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

webrtc/modules/audio_processing/utility/audio_ring_buffer.cc

Index: webrtc/modules/audio_processing/utility/audio_ring_buffer.cc
diff --git a/webrtc/modules/audio_processing/utility/audio_ring_buffer.cc b/webrtc/modules/audio_processing/utility/audio_ring_buffer.cc
deleted file mode 100644
index 73f578f58b9508a6525460333182ca69eb6dd435..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_processing/utility/audio_ring_buffer.cc
+++ /dev/null
@@ -1,75 +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 "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
-
-// This is a simple multi-channel wrapper over the ring_buffer.h C interface.
-
-namespace webrtc {
-
-AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) {
-  buffers_.reserve(channels);
-  for (size_t i = 0; i < channels; ++i)
-    buffers_.push_back(WebRtc_CreateBuffer(max_frames, sizeof(float)));
-}
-
-AudioRingBuffer::~AudioRingBuffer() {
-  for (auto buf : buffers_)
-    WebRtc_FreeBuffer(buf);
-}
-
-void AudioRingBuffer::Write(const float* const* data, size_t channels,
-                            size_t frames) {
-  RTC_DCHECK_EQ(buffers_.size(), channels);
-  for (size_t i = 0; i < channels; ++i) {
-    const size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames);
-    RTC_CHECK_EQ(written, frames);
-  }
-}
-
-void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) {
-  RTC_DCHECK_EQ(buffers_.size(), channels);
-  for (size_t i = 0; i < channels; ++i) {
-    const size_t read =
-        WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames);
-    RTC_CHECK_EQ(read, frames);
-  }
-}
-
-size_t AudioRingBuffer::ReadFramesAvailable() const {
-  // All buffers have the same amount available.
-  return WebRtc_available_read(buffers_[0]);
-}
-
-size_t AudioRingBuffer::WriteFramesAvailable() const {
-  // All buffers have the same amount available.
-  return WebRtc_available_write(buffers_[0]);
-}
-
-void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
-  for (auto buf : buffers_) {
-    const size_t moved =
-        static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames)));
-    RTC_CHECK_EQ(moved, frames);
-  }
-}
-
-void AudioRingBuffer::MoveReadPositionBackward(size_t frames) {
-  for (auto buf : buffers_) {
-    const size_t moved = static_cast<size_t>(
-        -WebRtc_MoveReadPtr(buf, -static_cast<int>(frames)));
-    RTC_CHECK_EQ(moved, frames);
-  }
-}
-
-}  // namespace webrtc