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
diff --git a/test/fake_audio_device.h b/test/fake_audio_device.h
new file mode 100644
index 0000000..40a7547
--- /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_