git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
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niklase@google.com
257
modules/video_coding/main/test/test_util.h
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257
modules/video_coding/main/test/test_util.h
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef TEST_UTIL_H
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#define TEST_UTIL_H
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#include "video_coding.h"
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#include "rtp_rtcp.h"
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#include "trace.h"
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#include "module_common_types.h"
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#include "tick_time.h"
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#include "test_macros.h"
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#include "test_util.h"
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#include <string.h>
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#include <fstream>
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#include <cstdlib>
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enum { kMaxWaitEncTimeMs = 100 };
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// Class used for passing command line arguments to tests
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class CmdArgs
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{
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public:
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CmdArgs() : codecName(""), codecType(webrtc::kVideoCodecVP8), width(-1),
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height(-1), bitRate(-1), frameRate(-1),
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inputFile(""), outputFile(""), testNum(-1)
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{}
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std::string codecName;
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webrtc::VideoCodecType codecType;
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int width;
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int height;
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int bitRate;
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int frameRate;
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std::string inputFile;
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std::string outputFile;
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int testNum;
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};
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// forward declaration
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int MTRxTxTest(CmdArgs& args);
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namespace webrtc
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{
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class RtpDump;
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}
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// definition of general test function to be used by VCM tester (mainly send side)
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/*
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Includes the following:
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1. General Callback definition for VCM test functions - no RTP.
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2. EncodeComplete callback:
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2a. Transfer encoded data directly to the decoder
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2b. Pass encoded data via the RTP module
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3. Caluclate PSNR from file function (for now: does not deal with frame drops)
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*/
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//Send Side - Packetization callback - send an encoded frame directly to the VCMReceiver
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class VCMEncodeCompleteCallback: public webrtc::VCMPacketizationCallback
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{
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public:
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// constructor input: file in which encoded data will be written, and test parameters
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VCMEncodeCompleteCallback(FILE* encodedFile);
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virtual ~VCMEncodeCompleteCallback();
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// Register transport callback
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void RegisterTransportCallback(webrtc::VCMPacketizationCallback* transport);
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// process encoded data received from the encoder, pass stream to the VCMReceiver module
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WebRtc_Word32 SendData(const webrtc::FrameType frameType,
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const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
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const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
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const webrtc::RTPFragmentationHeader& fragmentationHeader);
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// Register exisitng VCM. Currently - encode and decode with the same vcm module.
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void RegisterReceiverVCM(webrtc::VideoCodingModule *vcm) { _VCMReceiver = vcm; }
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// Return size of last encoded frame encoded data (all frames in the sequence)
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// Good for only one call - after which will reset value (to allow detection of frame drop)
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float EncodedBytes();
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// return encode complete (true/false)
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bool EncodeComplete();
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// Inform callback of codec used
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void SetCodecType(webrtc::RTPVideoCodecTypes codecType) { _codecType = codecType; }
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// inform callback of frame dimensions
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void SetFrameDimensions(WebRtc_Word32 width, WebRtc_Word32 height)
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{
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_width = width;
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_height = height;
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}
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//Initialize callback data
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void Initialize();
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void ResetByteCount();
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// conversion function for payload type (needed for the callback function)
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// RTPVideoVideoCodecTypes ConvertPayloadType(WebRtc_UWord8 payloadType);
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private:
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FILE* _encodedFile;
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float _encodedBytes;
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webrtc::VideoCodingModule* _VCMReceiver;
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webrtc::FrameType _frameType;
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WebRtc_UWord8* _payloadData;
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WebRtc_UWord8 _seqNo;
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bool _encodeComplete;
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WebRtc_Word32 _width;
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WebRtc_Word32 _height;
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webrtc::RTPVideoCodecTypes _codecType;
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WebRtc_UWord8 _layerPacketId;
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}; // end of VCMEncodeCompleteCallback
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//Send Side - Packetization callback - packetize an encoded frame via the RTP module
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class VCMRTPEncodeCompleteCallback: public webrtc::VCMPacketizationCallback
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{
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public:
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VCMRTPEncodeCompleteCallback(webrtc::RtpRtcp* rtp) :
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_seqNo(0), _encodedBytes(0), _RTPModule(rtp), _encodeComplete(false) {}
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virtual ~VCMRTPEncodeCompleteCallback() {}
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// process encoded data received from the encoder, pass stream to the RTP module
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WebRtc_Word32 SendData(const webrtc::FrameType frameType,
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const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
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const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
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const webrtc::RTPFragmentationHeader& fragmentationHeader);
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// Return size of last encoded frame. Value good for one call
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// (resets to zero after call to inform test of frame drop)
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float EncodedBytes();
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// return encode complete (true/false)
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bool EncodeComplete();
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// Inform callback of codec used
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void SetCodecType(webrtc::RTPVideoCodecTypes codecType) { _codecType = codecType; }
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// inform callback of frame dimensions
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void SetFrameDimensions(WebRtc_Word16 width, WebRtc_Word16 height)
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{
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_width = width;
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_height = height;
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}
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private:
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float _encodedBytes;
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webrtc::FrameType _frameType;
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WebRtc_UWord8* _payloadData;
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WebRtc_UWord16 _seqNo;
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bool _encodeComplete;
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webrtc::RtpRtcp* _RTPModule;
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WebRtc_Word16 _width;
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WebRtc_Word16 _height;
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webrtc::RTPVideoCodecTypes _codecType;
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}; // end of VCMEncodeCompleteCallback
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class VCMDecodeCompleteCallback: public webrtc::VCMReceiveCallback
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{
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public:
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VCMDecodeCompleteCallback(FILE* decodedFile) :
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_decodedFile(decodedFile), _decodedBytes(0) {}
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virtual ~VCMDecodeCompleteCallback() {}
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// will write decoded frame into file
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WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
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WebRtc_Word32 DecodedBytes();
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int PSNRLastFrame(const webrtc::VideoFrame& sourceFrame, double *YPSNRptr);
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private:
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FILE* _decodedFile;
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WebRtc_UWord32 _decodedBytes;
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webrtc::VideoFrame _lastDecodedFrame;
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}; // end of VCMDecodeCompleCallback class
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///
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class RTPSendCompleteCallback: public webrtc::Transport
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{
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public:
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// constructor input: (reeive side) rtp module to send encoded data to
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RTPSendCompleteCallback(webrtc::RtpRtcp* rtp,
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const char* filename = NULL);
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virtual ~RTPSendCompleteCallback();
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// Send Packet to receive side RTP module
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virtual int SendPacket(int channel, const void *data, int len);
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// Send RTCP Packet to receive side RTP module
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virtual int SendRTCPPacket(int channel, const void *data, int len);
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// Set percentage of channel loss in the network
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void SetLossPct(double lossPct);
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// return send count
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int SendCount() { return _sendCount; }
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private:
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// randomly decide weather to drop a packet or not, based on the channel model
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bool PacketLoss(double lossPct);
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WebRtc_UWord32 _sendCount;
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webrtc::RtpRtcp* _rtp;
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double _lossPct;
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webrtc::RtpDump* _rtpDump;
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};
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// used in multi thread test
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class SendSharedState
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{
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public:
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SendSharedState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp,
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CmdArgs args) :
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_rtp(rtp), _vcm(vcm), _args(args), _sourceFile(NULL), _frameCnt(0),
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_timestamp(0) {}
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webrtc::VideoCodingModule& _vcm;
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webrtc::RtpRtcp& _rtp;
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CmdArgs _args;
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FILE* _sourceFile;
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WebRtc_Word32 _frameCnt;
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WebRtc_Word32 _timestamp;
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};
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class PacketRequester: public webrtc::VCMPacketRequestCallback
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{
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public:
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PacketRequester(webrtc::RtpRtcp& rtp) :
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_rtp(rtp) {}
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WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers,
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WebRtc_UWord16 length);
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private:
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webrtc::RtpRtcp& _rtp;
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};
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// PSNR & SSIM calculations
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WebRtc_Word32
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PSNRfromFiles(const WebRtc_Word8 *refFileName,
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const WebRtc_Word8 *testFileName, WebRtc_Word32 width,
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WebRtc_Word32 height, double *YPSNRptr);
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WebRtc_Word32
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SSIMfromFiles(const WebRtc_Word8 *refFileName,
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const WebRtc_Word8 *testFileName, WebRtc_Word32 width,
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WebRtc_Word32 height, double *SSIMptr);
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// codec type conversion
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webrtc::RTPVideoCodecTypes
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ConvertCodecType(const char* plname);
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class SendStatsTest: public webrtc::VCMSendStatisticsCallback
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{
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public:
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SendStatsTest() : _frameRate(15) {}
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WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bitRate,
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const WebRtc_UWord32 frameRate);
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void SetTargetFrameRate(WebRtc_UWord32 frameRate) { _frameRate = frameRate; }
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private:
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WebRtc_UWord32 _frameRate;
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};
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class KeyFrameReqTest: public webrtc::VCMFrameTypeCallback
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{
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public:
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WebRtc_Word32 FrameTypeRequest(const webrtc::FrameType frameType);
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};
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#endif
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