modules/video_coding refactorings

The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.

To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).

Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417283007 .

Cr-Commit-Position: refs/heads/master@{#10694}
diff --git a/webrtc/modules/video_coding/test/test_util.h b/webrtc/modules/video_coding/test/test_util.h
new file mode 100644
index 0000000..30f337d
--- /dev/null
+++ b/webrtc/modules/video_coding/test/test_util.h
@@ -0,0 +1,86 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
+
+/*
+ * General declarations used through out VCM offline tests.
+ */
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+
+enum { kMaxNackListSize = 250 };
+enum { kMaxPacketAgeToNack = 450 };
+
+class NullEvent : public webrtc::EventWrapper {
+ public:
+  virtual ~NullEvent() {}
+
+  virtual bool Set() { return true; }
+
+  virtual bool Reset() { return true; }
+
+  virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) {
+    return webrtc::kEventTimeout;
+  }
+
+  virtual bool StartTimer(bool periodic, unsigned long time) { return true; }
+
+  virtual bool StopTimer() { return true; }
+};
+
+class NullEventFactory : public webrtc::EventFactory {
+ public:
+  virtual ~NullEventFactory() {}
+
+  virtual webrtc::EventWrapper* CreateEvent() {
+    return new NullEvent;
+  }
+};
+
+class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
+ public:
+  FileOutputFrameReceiver(const std::string& base_out_filename, uint32_t ssrc);
+  virtual ~FileOutputFrameReceiver();
+
+  // VCMReceiveCallback
+  virtual int32_t FrameToRender(webrtc::VideoFrame& video_frame);
+
+ private:
+  std::string out_filename_;
+  FILE* out_file_;
+  FILE* timing_file_;
+  int width_;
+  int height_;
+  int count_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FileOutputFrameReceiver);
+};
+
+class CmdArgs {
+ public:
+  CmdArgs();
+
+  std::string codecName;
+  webrtc::VideoCodecType codecType;
+  int width;
+  int height;
+  int rtt;
+  std::string inputFile;
+  std::string outputFile;
+};
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_