Change default timestamp to 64 bits in all webrtc directories.

webrtc/api/test/fakeaudiocapturemodule.h

 /*
  *  Copyright 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.
  */
 
@@ skipping 209 matching lines @@
   // Periodcally called function that ensures that frames are pulled and pushed
   // periodically if enabled/started.
   void ProcessFrameP();
   // Pulls frames from the registered webrtc::AudioTransport.
   void ReceiveFrameP();
   // Pushes frames to the registered webrtc::AudioTransport.
   void SendFrameP();
 
   // The time in milliseconds when Process() was last called or 0 if no call
   // has been made.
-  uint32_t last_process_time_ms_;
+  int64_t last_process_time_ms_;
 
   // Callback for playout and recording.
   webrtc::AudioTransport* audio_callback_;
 
   bool recording_; // True when audio is being pushed from the instance.
   bool playing_; // True when audio is being pulled by the instance.
 
   bool play_is_initialized_; // True when the instance is ready to pull audio.
   bool rec_is_initialized_; // True when the instance is ready to push audio.
 
   // Input to and output from RecordedDataIsAvailable(..) makes it possible to
   // modify the current mic level. The implementation does not care about the
   // mic level so it just feeds back what it receives.
   uint32_t current_mic_level_;
 
   // next_frame_time_ is updated in a non-drifting manner to indicate the next
   // wall clock time the next frame should be generated and received. started_
   // ensures that next_frame_time_ can be initialized properly on first call.
   bool started_;
-  uint32_t next_frame_time_;
+  int64_t next_frame_time_;
 
   std::unique_ptr<rtc::Thread> process_thread_;
 
   // Buffer for storing samples received from the webrtc::AudioTransport.
   char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
   // Buffer for samples to send to the webrtc::AudioTransport.
   char send_buffer_[kNumberSamples * kNumberBytesPerSample];
 
   // Counter of frames received that have samples of high enough amplitude to
   // indicate that the frames are not faked somewhere in the audio pipeline
   // (e.g. by a jitter buffer).
   int frames_received_;
 
   // Protects variables that are accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_;
   // Protects |audio_callback_| that is accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_callback_;
 };
 
 #endif  // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_