Delete all use of tick_util.h.

webrtc/modules/audio_processing/test/audio_file_processor.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
 
 #include <algorithm>
 #include <limits>
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
 #else
 #include "webrtc/modules/audio_processing/debug.pb.h"
 #endif
 
 namespace webrtc {
 
 // Holds a few statistics about a series of TickIntervals.
 struct TickIntervalStats {
   TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
-  TickInterval sum;
-  TickInterval max;
-  TickInterval min;
+  int64_t sum;
+  int64_t max;
+  int64_t min;
 };
 
 // Interface for processing an input file with an AudioProcessing instance and
 // dumping the results to an output file.
 class AudioFileProcessor {
  public:
   static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
 
   virtual ~AudioFileProcessor() {}
 
   // Processes one AudioProcessing::kChunkSizeMs of data from the input file and
   // writes to the output file.
   virtual bool ProcessChunk() = 0;
 
   // Returns the execution time of all AudioProcessing calls.
   const TickIntervalStats& proc_time() const { return proc_time_; }
 
  protected:
   // RAII class for execution time measurement. Updates the provided
   // TickIntervalStats based on the time between ScopedTimer creation and
   // leaving the enclosing scope.
   class ScopedTimer {
    public:
     explicit ScopedTimer(TickIntervalStats* proc_time)
-        : proc_time_(proc_time), start_time_(TickTime::Now()) {}
+      : proc_time_(proc_time), start_time_(rtc::TimeNanos()) {}
 
     ~ScopedTimer() {
-      TickInterval interval = TickTime::Now() - start_time_;
+      int64_t interval = rtc::TimeNanos() - start_time_;
       proc_time_->sum += interval;
       proc_time_->max = std::max(proc_time_->max, interval);
       proc_time_->min = std::min(proc_time_->min, interval);
     }
 
    private:
     TickIntervalStats* const proc_time_;
-    TickTime start_time_;
+    int64_t start_time_;
   };
 
   TickIntervalStats* mutable_proc_time() { return &proc_time_; }
 
  private:
   TickIntervalStats proc_time_;
 };
 
 // Used to read from and write to WavFile objects.
 class WavFileProcessor final : public AudioFileProcessor {
@@ skipping 53 matching lines @@
   ChannelBuffer<float> out_buf_;
   StreamConfig input_config_;
   StreamConfig reverse_config_;
   const StreamConfig output_config_;
   ChannelBufferWavWriter buffer_writer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_