Removes usage of system_wrappers/include/clock.h in audio_device/

webrtc/modules/audio_device/android/audio_device_unittest.cc

Index: webrtc/modules/audio_device/android/audio_device_unittest.cc
diff --git a/webrtc/modules/audio_device/android/audio_device_unittest.cc b/webrtc/modules/audio_device/android/audio_device_unittest.cc
index c31e14e1c6fd0c8dc3657407c80071e3896f1b70..2328929b6d796350bfe37c83722236e2d26ebfec 100644
--- a/webrtc/modules/audio_device/android/audio_device_unittest.cc
+++ b/webrtc/modules/audio_device/android/audio_device_unittest.cc
@@ -20,6 +20,7 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/build_info.h"
@@ -27,7 +28,6 @@
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/include/mock_audio_transport.h"
-#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/gmock.h"
@@ -252,8 +252,7 @@ class FifoAudioStream : public AudioStreamInterface {
 class LatencyMeasuringAudioStream : public AudioStreamInterface {
  public:
   explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
-      : clock_(Clock::GetRealTimeClock()),
-        frames_per_buffer_(frames_per_buffer),
+      : frames_per_buffer_(frames_per_buffer),
         bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
         play_count_(0),
         rec_count_(0),
@@ -270,7 +269,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
     memset(destination, 0, bytes_per_buffer_);
     if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
       if (pulse_time_ == 0) {
-        pulse_time_ = clock_->TimeInMilliseconds();
+        pulse_time_ = rtc::TimeMillis();
       }
       PRINT(".");
       const int16_t impulse = std::numeric_limits<int16_t>::max();
@@ -301,7 +300,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
                                      max));
     if (max > kImpulseThreshold) {
       PRINTD("(%d,%d)", max, index_of_max);
-      int64_t now_time = clock_->TimeInMilliseconds();
+      int64_t now_time = rtc::TimeMillis();
       int extra_delay = IndexToMilliseconds(static_cast<double> (index_of_max));
       PRINTD("[%d]", static_cast<int> (now_time - pulse_time_));
       PRINTD("[%d]", extra_delay);
@@ -356,7 +355,6 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
   }
 
  private:
-  Clock* clock_;
   const size_t frames_per_buffer_;
   const size_t bytes_per_buffer_;
   size_t play_count_;
@@ -999,7 +997,7 @@ TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
 // - Store time differences in a vector and calculate min, max and average.
 // This test requires a special hardware called Audio Loopback Dongle.
 // See http://source.android.com/devices/audio/loopback.html for details.
-TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
+TEST_F(AudioDeviceTest, MeasureLoopbackLatency) {

[nisse-webrtc, 2016/11/15 08:12:33]

Nice it the test can now be enabled, but it's unclear why this change makes that
possible?

   EXPECT_EQ(record_channels(), playout_channels());
   EXPECT_EQ(record_sample_rate(), playout_sample_rate());
   NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);