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/transient/transient_detector_unittest.cc

@@ -0,0 +1,104 @@
+/*
+ *  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_processing/transient/transient_detector.h"
+
+#include <memory>
+#include <sstream>
+#include <string>
+
+#include "webrtc/modules/audio_processing/transient/common.h"
+#include "webrtc/modules/audio_processing/transient/file_utils.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/test/gtest.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+static const int kSampleRatesHz[] = {ts::kSampleRate8kHz,
+                                     ts::kSampleRate16kHz,
+                                     ts::kSampleRate32kHz,
+                                     ts::kSampleRate48kHz};
+static const size_t kNumberOfSampleRates =
+    sizeof(kSampleRatesHz) / sizeof(*kSampleRatesHz);
+
+// This test is for the correctness of the transient detector.
+// Checks the results comparing them with the ones stored in the detect files in
+// the directory: resources/audio_processing/transient/
+// The files contain all the results in double precision (Little endian).
+// The audio files used with different sample rates are stored in the same
+// directory.
+#if defined(WEBRTC_IOS)
+TEST(TransientDetectorTest, DISABLED_CorrectnessBasedOnFiles) {
+#else
+TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
+#endif
+  for (size_t i = 0; i < kNumberOfSampleRates; ++i) {
+    int sample_rate_hz = kSampleRatesHz[i];
+
+    // Prepare detect file.
+    std::stringstream detect_file_name;
+    detect_file_name << "audio_processing/transient/detect"
+                     << (sample_rate_hz / 1000) << "kHz";
+
+    std::unique_ptr<FileWrapper> detect_file(FileWrapper::Create());
+
+    detect_file->OpenFile(
+        test::ResourcePath(detect_file_name.str(), "dat").c_str(),
+        true);  // Read only.
+
+    bool file_opened = detect_file->is_open();
+    ASSERT_TRUE(file_opened) << "File could not be opened.\n"
+          << detect_file_name.str().c_str();
+
+    // Prepare audio file.
+    std::stringstream audio_file_name;
+    audio_file_name << "audio_processing/transient/audio"
+                    << (sample_rate_hz / 1000) << "kHz";
+
+    std::unique_ptr<FileWrapper> audio_file(FileWrapper::Create());
+
+    audio_file->OpenFile(
+        test::ResourcePath(audio_file_name.str(), "pcm").c_str(),
+        true);  // Read only.
+
+    // Create detector.
+    TransientDetector detector(sample_rate_hz);
+
+    const size_t buffer_length = sample_rate_hz * ts::kChunkSizeMs / 1000;
+    std::unique_ptr<float[]> buffer(new float[buffer_length]);
+
+    const float kTolerance = 0.02f;
+
+    size_t frames_read = 0;
+
+    while (ReadInt16FromFileToFloatBuffer(audio_file.get(),
+                                          buffer_length,
+                                          buffer.get()) == buffer_length) {
+      ++frames_read;
+
+      float detector_value =
+          detector.Detect(buffer.get(), buffer_length, NULL, 0);
+      double file_value;
+      ASSERT_EQ(1u, ReadDoubleBufferFromFile(detect_file.get(), 1, &file_value))
+          << "Detect test file is malformed.\n";
+
+      // Compare results with data from the matlab test file.
+      EXPECT_NEAR(file_value, detector_value, kTolerance) << "Frame: "
+          << frames_read;
+    }
+
+    detect_file->CloseFile();
+    audio_file->CloseFile();
+  }
+}
+
+}  // namespace webrtc