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

2011-05-30 11:22:19 +00:00
parent 01813fe945
commit 77ae29bc81
1153 changed files with 404089 additions and 0 deletions
--- a/modules/video_coding/main/test/generic_codec_test.h
+++ b/modules/video_coding/main/test/generic_codec_test.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2011 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_GENERIC_CODEC_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
+
+#include "video_coding.h"
+#include "test_macros.h"
+#include "test_util.h"
+
+#include <string.h>
+#include <fstream>
+
+/*
+Test consists of:
+1. Sanity checks
+2. Bit rate validation
+3. Encoder control test / General API functionality
+4. Decoder control test / General API functionality
+
+*/
+int VCMGenericCodecTest(CmdArgs& args);
+
+class GenericCodecTest
+{
+public:
+    GenericCodecTest(webrtc::VideoCodingModule* vcm);
+    ~GenericCodecTest();
+    static int RunTest(CmdArgs& args);
+    WebRtc_Word32 Perform(CmdArgs& args);
+    float WaitForEncodedFrame() const;
+
+private:
+    void Setup(CmdArgs& args);
+    void Print();
+    WebRtc_Word32 TearDown();
+    void IncrementDebugClock(float frameRate);
+
+    webrtc::VideoCodingModule*           _vcm;
+    webrtc::VideoCodec                   _sendCodec;
+    webrtc::VideoCodec                   _receiveCodec;
+    std::string                      _inname;
+    std::string                      _outname;
+    std::string                      _encodedName;
+    WebRtc_Word32                      _sumEncBytes;
+    FILE*                            _sourceFile;
+    FILE*                            _decodedFile;
+    FILE*                            _encodedFile;
+    WebRtc_UWord16                     _width;
+    WebRtc_UWord16                     _height;
+    float                            _frameRate;
+    WebRtc_UWord32                     _lengthSourceFrame;
+    WebRtc_UWord32                     _timeStamp;
+    int                              vcmMacrosTests;
+    int                              vcmMacrosErrors;
+    VCMDecodeCompleteCallback*         _decodeCallback;
+    VCMEncodeCompleteCallback*       _encodeCompleteCallback;
+
+}; // end of GenericCodecTest class definition
+
+class RTPSendCallback_SizeTest : public webrtc::Transport
+{
+public:
+    // constructor input: (receive side) rtp module to send encoded data to
+    RTPSendCallback_SizeTest() : _maxPayloadSize(0), _payloadSizeSum(0), _nPackets(0) {}
+    virtual int SendPacket(int channel, const void *data, int len);
+    virtual int SendRTCPPacket(int channel, const void *data, int len) {return 0;}
+    void SetMaxPayloadSize(WebRtc_UWord32 maxPayloadSize);
+    void Reset();
+    float AveragePayloadSize() const;
+private:
+    WebRtc_UWord32         _maxPayloadSize;
+    WebRtc_UWord32         _payloadSizeSum;
+    WebRtc_UWord32         _nPackets;
+};
+
+class VCMEncComplete_KeyReqTest : public webrtc::VCMPacketizationCallback
+{
+public:
+    VCMEncComplete_KeyReqTest(webrtc::VideoCodingModule &vcm) : _vcm(vcm), _seqNo(0), _timeStamp(0) {}
+    WebRtc_Word32 SendData(const webrtc::FrameType frameType,
+                                 const WebRtc_UWord8 payloadType,
+                                 const WebRtc_UWord32 timeStamp,
+                                 const WebRtc_UWord8* payloadData,
+                                 const WebRtc_UWord32 payloadSize,
+                                 const webrtc::RTPFragmentationHeader& fragmentationHeader);
+private:
+    webrtc::VideoCodingModule& _vcm;
+    WebRtc_UWord16 _seqNo;
+    WebRtc_UWord32 _timeStamp;
+}; // end of VCMEncodeCompleteCallback
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_