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/performance_timer.cc

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2016 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/performance_timer.h"
+
+#include <math.h>
+
+#include <numeric>
+
+#include "webrtc/rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+PerformanceTimer::PerformanceTimer(int num_frames_to_process)
+    : clock_(webrtc::Clock::GetRealTimeClock()) {
+  timestamps_us_.reserve(num_frames_to_process);
+}
+
+PerformanceTimer::~PerformanceTimer() = default;
+
+void PerformanceTimer::StartTimer() {
+  start_timestamp_us_ = rtc::Optional<int64_t>(clock_->TimeInMicroseconds());
+}
+
+void PerformanceTimer::StopTimer() {
+  RTC_DCHECK(start_timestamp_us_);
+  timestamps_us_.push_back(clock_->TimeInMicroseconds() - *start_timestamp_us_);
+}
+
+double PerformanceTimer::GetDurationAverage() const {
+  return GetDurationAverage(0);
+}
+
+double PerformanceTimer::GetDurationStandardDeviation() const {
+  return GetDurationStandardDeviation(0);
+}
+
+double PerformanceTimer::GetDurationAverage(
+    size_t number_of_warmup_samples) const {
+  RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples);
+  const size_t number_of_samples =
+      timestamps_us_.size() - number_of_warmup_samples;
+  return static_cast<double>(
+             std::accumulate(timestamps_us_.begin() + number_of_warmup_samples,
+                             timestamps_us_.end(), static_cast<int64_t>(0))) /
+         number_of_samples;
+}
+
+double PerformanceTimer::GetDurationStandardDeviation(
+    size_t number_of_warmup_samples) const {
+  RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples);
+  const size_t number_of_samples =
+      timestamps_us_.size() - number_of_warmup_samples;
+  RTC_DCHECK_GT(number_of_samples, 0);
+  double average_duration = GetDurationAverage(number_of_warmup_samples);
+
+  double variance = std::accumulate(
+      timestamps_us_.begin() + number_of_warmup_samples, timestamps_us_.end(),
+      0.0, [average_duration](const double& a, const int64_t& b) {
+        return a + (b - average_duration) * (b - average_duration);
+      });
+
+  return sqrt(variance / number_of_samples);
+}
+
+}  // namespace test
+}  // namespace webrtc