takes. For
+// example, most FFTs work best when the fft-size is a power of 2, but suppose
+// we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames
+// of audio, which is not a power of 2. Blocker allows us to specify the
+// transform and all other necessary processing via the Process() callback
+// function without any constraints on the transform-size
+// (read: `block_size_`) or received-audio-size (read: `chunk_size_`).
+// We handle this for the multichannel audio case, allowing for different
+// numbers of input and output channels (for example, beamforming takes 2 or
+// more input channels and returns 1 output channel). Audio signals are
+// represented as deinterleaved floats in the range [-1, 1].
+//
+// Blocker is responsible for:
+// - blocking audio while handling potential discontinuities on the edges
+//   of chunks
+// - windowing blocks before sending them to Process()
+// - windowing processed blocks, and overlap-adding them together before
+//   sending back a processed chunk
+//
+// To use blocker:
+// 1. Impelment a BlockerCallback object `bc`.
+// 2. Instantiate a Blocker object `b`, passing in `bc`.
+// 3. As you receive audio, call b.ProcessChunk() to get processed audio.
+//
+// A small amount of delay is added to the first received chunk to deal with
+// the difference in chunk/block sizes. This delay is <= chunk_size.
+//
+// Ownership of window is retained by the caller.  That is, Blocker makes a
+// copy of window and does not attempt to delete it.
+class Blocker {
+ public:
+  Blocker(size_t chunk_size,
+          size_t block_size,
+          size_t num_input_channels,
+          size_t num_output_channels,
+          const float* window,
+          size_t shift_amount,
+          BlockerCallback* callback);
+  ~Blocker();
+
+  void ProcessChunk(const float* const* input,
+                    size_t chunk_size,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output);
+
+  size_t initial_delay() const { return initial_delay_; }
+
+ private:
+  const size_t chunk_size_;
+  const size_t block_size_;
+  const size_t num_input_channels_;
+  const size_t num_output_channels_;
+
+  // The number of frames of delay to add at the beginning of the first chunk.
+  const size_t initial_delay_;
+
+  // The frame index into the input buffer where the first block should be read
+  // from. This is necessary because shift_amount_ is not necessarily a
+  // multiple of chunk_size_, so blocks won't line up at the start of the
+  // buffer.
+  size_t frame_offset_;
+
+  // Since blocks nearly always overlap, there are certain blocks that require
+  // frames from the end of one chunk and the beginning of the next chunk. The
+  // input and output buffers are responsible for saving those frames between
+  // calls to ProcessChunk().
+  //
+  // Both contain |initial delay| + `chunk_size` frames. The input is a fairly
+  // standard FIFO, but due to the overlap-add it's harder to use an
+  // AudioRingBuffer for the output.
+  AudioRingBuffer input_buffer_;
+  ChannelBuffer<float> output_buffer_;
+
+  // Space for the input block (can't wrap because of windowing).
+  ChannelBuffer<float> input_block_;
+
+  // Space for the output block (can't wrap because of overlap/add).
+  ChannelBuffer<float> output_block_;
+
+  std::unique_ptr<float[]> window_;
+
+  // The amount of frames between the start of contiguous blocks. For example,
+  // `shift_amount_` = `block_size_` / 2 for a Hann window.
+  size_t shift_amount_;
+
+  BlockerCallback* callback_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_