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

modules/rtp_rtcp/test/testRateControl/testRateControl.cpp

@@ -0,0 +1,271 @@
+/*
+ *  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.
+ */
+
+#include <cassert>
+#include <windows.h>
+#include <iostream>
+#include <tchar.h>
+
+#include "rtp_rtcp.h"
+#include "common_types.h"
+#include "RateControlDetector.h"
+/*#include "rtcp_utility.h"
+#include "tmmbr_help.h"*/
+
+#define TEST_STR "Test RateControl."
+#define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
+#define PRINT_LINE std::cout << "------------------------------------------" << std::endl;
+
+
+const int maxFileLen = 200;
+WebRtc_UWord8* dataFile[maxFileLen];
+
+
+struct InputSet
+{
+    WebRtc_UWord32 TMMBR;
+    WebRtc_UWord32 packetOH;
+    WebRtc_UWord32 SSRC;
+};
+
+const InputSet set0   = {220,  80, 11111};  // bitRate, packetOH, ssrc
+const InputSet set1   = {180,  90, 22222};
+const InputSet set2   = {100, 210, 33333};
+const InputSet set3   = { 35,  40, 44444};
+const InputSet set4   = { 40,  60, 55555};
+const InputSet set4_1 = {100,  60, 55555};
+const InputSet set4_2 = { 10,  60, 55555};
+const InputSet set5   = {200,  40, 66666};
+const InputSet set00  = {  0,  40, 66666};
+
+
+
+
+WebRtc_Word32 GetFile(char* fileName)
+{
+    if (!fileName[0])
+    {
+        return 0;
+    }
+
+    FILE* openFile = fopen(fileName, "rb");
+    assert(openFile != NULL);
+    fseek(openFile, 0, SEEK_END);
+    int len = (WebRtc_Word16)(ftell(openFile));
+    rewind(openFile);
+    assert(len > 0 && len < maxFileLen);
+    fread(dataFile, 1, len, openFile);
+    fclose(openFile);
+    return len;
+};
+
+
+class LoopBackTransport2 : public webrtc::Transport
+{
+public:
+    LoopBackTransport2(RtpRtcp* rtpRtcpModule)  :
+      _rtpRtcpModule(rtpRtcpModule),
+      _cnt(0)
+    {
+    }
+    virtual int SendPacket(int channel, const void *data, int len)
+    {
+        return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len);
+    }
+    virtual int SendRTCPPacket(int channel, const void *data, int len)
+    {
+        char fileName[256] = {0};
+
+
+        // Get stored rtcp packet w/ TMMBR
+        len = GetFile(fileName);
+        if (len == 0)
+        {
+            return 0;
+        }
+
+        // Send in bitrate request
+        return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len);
+    }
+    RtpRtcp* _rtpRtcpModule;
+    WebRtc_UWord32       _cnt;
+};
+
+
+class LoopBackTransportVideo : public webrtc::Transport
+{
+public:
+    LoopBackTransportVideo(RtpRtcp* rtpRtcpModule)  :
+      _rtpRtcpModule(rtpRtcpModule),
+      _cnt(0)
+    {
+    }
+    virtual int SendPacket(int channel, const void *data, int len)
+    {
+        return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len);
+    }
+    virtual int SendRTCPPacket(int channel, const void *data, int len)
+    {
+        char fileName[256] = {0};
+
+        strcpy(fileName, "RTCPPacketTMMBR0.bin");
+
+        ++_cnt;
+
+        // Get stored rtcp packet w/ TMMBR
+        len = GetFile(fileName);
+        if (len == 0)
+        {
+            return 0;
+        }
+
+        // Send in bitrate request*/
+        return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len);
+    }
+
+    RtpRtcp* _rtpRtcpModule;
+    WebRtc_UWord32       _cnt;
+};
+
+class TestRateControl : private RateControlDetector
+{
+public:
+    TestRateControl():RateControlDetector(0)
+    {
+    }
+    ~TestRateControl()
+    {
+    }
+    void Start()
+    {
+        //Test perfect conditions
+        // But only one packet per frame
+        SetLastUsedBitRate(500);
+        WebRtc_UWord32 rtpTs=1234*90;
+        WebRtc_UWord32 framePeriod=33; // In Ms
+        WebRtc_UWord32 rtpDelta=framePeriod*90;
+        WebRtc_UWord32 netWorkDelay=10;
+        WebRtc_UWord32 arrivalTime=rtpTs/90+netWorkDelay;
+        WebRtc_UWord32 newBitRate=0;
+        for(WebRtc_UWord32 k=0;k<10;k++)
+        {
+            // Receive 10 packets
+            for(WebRtc_UWord32 i=0;i<10;i++)
+            {
+                NotifyNewArrivedPacket(rtpTs,arrivalTime);
+                rtpTs+=rtpDelta;
+                arrivalTime=rtpTs/90+netWorkDelay;
+            }
+            newBitRate=RateControl(2*netWorkDelay);
+            SetLastUsedBitRate(newBitRate);
+            Sleep(10*framePeriod);
+            std::cout << "RTCP Packet " << k << " new bitrate " << newBitRate << std::endl;
+        }
+        Reset();
+
+
+        //Test increasing RTT
+        std::cout << "Test increasing RTT - No Receive timing changes" << std::endl;
+        SetLastUsedBitRate(500);
+
+        for(WebRtc_UWord32 k=0;k<10;k++)
+        {
+            // Receive 10 packets
+            for(WebRtc_UWord32 i=0;i<10;i++)
+            {
+                NotifyNewArrivedPacket(rtpTs,arrivalTime);
+                rtpTs+=rtpDelta;
+                arrivalTime=rtpTs/90+netWorkDelay;
+            }
+            WebRtc_UWord32 rtt=2*netWorkDelay+k*20;
+            newBitRate=RateControl(rtt);
+            Sleep(10*framePeriod);
+            SetLastUsedBitRate(newBitRate);
+            std::cout << "RTCP Packet " << k << " RTT "<< rtt << " new bitrate " << newBitRate << std::endl;
+
+        }
+
+        Reset();
+
+
+        //Test increasing RTT
+        std::cout << "Test increasing RTT - Changed receive timing" << std::endl;
+        SetLastUsedBitRate(500);
+
+        for(WebRtc_UWord32 k=0;k<10;k++)
+        {
+            // Receive 10 packets
+            for(WebRtc_UWord32 i=0;i<10;i++)
+            {
+                NotifyNewArrivedPacket(rtpTs,arrivalTime);
+                rtpTs+=rtpDelta;
+                arrivalTime=rtpTs/90+netWorkDelay+i+(k*20);
+            }
+            WebRtc_UWord32 rtt=2*netWorkDelay+k*20;
+            newBitRate=RateControl(rtt);
+            Sleep(10*framePeriod);
+            SetLastUsedBitRate(newBitRate);
+            std::cout << "RTCP Packet " << k << " RTT "<< rtt << " new bitrate " << newBitRate << std::endl;
+
+        }
+
+
+
+    };
+};
+
+class NULLDataZink: public RtpData
+{
+    virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+                                                const WebRtc_UWord16 payloadSize,
+                                                const webrtc::WebRtcRTPHeader* rtpHeader)
+    {
+        return 0;
+    };
+};
+
+
+int _tmain(int argc, _TCHAR* argv[])
+{
+
+    std::string str;
+    std::cout << "------------------------" << std::endl;
+    std::cout << "---Test RateControl ----" << std::endl;
+    std::cout << "------------------------" << std::endl;
+    std::cout << "  "  << std::endl;
+
+    // --------------------
+    // Test TMMBRHelp class
+
+    // --------------------
+    TestRateControl test;
+    test.Start();
+
+    printf("RateControl-class test done.\n");
+
+    // ------------------------
+    // Test RateControl single module
+    // ------------------------
+    RtpRtcp* rtpRtcpModuleVideo = RtpRtcp::CreateRtpRtcp(0, false);
+
+    LoopBackTransportVideo* myLoopBackTransportVideo = new LoopBackTransportVideo(rtpRtcpModuleVideo);
+    assert(0 == rtpRtcpModuleVideo->RegisterSendTransport(myLoopBackTransportVideo));
+    printf("Multi module test done.\n");
+
+
+    RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo);
+    delete myLoopBackTransportVideo;
+
+    TEST_PASSED();
+    ::Sleep(5000);
+
+    return 0;
+}
+