Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/.

webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc

 /*
  *  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/audio_processing_impl.h"
 
 #include <algorithm>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/random.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
-#include "webrtc/test/random.h"
 
 namespace webrtc {
 
 namespace {
 
 class AudioProcessingImplLockTest;
 
 // Sleeps a random time between 0 and max_sleep milliseconds.
-void SleepRandomMs(int max_sleep, test::Random* rand_gen) {
+void SleepRandomMs(int max_sleep, Random* rand_gen) {
   int sleeptime = rand_gen->Rand(0, max_sleep);
   SleepMs(sleeptime);
 }
 
 // Populates a float audio frame with random data.
 void PopulateAudioFrame(float** frame,
                         float amplitude,
                         size_t num_channels,
                         size_t samples_per_channel,
-                        test::Random* rand_gen) {
+                        Random* rand_gen) {
   for (size_t ch = 0; ch < num_channels; ch++) {
     for (size_t k = 0; k < samples_per_channel; k++) {
       // Store random 16 bit quantized float number between +-amplitude.
       frame[ch][k] = amplitude * (2 * rand_gen->Rand<float>() - 1);
     }
   }
 }
 
 // Populates an audioframe frame of AudioFrame type with random data.
 void PopulateAudioFrame(AudioFrame* frame,
                         int16_t amplitude,
-                        test::Random* rand_gen) {
+                        Random* rand_gen) {
   ASSERT_GT(amplitude, 0);
   ASSERT_LE(amplitude, 32767);
   for (int ch = 0; ch < frame->num_channels_; ch++) {
     for (int k = 0; k < static_cast<int>(frame->samples_per_channel_); k++) {
       // Store random 16 bit number between -(amplitude+1) and
       // amplitude.
       frame->data_[k * ch] = rand_gen->Rand(2 * amplitude + 1) - amplitude - 1;
     }
   }
 }
@@ skipping 257 matching lines @@
 
  private:
   rtc::CriticalSection crit_;
   bool capture_side_called_ GUARDED_BY(crit_) = false;
 };
 
 // Class for handling the capture side processing.
 class CaptureProcessor {
  public:
   CaptureProcessor(int max_frame_size,
-                   test::Random* rand_gen,
+                   Random* rand_gen,
                    FrameCounters* shared_counters_state,
                    CaptureSideCalledChecker* capture_call_checker,
                    AudioProcessingImplLockTest* test_framework,
                    TestConfig* test_config,
                    AudioProcessing* apm);
   bool Process();
 
  private:
   static const int kMaxCallDifference = 10;
   static const float kCaptureInputFloatLevel;
   static const int kCaptureInputFixLevel = 1024;
 
   void PrepareFrame();
   void CallApmCaptureSide();
   void ApplyRuntimeSettingScheme();
 
-  test::Random* rand_gen_ = nullptr;
+  Random* rand_gen_ = nullptr;
   FrameCounters* frame_counters_ = nullptr;
   CaptureSideCalledChecker* capture_call_checker_ = nullptr;
   AudioProcessingImplLockTest* test_ = nullptr;
   TestConfig* test_config_ = nullptr;
   AudioProcessing* apm_ = nullptr;
   AudioFrameData frame_data_;
 };
 
 // Class for handling the stats processing.
 class StatsProcessor {
  public:
-  StatsProcessor(test::Random* rand_gen,
+  StatsProcessor(Random* rand_gen,
                  TestConfig* test_config,
                  AudioProcessing* apm);
   bool Process();
 
  private:
-  test::Random* rand_gen_ = nullptr;
+  Random* rand_gen_ = nullptr;
   TestConfig* test_config_ = nullptr;
   AudioProcessing* apm_ = nullptr;
 };
 
 // Class for handling the render side processing.
 class RenderProcessor {
  public:
   RenderProcessor(int max_frame_size,
-                  test::Random* rand_gen,
+                  Random* rand_gen,
                   FrameCounters* shared_counters_state,
                   CaptureSideCalledChecker* capture_call_checker,
                   AudioProcessingImplLockTest* test_framework,
                   TestConfig* test_config,
                   AudioProcessing* apm);
   bool Process();
 
  private:
   static const int kMaxCallDifference = 10;
   static const int kRenderInputFixLevel = 16384;
   static const float kRenderInputFloatLevel;
 
   void PrepareFrame();
   void CallApmRenderSide();
   void ApplyRuntimeSettingScheme();
 
-  test::Random* rand_gen_ = nullptr;
+  Random* rand_gen_ = nullptr;
   FrameCounters* frame_counters_ = nullptr;
   CaptureSideCalledChecker* capture_call_checker_ = nullptr;
   AudioProcessingImplLockTest* test_ = nullptr;
   TestConfig* test_config_ = nullptr;
   AudioProcessing* apm_ = nullptr;
   bool first_render_side_call_ = true;
   AudioFrameData frame_data_;
 };
 
 class AudioProcessingImplLockTest
@@ skipping 46 matching lines @@
     stats_thread_->SetPriority(kNormalPriority);
   }
 
   // Event handler for the test.
   const rtc::scoped_ptr<EventWrapper> test_complete_;
 
   // Thread related variables.
   rtc::scoped_ptr<ThreadWrapper> render_thread_;
   rtc::scoped_ptr<ThreadWrapper> capture_thread_;
   rtc::scoped_ptr<ThreadWrapper> stats_thread_;
-  mutable test::Random rand_gen_;
+  mutable Random rand_gen_;
 
   rtc::scoped_ptr<AudioProcessing> apm_;
   TestConfig test_config_;
   FrameCounters frame_counters_;
   CaptureSideCalledChecker capture_call_checker_;
   RenderProcessor render_thread_state_;
   CaptureProcessor capture_thread_state_;
   StatsProcessor stats_thread_state_;
 };
 
@@ skipping 77 matching lines @@
     apm_->SetExtraOptions(config);
   }
 }
 
 void AudioProcessingImplLockTest::TearDown() {
   render_thread_->Stop();
   capture_thread_->Stop();
   stats_thread_->Stop();
 }
 
-StatsProcessor::StatsProcessor(test::Random* rand_gen,
+StatsProcessor::StatsProcessor(Random* rand_gen,
                                TestConfig* test_config,
                                AudioProcessing* apm)
     : rand_gen_(rand_gen), test_config_(test_config), apm_(apm) {}
 
 // Implements the callback functionality for the statistics
 // collection thread.
 bool StatsProcessor::Process() {
   SleepRandomMs(100, rand_gen_);
 
   EXPECT_EQ(apm_->echo_cancellation()->is_enabled(),
@@ skipping 13 matching lines @@
   apm_->noise_suppression()->speech_probability();
   apm_->voice_detection()->is_enabled();
 
   return true;
 }
 
 const float CaptureProcessor::kCaptureInputFloatLevel = 0.03125f;
 
 CaptureProcessor::CaptureProcessor(
     int max_frame_size,
-    test::Random* rand_gen,
+    Random* rand_gen,
     FrameCounters* shared_counters_state,
     CaptureSideCalledChecker* capture_call_checker,
     AudioProcessingImplLockTest* test_framework,
     TestConfig* test_config,
     AudioProcessing* apm)
     : rand_gen_(rand_gen),
       frame_counters_(shared_counters_state),
       capture_call_checker_(capture_call_checker),
       test_(test_framework),
       test_config_(test_config),
@@ skipping 247 matching lines @@
   // Restric the number of output channels not to exceed
   // the number of input channels.
   frame_data_.output_number_of_channels =
       std::min(frame_data_.output_number_of_channels,
                frame_data_.input_number_of_channels);
 }
 
 const float RenderProcessor::kRenderInputFloatLevel = 0.5f;
 
 RenderProcessor::RenderProcessor(int max_frame_size,
-                                 test::Random* rand_gen,
+                                 Random* rand_gen,
                                  FrameCounters* shared_counters_state,
                                  CaptureSideCalledChecker* capture_call_checker,
                                  AudioProcessingImplLockTest* test_framework,
                                  TestConfig* test_config,
                                  AudioProcessing* apm)
     : rand_gen_(rand_gen),
       frame_counters_(shared_counters_state),
       capture_call_checker_(capture_call_checker),
       test_(test_framework),
       test_config_(test_config),
@@ skipping 236 matching lines @@
     DISABLED_AudioProcessingImplLockExtensive,
     AudioProcessingImplLockTest,
     ::testing::ValuesIn(TestConfig::GenerateExtensiveTestConfigs()));
 
 INSTANTIATE_TEST_CASE_P(
     DISABLED_AudioProcessingImplLockBrief,
     AudioProcessingImplLockTest,
     ::testing::ValuesIn(TestConfig::GenerateBriefTestConfigs()));
 
 }  // namespace webrtc