Delete all use of tick_util.h.

webrtc/modules/utility/source/file_recorder_impl.h

 /*
  *  Copyright (c) 2012 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.
  */
 
 // This file contains a class that can write audio and/or video to file in
 // multiple file formats. The unencoded input data is written to file in the
 // encoded format specified.
 
 #ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
 #define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
 
 #include <list>
 
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 // The largest decoded frame size in samples (60ms with 32kHz sample rate).
 enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
 enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
 enum { kMaxAudioBufferQueueLength = 100 };
 
 class CriticalSectionWrapper;
 
@@ skipping 10 matching lines @@
         const char* fileName,
         const CodecInst& codecInst,
         uint32_t notificationTimeMs) override;
     int32_t StartRecordingAudioFile(
         OutStream& destStream,
         const CodecInst& codecInst,
         uint32_t notificationTimeMs) override;
     int32_t StopRecording() override;
     bool IsRecording() const override;
     int32_t codec_info(CodecInst& codecInst) const override;
-    int32_t RecordAudioToFile(
-        const AudioFrame& frame,
-        const TickTime* playoutTS = NULL) override;
+    int32_t RecordAudioToFile(const AudioFrame& frame) override;
     int32_t StartRecordingVideoFile(
         const char* fileName,
         const CodecInst& audioCodecInst,
         const VideoCodec& videoCodecInst,
         bool videoOnly = false) override
     {
         return -1;
     }
     int32_t RecordVideoToFile(const VideoFrame& videoFrame) override {
         return -1;
@@ skipping 10 matching lines @@
     MediaFile* _moduleFile;
 
 private:
     CodecInst codec_info_;
     int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
     AudioCoder _audioEncoder;
     Resampler _audioResampler;
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_