Change default timestamp to 64 bits in all webrtc directories.

webrtc/api/test/fakeaudiocapturemodule.cc

 /*
  *  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.
  */
 
 #include "webrtc/api/test/fakeaudiocapturemodule.h"
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
 
 // Audio sample value that is high enough that it doesn't occur naturally when
 // frames are being faked. E.g. NetEq will not generate this large sample value
 // unless it has received an audio frame containing a sample of this value.
 // Even simpler buffers would likely just contain audio sample values of 0.
 static const int kHighSampleValue = 10000;
 
 // Same value as src/modules/audio_device/main/source/audio_device_config.h in
 // https://code.google.com/p/webrtc/
-static const uint32_t kAdmMaxIdleTimeProcess = 1000;
+static const int kAdmMaxIdleTimeProcess = 1000;
 
 // Constants here are derived by running VoE using a real ADM.
 // The constants correspond to 10ms of mono audio at 44kHz.
 static const int kTimePerFrameMs = 10;
 static const uint8_t kNumberOfChannels = 1;
 static const int kSamplesPerSecond = 44000;
 static const int kTotalDelayMs = 0;
 static const int kClockDriftMs = 0;
 static const uint32_t kMaxVolume = 14392;
 
@@ skipping 29 matching lines @@
   }
   return capture_module;
 }
 
 int FakeAudioCaptureModule::frames_received() const {
   rtc::CritScope cs(&crit_);
   return frames_received_;
 }
 
 int64_t FakeAudioCaptureModule::TimeUntilNextProcess() {
-  const uint32_t current_time = rtc::Time();
+  const int64_t current_time = rtc::TimeMillis();
   if (current_time < last_process_time_ms_) {
     // TODO: wraparound could be handled more gracefully.
     return 0;
   }
-  const uint32_t elapsed_time = current_time - last_process_time_ms_;
+  const int64_t elapsed_time = current_time - last_process_time_ms_;
   if (kAdmMaxIdleTimeProcess < elapsed_time) {
     return 0;
   }
   return kAdmMaxIdleTimeProcess - elapsed_time;
 }
 
 void FakeAudioCaptureModule::Process() {
-  last_process_time_ms_ = rtc::Time();
+  last_process_time_ms_ = rtc::TimeMillis();
 }
 
 int32_t FakeAudioCaptureModule::ActiveAudioLayer(
     AudioLayer* /*audio_layer*/) const {
   ASSERT(false);
   return 0;
 }
 
 webrtc::AudioDeviceModule::ErrorCode FakeAudioCaptureModule::LastError() const {
   ASSERT(false);
@@ skipping 483 matching lines @@
       ASSERT(false);
   }
 }
 
 bool FakeAudioCaptureModule::Initialize() {
   // Set the send buffer samples high enough that it would not occur on the
   // remote side unless a packet containing a sample of that magnitude has been
   // sent to it. Note that the audio processing pipeline will likely distort the
   // original signal.
   SetSendBuffer(kHighSampleValue);
-  last_process_time_ms_ = rtc::Time();
+  last_process_time_ms_ = rtc::TimeMillis();
   return true;
 }
 
 void FakeAudioCaptureModule::SetSendBuffer(int value) {
   Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_);
   const size_t buffer_size_in_samples =
       sizeof(send_buffer_) / kNumberBytesPerSample;
   for (size_t i = 0; i < buffer_size_in_samples; ++i) {
     buffer_ptr[i] = value;
   }
@@ skipping 38 matching lines @@
   if (started_) {
     // Already started.
     return;
   }
   ProcessFrameP();
 }
 
 void FakeAudioCaptureModule::ProcessFrameP() {
   ASSERT(process_thread_->IsCurrent());
   if (!started_) {
-    next_frame_time_ = rtc::Time();
+    next_frame_time_ = rtc::TimeMillis();
     started_ = true;
   }
 
   {
     rtc::CritScope cs(&crit_);
     // Receive and send frames every kTimePerFrameMs.
     if (playing_) {
       ReceiveFrameP();
     }
     if (recording_) {
       SendFrameP();
     }
   }
 
   next_frame_time_ += kTimePerFrameMs;
-  const uint32_t current_time = rtc::Time();
-  const uint32_t wait_time =
+  const int64_t current_time = rtc::TimeMillis();
+  const int64_t wait_time =
       (next_frame_time_ > current_time) ? next_frame_time_ - current_time : 0;
   process_thread_->PostDelayed(wait_time, this, MSG_RUN_PROCESS);
 }
 
 void FakeAudioCaptureModule::ReceiveFrameP() {
   ASSERT(process_thread_->IsCurrent());
   {
     rtc::CritScope cs(&crit_callback_);
     if (!audio_callback_) {
       return;
@@ skipping 36 matching lines @@
                                               kNumberOfChannels,
                                               kSamplesPerSecond, kTotalDelayMs,
                                               kClockDriftMs, current_mic_level,
                                               key_pressed,
                                               current_mic_level) != 0) {
     ASSERT(false);
   }
   SetMicrophoneVolume(current_mic_level);
 }