Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/video_coding/main/test/test_util.h

@@ -0,0 +1,75 @@
+/*
+ *  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.h>
+#include <fstream>
+#include <cstdlib>
+
+#include "module_common_types.h"
+#include "testsupport/fileutils.h"
+
+// Class used for passing command line arguments to tests
+class CmdArgs
+{
+ public:
+  CmdArgs()
+      : codecName("VP8"),
+        codecType(webrtc::kVideoCodecVP8),
+        width(352),
+        height(288),
+        bitRate(500),
+        frameRate(30),
+        packetLoss(0),
+        rtt(0),
+        protectionMode(0),
+        camaEnable(0),
+        inputFile(webrtc::test::ProjectRootPath() +
+                  "/resources/foreman_cif.yuv"),
+        outputFile(webrtc::test::OutputPath() +
+                   "video_coding_test_output_352x288.yuv"),
+        testNum(11) {}
+     std::string codecName;
+     webrtc::VideoCodecType codecType;
+     int width;
+     int height;
+     int bitRate;
+     int frameRate;
+     int packetLoss;
+     int rtt;
+     int protectionMode;
+     int camaEnable;
+     std::string inputFile;
+     std::string outputFile;
+     int testNum;
+};
+
+// forward declaration
+int MTRxTxTest(CmdArgs& args);
+double NormalDist(double mean, double stdDev);
+
+struct RtpPacket {
+  WebRtc_Word8 data[1650]; // max packet size
+  WebRtc_Word32 length;
+  WebRtc_Word64 receiveTime;
+};
+
+
+// Codec type conversion
+webrtc::RTPVideoCodecTypes
+ConvertCodecType(const char* plname);
+
+#endif