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_processing/test/audio_buffer_tools.cc

@@ -0,0 +1,55 @@
+/*
+ *  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/test/audio_buffer_tools.h"
+
+#include <string.h>
+
+namespace webrtc {
+namespace test {
+
+void SetupFrame(const StreamConfig& stream_config,
+                std::vector<float*>* frame,
+                std::vector<float>* frame_samples) {
+  frame_samples->resize(stream_config.num_channels() *
+                        stream_config.num_frames());
+  frame->resize(stream_config.num_channels());
+  for (size_t ch = 0; ch < stream_config.num_channels(); ++ch) {
+    (*frame)[ch] = &(*frame_samples)[ch * stream_config.num_frames()];
+  }
+}
+
+void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
+                             rtc::ArrayView<const float> source,
+                             AudioBuffer* destination) {
+  std::vector<float*> input;
+  std::vector<float> input_samples;
+
+  SetupFrame(stream_config, &input, &input_samples);
+
+  RTC_CHECK_EQ(input_samples.size(), source.size());
+  memcpy(input_samples.data(), source.data(),
+         source.size() * sizeof(source[0]));
+
+  destination->CopyFrom(&input[0], stream_config);
+}
+
+void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config,
+                                  AudioBuffer* source,
+                                  std::vector<float>* destination) {
+  std::vector<float*> output;
+
+  SetupFrame(stream_config, &output, destination);
+
+  source->CopyTo(stream_config, &output[0]);
+}
+
+}  // namespace test
+}  // namespace webrtc