Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t.

webrtc/modules/audio_device/audio_device_buffer.cc

 /*
  *  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.
  */
 
@@ skipping 93 matching lines @@
   }
   LOG(INFO) << __FUNCTION__;
   playout_thread_checker_.DetachFromThread();
   // Clear members tracking playout stats and do it on the task queue.
   task_queue_.PostTask([this] { ResetPlayStats(); });
   // Start a periodic timer based on task queue if not already done by the
   // recording side.
   if (!recording_) {
     StartPeriodicLogging();
   }
-  const uint64_t now_time = rtc::TimeMillis();
+  const int64_t now_time = rtc::TimeMillis();
   // Clear members that are only touched on the main (creating) thread.
   play_start_time_ = now_time;
   playing_ = true;
 }
 
 void AudioDeviceBuffer::StartRecording() {
   RTC_DCHECK_RUN_ON(&main_thread_checker_);
   if (recording_) {
     return;
   }
@@ skipping 400 matching lines @@
                                         size_t samples_per_channel) {
   RTC_DCHECK_RUN_ON(&task_queue_);
   ++play_callbacks_;
   play_samples_ += samples_per_channel;
   if (max_abs > max_play_level_) {
     max_play_level_ = max_abs;
   }
 }
 
 }  // namespace webrtc