Replace Clock with timeutils in AudioEncoder.

webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc

 /*
  *  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/base/format_macros.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 namespace {
 int64_t RunComplexityTest(const AudioEncoderOpus::Config& config) {
   // Create encoder.
   AudioEncoderOpus encoder(config);
   // Open speech file.
   const std::string kInputFileName =
       webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
   test::AudioLoop audio_loop;
   constexpr int kSampleRateHz = 48000;
   EXPECT_EQ(kSampleRateHz, encoder.SampleRateHz());
   constexpr size_t kMaxLoopLengthSamples =
       kSampleRateHz * 10;  // 10 second loop.
   constexpr size_t kInputBlockSizeSamples =
       10 * kSampleRateHz / 1000;  // 60 ms.
   EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
                               kInputBlockSizeSamples));
   // Encode.
-  webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
-  const int64_t start_time_ms = clock->TimeInMilliseconds();
+  const int64_t start_time_ms = rtc::TimeMillis();
   AudioEncoder::EncodedInfo info;
   rtc::Buffer encoded(500);
   uint32_t rtp_timestamp = 0u;
   for (size_t i = 0; i < 10000; ++i) {
     encoded.Clear();
     info = encoder.Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
     rtp_timestamp += kInputBlockSizeSamples;
   }
-  return clock->TimeInMilliseconds() - start_time_ms;
+  return rtc::TimeMillis() - start_time_ms;
 }
 }  // namespace
 
 // This test encodes an audio file using Opus twice with different bitrates
 // (~11 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
 // between the two is calculated and tracked. This test explicitly sets the
 // low_rate_complexity to 9. When running on desktop platforms, this is the same
 // as the regular complexity, and the expectation is that the resulting ratio
 // should be less than 100% (since the encoder runs faster at lower bitrates,
 // given a fixed complexity setting). On the other hand, when running on
@@ skipping 30 matching lines @@
   int64_t runtime_10999bps = RunComplexityTest(config);
 
   config.bitrate_bps = rtc::Optional<int>(15500);
   int64_t runtime_15500bps = RunComplexityTest(config);
 
   test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off",
                     100.0 * runtime_10999bps / runtime_15500bps, "percent",
                     true);
 }
 }  // namespace webrtc