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

webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc

Index: webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc
diff --git a/webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc b/webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc
deleted file mode 100644
index c2d5e7a64cb84895206636c6a7c798d93597a77f..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- *  Copyright (c) 2015 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.
- */
-
-#include <memory>
-#include <tuple>
-
-#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common_audio/channel_buffer.h"
-
-namespace webrtc {
-
-class AudioRingBufferTest :
-    public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
-};
-
-void ReadAndWriteTest(const ChannelBuffer<float>& input,
-                      size_t num_write_chunk_frames,
-                      size_t num_read_chunk_frames,
-                      size_t buffer_frames,
-                      ChannelBuffer<float>* output) {
-  const size_t num_channels = input.num_channels();
-  const size_t total_frames = input.num_frames();
-  AudioRingBuffer buf(num_channels, buffer_frames);
-  std::unique_ptr<float* []> slice(new float*[num_channels]);
-
-  size_t input_pos = 0;
-  size_t output_pos = 0;
-  while (input_pos + buf.WriteFramesAvailable() < total_frames) {
-    // Write until the buffer is as full as possible.
-    while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
-      buf.Write(input.Slice(slice.get(), input_pos), num_channels,
-                num_write_chunk_frames);
-      input_pos += num_write_chunk_frames;
-    }
-    // Read until the buffer is as empty as possible.
-    while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
-      EXPECT_LT(output_pos, total_frames);
-      buf.Read(output->Slice(slice.get(), output_pos), num_channels,
-               num_read_chunk_frames);
-      output_pos += num_read_chunk_frames;
-    }
-  }
-
-  // Write and read the last bit.
-  if (input_pos < total_frames) {
-    buf.Write(input.Slice(slice.get(), input_pos), num_channels,
-              total_frames - input_pos);
-  }
-  if (buf.ReadFramesAvailable()) {
-    buf.Read(output->Slice(slice.get(), output_pos), num_channels,
-             buf.ReadFramesAvailable());
-  }
-  EXPECT_EQ(0u, buf.ReadFramesAvailable());
-}
-
-TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
-  const size_t kFrames = 5000;
-  const size_t num_channels = ::testing::get<3>(GetParam());
-
-  // Initialize the input data to an increasing sequence.
-  ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
-  for (size_t i = 0; i < num_channels; ++i)
-    for (size_t j = 0; j < kFrames; ++j)
-      input.channels()[i][j] = (i + 1) * (j + 1);
-
-  ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
-  ReadAndWriteTest(input,
-                   ::testing::get<0>(GetParam()),
-                   ::testing::get<1>(GetParam()),
-                   ::testing::get<2>(GetParam()),
-                   &output);
-
-  // Verify the read data matches the input.
-  for (size_t i = 0; i < num_channels; ++i)
-    for (size_t j = 0; j < kFrames; ++j)
-      EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
-}
-
-INSTANTIATE_TEST_CASE_P(
-    AudioRingBufferTest, AudioRingBufferTest,
-    ::testing::Combine(::testing::Values(10, 20, 42),  // num_write_chunk_frames
-                       ::testing::Values(1, 10, 17),   // num_read_chunk_frames
-                       ::testing::Values(100, 256),    // buffer_frames
-                       ::testing::Values(1, 4)));      // num_channels
-
-TEST_F(AudioRingBufferTest, MoveReadPosition) {
-  const size_t kNumChannels = 1;
-  const float kInputArray[] = {1, 2, 3, 4};
-  const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray);
-  ChannelBuffer<float> input(kNumFrames, kNumChannels);
-  input.SetDataForTesting(kInputArray, kNumFrames);
-  AudioRingBuffer buf(kNumChannels, kNumFrames);
-  buf.Write(input.channels(), kNumChannels, kNumFrames);
-
-  buf.MoveReadPositionForward(3);
-  ChannelBuffer<float> output(1, kNumChannels);
-  buf.Read(output.channels(), kNumChannels, 1);
-  EXPECT_EQ(4, output.channels()[0][0]);
-  buf.MoveReadPositionBackward(3);
-  buf.Read(output.channels(), kNumChannels, 1);
-  EXPECT_EQ(2, output.channels()[0][0]);
-}
-
-}  // namespace webrtc