Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/neteq/tools/audio_loop.cc

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2013 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_coding/neteq/tools/audio_loop.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <string.h>
+
+namespace webrtc {
+namespace test {
+
+bool AudioLoop::Init(const std::string file_name,
+                     size_t max_loop_length_samples,
+                     size_t block_length_samples) {
+  FILE* fp = fopen(file_name.c_str(), "rb");
+  if (!fp) return false;
+
+  audio_array_.reset(new int16_t[max_loop_length_samples +
+                                 block_length_samples]);
+  size_t samples_read = fread(audio_array_.get(), sizeof(int16_t),
+                              max_loop_length_samples, fp);
+  fclose(fp);
+
+  // Block length must be shorter than the loop length.
+  if (block_length_samples > samples_read) return false;
+
+  // Add an extra block length of samples to the end of the array, starting
+  // over again from the beginning of the array. This is done to simplify
+  // the reading process when reading over the end of the loop.
+  memcpy(&audio_array_[samples_read], audio_array_.get(),
+         block_length_samples * sizeof(int16_t));
+
+  loop_length_samples_ = samples_read;
+  block_length_samples_ = block_length_samples;
+  next_index_ = 0;
+  return true;
+}
+
+rtc::ArrayView<const int16_t> AudioLoop::GetNextBlock() {
+  // Check that the AudioLoop is initialized.
+  if (block_length_samples_ == 0)
+    return rtc::ArrayView<const int16_t>();
+
+  const int16_t* output_ptr = &audio_array_[next_index_];
+  next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
+  return rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
+}
+
+
+}  // namespace test
+}  // namespace webrtc