Hook up audio/video sync to Call.

Adds an end-to-end audio/video sync test.

BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5128 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 69 insertions, 0 deletions

diff --git a/test/fake_audio_device.h b/test/fake_audio_device.h
new file mode 100644
index 00000000..40a7547d
--- /dev/null
+++ b/test/fake_audio_device.h
@@ -0,0 +1,69 @@
+/*
+ *  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.
+ */
+#ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
+#define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
+
+#include <string>
+
+#include "webrtc/modules/audio_device/include/fake_audio_device.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Clock;
+class CriticalSectionWrapper;
+class EventWrapper;
+class FileWrapper;
+class ModuleFileUtility;
+class ThreadWrapper;
+
+namespace test {
+
+class FakeAudioDevice : public FakeAudioDeviceModule {
+ public:
+  FakeAudioDevice(Clock* clock, const std::string& filename);
+
+  virtual ~FakeAudioDevice();
+
+  virtual int32_t Init() OVERRIDE;
+  virtual int32_t RegisterAudioCallback(AudioTransport* callback) OVERRIDE;
+
+  virtual bool Playing() const OVERRIDE;
+  virtual int32_t PlayoutDelay(uint16_t* delay_ms) const OVERRIDE;
+  virtual bool Recording() const OVERRIDE;
+
+  void Start();
+  void Stop();
+
+ private:
+  static bool Run(void* obj);
+  void CaptureAudio();
+
+  static const uint32_t kFrequencyHz = 16000;
+  static const uint32_t kBufferSizeBytes = 2 * kFrequencyHz;
+
+  AudioTransport* audio_callback_;
+  bool capturing_;
+  int8_t captured_audio_[kBufferSizeBytes];
+  int8_t playout_buffer_[kBufferSizeBytes];
+  int64_t last_playout_ms_;
+
+  Clock* clock_;
+  scoped_ptr<EventWrapper> tick_;
+  scoped_ptr<CriticalSectionWrapper> lock_;
+  scoped_ptr<ThreadWrapper> thread_;
+  scoped_ptr<ModuleFileUtility> file_utility_;
+  scoped_ptr<FileWrapper> input_stream_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_