Added A/V sync tests with drifting clocks.

webrtc/test/fake_audio_device.cc

 /*
  *  Copyright (c) 2013 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/test/fake_audio_device.h"
 
 #include <algorithm>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/media_file/media_file_utility.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 namespace webrtc {
 namespace test {
 
-FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
+FakeAudioDevice::FakeAudioDevice(Clock* clock,
+                                 const std::string& filename,
+                                 float speed)
     : audio_callback_(NULL),
       capturing_(false),
       captured_audio_(),
       playout_buffer_(),
+      speed_(speed),
       last_playout_ms_(-1),
-      clock_(clock),
+      clock_(clock, speed),
       tick_(EventTimerWrapper::Create()),
       thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
       file_utility_(new ModuleFileUtility(0)),
       input_stream_(FileWrapper::Create()) {
   memset(captured_audio_, 0, sizeof(captured_audio_));
   memset(playout_buffer_, 0, sizeof(playout_buffer_));
   // Open audio input file as read-only and looping.
   EXPECT_EQ(0, input_stream_->OpenFile(filename.c_str(), true, true))
       << filename;
 }
 
 FakeAudioDevice::~FakeAudioDevice() {
   Stop();
 
   thread_.Stop();
 }
 
 int32_t FakeAudioDevice::Init() {
   rtc::CritScope cs(&lock_);
   if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
     return -1;
 
-  if (!tick_->StartTimer(true, 10))
+  if (!tick_->StartTimer(true, 10 / speed_))
     return -1;
   thread_.Start();
   thread_.SetPriority(rtc::kHighPriority);
   return 0;
 }
 
 int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
   rtc::CritScope cs(&lock_);
   audio_callback_ = callback;
   return 0;
@@ skipping 35 matching lines @@
                                                          num_samples,
                                                          2,
                                                          1,
                                                          kFrequencyHz,
                                                          0,
                                                          0,
                                                          0,
                                                          false,
                                                          new_mic_level));
       size_t samples_needed = kFrequencyHz / 100;
-      int64_t now_ms = clock_->TimeInMilliseconds();
+      int64_t now_ms = clock_.TimeInMilliseconds();
       uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
       if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
         samples_needed = std::min(
             static_cast<size_t>(kFrequencyHz / time_since_last_playout_ms),
             kBufferSizeBytes / 2);
       }
       size_t samples_out = 0;
       int64_t elapsed_time_ms = -1;
       int64_t ntp_time_ms = -1;
       EXPECT_EQ(0,
@@ skipping 14 matching lines @@
   rtc::CritScope cs(&lock_);
   capturing_ = true;
 }
 
 void FakeAudioDevice::Stop() {
   rtc::CritScope cs(&lock_);
   capturing_ = false;
 }
 }  // namespace test
 }  // namespace webrtc