Revert of Moved ring-buffer related files from common_audio to audio_processing"

webrtc/modules/audio_processing/utility/blocker.h

Index: webrtc/modules/audio_processing/utility/blocker.h
diff --git a/webrtc/modules/audio_processing/utility/blocker.h b/webrtc/modules/audio_processing/utility/blocker.h
deleted file mode 100644
index 7d9bf66e48fe40168d972e1b84858778002e79ac..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_processing/utility/blocker.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- *  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_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
-
-#include <memory>
-
-#include "webrtc/common_audio/channel_buffer.h"
-#include "webrtc/modules/audio_processing/utility/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);
-
-  void ProcessChunk(const float* const* input,
-                    size_t chunk_size,
-                    size_t num_input_channels,
-                    size_t num_output_channels,
-                    float* const* output);
-
- 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  // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_